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Administrator and User Manual

Mediatrix Boss

Document Revision 08 January 5, 2012

Media5 Corporation Administrator and User Manual

Media5 Corporation4229 Garlock StreetSherbrooke, Québec, Canada J1L 2C8

Mediatrix Boss Administrator and User Manual

© 2012, Media5 Corporation

All rights reserved. No part of this publication may be reproduced or used in any form or by any means – graphic, electronic, or mechanical, including photocopying, recording, taping, or information storage and retrieval systems – without the express written permission of the publisher.

Media5 Corporation reserves the right to revise this publication and make changes at any time and without the obligation to notify any person and/or entity of such revisions and/or changes.

Trademarks

Microsoft and Windows are registered trademarks of Microsoft Corporation.

Adobe and Acrobat are registered trademarks of Adobe Systems Incorporated.

All other trademarks and registered trademarks are the property of their respective owners.

Third-Party Software Copyright Information

The Mediatrix Boss firmware aggregates some third-party software modules (open source and commercial) that are distributed to you in accordance with their respective licenses. Refer to the Third Party Software Copyright Information addendum available on the Mediatrix Download Portal, which lists the third-party software modules along with any copyright and license information.

Software Configuration Guide

Contents

System Component

Chapter 1

Overview ..............................................................................................................................3

Scope of delivery ..........................................................................................................................................3Performance and Usage Recommendation..................................................................................................3

BHCA .......................................................................................................................................................................... 4Recommendation ........................................................................................................................................................ 4

Chapter 2

Basic configuration.............................................................................................................5

Establish a connection to the Mediatrix Boss ...............................................................................................5Architecture of the WebGUI ........................................................................................................................................ 6GUI Considerations ..................................................................................................................................................... 7Mediatrix Boss............................................................................................................................................................. 7

Internal Architecture of the Integrated Gateway Products ............................................................................8Typical Configuration................................................................................................................................................... 8Autoattendent Configuration........................................................................................................................................ 9Conferencing ............................................................................................................................................................. 10

Advanced System Introduction ...................................................................................................................11

Chapter 3

SYSTEM .............................................................................................................................13

Basic Settings .............................................................................................................................................13Time and Date:.......................................................................................................................................................... 13Time zone:................................................................................................................................................................. 14Domain Name Server: ............................................................................................................................................... 14System name: ........................................................................................................................................................... 14

Change Password ......................................................................................................................................15Install Update..............................................................................................................................................16

Installing an update for the Integrated Voice Gateway.............................................................................................. 17Provisioning ................................................................................................................................................17Configuration Backup .................................................................................................................................17

forceconfiguration...................................................................................................................................................... 18Configuration Rollback................................................................................................................................18

Number of Configurations ......................................................................................................................................... 19Previous Configuration .............................................................................................................................................. 19Configuration Save now ............................................................................................................................................ 19

Remote Backup ..........................................................................................................................................20Restart System ...........................................................................................................................................22Port Numbers..............................................................................................................................................23Restore Factory Defaults ............................................................................................................................24

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Contents

Licenses......................................................................................................................................................24

Chapter 4

NETWORK..........................................................................................................................25

WAN Interface ............................................................................................................................................25Connection via IP ...................................................................................................................................................... 25Connection via PPPoE / PPTP.................................................................................................................................. 25

LAN Interface 1...........................................................................................................................................26Dynamic IP address .................................................................................................................................................. 26Static IP address ....................................................................................................................................................... 27

LAN Interface 2...........................................................................................................................................29Deactivated ............................................................................................................................................................... 29Internal Network ........................................................................................................................................................ 29

DMZ Interface .............................................................................................................................................30Virtual Interfaces.........................................................................................................................................31Dynamic DNS .............................................................................................................................................32Name Server...............................................................................................................................................33Routing .......................................................................................................................................................35Bandwidth Management .............................................................................................................................36

Parent class............................................................................................................................................................... 38Description ................................................................................................................................................................ 38Rate........................................................................................................................................................................... 38Ceiling ....................................................................................................................................................................... 38Burst and Ceiling Burst.............................................................................................................................................. 38Priority ....................................................................................................................................................................... 38Leaf Queuing Discipline ............................................................................................................................................ 38MTU........................................................................................................................................................................... 39Filter .......................................................................................................................................................................... 39

CAC Bandwidth Sampling ..........................................................................................................................40Sample rate ............................................................................................................................................................... 40Number of samples ................................................................................................................................................... 40Log level .................................................................................................................................................................... 40

NTP.............................................................................................................................................................41VRRP..........................................................................................................................................................42

Additional Background Information ........................................................................................................................... 43QoS.............................................................................................................................................................44

Quality of Service Configuration................................................................................................................................ 44Traffic tagging methods:............................................................................................................................................ 44

Chapter 5

VPN.....................................................................................................................................45

Certificates..................................................................................................................................................45Preparations .............................................................................................................................................................. 45Managing CA certificates .......................................................................................................................................... 47Managing X.509 certificates ...................................................................................................................................... 47Creating X.509 certificates ........................................................................................................................................ 47

OpenVPN Shared Secrets..........................................................................................................................48Connections................................................................................................................................................49

Current Connections ................................................................................................................................................. 49Authentification Details: ............................................................................................................................................. 49Authentication Details:............................................................................................................................................... 51

L2TP Server................................................................................................................................................52

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Chapter 6

SECURITY ..........................................................................................................................55

Security Level .............................................................................................................................................55Firewall .......................................................................................................................................................56

Activate Basic Services ............................................................................................................................................. 56Custom Rules............................................................................................................................................................ 57

Port Forwarding ..........................................................................................................................................58

Chapter 7

VOICE .................................................................................................................................59

Survivability and Call Admission Control Template ....................................................................................60Survivability Basic Settings ....................................................................................................................................... 60PBX Addresses ......................................................................................................................................................... 61Survivability and Features ......................................................................................................................................... 63SIP Settings............................................................................................................................................................... 65Port Settings.............................................................................................................................................................. 70Integrated GW Settings ............................................................................................................................................. 70Domains .................................................................................................................................................................... 71Gateways .................................................................................................................................................................. 72CAC........................................................................................................................................................................... 74Diagnostics................................................................................................................................................................ 78

Branch Session Border Controller (SBC) Functionality ..............................................................................78Branch SBC Basic Settings....................................................................................................................................... 80PBX Addresses ......................................................................................................................................................... 80Survivability and Features ......................................................................................................................................... 80SIP Settings............................................................................................................................................................... 81Port Settings.............................................................................................................................................................. 81Integrated Gateway Settings ..................................................................................................................................... 81SBC Settings ............................................................................................................................................................. 81Domains .................................................................................................................................................................... 82Gateways .................................................................................................................................................................. 82CAC........................................................................................................................................................................... 82Diagnostics................................................................................................................................................................ 82

SBC (Session Border Controller) Template................................................................................................83Explanation of NAT ................................................................................................................................................... 84Basic Settings............................................................................................................................................................ 85PBX Addresses ......................................................................................................................................................... 85Port Settings.............................................................................................................................................................. 86Integrated Gateway Settings ..................................................................................................................................... 86SBC Settings ............................................................................................................................................................. 86Domains .................................................................................................................................................................... 88Security: .................................................................................................................................................................... 88Diagnostics................................................................................................................................................................ 88

SIP Trunking Template ...............................................................................................................................89Basic Settings............................................................................................................................................................ 90PBX Addresses ......................................................................................................................................................... 91SIP Settings............................................................................................................................................................... 91Port Settings.............................................................................................................................................................. 92SBC Settings ............................................................................................................................................................. 92Quality of Service - TOS value for SIP/RTP traffic .................................................................................................... 92Diagnostics................................................................................................................................................................ 92

Standalone Template..................................................................................................................................93Basic Settings............................................................................................................................................................ 93PBX Addresses/ Failover Convergence IP................................................................................................................ 94Port Settings.............................................................................................................................................................. 94

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Contents

Domains .................................................................................................................................................................... 94Gateways .................................................................................................................................................................. 95Debug........................................................................................................................................................................ 95

ENUM Template .........................................................................................................................................95Basic Settings............................................................................................................................................................ 95PBX Addresses ......................................................................................................................................................... 97Port Settings.............................................................................................................................................................. 97SBC Settings ............................................................................................................................................................. 98Debug........................................................................................................................................................................ 98

Custom Template .......................................................................................................................................98Registered Users ........................................................................................................................................98SIP TLS Configuration ................................................................................................................................98

Upload Private Key and Certificate ......................................................................................................................... 100Create default private key and certificate ................................................................................................................ 100

B2BUA ......................................................................................................................................................100Configuration Concept Introduction......................................................................................................................... 101General Options ...................................................................................................................................................... 101SIP Carrier Accounts............................................................................................................................................... 102SIP Peers ................................................................................................................................................................ 103Call Routing Rules................................................................................................................................................... 104Diagnostics.............................................................................................................................................................. 105

Autoattendant Basic Settings....................................................................................................................105Basic Settings.......................................................................................................................................................... 105

Announcements........................................................................................................................................106Announcement Extensions...................................................................................................................................... 107Configure Prompts for Announcement Extensions ................................................................................................. 108Active Times............................................................................................................................................................ 110

Conferencing ............................................................................................................................................112Create a conference room....................................................................................................................................... 112Create a conference moderator .............................................................................................................................. 113

Voicemail ..................................................................................................................................................114PBX Extension Registration......................................................................................................................115Hunt Group Configuration.........................................................................................................................116

Simultaneous Ringing ............................................................................................................................................. 116Call Queuing............................................................................................................................................................ 117

Upload and Deletion of hold music ...........................................................................................................118Upload and Deletion of Announcements ..................................................................................................119

Chapter 8

DIAGNOSTICS .................................................................................................................121

Syslog-File ................................................................................................................................................121Logging .....................................................................................................................................................122Debugging Tools.......................................................................................................................................123

Ping ......................................................................................................................................................................... 123Traceroute ............................................................................................................................................................... 123

SNMP .......................................................................................................................................................124SNMP Basic Settings .............................................................................................................................................. 124Trap destination....................................................................................................................................................... 124Proxy ....................................................................................................................................................................... 124

Firewall Report..........................................................................................................................................126Network Trace ..........................................................................................................................................126Support Trace ...........................................................................................................................................128

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Chapter 9

Command Line Interface ................................................................................................129

Access Methods .......................................................................................................................................130CLI Console............................................................................................................................................................. 130Serial Line Console ................................................................................................................................................. 130SSH Access ............................................................................................................................................................ 130

Initial Setup using the Command Line Interface .......................................................................................131Physical Network Settings....................................................................................................................................... 132

Configuration utilities ................................................................................................................................133applyconfig .............................................................................................................................................................. 133restartservices ......................................................................................................................................................... 133chconfig ................................................................................................................................................................... 134lockconfig and unlockconfig .................................................................................................................................... 134apply_flush_firewall ................................................................................................................................................. 134saveconfiguration / restoreconfiguration ................................................................................................................. 135

The "baseconfig" configuration file ...........................................................................................................135Syntax ..................................................................................................................................................................... 135

Parameter Reference for baseconfig........................................................................................................136System (system_).................................................................................................................................................... 136Network ................................................................................................................................................................... 136Network related Topics............................................................................................................................................ 140Security (including VPN) ......................................................................................................................................... 143Diagnostics.............................................................................................................................................................. 152Voice (ser_) ............................................................................................................................................................. 153

Factory and Password Reset....................................................................................................................175Factory Reset and Password Reset via SSH.......................................................................................................... 175Password Reset via RS232..................................................................................................................................... 175Factory reset via RS232.......................................................................................................................................... 175

Additional Information ...............................................................................................................................176Traffic Control / Quality of Service........................................................................................................................... 176Testing Behavior with force_send_notifies.............................................................................................................. 176Control Script: serctlc .............................................................................................................................................. 176Improving VRRP...................................................................................................................................................... 178Call Detail Records (CDRs)..................................................................................................................................... 179Using Radius ........................................................................................................................................................... 180Regular expressions................................................................................................................................................ 182

Chapter 10

Sample scenarios............................................................................................................185

Survivability...............................................................................................................................................185Branch Connectivity and Branch Session Border Controller ....................................................................186Hosted Communication ............................................................................................................................187Scenarios with Integrated Voice Gateways ..............................................................................................187SIP Trunking .............................................................................................................................................187

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Contents

Gateway Component

Chapter 11

System Overview ............................................................................................................193

Introduction ...............................................................................................................................................193Mediatrix 3000 Series.............................................................................................................................................. 193

DSP Limitation ..........................................................................................................................................195Mediatrix 3200 Series.............................................................................................................................................. 195Mediatrix 3300 Series.............................................................................................................................................. 196Mediatrix 3400 Series.............................................................................................................................................. 196Mediatrix 3600 Series.............................................................................................................................................. 196Mediatrix 3731......................................................................................................................................................... 197Mediatrix 3732......................................................................................................................................................... 197Mediatrix 3734......................................................................................................................................................... 197Mediatrix 3741......................................................................................................................................................... 197Mediatrix 3742......................................................................................................................................................... 198

RESET/DEFAULT Button .........................................................................................................................198At Run-Time ............................................................................................................................................................ 198At Start-Time ........................................................................................................................................................... 199Partial Reset............................................................................................................................................................ 200Factory Reset .......................................................................................................................................................... 201

Chapter 12

Web Interface Configuration ..........................................................................................203

Introduction ...............................................................................................................................................203Using the Web Interface ...........................................................................................................................204

Menu Items.............................................................................................................................................................. 205Submitting Changes .................................................................................................................................207Where to Go From Here? .........................................................................................................................207

Using Secure Communication................................................................................................................................. 208

System Parameters

Chapter 13

Services ...........................................................................................................................211

Services Table ..........................................................................................................................................211

Chapter 14

Hardware Parameters .....................................................................................................215

Hardware Card Configuration ...................................................................................................................215BRI Hardware Configuration ................................................................................................................................... 215PRI Hardware Configuration ................................................................................................................................... 216

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Chapter 15

Syslog Configuration......................................................................................................219

Syslog Daemon Configuration ..................................................................................................................219Configuring PCM Capture ....................................................................................................................................... 221Configuring the Syslog Daemon Application ........................................................................................................... 222

Network Parameters

Chapter 16

Local QoS (Quality of Service) Configuration ..............................................................225

Introduction ...............................................................................................................................................225Specific Service Class Configuration........................................................................................................225

ISDN Parameters

Chapter 17

ISDN Configuration .........................................................................................................229

Introduction ...............................................................................................................................................229ISDN Reference Points ........................................................................................................................................... 229Inband Tones Generation........................................................................................................................................ 230

Setting PRI Hardware Parameters ...........................................................................................................231PRI Configuration .....................................................................................................................................232BRI Configuration .....................................................................................................................................239

Bypass Feature (Mediatrix 3404/3408/3734/3741/3742 Models)............................................................................ 246Interop Parameters Configuration.............................................................................................................247ISDN Timers Configuration.......................................................................................................................250Services Configuration..............................................................................................................................251

CAS Parameters

Chapter 18

R2 CAS Configuration.....................................................................................................259

Introduction ...............................................................................................................................................259Line Signals for the Digital Version of MFC/R2 ....................................................................................................... 259Interregister Signals ................................................................................................................................................ 259

Selecting the R2 Signaling Protocol .........................................................................................................260R2 Channel Associated Signaling ............................................................................................................261R2 Signaling Variants ...............................................................................................................................264

Override Default Country Settings........................................................................................................................... 265R2 Signaling Variants.............................................................................................................................................. 266

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Contents

R2 Timers Variants ...................................................................................................................................268Override Default Country Settings........................................................................................................................... 269R2 Timers Variants.................................................................................................................................................. 270

R2 Digit Timers Variants...........................................................................................................................272Override Default Country Settings........................................................................................................................... 272R2 Digit Timers Variants ......................................................................................................................................... 273

R2 Link Timers Variants ...........................................................................................................................274Override Default Country Settings........................................................................................................................... 274R2 Link Timers Variants .......................................................................................................................................... 275

R2 Tones Variants ....................................................................................................................................275Override Default Country Settings........................................................................................................................... 277R2 Tones Forward Groups...................................................................................................................................... 278R2 Tones Backward Groups ................................................................................................................................... 279

POTS Parameters

Chapter 19

POTS Configuration........................................................................................................287

POTS Status.............................................................................................................................................287Line Status .............................................................................................................................................................. 287FXO Line Status ...................................................................................................................................................... 288

General POTS Configuration....................................................................................................................288Caller ID Information ............................................................................................................................................... 290

FXS Configuration ....................................................................................................................................291FXS Country Customization .....................................................................................................................293

Calling Party Name of the Caller ID ........................................................................................................................ 294FXS Bypass ..............................................................................................................................................295FXO Configuration ....................................................................................................................................296

FXO Dialing Configuration....................................................................................................................................... 297FXO Answering Configuration................................................................................................................................. 298FXO Incoming Call Configuration............................................................................................................................ 299FXO Line Verification .............................................................................................................................................. 299FXO Force End of Call ............................................................................................................................................ 300

SIP Parameters

Chapter 20

SIP Gateways...................................................................................................................305

SIP Gateways Configuration.....................................................................................................................305

Chapter 21

SIP Servers ......................................................................................................................307

Introduction ...............................................................................................................................................307TLS Persistent Connections Status ..........................................................................................................308SIP Servers Configuration ........................................................................................................................309

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Multiple SIP Gateways..............................................................................................................................310SIP Gateway Specific Registrar Servers................................................................................................................. 310SIP Gateway Specific Messaging Servers .............................................................................................................. 310SIP Gateway Specific Proxy Servers ...................................................................................................................... 311

Outbound Proxy Loose Router Configuration...........................................................................................312

Chapter 22

SIP Registration...............................................................................................................315

Endpoints Registration..............................................................................................................................315Unit Registration .......................................................................................................................................317Registration Configuration ........................................................................................................................317Number of Registrations ...........................................................................................................................319Registration Refresh Parameters .............................................................................................................320

Default vs. Specific Configurations.......................................................................................................................... 320Registration Refresh................................................................................................................................................ 320Registration Expiration ............................................................................................................................................ 320Unregistered Endpoint Behaviour ........................................................................................................................... 321

SIP User Agent Header ............................................................................................................................322

Chapter 23

Endpoints State Configuration ......................................................................................323

Endpoints Configuration ...........................................................................................................................323Administration ...........................................................................................................................................324Session Timers .........................................................................................................................................325

Background Information .......................................................................................................................................... 326

Chapter 24

SIP Authentication ..........................................................................................................327

Authentication Configuration.....................................................................................................................327Creating/Editing an Authentication Entry................................................................................................................. 328Moving an Authentication Entry............................................................................................................................... 330Deleting an Authentication Entry ............................................................................................................................. 330

Chapter 25

SIP Transport Parameters ..............................................................................................331

SIP Transport Type...................................................................................................................................331Additional Transport Parameters ..............................................................................................................333

UDP Source Port Behaviour.................................................................................................................................... 333TLS Client Authentication........................................................................................................................................ 334

Chapter 26

Interop Parameters..........................................................................................................335

Behavior on T.38 INVITE Not Accepted ...................................................................................................335SIP Interop................................................................................................................................................336SDP Interop ..............................................................................................................................................339TLS Interop ...............................................................................................................................................342

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Contents

Misc Interop ..............................................................................................................................................343Additional Interop Parameters ..................................................................................................................344

Call Waiting Private Number Criteria for SIP INFO ................................................................................................. 344Max-Forwards Header............................................................................................................................................. 344Direction Attributes in a Media Stream.................................................................................................................... 345Local Ring Behaviour on Provisional Response ..................................................................................................... 347Session ID and Session Version Number in the Origin Field of the SDP................................................................ 347Register Home Domain Override ............................................................................................................................ 348Listening for Early RTP ........................................................................................................................................... 348

Chapter 27

Miscellaneous SIP Parameters ......................................................................................351

SIP Penalty Box........................................................................................................................................351Penalty Box vs Transport Types ............................................................................................................................. 351Penalty Box Configuration....................................................................................................................................... 352

Error Mapping ...........................................................................................................................................353SIP to Cause Error Mapping ................................................................................................................................... 355Cause to SIP Error Mapping ................................................................................................................................... 356

SIP Keep Alive..........................................................................................................................................359PRACK & UPDATE ..................................................................................................................................360SIP Gateway Configuration ......................................................................................................................361Diversion Configuration ............................................................................................................................361DNS Configuration....................................................................................................................................362

Telephony Parameters

Chapter 28

DTMF Maps Configuration..............................................................................................365

Introduction ...............................................................................................................................................365Syntax.......................................................................................................................................................365

Special Characters .................................................................................................................................................. 366How to Use a DTMF Map........................................................................................................................................ 366

General DTMF Maps Parameters.............................................................................................................368Configuring Timeouts per Endpoint......................................................................................................................... 369

Allowed DTMF Maps ................................................................................................................................370Refused DTMF Maps................................................................................................................................372

Chapter 29

Voice & Fax Codecs Configuration ...............................................................................373

Codec Descriptions...................................................................................................................................373G.711 A-Law and µ-Law.......................................................................................................................................... 373G.723.1.................................................................................................................................................................... 374G.726....................................................................................................................................................................... 374G.729....................................................................................................................................................................... 375Clear Mode.............................................................................................................................................................. 375Clear Channel ......................................................................................................................................................... 376X-CCD Clear Channel ............................................................................................................................................. 377T.38 ......................................................................................................................................................................... 377

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Codec Parameters....................................................................................................................................378Codec vs. Bearer Capabilities Mapping....................................................................................................379Miscellaneous Parameters .......................................................................................................................380

About Changing Jitter Buffer Values ....................................................................................................................... 387Starting a Call in Voiceband Data Mode ................................................................................................................. 387DTMF Transport over the SIP Protocol ................................................................................................................... 388DTMF Detection ...................................................................................................................................................... 389Using the Payload Type Found in the Answer ........................................................................................................ 390Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP .......................................................... 391

Security.....................................................................................................................................................391Enforcing Symmetric RTP ....................................................................................................................................... 393

G.711 Codec Parameters .........................................................................................................................394G.723 Codec Parameters .........................................................................................................................395G.726 Codecs Parameters .......................................................................................................................397G.729 Codec Parameters .........................................................................................................................399Clear Mode Codec Parameters ................................................................................................................400Clear Channel Codec Parameters............................................................................................................402X-CCD Clear Channel Codec Parameters ...............................................................................................404Fax Parameters ........................................................................................................................................405

Clear Channel Fax .................................................................................................................................................. 406T.38 Fax .................................................................................................................................................................. 406T.38 Parameters Configuration ............................................................................................................................... 407Data Codec Selection Procedure ............................................................................................................................ 409

Chapter 30

Call Forward Configuration............................................................................................411

Call Forward On Busy...............................................................................................................................411Configuring Call Forward on Busy via Handset....................................................................................................... 413

Call Forward On No Answer .....................................................................................................................414Configuring Call Forward on Answer via Handset................................................................................................... 415

Call Forward Unconditional.......................................................................................................................416Configuring Call Forward on Unconditional via Handset......................................................................................... 417

Chapter 31

Telephony Services Configuration................................................................................419

General Configuration...............................................................................................................................419Automatic Call ......................................................................................................................................................... 421Call Completion ....................................................................................................................................................... 422Call Transfer............................................................................................................................................................ 426Call Waiting ............................................................................................................................................................. 427Conference.............................................................................................................................................................. 429Delayed Hot Line..................................................................................................................................................... 431Direct IP Address Call ............................................................................................................................................. 432Call Hold.................................................................................................................................................................. 433Second Call ............................................................................................................................................................. 434

Message Waiting Indicator .......................................................................................................................435

Chapter 32

Tone Customization Parameters Configuration...........................................................437

Current Tone Definition.............................................................................................................................437Tone Override...........................................................................................................................................438

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Contents

Chapter 33

Music on Hold Parameters Configuration.....................................................................441

Music on Hold Configuration.....................................................................................................................441

Chapter 34

RTP Statistics Configuration..........................................................................................443

Statistics Displayed...................................................................................................................................443Statistics Configuration .............................................................................................................................445

Chapter 35

Country Parameters Configuration ...............................................................................447

Country Configuration...............................................................................................................................447Additional Country Settings ......................................................................................................................448

Default vs. Specific Configurations.......................................................................................................................... 448Input/Output User Gain ........................................................................................................................................... 448Dialing Settings ....................................................................................................................................................... 449Fax Calling Tone Detection ..................................................................................................................................... 450

Call Router Parameters

Chapter 36

Call Router Configuration...............................................................................................453

Introduction ...............................................................................................................................................453Limitations ............................................................................................................................................................... 454Regular Expressions ............................................................................................................................................... 454Routing Type ........................................................................................................................................................... 456Call Properties Parameters ..................................................................................................................................... 462SIP/ISDN Call Default Values ................................................................................................................................. 468Call Routing Status.................................................................................................................................................. 470

Routes ......................................................................................................................................................470Creating/Editing a Route ......................................................................................................................................... 471Moving a Route ....................................................................................................................................................... 476Deleting a Route...................................................................................................................................................... 476

Mappings ..................................................................................................................................................476Creating/Editing a Mapping Type ............................................................................................................................ 477Creating/Editing a Mapping Expression .................................................................................................................. 478Moving a Mapping Type or Expression Row........................................................................................................... 486Deleting a Mapping Type or Expression Row ......................................................................................................... 486

Signalling Properties.................................................................................................................................486Creating/Editing a Signalling Property..................................................................................................................... 487Moving a Signalling Property Row .......................................................................................................................... 490Deleting a Signalling Property Row......................................................................................................................... 490

SIP Headers Translations.........................................................................................................................490Creating/Editing a SIP Headers Translation............................................................................................................ 490Moving a SIP Headers Translation Row ................................................................................................................. 492Deleting a SIP Headers Translation Row................................................................................................................ 493

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Call Properties Translations......................................................................................................................493Creating/Editing a Call Properties Translation ........................................................................................................ 493Moving a Call Properties Translation Row .............................................................................................................. 496Deleting a Call Properties Translation Row............................................................................................................. 496

Hunt Service .............................................................................................................................................496Creating/Editing a Hunt ........................................................................................................................................... 496Call Rejection (Drop) Causes.................................................................................................................................. 499Moving a Hunt ......................................................................................................................................................... 503Deleting a Hunt........................................................................................................................................................ 503

Hairpinning................................................................................................................................................504Configuration Examples............................................................................................................................505

Chapter 37

Auto-Routing Configuration...........................................................................................507

Auto-Routing.............................................................................................................................................507Manual Routing ....................................................................................................................................................... 510

Management Parameters

Chapter 38

Configuration Script........................................................................................................513

Configuration Script Server.......................................................................................................................513Configuring the TFTP Server .................................................................................................................................. 513Configuring the SNTP Server.................................................................................................................................. 514Configuring the HTTP Server .................................................................................................................................. 514Configuring the HTTPS Server................................................................................................................................ 514

Generating a Configuration Script from the Running Configuration .........................................................515Configuration Scripts Server Settings.......................................................................................................518

Certificate Validation ............................................................................................................................................... 521Configuration Download Procedure..........................................................................................................522Automatic Configuration Update...............................................................................................................522

Automatic Update on Restart .................................................................................................................................. 523Automatic Update at a Specific Time Interval ......................................................................................................... 523DHCP Auto-Provisioning ......................................................................................................................................... 524Number of Retries ................................................................................................................................................... 525

Creating a Configuration Script.................................................................................................................526

Chapter 39

Certificates Management................................................................................................527

Introduction ...............................................................................................................................................527Managing Certificates ...............................................................................................................................528Transferring a Certificate ..........................................................................................................................528

Transferring a Certificate via Configuration Script................................................................................................... 529Host Certificate Associations ....................................................................................................................530

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Contents

Appendices

Appendix A

Country-Specific Parameters .........................................................................................533

Definitions .................................................................................................................................................533Conventions ............................................................................................................................................................ 533Distinctive Ring........................................................................................................................................................ 535

Austria.......................................................................................................................................................536Czech Republic.........................................................................................................................................537France.......................................................................................................................................................538Germany ...................................................................................................................................................539

Germany 1............................................................................................................................................................... 539Germany2................................................................................................................................................................ 540

Italy ...........................................................................................................................................................541North America...........................................................................................................................................542

North America 1 ...................................................................................................................................................... 542Spain.........................................................................................................................................................543Switzerland ...............................................................................................................................................544United Arab Emirates................................................................................................................................545

Appendix B

Scripting Language.........................................................................................................547

General Scripting Language Syntax .........................................................................................................547Supported Characters ............................................................................................................................................. 548

Assigning Scalar Values ...........................................................................................................................549Assigning Table Cell Values .....................................................................................................................549Executing Commands...............................................................................................................................550

Normal Commands ................................................................................................................................................. 550Row Commands...................................................................................................................................................... 551DeleteAllRows Command ....................................................................................................................................... 551

Call Router Specific Information ...............................................................................................................551Examples ..................................................................................................................................................552

Basic Call ................................................................................................................................................................ 552Debugging ............................................................................................................................................................... 553Advanced Configuration .......................................................................................................................................... 554

Appendix C

Glossary...........................................................................................................................557

Appendix D

List of Acronyms.............................................................................................................567

Appendix E

List of Standards Supported..........................................................................................569

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System Component

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C H A P T E R

1 Overview

Scope of delivery

1. Mediatrix Boss with built-in power supply (the ports vary on what model you purchased varying from 4+1 BRI ports to 2 E1/T1 ports)

2. Power cord

3. Network cable

4. Crossover network cable

5. Mounting brackets for 19" racks

6. RS232 console cable

7. Quick reference sheet

Figure 1: Media5 Boss

Performance and Usage Recommendation

The following table shows the expected performance for the Media5 Boss.

Note: These devices are configured through an HTTP interface. The standard configuration GUI is running on port 80. From there it is possible to configure everything including the gateway components. It is possible to access the gateway configuration via a separate GUI running on port 8080. Note however that any network setting you make there will have no effect.

Table 1: Performance and Usage Recommendation

FeatureMaximum

Concurrent CallsResource Usage

per Call

Number of registered user 150 N/A

Number of calls to/from analog/digital ports 42 2.4 %

Number of calls SIP to SIP 100 1 %

IVR playback to analog/digital port 22 4 %

IVR playback to SIP client/server 32 3 %

Voicemail playback to analog/digital port 25 4 %

Media5 Boss 3

Chapter 1 - Overview Performance and Usage Recommendation

BHCA

Busy hour call attempts (BHCA) SIP to analog/digital port: 7000

Busy hour call attempts (BHCA) SIP to SIP: 20000

Recommendation

Number of users: 150

Number of voice mail boxes: 100

Do not exceed 100% resources usage

Note that the storage capacity may further limit the number of voice mail boxes

Voicemail playback to SIP client/server 32 3 %

Voicemail recording from analog/digital port 5a 4.3 %

Voicemail recording from SIP client/server 5a 2.5 %

a. Maximum of 5 concurrent voicemail recording from any source.

Table 1: Performance and Usage Recommendation (Continued)

FeatureMaximum

Concurrent CallsResource Usage

per Call

4 Media5 Boss

C H A P T E R

2 Basic configuration

The Mediatrix Boss is a browser-based application. You can access it via any common browser. Therefore, the Mediatrix Boss configuration is independent of the operating system, no matter if it is Linux, Macintosh or Microsoft Windows. It is also not necessary to install any special tools, unless you want to copy files or have command-line access. Please refer to “Chapter 9 - Command Line Interface” on page 129 for more information about this possibility.

For the web configuration, your web browser must support frames, but nowadays this applies to almost every modern browser. For checking your input, on some pages JavaScript support is also required. Please make sure this is enabled in your web browser.

The LAN1 interface of the Mediatrix Boss is configured to the following parameters by default.

Therefore, in order to be able to communicate with the Mediatrix Boss , you need to configure your PC to an address in the same subnet, as for example 10.0.0.206 .

Establish a connection to the Mediatrix Boss

1. To enter the graphical user interface (GUI) of the Mediatrix Boss and to begin the configuration, enter https://10.0.0.205 in your browser's address field. For some products, the correct protocol is HTTP, so you have to enter

http://10.0.0.205 to connect correctly.

Check the Quick Reference Sheet for your Mediatrix Boss for more information about this.

In order for this to work, the IP address of your PC has to be set up to be in this subnet. Please consult the documentation of your operating system for more information on how to accomplish this.

Possibly your browser will ask you to accept a certificate and warn you that it is self-signed. To continue you will have to accept this certificate for establishing an encrypted connection to the Mediatrix Boss .

2. Enter the username and password. The defaults values are as follows:

Username: admin

Password: sesam

For security reasons, you should change the default password as soon as possible. The menu for password modification is described in section “Change Password” on page 15.

3. Select your language.

After having logged in successfully you should see the HTML frontend of your Mediatrix Boss. You can change the language anytime on the Head.

Table 2: LAN1 Properties

LAN1 Properties Details

Network protocol TCP/IP

IP-Address 10.0.0.205

Subnetmask 255.255.255.0

Mediatrix Boss 5

Chapter 2 - Basic configuration Establish a connection to the Mediatrix Boss

Architecture of the WebGUI

The WebGUI of the Mediatrix Boss consist of different parts. These are

Page Header

Top Menu Bar

Top Navigation

Main Frame

Page Header

The page header gives you information about the Mediatrix Boss you are dealing with.

Top Menu Bar

The Top Menu Bar is the top level navigation. This navigation gives you three major options to select:

1. Home: this page displays the status of all the network components of the Mediatrix Boss in alphabetical order. There are three possible status which are shown by LEDs (compare Figure 2 on page 6):

- active

- inactive

- disabled

It also offers the possibility to start, stop or restart individual services of the Mediatrix Boss by clicking , or , respectively. The latter (restart) is an alternative to Apply Configuration which leads to a complete restart of all services (see Figure 2 on page 6 below).

2. Apply Configuration: whenever you click on Apply Configuration the entered content will be written into the base configuration of the Mediatrix Boss. At this point running services will be interrupted due to a restart of the Mediatrix Boss. If you do not want to interrupt the operations of the Mediatrix Boss, the Status Page offers you a way to apply configuration changes for individual services only.

3. Language: The currently not selected language will be displayed here. Select it by clicking on it. If a menu is opened at that time you might have to reopen it to apply the new language setting.

Figure 2: Service Status

Note: Changing users and other options which are stored in the database does usually not require clicking Apply Configuration. However, if it is necessary, the Apply Configuration option will visually indicate this.

6 Mediatrix Boss

Establish a connection to the Mediatrix Boss Software Configuration Guide

Main Navigation (left)

The main navigation is a two level navigation. This manual is structured according to this navigation. Select any of the top (main) menu points (SYSTEM, NETWORK etc.). When you do that the list of available submenus for this main menu will open in the main frame. By clicking on the menu titles of this list, the configuration view for the selected menu item will open.

Main Frame

The main frame will always contain the currently selected configuration screen. Use the Main Navigation to select what part of the Mediatrix Boss you want to configure.

Bottom Menu Bar

You can find important product and software information on the bottom right hand side. Of this information, the most important is the software version which is a four digit number. This number can be followed by a dot which is followed by more digits. The digits after this dot indicate the patch set number which the product contains. Note that Updates (with or without included patches) install and behave exactly the same way.

This information simply indicates that this software version is based on another major revision, but contains additional patches.

GUI Considerations

The following explains a few points that you need to consider.

Default Values

There are both explicit as well as implicit default values. The explicit ones are set by default in the WebGUI. Implicit ones are fields that can simply be left empty. The system will select an appropriate default value for you. You can leave all fields except the mandatory ones empty.

Mandatory Values

Mandatory values have to be configured to use a certain feature. In the GUI, you can identify them by the * symbol behind the title. If left empty, you will receive an error message asking you to set the value.

Erroneous Values

The Mediatrix Boss checks the entered values for consistency and will not accept values failing this check. An error message will be displayed. This however cannot avoid erroneous configurations since only basic sanity and syntax checks of the entered values are performed.

These checks include type checks (string or number, etc.), range checks (do entered numbers make sense), and basic sanity checks (do certain parameters exclude each other, etc.).

Mediatrix Boss

Note that the product with integrated gateway devices have some special properties. First, as already mentioned in this manual, you have to enter http://10.0.0.205 to correctly connect. This means that the product is using the HTTP protocol as its default interface.

In addition to that, there is an additional menu entry in the configuration menu by the name of Gateway. This menu point will bring you to the VoIP gateway configuration. When clicking this menu, you will get to the configuration page shown in Figure 3 on page 8.

Here you will need to use the standard gateway credentials supplied with the gateway which is public as username with an empty password. This will bring you into the gateway configuration. In order to get a description of the parameters there, please refer to the Mediatrix 3000 Series Digital Gateway documentation. Note that all menu points not making sense in the integrated product have been made inaccessible.

Mediatrix Boss 7

Chapter 2 - Basic configuration Internal Architecture of the Integrated Gateway Products

Figure 3: Gateway Configuration

Internal Architecture of the Integrated Gateway Products

The Mediatrix Boss architecture is a very open one that consists of multiple parts that loosely interact. The internal communication between the different components is all handled in SIP.

The Mediatrix Boss therefore has a whole number that run on different ports and can be addressed internally as well as externally.

Figure 4 on page 9 should give an overview as to how the different components within this Mediatrix Boss work together.

SBC and Survivability Component (running on port 5060 for UDP/TCP SIP and on port 5061 for TLS SIP). This port is configurable because it is used for all outside communication

Autoattendent and Conferencing unit (running on port 5070, not configurable)

Media Gateway (By default port 5064, but you can create multiple virtual gateways)

Typical Configuration

The most interesting part of the configuration is the integration of the autoattendant. Let us however first begin with a general overview. A typical configuration will have a Softswitch somewhere in the network, where the Mediatrix Boss acts as a Voice Gateway and SBC / Survivability component. The SBC will be necessary only if there is a NAT between the central Softswitch / IP PBX and the phones connected behind the Mediatrix Boss . In those cases, the SBC component can mask all private IP addresses used on the LAN side of the Mediatrix Boss . This would mean that

If the above described network separation is not necessary, you will typically only have the LAN interface connected. All services are then reachable directly on their respective ports.

In that configuration, you can just configure the Survivability. In order to be able to use this also for inbound and outbound calls, you should point the SIP phones as well as the gateway to the local Survivability component as an outbound proxy. See (Figure 6 on page 10 ) for more information of how to configure the gateway.

Note: When using the Mediatrix Boss as a Branch SBC, you will need to configure your PBX in such a way, that all SIP messages intended for the gateway are directed towards the SBC component. The SBC component will mask them and forward them internally to the gateway component. In a single interface configuration, this is not necessary and you can have your PBX forward the SIP signalling directly to the port of the configured virtual gateway.

8 Mediatrix Boss

Internal Architecture of the Integrated Gateway Products Software Configuration Guide

Figure 4: BiaB Architecture

Autoattendent Configuration

The autoattendent will behave like a virtual gateway for the PBX. In order to guarantee a typical autoattendent functionality, incoming calls to a certain number must be answered by the autoattendent. This can be accomplished by defining a separate virtual gateway that uses the autoattendent as a SIP destination. Then you can use the call routing to route this number to the appropriate virtual gateway. Another virtual gateway can be used for all direct inward dialing numbers. Refer to Figure 5 on page 10 and Figure 6 on page 10 for more information on this.

Mediatrix Boss 9

Chapter 2 - Basic configuration Internal Architecture of the Integrated Gateway Products

Figure 5: SIP Gateway configuration

If configured as described above, the incoming calls will be picked up on the autoattendent. The autoattendent can then be configured to ask for an extension which in turn can be forwarded to the PBX. The autoattendent would then initiate a call to the PBX as if it as the gateway the call arrived over.

Conferencing

The conference rooms are also normal extensions on the local device. This can either be dialed from external users via the autoattendent (this could be accomplished by pointing the gateway component to the autoattendent as described above, or by configuring a prefix on your PBX to forward calls to the autoattendant), or by registering the extensions to the PBX. If an extension is registered to the PBX, a conference room could e.g. be dialed just like a regular user. The conference unit will pick up the call and ask for the room number or password if applicable. Please check the full autoattendent configuration for more information on this.

Figure 6: SIP Gateway server configuration

10 Mediatrix Boss

Advanced System Introduction Software Configuration Guide

Advanced System Introduction

The following sections assume familiarity with the concepts of IP networking and if applicable with those of VoIP.

You can find all configuration parameters structured in a two level menu. The configuration can be performed in any order.

Below is a list of all sections that can be found in the configuration.

This table gives you an overview as to where you can find more detailed information about a certain topic in this manual.

Table 3: Documentation Sections

SYSTEM

In the section, you can configure all the system basics, update the unit, factory reset the unit, as well as pull configuration backups, or restore saved configurations.

Basic settings “Basic Settings” on page 13

Change Password “Change Password” on page 15

Install Update “Install Update” on page 16

Provisioning “Provisioning” on page 17

Configuration Backup “Configuration Backup” on page 17

Configuration Rollback “Configuration Rollback” on page 18

Remote Backup “Remote Backup” on page 20

Restart System “Restart System” on page 22

Port Numbers “Port Numbers” on page 23

Restore Factory Defaults “Restore Factory Defaults” on page 24

NETWORK

The network part lets the user configure all interfaces, virtual interfaces, traffic shaping and other network services such as NTP.

WAN Interface “WAN Interface” on page 25

LAN Interface 1 “LAN Interface 1” on page 26

LAN Interface 2 “LAN Interface 2” on page 29

Dynamic DNS “Dynamic DNS” on page 32

Name Server “Name Server” on page 33

Routing “Routing” on page 35

Bandwidth Management “Bandwidth Management” on page 36

CAC Bandwidth Sampling “CAC Bandwidth Sampling” on page 40

NTP “NTP” on page 41

VRRP “VRRP” on page 42

QOS “QoS” on page 44

VPN

Here you can manage certificates and create VPN connections.

Certificates “Certificates” on page 45

OpenVPN Shared Secrets “OpenVPN Shared Secrets” on page 48

Mediatrix Boss 11

Chapter 2 - Basic configuration Advanced System Introduction

Connections “Connections” on page 49

L2TP Server “L2TP Server” on page 52

SECURITY

The security portion consists mostly of the firewall settings.

Security Level “Security Level” on page 55

Firewall “Firewall” on page 56

Port Forwarding “Port Forwarding” on page 58

VOICE

The Voice tab contains all templates for VoIP scenarios (Session Border Controller configurations, Survivability functionality etc.), as well as Autoattendant and B2BUA handling functions.

VoIP Templates from “Survivability and Call Admission Control Template” on page 60 to “Custom Template” on page 98

SIP Proxy and SBC Users “Registered Users” on page 98

Autoattendant “Autoattendant Basic Settings” on page 105

DIAGNOSTICS

This section contains information about such things as Logging and Tracing as well as other Diagnostic measures.

Syslog-File “Syslog-File” on page 121

Logging “Logging” on page 122

Debugging Tools “Debugging Tools” on page 123

SNMP “SNMP” on page 124

Firewall Report “Firewall Report” on page 126

Network Trace “Network Trace” on page 126

Support Trace “Support Trace” on page 128

GATEWAY

Part II of this manual describes all settings related to the gateway component of your Mediatrix Boss.

System “System Parameters” on page 209

Network “Network Parameters” on page 223

ISDN “ISDN Parameters” on page 227

R2 CAS “CAS Parameters” on page 257

POTS “POTS Parameters” on page 285

SIP “SIP Parameters” on page 303

Telephony “Telephony Parameters” on page 363

Call Router “Call Router Parameters” on page 451

Management “Management Parameters” on page 511

Table 3: Documentation Sections (Continued)

12 Mediatrix Boss

C H A P T E R

3 SYSTEM

Basic Settings

This menu contains all of the basic settings which are needed for the configuration of the system of your Mediatrix Boss.

This comprises:

date and time (any NTP settings have precedence over the time and date configured here, refer to “NTP” on page 41)

specification of a domain name server (DNS server)

host name for the Mediatrix Boss (the system name should be set for administrative purposes, but it is not a crucial setting)

Figure 7: Basic settings

Time and Date:

Date and time will be applied immediately after pressing the button Save, for all other SYSTEM options Apply configuration has to be selected. The system time and date are important for using for example the logging facilities. The settings done here will also be saved in the hardware clock of the system and thus not be lost with a reboot.

However, the configuration of a reliable NTP server, especially for logging, is recommended (refer to “NTP” on page 41).

Note: SYSTEM NAME

The system name is shown in the command prompt when you access the system via the command line interface where it is very helpful to see the system you are on.

Mediatrix Boss 13

Chapter 3 - SYSTEM Basic Settings

Time zone:

Usually NTP servers utilize GMT (Greenwich Mean Time). Therefore to get the correct time zone on your device, you need to set the time zone where the device is located. There are two types of time zones that are supported by the device. The GMT zones simply define a fixed offset from GMT / UTC time. However, this does not include daylight savings time.

The change between normal and daylight savings time occurs at different times in different parts of the world. These are coded in the regional time zones (e.g. CET for Central European Time). If you select such a time zone the switch to daylight savings time and back is done automatically by the device.

Domain Name Server:

In order to use debugging tools like ping, traceroute and telnet with domain names instead of addresses, and to specify domain names for the remote clients of VPN tunnels (e.g. if the remote client works with a dynamic DNS entry) a DNS server has to be specified.

The same holds true of course for all other services where the specification of host names is possible (e.g. VoIP). If you do not specify a DNS server here, you can only use IP addresses for all services configured on the Mediatrix Boss.

If your internet connection runs with PPPoE and PPTP, a DNS server is automatically assigned by your provider. The value you set here will then often be substituted with the provider's DNS Server. In this case you do not have to set this value.

System name:

The entered name must not contain any whitespace or special characters. This system name should be unique and has to correspond to domain name towards the DNS server that has been assigned to the Mediatrix Boss. Non-matching forward and reverse look ups as well as differing host names will cause problems with various services.

The settings will be applied when you click the button Save.

14 Mediatrix Boss

Change Password Software Configuration Guide

Change Password

With this menu item, the access password for the Mediatrix Boss can be changed. The default password is sesam. It is highly recommended to change it before connecting the appliance to the Internet.

The password has to have at least 5, but not more than 8 characters. Excess characters (more than 8) will be ignored silently. Alphanumeric characters and the following special characters are legal:

+*/=~_@#()<>[]!?.-

Figure 8: Change the password

Especially for security reasons, you now have the possibility to configure two separate administrator profiles with their own passwords simply by checking the box Separate CLI and GUI admin password and clicking Save.

The password configured here will then be limited to accessing the Web-GUI administration only and the password of the command line interface (CLI) profile will become the master password.

The password you enter here should not contain whitespace, since such a password might be mangled by the used browser. If you still want to set such a password, please use the CLI to avoid these browser problems (this is of course only sensible if Web-GUI and CLI administration have not been separated). If you want to use passwords with special characters, please check the Command Line Guide as well for information on how to safely accomplish this.

The new password will be applied immediately after clicking the button Save.

The new password applies for both access over the web interface and over command line interface (serial console, SSH), unless the box Separate CLI and GUI admin password has been checked.

Warning:

• Whitespaces and $ are not supported!

• The new password will be applied immediately. This means that right after changing the password with your browser, you will be prompted for it in order to continue with the configuration.

Mediatrix Boss 15

Chapter 3 - SYSTEM Install Update

Install Update

With this menu item, a local file on your PC can be selected to start the update process. Depending on the network connection, this upload might take several minutes. The update can even fail if done over a low bandwidth WAN link. It is not recommended to perform the update with a connection of less than 128 kbit/s. However, aborted updates do not harm the Mediatrix Boss or influence it in any way.

There is no automatic reboot or restart of services after a successful update; the Mediatrix Boss has to be rebooted manually. However, this can be performed at a later time. The update is simply installed into the flash memory which means that it has no effect on the running system.

Figure 9: Install an update

Warning: Old Configurations

You should not import old configurations into more recent versions of the Mediatrix Boss system. The configuration should be on the Mediatrix Boss prior to the update. During the update process, this configuration will be converted.

When installing an update that contains new configuration options, these must be checked and set after the installation since this is not done automatically.

Warning: Release Notes

Please refer to the Release Notes before performing an update/upgrade. These might contain special instructions to be considered before upgrading to a certain version.

Note: Downgrade

If you would like to execute a downgrade of your system, the steps to be taken are exactly the same as the ones mentioned above. This means that you can select an older version instead of a more recent file. Please note that you will have to reapply the configuration after rebooting the system after a downgrade.

Please note as well that possibly not all services will run properly after a reboot. This can only be ensured by reapplying the configuration. Only configurations compatible with older versions should be used. This can be accomplished by reverting to a saved configuration or by doing a factory reset.

Note: Timeout during Updates

A timeout can occur while uploading an update. The Mediatrix Boss has a fixed timeout of 2 hours. This means that any upload, if not completed, will be aborted after this time. Note that your browser might interrupt the upload sooner than that.

16 Mediatrix Boss

Provisioning Software Configuration Guide

Installing an update for the Integrated Voice Gateway

Since the Voice Gateway Software is not part of the system software, it must be updated separately. The Gateway software update may be packaged like a hotfix, but it is a full software update for the gateway component.

1. Download the Gateway update bundle (GW Software Update) and save it on your PC.

2. Unpack this ZIP archive It will contain the software file and maybe some documentation and release notes.

3. Select the downloaded update file in the Upload Update Web Interface.

4. Click the button Upload and wait for the update to complete.

5. Reboot the system in order for the update to take effect.

Provisioning

The Mediatrix Boss provides provisioning facilities that can be used for mass deployment.

Figure 10: Configuration of the Provisioning

By using provisioning it is possible to create a more or less standardized configuration, including a list of variables that are different for the Mediatrix Boss in concern. Provisioning is always a very project-specific topic where there is no common standard answer. Suffice to say that provisioning facilities are present and can be used.

For further information please contact your reseller or our support team.

Configuration Backup

In this section you can save or restore the configuration of the complete Mediatrix Boss. By clicking the button Save, the most important configuration files will be saved into an archive file configuration-<date>.cpio.gz which will be offered for download.

Restoring configuration

1. Select the location of the backup file stored on your PC

2. Click the button Send to upload it then click on Apply Configuration.

Mediatrix Boss 17

Chapter 3 - SYSTEM Configuration Rollback

Figure 11: Configuration backup

forceconfiguration

If the version of the backup and of the currently installed software are far apart, the Mediatrix Boss will inform you that this might lead to inconsistencies and/or failures and the restore will not take place. However, you have the possibility to force the Mediatrix Boss to except the backup configuration by changing its file name from

configuration-xxxxxx.cpio.gz to

forceconfiguration-xxxxxx.cpio.gz.

Configuration Rollback

The Configuration Rollback enables you to revert the configuration of your Mediatrix Boss to an earlier status. The configuration available for a roll back depends on when you last saved your configuration.

Note: This backup applies to all necessary parameters. Necessary parameters include also data regarding the Integrated Gateway component, e.g. the ISDN configuration etc.

Warning: INCONSISTENCIES

By using "force" the possibility of inconsistencies and failures is not eliminated! This merely forces the Mediatrix Boss to restore the contained data.

Warning: LANGUAGE

Another point worth mentioning here is that the localization of the web interface is also stored in the configuration archive (it is part of its configuration), i. e. if you restore a configuration later, the language might change. If this is the case, simply click on the language button in the top menu bar.

Warning: CONFIGURATION BACKUP IN GENERAL

This mechanism is intended for backing up and restoring configurations on the same device(s). Please do not use this for provisioning purposes by creating a default configuration and then replicating this to multiple devices. Please note that device-specific files which should not be replicated to arbitrary devices, will also be included in this archive. However, this does enable you to make a one-to-one restoration of a failed device. Saved configurations exclusively work on identical restoration devices. Do not use a saved configuration from one device on a different device type unless this procedure has been approved by the manufacturer!

18 Mediatrix Boss

Configuration Rollback Software Configuration Guide

After performing a rollback, you will always have to press Apply Configuration to activate it. This will also provide a chance to look at the restored configuration first, before activating it.

The naming convention of the stored configurations is as follows: DATE-TIME. The date will be formatted as yymmdd (two digits for the year, month and day), followed by a minus and the time in 24-hours format (hhmm). Figure 12 on page 19 shows an example file name of a backup, (080311-1423) done on March 11th 2008 at 14:23.

Number of Configurations

This field defines how many configurations should be stored on the system. You have a choice between '5', '15' or '25' stored configurations. Saving a configuration will fail if there is no space available on the system. The system has been designed to provide enough space for all of the above mentioned settings. However, this can not be guaranteed if additional data has been manually copied to the system in CLI mode.

Previous Configuration

Select the configuration you would like to restore by using the drop-down box. To activate this configuration, please click Apply Configuration in the top menu bar.

Configuration Save now

By Pressing Save you can trigger an immediate backup of the configuration. Note that such a backup can take several seconds. After the backup has completed you can refresh the page and see the backup in the Previous Configuration selection list.

Figure 12: Restore a previous configuration in this menu

Note: Before making bigger changes to your configuration, it is always wise to go to the Rollback page and save the current configuration. This will enable you to roll back to this configuration at a later point in time. Every time you use the Apply Configuration the current configuration will be stored.

Warning: A restored configuration will not automatically be active after a reboot. The correct procedure it to always press Apply Configuration after performing a restore. A reboot is not necessary.

Mediatrix Boss 19

Chapter 3 - SYSTEM Remote Backup

Remote Backup

The Mediatrix Boss also offers a feature for providing an automated configuration backup to a remote system. The backup can be done through FTP or SCP protocols in scheduled intervals. By default this functionality is disabled, i. e. you need to configure it first in order to use it.

Figure 13: Remote Backup Configuration

Specify the How, Where and When of your Remote Backup

1. Active: You need to check this in order to enable the functionality.

2. Type: You can choose between FTP and SCP. Note that you should use secure copy (SCP) whenever possible, because the backed up configuration also includes passwords. However, using SCP needs some manual setup which is described below.

3. Servername: Specifiy the server for your backup by entering either an IP address such as 192.168.1.1 or a hostname for example ftp.foo.bar here. If you are using host names, please make sure that a DNS server has been set on the Basic Settings page.

4. Path: Please specify the path on the server where the files should be stored. This path is usually identical to the default login directory of the specified user. This field needs to reflect the path only relative to the server root directory. This means if you specify no path, the backup will be made to the root directory of the server. If you do specify a path, it has to be an existing path under the root directory the FTP server, such as for example /backup/test. The first slash is optional.

5. Username: Please specify a user name here. Even if the FTP server in question does not require authentication, i. e. is public, you should still specify a random name here e.g. anonymous.

6. Password: Specify the password of the defined user on the server. If this setting is incorrect, the backup will fail. If this occurs you should check your server logs to see if there is an authentication mismatch. This setting does not apply to SCP !

20 Mediatrix Boss

Remote Backup Software Configuration Guide

7. Interval: You have a choice of different backup intervals. The following settings are possible:

• Daily: This will back up the configuration everyday. Set the hour in the Time/Day of Week field. Legal Values are integers between 1 and 24.

• OnChange: This will create a backup after every material change

• Hourly: This will create a backup every hour. There is no need for additional settings.

• Weekly: saves backups every week. Select the day via the Time/Day of Week field. Legal values are integers from 1 to 7; 1 for Monday, 7 for Sunday.

8. Time/Day of Week: The setting in this field always has to be an integer. It only has to be set if the Interval field is set to either Daily or Weekly.

Note: DIRECTORIES

Note that directories have to be specified with / slashes in the Mediatrix Boss web interface.

Note: ARCHIVE FORMAT

The archive format used for the automatic backup and the manual Configuration Backup are identical. In order to restore a backed-up configuration, you should hence use the Configuration Backup page.

See section “Configuration Backup” on page 17 for more information.

Note: SCP CONFIGURATION

To configure SCP, some manual setup is needed. First, you need to log on to your Mediatrix Boss via SSH. If not already done, run

ssh-keygen -t dsa

in the home directory of root to generate an SSH key pair (don’t assign a password to the private key). You will get the files id_dsa.pub (public key) and id_dsa (private key).

Copy the public key to the backup server by running:

scp /etc/root/id_dsa.pub [backupuser]@[backuphost]

You will then be prompted for the password for that account.

Then you need to set up a key-based login to your backup account on the backup host. Logon to the backup server: ssh backupuser@backupserver and create a directory /root/.ssh. Add the public key to the end of an "authorized_keys" file in the recently created .ssh directory:

cat /id_dsa.pub » /.ssh/authorized_keys.

If the file does not exist, it will be created.

The private key on the Mediatrix Boss (/etc/root/.ssh/id_dsa) should remain there and will be used by the system for an authorized login to the backup server. Once this is done, SCP is available for remote backups.

Mediatrix Boss 21

Chapter 3 - SYSTEM Restart System

Restart System

By clicking the button Restart, the Mediatrix Boss will be rebooted.

Figure 14: Restarting the System

It may take up to two minutes until the Mediatrix Boss is fully available and operating again. Depending on the configured WAN type and the number of configured VPN tunnels, it can take even longer until all tunnels are available again.

Generally there are two options: the Mediatrix Boss can be restarted instantly or, with the second option, the Mediatrix Boss can postpone its restart until there are no ongoing calls.

22 Mediatrix Boss

Port Numbers Software Configuration Guide

Port Numbers

On this page it is possible to define the HTTP(S) and the SSH port used for administration. The Default values are 22 for SSH, 443 for HTTPS (or port 80 for HTTP on systems that do not use HTTPS), as recommended by the IANA (Internet Assigned Numbers Authority). However, this menu gives you the possibility to set a different port.

Figure 15: Port Numbers

Note: Usually you do not need to change these values. Changing them is mostly done for security purposes if the administration through the WAN interface is enabled. Using non-standard ports makes it harder for scan programs to find these open ports.

Mediatrix Boss 23

http://www.iana.org/assignments/port-numbers

Chapter 3 - SYSTEM Restore Factory Defaults

Restore Factory Defaults

The 'Restore the basic configurations' page allows you to restore the basic configuration and / or reset the 'Dualmode' database (only FMC products) to factory settings. Refer to Figure 16 on page 24 for more information.

Restore Basic Configuration

With the exception of the 'Dualmode database' settings, all configured settings will be deleted from the Mediatrix Boss by restoring the basic configuration. To execute a restore click the first Reset button on the right side of the page.

Figure 16: Reset basic configurations

Licenses

This product contains Open Source Software. Most Open Source Software demands that the license of the package is included with the product containing the software. Therefore, this page lists all licenses of Open Source packages used in this Mediatrix Boss. A mouse click on the package name shows the complete license text.

Warning: REBOOT

Your system has to be rebooted after a restore / reset is done! Since the default IP address is 10.0.0.205 and subnet mask is 255.255.255.0 you will have to enter this information once more after a restore /reset!

24 Mediatrix Boss

C H A P T E R

4 NETWORK

WAN Interface

The internet (WAN) connection can be configured with the menu item WAN interface. The following parameters can be set up (compare Figure 17).

Figure 17: Configuration of the WAN interface

Connection via IP

If the Mediatrix Boss is connected to the internet via a leased line or an internet router (xDSL router, ISDN router, etc.), you have to choose IP and enter the information for your network.

For the field Gateway you should enter the address of your router (default gateway) for the WAN connection.

Connection via PPPoE / PPTP

If the Mediatrix Boss is directly connected to the internet via a DSL modem, you will have to select PPPoE and PPTP (depending on your provider) and enter the connection information given to you by your provider. Please also refer to your internet service provider (ISP) on applicable additional configurations necessary to set with DSL routers. Note that some providers do not want you to connect a router to connect multiple PCs. They will often insist you install special software on the PCs in order to access their network. If you experience problems with the WAN configuration, please make sure your ISP does not have such restrictions.

If the connection uses PPPoE, it's only the user name and password that you have to enter. Additional information is needed if the connection uses PPTP:

Mediatrix Boss 25

Chapter 4 - NETWORK LAN Interface 1

The internal IP address of your DSL modem, the related netmask (both should be known by your provider) and a matching IP address for your Mediatrix Boss are required. The IP of the Mediatrix Boss has to be in the same network as the IP address of the modem. Please ask your provider or consult your DSL modem/router manual about the IP settings of your modem.

A common IP address is 10.0.0.138.

Configuring WAN Interface

1. Select the connection type: IP, PPPoE, PPTP or choose Deactivated if you do not want to use WAN interface.

2. Provide configuration details for the selected section as described above.

3. Save changes by clicking on Save. Your changes will be applied after selecting Apply configuration.

LAN Interface 1

This menu item is used for configuring the primary LAN interface. You can choose to have a dynamic IP address assigned to it by a DHCP server in your network, or you may give it a Static IP address. The configuration templates available for LAN Interface 1 depend on the Intended Use that you select here. Do not forget to save your changes after configuration has been completed.

Figure 18: Configuration of the primary LAN interface

Dynamic IP address

Select this if you have a DHCP server in your LAN network that the Mediatrix Boss is to obtain its IP address from. This only applies to bigger networks where the Mediatrix Boss is dedicated to a certain functionality only (e.g. VoIP).

If the Mediatrix Boss's IP address is set to dynamic, you only have the option to activate or deactivate NAT handling under the Basic Settings. Further configuration options become available if the product has a static IP address. For example, with a static IP, the Mediatrix Boss itself can be used as a DHCP server. Please refer to section “Static IP address” on page 27 for the details.

26 Mediatrix Boss

LAN Interface 1 Software Configuration Guide

The NAT option should be chosen if address translation is required for the whole network traffic from this LAN to the Mediatrix Boss's WAN address. This only applies to outgoing traffic from the LAN to the WAN.

Generally this is needed, if your provider does not render official IP addresses for your LAN (the usual case) and your Mediatrix Boss is used as an internet router.

Static IP address

If Intended Use is set to Static IP address, further options become available. For once, the IP address itself has to be specified under Basic Settings. Furthermore, the product can be used as a DHCP server.

Figure 19: Static IP for primary LAN Interface

Basic Settings

IP Address and Netmask

Specify the Mediatrix Boss's IP address and Netmask.

NAT

As mentioned for the dynamic IP scenario, if NAT handling is required for your network, check this box.

Warning: Since the Mediatrix Boss is a server for certain protocols such as SIP, the IP address of the device must be known to the user. Assigning a truly dynamic IP address hence is not at all recommendable. DHCP, however, is frequently used as a tool for the central administration of IP addresses. You can have the concerned DHCP server assign a fixed IP address to a device. This approach is controlled by the MAC address that can be configured in the DHCP server.

Mediatrix Boss 27

Chapter 4 - NETWORK LAN Interface 1

DHCP server

If the primary LAN Interface has a static IP address the Mediatrix Boss is able to automatically assign IP addresses to computers connected to it. In the following we will explain how to set this up.

Configuration

1. DHCP IP Range from and to

If the Mediatrix Boss is intended to be used as DHCP server you have to define the address range the DHCP server will assign to clients requesting an IP address. This address range must not contain IP addresses of any other configured computer with a static IP address within this LAN.

Both addresses have to be in the same subnet as the Mediatrix Boss's IP address. The beginning and the end of the range of course also have to be in the same subnet.

2. Domain Name

Give a Domain name to the Mediatrix Boss's DHCP server functionality. This name will serve for the configured IP range as a whole.

3. Standard Gateway IP

By default, the Mediatrix Boss itself acts as a standard gateway. Another standard gateway for the DHCP functionality can be set by checking this box and assigning an IP address below.

4. Primary and Secondary DNS

You may specify up to two DNS server IPs to obtain domain names. Of course configuring two is safer than relying on a single one.

5. First, Second and Third NTP Server

Configure up to three Network Time Protocol servers here by simply entering their IP addresses.

Again configuring two is more reliable than only one, and three is even better. These addresses will be transmitted and assigned to the computers that are using DHCP.

Section “NTP” on page 41 about the general NTP server settings contains some more information about NTP and the product.

6. DHCP Fixed Addresses

This option lets you assign fixed IP addresses to certain network devices. Choose an appropriate Hostname for the device, configure it to the MAC address of said device, and choose the IP address (which has to be within the configured DHCP IP Range) for it.

Warning: If you would like to make use of this functionality, please make sure that there is no other DHCP server running in your network since this can have very adverse effects.

Warning: STANDARD GATEWAY IP

You cannot specify the default gateway for the system here. In order to specify a default gateway for the Mediatrix Boss as a whole, please configure it with the WAN interface. If you do not want to configure a WAN interface, but still have a default gateway, please configure a routing entry. Please see “Routing” on page 35 for more information.

28 Mediatrix Boss

LAN Interface 2 Software Configuration Guide

LAN Interface 2

There are several options for the secondary LAN interface:

Figure 20: Configuration of the secondary LAN interface

Deactivated

The secondary LAN interface will not be used.

Internal Network

If this is selected, the secondary LAN interface will be treated exactly the same way as the primary one.

The possibilities of configuration are therefore the same as in section “LAN Interface 1” on page 26.

Mediatrix Boss 29

Chapter 4 - NETWORK DMZ Interface

DMZ Interface

The DMZ interface can be used for the implementation of a so called demilitarized zone (DMZ). This term describes a secure network area, which is separate from the internet and the local network. Normally, this is used for servers that have to be reachable from both the internal network and the internet (e.g. mail server, web server, etc.). A DMZ will usually have official IP addresses.

By default the DMZ interface is deactivated. If you choose to activate it, just select it. However, to actually use the DMZ, you have to configure an IP address and a netmask. There are no more options left. Typically you will want to restrict access to the DMZ. Refer to section “Firewall” on page 56 for more information.

Configuring DMZ Interface

1. Select DMZ in order to activate DMZ Interface.

2. Provide an IP address and netmask for this DMZ.

3. Save changes by clicking on Save. Your changes will be applied after selecting Apply Configuration.

Figure 21: DMZ interface

Note: Port Forwarding rather than a DMZ is needed for providing services to the internet while having servers with private IP addresses. Please refer to section “Port Forwarding” on page 58 for more information about Port Forwarding.

30 Mediatrix Boss

Virtual Interfaces Software Configuration Guide

Virtual Interfaces

This menu serves for the configuration of virtual interfaces. Every virtual interface needs its separate number, IP address and netmask.

Figure 22: Define virtual interfaces

Configure a Virtual Interface

1. Local Interface: Choose to which interface of the Mediatrix Boss you would like to add another virtual interface. Further Virtual Interfaces can be assigned to all configured physical interfaces.

2. Number: This will be the ID of your virtual interface. It is arbitrary. If 802.1q is activated, this Number will be your VLAN ID. As mentioned before if you configure several virtual interfaces, each of them needs a unique ID or Number.

3. IP Address and Subnetmask: The IP address of your virtual interface is also arbitrary.

4. 802.1Q: The standard IEEE 802.1q is used to tag control information to the IP packet headers. In our case the VLAN ID, i.e. the configured Number will be displayed there. It will then be clear for all network devices that they are dealing with a virtual local area network and not with a physical one. Enabling this does not do any harm, even if your PBX does not support VLANs.

Warning: Ports for virtual interfaces have to be opened manually on the firewall! Use Custom Rules (refer to section “Custom Rules” on page 57) to open ports.

Mediatrix Boss 31

Chapter 4 - NETWORK Dynamic DNS

Dynamic DNS

With the help of this menu item, it is possible to make the Mediatrix Boss accessible from the internet even if a dynamic IP address is assigned by your provider.

The principle is very simple: if a DynDNS provider is set up, the Mediatrix Boss will post its current IP address each time it is assigned a new one. This is the only possibility to set up a VPN tunnel between two VPN routers without static IP addresses.

In order to use a dynamic DNS service, you have to register a name with such a service (most DynDNS providers offer this basic service free of charge). This name and the user name and password must be specified here together with the provider of the service. User name and password are essential for the before mentioned IP address updates.

Figure 23: Dynamic DNS

Table 4: Currently supported providers for dynamic DNS service. This list can vary depending on the software release!

Provider Register at

DHS http://www.dhs.org

DynDNS http://www.dyndns.org

DyNS http://www.dyns.cx

easyDNS http://www.easydns.com

EZ.net http://www.ez-ip.net

HN.org http://www.hn.org

JustLinux http://www.justlinix.com

ODS http://www.ods.org

TZO.com http://www.tzo.com

ZoneEdit http://www.zoneedit.com

32 Mediatrix Boss

http://www.dhs.org

http://www.dyndns.org

http://www.dyns.cx

http://www.easydns.com

http://www.ez-ip.net

http://www.hn.org

http://www.justlinix.com

http://www.ods.org

http://www.tzo.com

http://www.zoneedit.com

Name Server Software Configuration Guide

Configuring Dynamic DNS

1. Select the DynDNS Supplier.

2. Enter your Username and Password, and Hostname to be used.

3. Save changes by clicking on Save. Your changes will be applied after selecting Apply Configuration.

Name Server

The Mediatrix Boss can run a name server to forward and resolve DNS queries. On this page you can manage the list of forwarders to be used. As a DNS the Mediatrix Boss will act as a caching server, i. e. it will not forward each individual query. It will, however, query once for each domain name (requeries will be forwarded according to the expiry times specified in the DNS names).

Therefore, this functionality can also be used for performance improvement with applications doing a lot of name server lookups. A simple configuration would be to enter the DNS server assigned by your Internet provider. The PCs would then no longer contact this server for DNS, but the Mediatrix Boss which would then act as a caching forwarder.

Instead of configuring a forwarder for all requests, the DNS server can also be run as a slave server or as a forwarder for certain zones. As slave server, it would not only act as a caching proxy to its assigned zones, but as an authoritative name server.

Configuring a Name Server

1. Select a profile: there is Deactivated, Custom and Simple Configuration. The Custom profile allows you to apply an existing configuration via the command line (refer to “Network related Topics” on page 140). Use the Simple Configuration to use this service via the web interface. Choosing Deactivated will of course disable this service.

2. Provide the IP addresses of the Forwarders to be used. This is an optional feature, therefore, the DNS feature does not depend on it. A forwarder is a server in a domain name system (DNS) which forwards queries for external DNS names to external DNS servers.

3. Add Zones and Forwarded Zones that should be managed by the Mediatrix Boss. This is also optional. Specify the zone and its master (IP or domain name) and the name of the zone file as it is provided by the master. Please note that the Mediatrix Boss acts as a slave server and that you have to provide exact names of the zone files as they are set on the master server. These options are usually not required for standard scenarios and should only be used by administrators familiar with the DNS protocol.

4. Save your changes by clicking on Save; they will be applied after selecting Apply configuration.

Mediatrix Boss 33

Chapter 4 - NETWORK Name Server

Figure 24: Name Server Configuration

34 Mediatrix Boss

Routing Software Configuration Guide

Routing

With this menu item you can configure additional routing entries. Standard routes like the default route (only on the WAN interface) and routing between the different configured interfaces are generated automatically. Hence, this menu only has to be used if any additional routes are desired. Note that you only set the routes for outgoing IP packets here. In order to make the network behind the Mediatrix Boss accessible from your network, you have to make sure that the network towards the Mediatrix Boss is handled correctly.

Use to edit, or to delete an existing entry.

New routes are created by entering the matching values in the last row and can be applied by clicking the button New.

Figure 25: Routing

Adding a new routing entry

1. Click Add to create a new entry and specify configuration parameters as follows.

2. Destination: The destination network for the desired route. This can also be a single host.

3. Netmask: The netmask of the destination network. Note that 255.255.255.255 will signify a host route.

4. Gateway: The address of the gateway router that connects the destination network with the local network. This address must be accessible via a local interface. The gateway must hence represent the next hop that can be used for routing.

5. Interface: The interface to which the gateway is connected (directly or indirectly via one or more switches).

Mediatrix Boss 35

Chapter 4 - NETWORK Bandwidth Management

6. Click Save to save the changes. Your changes will be applied after selecting Apply Configuration. Compare Figure 25.

Bandwidth Management

QoS and Bandwidth management makes it possible for you to fully control your network traffic. For this purpose, the allocation of bandwidth for different services and computers in the network can be controlled with the available bandwidth management features. This functionality is also used in conjunction with other available features, for example Call Admission Control, which will be described later.

The following example will illustrate the concept:

Normally precedence should be given to realtime applications like Voice-over-IP or Video Conferencing, because maintaining a constant and low-jitter bandwidth is very critical for the quality of these services. At the same time, other internet traffic (browsing web pages etc.) should not be shut down completely by too many concurrent real-time sessions. It must of course by no means interfere with the performance of the prioritized service(s). Traffic Shaping and QoS is a feature for advanced users and administrators, because a misconfiguration here can easily prevent applications from functioning properly. A lot of information about QoS and Traffic Shaping is available in the internet.

An in-depth explanation of the concepts would be beyond the scope of this document.

The QoS configuration of the Mediatrix Boss happens in so called classes in hierarchical order. This approach is called hierarchical token bucket packet scheduling (HTB).

The classes' names are composed as follows:

<network interface>-<parentclass ids>:<class id>

<parentclass ids> ::= (:<class id>)*

The class id has to be unique for all classes!

As long as the interface is configured, there will always be the sections for network interface configuration. It can therefore only be changed and never deleted.

Note: DEFAULT ROUTE

Please be careful when setting a default route here. Usually, the default route is set automatically when the WAN interface is configured. In that case, the default route points to the gateway specified in the WAN interface configuration. If an ADSL connection is established, the default gateway is usually assigned via DHCP. If no WAN interface is configured, you should specify a default gateway here. Specifying a default gateway works by using destination 0.0.0.0 with netmask 0.0.0.0

Note: RESTART

Please restart the system after editing/adding routing entries.

36 Mediatrix Boss

Bandwidth Management Software Configuration Guide

Figure 26: Bandwidth management overview

The only parameter of a network interface is the default class, which the last number of its class ID. It will contain all of the data traffic not matched by any other class. Often this class only has a small amount of bandwidth for itself since everything else is used by its children, but it can exploit excess bandwidth.

Starting from any class in the list, you can arbitrarily create subclasses with the button on the right hand side of the respective class. Now press the following button on the right hand side of the respective class to get to the following mask.

Figure 27: Add new class

The table for entering the configuration data contains the following fields:

Mediatrix Boss 37

Chapter 4 - NETWORK Bandwidth Management

Parent class

Shows the name of the parent class for this class. The traffic limit of a child must be less than that of the parent. If a parent has multiple children, the sum of the bandwidths used by the children should not exceed that of the parent class.

Description

You can enter any description here to make the setup easier to understand and maintain.

Rate

The rate specifies the bandwidth that will be provided for this class. All traffic, that passes through this class will be shaped so that it does not exceed this configured bandwidth limit. The unit is by default "kilobit per second" but you can also enter Kbit, Mbit, bps, Kbps, Mbps. Note: bps stands for "bytes per second".

Ceiling

This field defines the maximum amount of bandwidth this class can use in total. The difference between rate and ceiling is the bandwidth that this class is allowed to "borrow" if there is unused bandwidth available in the parent class. If this field is left blank, this class will never borrow any bandwidth. If more than one class is demanding unused bandwidth, every class gets allotted a contingent that is proportional to their own rate. In root classes, defining a ceiling is of no use, since they do not have parent class to borrow from.

Burst and Ceiling Burst

These parameters define how much data is allowed to be sent with maximum (hardware) speed, before trying to serve data from a different class. This parameter will have a significant influence on the jitter behavior: The larger this value, the more jitter can possibly be induced.

If you set Ceiling Burst to a small value (e.g. the size of a packet), it can be used to cap bursts effectively (? a high data rate at the beginning of a connection). This way the data rate will not exceed the value for the Ceiling, not even for a very short period of time. However, small values can lead to packet fragmentation or long delays for very big packets.

Priority

This field is used to set the priority of the traffic for this class in relation to the classes on the same level (sister classes). The higher the number, the lower the priority. Furthermore, classes with higher bandwidth rates are preferred in the assignment of available bandwidth.

Leaf Queuing Discipline

The Leaf Queuing Discipline defines the queuing behavior for a class. Possible values are:

None, SFQ, Packet FIFO, Byte FIFO.

If you choose None, all inbound packets of this class are forwarded in order of their arrival.

SQF is a simple, efficiently computed, probabilistic algorithm. It is a useful method if you want to have one class' traffic distributed equally over several different hosts and/or services in a fair fashion. No single host will be able to use all bandwidth provided for a certain service then. With the parameter "Quantum", you can specify how many bytes are allowed to be taken from the queue (dequeued) until it is the next host's turn. By default, this is the value of the Maximum Transmission Unit (MTU). MTU defines the maximum size a package is allowed to have without being fragmented.

The parameter "Perturbation" defines the interval (in seconds) in which SQF renews its hash function for queuing incoming packets randomly. The default value is 10.

Warning: Never set a value smaller than MTU, since it would prevent larger packets from being forwarded in one piece!

38 Mediatrix Boss

Bandwidth Management Software Configuration Guide

Both FIFO queues have the parameter "Limit", which indicates the length of the queue. With Byte FIFO you can specify the value using the unit "'bytes"' and with Packet FIFO the number of packets.

MTU

The maximum size of a packet, used for transfer rate calculation. By default a value of 1500 is used, which is appropriate for Ethernet based networks (default MTU value).

Filter

These rules define how packets are classified. If the conditions defined here match, the packet will be assumed to be in the respective class. The algorithm used here is first match. A packet will be assigned to the first class for which the conditions match.

You can for example check the source and destination information of the packets. The format of such an entry is:

[IP address[/subnet prefix]][:port[/port mask]]

Legal values for instance are:

10.1.1.0/24:80 all packets of the network 10.1.1.0 with port 80

:50 all packets with port 50

Compare Figure 28 on page 39.

Figure 28: Add new Filter

An additional possibility to classify packets is a mark that can be assigned to a packet by the firewall. However, this is a very powerful mechanism that can be very complex to use. With this mechanism, it is possible to base classification on for example DSCP tag and the like, or the layer 7 protocol. This can be useful for limiting FTP connections, because they use dynamic ports and can use arbitrary IP addresses. It is hence possible to limit the traffic used by FTP, or any other protocol, to a certain value.

Mediatrix Boss 39

Chapter 4 - NETWORK CAC Bandwidth Sampling

CAC Bandwidth Sampling

The bandwidth daemon monitors the network traffic in order to determine the amount of free bandwidth. This allows for dynamic allocation of bandwidth for different services. Please see section “Bandwidth Management” on page 36 for further information on bandwidth management. Refer to section “QoS” on page 44.

In this section you can specify three parameters that affect the bandwidth daemon. These are sample rate, number of samples and log-level. However, please note that in most cases the bandwidth deamon's default values do not need to be changed.

Sample rate

This specifies the frequency of checks for available bandwidth in tenths of seconds. The default value for this setting is 1, i. e. the daemon performs checks every 0.1 seconds.

Number of samples

This is the number of samples to be taken in order to actually calculate the current average bandwidth. 30 samples is the default. The number of samples together with the default sample rate value give a time window of 3 seconds.

Log level

You can view log messages created by the bandwidth daemon in the system logfiles by setting the log level to error, warning, info, or debug. Choosing none disables logging the bandwidth daemon's messages.

Logging configuration for Mediatrix Boss is described in section “Logging” on page 122.

Finally, save your changes by clicking Save. Your changes will be applied as soon as you apply the current configuration.

Figure 29: CAC Bandwidth Sampling

40 Mediatrix Boss

NTP Software Configuration Guide

NTP

The table on this site contains one or more NTP servers that the Mediatrix Boss should consult in order to set the system time.

Figure 30: Configuration of NTP Server

To add a new entry to the list simply fill in the name or IP address of an NTP server and click Add. An NTP server can also be used with preference by checking preferred. This means that an answer received from such server won't be dropped if its answers differ significantly from others. Please note that only very reliable and stable NTP servers should be given a preference.

Setting up NTP

1. Checkbox: mark this box to activate NTP and click on Save.

2. Add one or more NTP servers which the Mediatrix Boss should use. You have to provide its IP address. Optionally you can check the box "Preferred" to grant priority to time information received from this NTP server. Use to edit, or to delete an existing entry.

3. Your changes will be applied after selecting Apply Configuration.

Warning: WINDOWS SERVER 2003

The Mediatrix Boss uses full NTP4 which can cause problems when trying to use the NTP service running on Windows Server 2003 (w32time) as a clock source. The w32time service has a few known issues which can lead to incorrect synchronization with the NTP server of the Mediatrix Boss while SNTP clients can often successfully get their time from the Windows Server 2003. The problem is that SNTP as used in some SIP phones and Windows clients is a stripped-down version of NTP which is less reliable and less accurate. However, this makes it seem like the Windows Server 2003 is running correctly while it is in fact not set up correctly to serve as a reliable NTP clock source.

It is essential that the Windows Server 2003 itself is correctly synchronized with a reliable clock source. The internal system clock is not sufficient. For more information about configuring w32time see the MicroSoft knowledge base entry 816042.

Mediatrix Boss 41

Chapter 4 - NETWORK VRRP

VRRP

This menu gives you the possibility to configure two Mediatrix Boss units for a failover scenario and possibly other individual VRRP scenarios (this depends on your specific implementation).

Figure 31: VRRP Configuration

VRRP Configuration

1. Go to General to activate the feature.

2. Click the Add -button to add a virtual server. Use to edit, or to delete an existing entry.

3. Specify the Interface that this VRRP virtual address should be assigned to. Currently, the following is supported:

• VRRP on local interface LAN 1

• VRRP on local interface LAN 1 and WAN Interface

4. Set the VSID for the virtual service identification.

Background: VRRP allows you to define groups of virtual servers / routers that can provide redundancy for each other. This must be a number, and this number must be unique for the group of routers you want to provide redundancy for.

5. Specify the Priority field. specifies the sending VRRP router's priority for the virtual router. Higher values equal higher priority. The priority must be between 0 and 255. For the master-Mediatrix Boss the value "253" is usually a reasonable choice.

Background: The priority value for the virtual router that owns the IP address(es) (IP Address field in this menu) is always "255" (decimal). VRRP routers backing up a virtual router MUST use priority values between 1-254 (decimal). The default priority value for VRRP routers backing up a virtual router is 100 (decimal).

The priority value zero (0) has special meaning indicating that the current master has stopped participating in VRRP. This is used to trigger backup routers to quickly execute a transition to master without having to wait for the current master to timeout.

6. Enter the IP address for your virtual router. Please note that you have to set this exact IP address on all devices with the same VSID in the same network segment.

42 Mediatrix Boss

VRRP Software Configuration Guide

7. Click Save to save your changes. Your changes will be applied after selecting Apply Configuration.

Additional Background Information

Virtual Router Redundancy Protocol as described in RFC 2338 is used to advertise a "virtual" network device. This virtual network device represents a group of other network devices. This group consists of one acting master device and at least one or more slave devices.

If the currently active Mediatrix Boss fails, an arrangement is made for the other Mediatrix Boss to automatically replace it. The Mediatrix Boss currently forwarding data on behalf of the virtual one is called master.

VRRP is primarily intended to provide redundancy between two Mediatrix Boss s, but it can also be used in conjunction with any router providing this functionality such as Cisco, Huawei, Juniper and others.

VRRP specifies an election protocol to provide the virtual router function described before. All protocol messaging is performed using IP-multicast datagrams, thus the protocol can operate over a variety of multi-access LAN technologies supporting IP multicast. Each VRRP virtual router has a single well-known MAC address allocated to it. This document currently only details the mapping to networks using the IEEE 802 48-bit MAC address. The virtual router MAC address is used as the source in all periodic VRRP messages sent by the master router to enable bridge learning in an extended LAN.

A virtual router is defined by its virtual router identifier (VSID) and a set of IP addresses. It may associate a virtual router with its real addresses on an interface, and may also be configured with additional virtual router mappings and priority for virtual routers it is willing to backup. The mapping between VSID and IP addresses must be coordinated among all VRRP routers on a LAN.

To minimize network traffic, only the master of each virtual router sends periodic VRRP- advertisement messages. A backup router will not attempt to pre-empt the master unless it has higher priority. This eliminates service disruption unless a more preferred path becomes available. It's also possible to administratively prohibit all preemption attempts. The only exception is that a VRRP router always becomes master of any virtual router associated with addresses belonging to it. If the master becomes unavailable the highest priority backup router will transform into the master after a short delay, providing a controlled transition of the virtual router responsibility with minimal service interruption.

The VRRP protocol design provides rapid transition from backup to master to minimize service interruption, and incorporates optimizations reducing protocol complexity while guaranteeing controlled master transition for typical operational scenarios. The optimizations result in an election protocol with minimal runtime state requirements, minimal active protocol states, and a single message type and sender. The expected duration of master election (from the pool of backup routers) if a failure occurs is quite small ( less than one second ).

Warning: SWITCHBACK

When the master-Mediatrix Boss becomes active again after a failure, it tries to get the call status of the slave-Mediatrix Boss. This is done every 60 seconds. If there is no active call, the switchback will be done. If the slave-Mediatrix Boss is not reachable (via SSH over LAN Interface 1) the master-Mediatrix Boss will become active immediately.

Warning: MASTER

Please note that whenever the master in a VRRP group is back online, it will immediately assume the virtual IP address configured here. This may have undesired effects, because any open session (e.g. SSH) to the currently active node will immediately break at that very moment.

Note: BEHAVIOR WHEN APPLYING THE CONFIGURATION

After "Apply Configuration" the master-Mediatrix Boss goes down (due to the restart of all services) and becomes active again if the slave-Mediatrix Boss has no active calls running. An "Apply Configuration" on the slave-Mediatrix Boss will not affect the operation on the primary server.

Mediatrix Boss 43

http://www.ietf.org/rfc/rfc2338.txt?number=2338

Chapter 4 - NETWORK QoS

QoS

This page allows you to configure layer-3 QoS tagging by using DSCP/ToS byte in the IP header. Simply configure the correct interfaces here. Note that this tagging only makes sense for outbound traffic.

There are three different ways to specify the traffic you want to mark:

Interface

IP and Application Protocol

TCP / UDP Ports

Quality of Service Configuration

There are several configuration options for each criteria:

1. Protocol: Select a Protocol, either UDP, TCP, ICMP or ANY. ANY will match all packets.

2. Received on Interface: This will match packets arriving on a specified interface. This option only applies if the traffic is simply forwarded and the Mediatrix Boss hence acts as a router. This rule or traffic terminated on the Mediatrix Boss.

3. Sent over Interface: Will match all packets sent out over the given interface. It does not matter if the packets were forwarded or generated by the Mediatrix Boss.

4. Destination Port: Matches all packets addressed to the given destination port.

5. L7 Protocol: Stands for Layer-7 Protocol and will actually analyze the payload of each packet to match a certain protocol. Only a subset of the supported protocols is given as a choice here, namely ANY, SIP, RTP, TLS. More protocols are available via the CLI.

Traffic tagging methods:

DSCP: Stands for Differentiated Services Code Point and is specified in the RFCs 2474 and 2475. It signals the priority of a package with 6 bits, where The DSCP or DiffServ standard supersedes the original specification for defining packet priority described in RFC 791.

TOS: Type of Service as described in RFC 791

Both set a QoS tag into the same byte field in the IP header. If this byte field is interpreted as DSCP or TOS, depends on network infrastructure settings. A value can be entered in decimal (e.g. "38") or hexadecimal (e.g. "0x26").

The following configuration example tags RTP traffic (with DSCP tag 46) and SIP traffic (all traffic on port 5060 to be precise; with DSCP tag 48).

Figure 32: Quality of Service Configuration

Warning: The use of Layer-7 Filtering is discouraged for Mediatrix Boss units needing to handle high data rates (in excess of 5 MBit constant load), because analyzing each packet can become quite resource consuming.

Note: It is only possible to specify one tagging method (i.e. DSCP or TOS) for any given tagging rule.

44 Mediatrix Boss




C H A P T E R

5 VPN

In this tab of the webGUI you have the possibility to configure VPN connections, i. e. secure connections through the WAN to another network, via IPsec or OpenVPN. If you do not need this kind of connection, you do not have to configure anything here. Please find the descriptions of the menu items below in their logical order for configuration.

Certificates

Before establishing a VPN connection using certificates you have to have a X.509 certificate in place. The menu item Certificates is used for the administration of these certificates which can be used for both IPsec connections and OpenVPN connections.

You have to differentiate between plain certificates, containing a public key (signed by a certification authority or self-signed), and complete certificate-key pairs, containing a public key and the matching private key.

Preparations

Before using certificates for VPN tunnels, there are several steps to be taken:

1. Create a self-signed or officially signed (i.e. signed by an official certification authority) X.509 certificate-key pair.

2. Distribute the certificate (not the key!) to all VPN remote stations.

3. Create an X.509 certificate-key pair for every VPN tunnel terminal station and sign them using the CA-key.

4. Distribution of the certificate-key pairs to all VPN terminal stations.

For setting up a VPN tunnel between two Mediatrix Boss units proceed as follows:

1. Login via SSH to one of the two Mediatrix Boss units (let's call it Box A and the other one Box B) (refer to the Command Line chapter “Access Methods” on page 130 for more information).

2. Execute the script /etc/ssl/mkCAcert.sh to create a self-signed CA cerificate-key pair. The certificate and the key are automatically saved in the appropriate location.

Example:

/etc/ssl/mkCAcert.sh "RootCA company" "/CN=Root-CA/O=company/C=DE"

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Chapter 5 - VPN Certificates

Figure 33: Certificate management

3. Download the CA certificate onto the local PC.

4. Login to the web interface of Box A and select the menu item Certificates. The CA certificate that has just been created is shown in the list of installed CA certificates. Furthermore, a form is shown below in which a new X.509 certificate can be created.

5. In this form, a certificate-key pair for each Mediatrix Boss (Box A and Box B) has to be created. Thereby at least Common name, Company/Organisation and Country must be entered. Click on Create certificate.

6. Download the certificate-key pair for Box B onto the local PC.

7. Login to the web interface of Box B and select the menu item Certificates.

8. Upload the CA certificate previously saved on your local PC (see step 3).

9. Upload the X.509 certificate-key pair saved on your local PC (see step 6).

After you have finished all these tasks, the certificates needed for the connection setup are available on both Mediatrix Boss units.

On the menu page Certificates you will find the following forms:

Installed CA certificates

Installed X.509 certificates (i.e. certificate-key pairs)

Create X.509 certificate (only if a private CA key is present on the Mediatrix Boss, see step 2 above)

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Managing CA certificates

CA certificates (i.e. root certificates) are needed to create and to verify X.509 certificates. To ensure proper authentication, the CA certificates stored on all VPN partner hosts must be the same.

In this form, CA certificates can be imported, exported or deleted. For the creation of a CA certificate the script mkCAcert.sh is provided on the Mediatrix Boss. It is executed via the CLI. This script creates two files – a private CA key and a public certificate. You can of course also use third party tools to create these certificates, or use existing ones.

If the certificate was directly created on the Mediatrix Boss, it can be exported by clicking on on the right side of the listed certificate. After that, it can be distributed to all other VPN partners.

All these partners can – if they are also Mediatrix Boss appliances – import the CA certificate via this menu of the web interface.

Managing X.509 certificates

X.509 certificates are used by the Mediatrix Boss when performing the authentication of VPN connections. The X.509 certificates of all VPN partners must be signed with the same CA key to ensure proper authentication.

X.509 certificates can be imported, exported and deleted in this part of the menu. X.509 certificates can be created within this menu page if a CA key is available on the Mediatrix Boss (see next subsection).

Creating X.509 certificates

This part of the menu is only available if there is a complete CA certificate-key pair present on the Mediatrix Boss (i. e. it has been created with mkCAcert.sh). Normally this needs to be done only once within an organization.

For the creation of the X.509 certificate at least Common name, Company/Organisation and Country must be entered.

The Common name should name the object (e.g. computer, network, campus) to be protected by the Mediatrix Boss and for which the certificate is being created.

After creating the certificate, it is available in the list of the installed certificates. It can now be used for the configuration of VPN connections.

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Chapter 5 - VPN OpenVPN Shared Secrets

OpenVPN Shared Secrets

OpenVPN shared secrets are simple Shared Secrets protecting an OpenVPN connection without the need of using certificates. This is a very simple, secure, and effective way for a small organization to set up a few VPN tunnels. It is necessary to create a Shared Secret before setting up the first VPN connection.

An OpenVPN Shared Secret file is created on the first Mediatrix Boss involved. Thereafter it is being copied to the local PC via the web browser from where it is being uploaded to a second Mediatrix Boss involved. You should use separate shared keys for each VPN connection to improve security.

All these steps are taken in this menu:

Figure 34: OpenVPN Shared Secret Files

The list Current shared secret files contains all locally available OpenVPN Shared Secret files. They can be exported by clicking on or deleted by clicking on .

Adding a shared secret file

1. Provide a user-defined name and click on Create new file. The new OpenVPN Shared Secret file is created with a random encryption key.

2. Alternatively, click on Browse and select an existing OpenVPN Shared Secret file on the local PC (previously created by a different Mediatrix Boss). Click Load file to upload the file.

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Connections Software Configuration Guide

Connections

Current Connections

In this menu the connection information of all configured VPN tunnels to remote hosts (other Mediatrix Boss units or third party products) is managed.

The list shown on this page contains all VPN tunnels configured so far.

Figure 35: Managing VPN connections

The active status LED indicates whether the connection is active or inactive. Connections can be deactivated and activated by clicking on the status LED.

If the connection is inactive the LED changes accordingly: Use to edit, or to delete an existing entry.

Setting up a new IPsec connection

1. Click on New IPsec Connection and provide configuration details as follows.

2. Connection name: this will be the custom name of the new connection.

3. Partner IP, Partner name, Partner unknown: with these radio-buttons, you can define how the VPN-remote host can be located: via its IP address or its name. If the remote station uses dynamic IP addresses, the option Partner unknown is the right one to choose (this is not possible if shared secrets are used, see warning below).

When configuring an IPSec VPN connection, only one partner should use dynamic IP addresses. If both partners are using dynamic IP addresses (e.g. dial-in, ADSL) it is recommended to use OpenVPN in conjunction with a service for dynamic DNS (refer to “Dynamic DNS” on page 32).

4. Partner network: the local network behind the VPN remote host. This network will be protected by IPSec and can be accessed only through the tunnel.

5. VPN is locally connected to: this allows you to choose whether the whole network behind LAN interface 1 is allowed to use the IPSec connection or just a part of the network (defined IP range). With this function you can also restrict the use of a VPN tunnel to just a single host.

6. Use compression: this checkbox enables compression.

7. Authentication: IPSec can use two methods for authentication of VPN connections, namely certificates or common passwords (Shared Secrets).

Authentification Details:

X.509 certificate: When choosing the authentication via X.509 certificates, both partners must have X.509 certificates signed with the same CA key. Furthermore, the CA certificate in concern must be available on both partner stations. The procedure to ensure this, is described

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Chapter 5 - VPN Connections

in section “Certificates” on page 45. You should, however, have a basic understanding of X.509 certificates when trying to set up such a connection in order to avoid potential security holes.

For the configuration of an IPSec connection, the corresponding certificate is selected and the label of the certificate stored on the remote station. This label can be looked up in the web interface of the remote station in the menu Certificates if it's a Mediatrix Boss (see section “Certificates” on page 45). If you are using third-party equipment, consult its documentation for how to handle certificates.

Shared secret: When choosing the authentication via a Shared Secret, simply use the same password and type it in on both partner stations.

Setting up a new OpenVPN connection

1. Click on New OpenVPN Connection and provide configuration details as follows.

2. Connection name: this is a custom name for the new connection.

Figure 36: New IPsec Connection

3. Partner IP, Partner name, Partner unknown: with these radio-buttons, you can define how the VPN remote host can be located: via its IP address or its name. If the remote station uses dynamic IP addresses, the option Partner unknown is the right one to choose.

When configuring a VPN connection, only one partner may have the parameter Partner unknown. If both partners use dynamic IP addresses (e.g. dial-in, ADSL), a name has to be assigned to at least one of them by using a dynamic DNS service (see section “Dynamic DNS” on page 32).

4. Partner network: the local network behind the VPN remote host. This network will be protected by OpenVPN. This is the network that can be accessed through the VPN tunnel.

5. Transfer net: enter two IP addresses that are to be used internally for the transfer of the VPN packets. This transfer network can be used for traffic shaping or firewalling. It is important that both addresses entered here lie within a single network. It is also important to have matching entries on

Warning: Unlike OpenVPN that supports both certificates and shared secrets in any combination with static and dynamic IP addresses, IPsec has limitations concerning the use of shared secrets. Both partners have to use either static IP addresses or names. The option Partner unknown is not applicable in this case!

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both sides of the VPN tunnel. The addresses of the transfer network have to be reciprocally identical for two VPN hosts – i. e. the local IP address of one partner has to be the remote IP address of the other.

6. OpenVPN port: OpenVPN works port-based. That means every OpenVPN connection must have a unique port number. All packets for this connection will then be exchanged via this port.

This port number must be the same on both connection partners.

7. Authentication: OpenVPN can use two methods of authentication for VPN connections, i.e. certificates, or a common password file.

Authentication Details:

X.509 certificate When choosing the authentication via X.509 certificates, both partners must have X.509 certificates signed with the same CA key. Furthermore, the corresponding CA certificate must be available on both partner stations.

The procedure to ensure this situation is described in the IPSec section.

For the setup of an OpenVPN connection using certificate authentication, one of the connection partners has to be configured as TLS server and the other one as TLS client.

Shared secret: When choosing the authentication via a Shared secret, the same Open-VPN Shared secret file simply has to be selected on both partners (see section “OpenVPN Shared Secrets” on page 48).

Figure 37: New OpenVPN connection

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Chapter 5 - VPN L2TP Server

L2TP Server

L2TP allows you to easily connect client hosts via VPN. It is hence mostly used for Roadwarrior scenarios. Using L2TP for such configurations is especially comfortable for clients running MS Windows 2000 or Windows XP/Vista as this type of connection is already part of the standard network connection schema of those operating systems. In this menu the server-side configuration is set. Besides this the login-data for each user and PC is entered.

Configuring L2TP server

1. L2TP active

With this checkbox, L2TP can be activated and deactivated without needing to reenter all data. This can be convenient to temporarily shut down L2TP service for some time.

2. L2TP interface address

3. L2TP netmask

4. IP range for clients (from and to address)

These four parameters define the L2TP access network on the Mediatrix Boss. The IP range has to be within the subnet defined by the netmask. This subnet must not be assigned anywhere else within the IP range of the clients. You should hence use a private IP address range that is not in use for any interface or VPN tunnel.

5. DNS server for clients

6. WINS server for clients

With these two parameters, a DNS server and/or a WINS server can be assigned to clients connecting via L2TP. Using this option, the clients can access internal resources via name instead of IP address.

7. Click Save to save the parameters. Your changes will be applied after selecting Apply configuration.

Adding a new L2TP user

1. To create a new user account click on New user and provide configuration details as follows (compare Figure 39 on page 54).

2. Login-data

To setup a new user, the user name, password and the assigned IP address (this must be part of the previously defined L2TP subnet) have to be entered. Entering the IP address is optional, but assigning a fixed IP address to a specified user is highly recommended since it enables individually restricting the access with the help of the firewall.

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Figure 38: Configuring the L2TP server

3. IPsec authentication

When establishing a connection to computers running Windows 2000/Windows XP, the L2TP packets are encrypted with IPsec. Corresponding to normal IPsec connections both (pre-)shared secrets and X.509 certificates can be used. There are also L2TP/IPsec clients available for other platforms as for example PocketPC. Please refer to the documentation of your device for more information.

Windows XP supports both possibilities whereas Windows 2000 only supports X.509 certificates.

4. X.509 certificate

If you select X.509 Certificate in the menu IPsec authentication the X.509 certificates on both sides have to be signed by the same CA key. Furthermore, the corresponding CA certificate must be available on the client computer. Please refer to your Windows OS documentation on how to handle certificates on Windows systems. Unless you have special tools simplifying the process, you will have to use the Microsoft Management Console.

5. Pre-Shared Secret

If you select Shared Secret in the menu IPsec authentication, a password has to be entered which must be the same on all clients, running the same Windows OS, connecting to this Mediatrix Boss. The hitch is that authentication in IPsec is done via the remote IP address. Since this address is arbitrary in our case, we cannot identify the remote computer at the IPSec level. Therefore, all computers have to authenticate with the same shared secret.

6. Click Save to save account data. Your changes will be applied after selecting Apply configuration.

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Chapter 5 - VPN L2TP Server

Figure 39: Adding a new L2TP user

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C H A P T E R

6 SECURITY

Security Level

The fundamental level of security for the Mediatrix Boss to use is defined in this menu. Security as defined here only applies to the accessibility of the Mediatrix Boss and the firewall settings.

Figure 40: Adjusting the security level

By selecting Highest security, access to the Mediatrix Boss is limited to the permitted administration ports and the defined VPN tunnels. With this setting it is by no means possible to access the Internet via the Mediatrix Boss to set up own firewall or port-forwarding rules.

We recommend using this setting if the location secured by the Mediatrix Boss is supposed to have access to the Internet only via a tunnel connected to the remote partner-network. This can be used for very restrictive branch connectivity solutions.

The option User defined allows you to make significantly more advanced and flexible configurations. In particular, access to the internet can be granted, ports can be forwarded to certain computers connected to the LAN and your own custom firewall rules can be established.

The configuration matrix "'Open administration ports via"' enables the exact definition of the allowed locations from which configuration access shall be enabled. The opening of the administrative ports can be done on a per-interface basis.

"for HTTPS" allows and denies configuration via web browser; "for SSH" allows and denies access via command line using the SSH protocol.

Attention: The menu items Firewall and Port forwarding are only available if the security option User defined is selected. The corresponding settings require intermediate knowledge of the TCP/IP protocol.

Note: INTERFACES – LAN1

The Administration through LAN1 interface is enabled by default. It is recommended that this access is kept open, because you might lock yourself out of the WebGUI by disabling this.

Note: INTERFACES – WAN

Administration on the WAN interface should only be enabled if it is absolutely necessary. In conjunction with an insecure password, this could put the device in jeopardy for attacks. In order to increase security, only the SSH connection could be opened. It is possible to tunnel HTTPS access through SSH.

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Chapter 6 - SECURITY Firewall

By switching the security option from User defined to Highest security, firewall rules and port-forwardings that may already have been configured are deactivated, but not deleted. By switching back to User defined they are available again.

Firewall

The firewall of the Mediatrix Boss can be configured in two ways (also in combination). There are default rules available, but you may also configure custom rules.

Figure 41: Firewall configuration

Activate Basic Services

The basic services are defined for data traffic between two network interfaces. Thereby, different outgoing services (i.e. from the internal interface to the Internet) can be allowed, as well as data transfer between the two local networks.

LAN > WAN

This section deals with outbound connections. You can also choose to permit all outbound connections. It is possible to select the most important services needed. All necessary firewall rules are then created automatically.

Allow incoming VoIP connections

These are the more security-relevant setting. By default, these are disabled on the WAN interface. You can selectively enable the protocols. Note that you need at least one signaling protocol (either TLS or SIP) plus the media stream protocol (RTP) for a functional setup.

Note: These options are only shown for LAN Interface 2 if this interface is configured (i.e. is not deactivated).

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Redirect multiport range to local SIP port

This setting is mandatory if:

1. Multiport Mapping is activated (refer to "multiport" in “Basic Settings” on page 85) and

2. if there are components within your network which do not honor the path header.

Custom Rules

Besides the rules available by default, it is also possible to configure your own, very detailed rules. This way of configuration is recommended to be used only by experienced users and administrators. Faulty firewall rules can cause network applications to stop working. Customizing rules hence requires a lot of background knowledge and experience.

The custom rules of the firewall are executed according to their hierarchy. By clicking on the arrow-buttons ( and ) you can rearrange the firewall rules. Registered rules can be edited by clicking on or can be deleted by clicking on .

Creating a new custom firewall rule

1. Click Create rule at the appropriate position shown in Figure 42.

2. Define the rule. Then specify Protocol (TCP, UDP, ICMP, other), Interfaces and IP details that this rule shall match.

Figure 42: Add a new custom firewall rule

3. Choose the rule target (Filter policy) in order to specify what should happen to the traffic that applies to this rule.

4. Click Create rule to save your changes. Your changes will be applied after selecting Apply configuration.

Note: Network which do not honor the path header will send messages to the multiport range instead of port 5060!

Warning: This applies only to UDP traffic!

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Chapter 6 - SECURITY Port Forwarding

Port Forwarding

Port forwarding allows you to forward incoming data packets arriving at the Mediatrix Boss to a certain a computer within the local network. It is being used in conjunction with NAT and is the only way to access a computer connected to the private LAN directly from the Internet. Port forwarding is also often described as virtual server functionality.

To reach several computers within the private LAN via the same protocol it is also possible to change the ports.

Example:

You would like to let three computers within the LAN be accessed via SSH. SSH normally uses port 22, but it's impossible to reach all three computers via port 22 of the Mediatrix Boss.

With Port Forwarding you can now define that e.g. port 2022 of the Mediatrix Boss is forwarded to port 22 of the first computer, port 2023 is forwarded to port 22 of the second computer and so on. An SSH client connecting to port 2022 of the Mediatrix Boss would then actually be connected to port 22 of the first computer in the LAN.

Setting a new port forwarding rule

1. Click Add in order to define a new port forwarding rule and specify Local port, Destination IP, Destination port (compare Figure 43 on page 58).

2. Local port is the port that will be provided by the Mediatrix Boss .

3. Destination IP is the IP address of the internal computer to which the data is to be forwarded.

4. Destination port is the port of the internal computer to which the data is to be forwarded.

5. Click Save to save your changes. Your changes will be applied after selecting Apply configuration.

Figure 43: Port forwarding

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7 VOICE

The services of the VOICE section are at the core of your Mediatrix Boss. These services are rendered by the SIP Proxy component in the Mediatrix Boss and can be employed in various ways.

Survivability

The most frequent use is as a SIP Proxy in front of another SIP server. In the WebGUI, this can be found under the Survivability template. For this kind of usage scenario, the WebGUI has been geared towards telephony applications. When acting in front of another SIP server, it is also possible to limit the number of permitted sessions based on bandwidth restrictions (Call Admission Control).

Session Border Controller (SBC)

The other rather frequent area of employment is as SBC (Session Border Controller). An SBC typically separates an untrusted and a trusted network. Usually, SBCs are made to protect a data center (trusted network) from Internet (untrusted (untrusted)) users.The Mediatrix Boss will then not only perform SIP-signaling interworking, but it will also perform media handling for any SIP-initiated sessions. This feature can be used to achieve a connection between two networks that do not have a routing connection (as for example a customer premise network and data center network), handle NAT between networks, or protect a network from the outside world without losing the ability of VoIP connectivity. The SBC templates can be used to voice-enable existing networks that have firewalls to separate segments, use devices from behind a NAT (for example a NAT router on the Internet) or facilitate deployments in hosted scenarios. Another feature of the SBC templates is topology hiding which is one of the factors that make attacks more difficult. What topology hiding does is that all IP addresses of the data center appearing in the SIP message as for example "To:" and "From:" headers are rewritten with the external IP address of the SBC thus concealing the network architecture of the data center.

Generally the Mediatrix Boss supports two different types of SBC scenarios, called SBC and Branch SBC. Branch SBCs are slightly different since the users are not really Internet users, but well known branch users. Nevertheless, the SBC should protect the hosts in the data center by isolating them from these users.

Derived SBC Scenarios

Other possible SBC scenarios are SIP Trunking as well as ENUM. With its ENUM functionality the Mediatrix Boss acts as an ENUM-lookup gateway and as a Session Border Controller controlling calls coming in over the Internet.

The SIP Trunking is a derived SBC scenario consisting of only a single logical connection (that could of course consist of many individual SIP sessions) between an IP PBX and a Trunking Server usually offered by a carrier. This setup replaces a local PSTN gateway on the customer premise, and is hence a step towards a full IP-only customer premise connectivity.

Other scenarios include:

Standalone: in this scenario the Mediatrix Boss acts as a small SIP Server and forwards certain sessions to another server or to a SIP gateway. From version 6719.19 on NAT handling is always active if this template is selected.

Custom: it is also possible to apply a custom scenario with the SIP Proxy component. However, this requires knowledge and experience beyond the scope of this manual.

Check the Command Line Reference (“Chapter 9 - Command Line Interface” on page 129) for more information.

Scenario Selection

Before any parameters for the SIP Proxy can be configured, you need to choose among the Survivability, Branch SBC, SBC, SIP Trunking, Standalone, ENUM, or Custom template configurations in the first menu item of the Voice tab, Intended use. For all types of branch office scenarios without the need for media handling, you will most likely want go for the Survivability Template. The Branch SBC scenario is suited for branch connectivity setups where media handling or NAT traversal is a requirement. The SBC is usually used on the way from a Branch SBC to an IP PBX to act as a media relay. Both, SIP messages as well as media will then pass through two hosts. Explaining all usage scenarios would be far beyond the scope of this manual, however, you can find a brief description for each of them in the sections below.

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Chapter 7 - VOICE Survivability and Call Admission Control Template

Just pick a template under Intended Use and press Select to activate it. You can switch templates without losing your configured parameters.

Survivability and Call Admission Control Template

The typical survivability scenario is explained rather quickly. Assume having a small/medium office with a WAN link. The central SIP server is normally an IP PBX and used for VoIP telephony. This IP PBX is located in the data center, at the other side of this WAN link. The WAN link can either be an Internet line or an internal leased-line to the data center network. In the latter case, using a single interface of the Mediatrix Boss can be enough. The interface configuration has a significant impact on the parameters you need to set for the SIP Proxy component of the Mediatrix Boss. Both scenarios (single and multiple interface) are mentioned in the appropriate places in the following.

The Survivability scenario provides telephony functionality to a small/medium office even if the central SIP server is unavailable due to a WAN failure. In such cases, the Mediatrix Boss will take over the functionality of the SIP server. Once the SIP server is reachable again, the operation returns to normal.

The current status of the product can be seen on the Status Page (simply click on HOME in the Top Menu Bar). Besides the VoIP Router Status display, you will see the text Survivability Mode, if the product has detected a failure of the central SIP server. Once operation returns to normal, this text will disappear again.

The second side of this scenario is Call Admission Control. Since all SIP messages are routed through the appliance, it can freely decide on how to act upon certain requests. Call Admission Control is about denying certain requests based on the defined bandwidth constraints. This mechanism is essential for providing Quality of Service (QoS) for real-time communications.

Survivability Basic Settings

The following explains the mandatory as well as the optional entries for this menu item.

Own number (without direct dial)

(Figure 44 on page 62)

The Own Number field is intended for entering the phone number of the branch office without the extension.

Common Scenario: phones can be reached via two different numbers; their complete PSTN number and internally by simply dialing an extension.//

Example: the branch office can be reached via the "+49 89 4711" from the outside. Internally there are the extensions "100" and "110". This means that from the PSTN you can reach the phones by dialing +49 89 4711 100 + 100 and 110 . It is your choice whether you use complete E.164 numbers here or work with e.g. area code only. In any case, the registration with the SIP server and the SIP Proxy is performed by the Mediatrix Boss with the full number. This is very important, because the central SIP server can have many phones with the extension "100" if handling multiple branches.

In Survivability where there is no connection to the off-campus network the Mediatrix Boss can then prepend the number of the branch and find the phone in the Registration database. This way extension dialing is still possible in Survivability.

If you do not want to call the internal phones with a shortcut, simply leave this field blank.

Prefix number

This is the prefix used for outgoing calls.

Usually, this is just a simple digit as for example a "0" for getting an outgoing line. However, it is also possible to use more complex expressions. You can enter arbitrary regular expressions here. Only numbers having the correct prefix will be routed to the gateway(s) (“Gateways” on page 72) in the Survivability case. This prevents unavailable internal numbers from ending up in the PSTN and going somewhere unintentionally.

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Timeout

The Timeout value is one of the most central parameters in the Survivability scenario. Usual values for this timeout parameter range between 2-4 seconds. The value entered here will be used to determine whether the IP PBX, Failover PBX and possibly gateway(s) (“Gateways” on page 72) are online or not.

First of all this timeout is of course used to check for the IP PBX. If it does not respond to a request within this timeout period, the Mediatrix Boss will go to Survivability mode. It is the maximum time the Mediatrix Boss will wait for a response upon a request before falling to survivability mode. There is a background checking algorithm running that will decide whether the IP PBX is available or not.

Ignore Gateway during Register

This feature is only required if the Mediatrix Boss modifies contact headers of SIP messages, because of topology hiding features (if e.g. Force Return route, Multiport Mapping are used).

In these cases, especially if there is a routing connection from the central SIP server to the branch voice gateway, it might be desirable to have the Mediatrix Boss handle the messages from a local voice gateway without performing contact header manipulation on them. The mentioned routing connection can be implemented in various ways (for example port forwarding).

Some SIP servers also require this because they treat such gateways differently than regular clients.

Enable Hunting

You have the possibility to configure several gateways in the menu described in “Gateways” on page 72. These gateways can then be grouped (with Access Codes) and given a Hunt Priority. If the Enable Hunting option is checked, gateway hunting is activated globally, i.e. for all your configured gateway groups. If this parameter is not set, only the first gateway will be considered in Survivability mode or for CAC redirection.

Hunting can be used for a variety of scenarios as for example stacking gateways or having gateways in multiple locations and applying Call Admission Control Parameters to them. The decision parameters for the gateway hunting are configured in section “Gateways” on page 72.

TLS server and TLS client authentication

If these two boxes are checked, the TLS certificates, which can be uploaded under SIP TLS Configuration in section “SIP TLS Configuration” on page 98, are applied for authentication for the Mediatrix Boss and the clients (registered telephones) resulting in a more secure connection.

PBX Addresses


PBX Address

As we have already described above, in Survivability Mode, the SIP Proxy component of the Mediatrix Boss is only active if the central PBX is not reachable.

Please enter the address of the central PBX here. It can either be entered in the form of an IP address, a host name or even as a DNS SRV address. Please note that entering a DNS or DNS SRV name can introduce more latency in forwarding messages to the IP PBX since the host name has to be looked up first. It is also important to know that if you enter a DNS name, the DNS server has to be available upon a WAN failure or any other conditions under which the Survivability mode is intended be active.

SIP Transmit Type

The SIP Transmit Type parameter is the transmission type the Mediatrix Boss has to use to contact the central SIP server. Three different SIP transport modes are supported, namely UDP, TCP, and the encrypted TLS variant. The type to use depends first and foremost on the used SIP server. The TCP and UDP types are straightforward and just specify the used protocol type.

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The only type requiring further explanation is TLS. For TLS additional items like the encryption keys are required. Since this is a very advanced topic, this cannot be configured via the WebGUI as of yet. The Mediatrix Boss will automatically generate TLS keys which can be downloaded for the use in servers/phones via SCP. Please consult the Command Line Guide for more information on how to do that.

Figure 44: Survivability: Basic Settings

PBX Port

Please define here which port the Mediatrix Boss has to use to contact the SIP server. If left empty, port 5060 will be assumed for the transmit types UDP and TCP, 5061 for the transmit type TLS.

PBX Alias

The Survivability and Call Admission Control function will only be enabled for SIP messages that fall into the domain of this PBX. Please also refer to the Domains parameter described in section “Domains” on page 71.

It is necessary to configure all source IP addresses or host names where SIP messages of the PBX can originate from.

To classify the origin of SIP messages (server or client) and hence to determine the direction of the message flow, the Mediatrix Boss must know which messages are coming from the PBX. These can be messages coming from the IP address of the PBX and Failover PBX, but it might also be other IP addresses if the servers are especially clustered or load balanced.

Simply add all of the alternative domain names and/or IP addresses of your PBX.

Failover PBX

If you have a failover system with a failover PBX(s), insert up to three IP addresses or host names here. Separate them with commas.

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Failover Convergence IP

Just like the Failover PBX: if you have a second Mediatrix Boss available for a failover scenario just enter its IP address or host name here.

Survivability and Features


The settings which will determine the behavior of your Mediatrix Boss in Survivability Mode are listed below.

Call Forwarding (CFW)

CFW in Survivability Mode

Enter a number or a URI here that should be mapped to the forwarding target (see below) in survivability mode. It is possible to use regular expressions here.

Please be careful with special characters: everything will be interpreted as a regular expression!

Figure 45: PBX Addresses

Forwarding Target

This is the destination of the CFW. It has to be the number of an actually registered phone. If you have multiple phones registered with the same number, the proxy will perform SIP forking as per RFC 3261.

The forwarding and number mapping will only be done in Survivability mode. The normal operation will not be impacted at all by this option.

Map Prefix

This feature is used for both survivability and Call Admission Control scenarios. It is most often used in pure telephony scenarios where the user part of a SIP URI is always a number. The Map-to-office prefix function consists of two fields that must both be filled in.

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Short Prefix

The Prefix that has to match an incoming request. If such a match is found, the Mediatrix Boss will map this prefix to the long office prefix as described below.

Example

Assumption 1: the number range for the phones connected to the Mediatrix Boss was 123456789xx (xx stands for an arbitrary extension).

Since 123456789 is a very long prefix, there might be additional prefixes configured in your central softswitch/PBX, that you also want to be considered correctly for Survivability and CAC.

Assumption 2: there is a second configured prefix, say 58000xx (xx is again an extension).

The Mediatrix Boss will then map this short prefix to the long version above, and only then perform all checks concerning CAC and Survivability.

Map to Office Prefix

Long office prefixes that the above described short one will be mapped to. If configured correctly, this Office Prefix plus the extension will then match an actually registered phone. This makes the survivability function work properly.

However, it could also be mapped to a call that should be routed out through a gateway in the survivability case (refer to Access Codes in the Gateway menu; section “Gateways” on page 72).

Class of Service

Sometimes it is useful to set some restrictions to certain phone numbers. With the Class of Service enhancements parameters it is possible to deny or allow target numbers for special phones. You can allow or deny for example national or international calls or even special numbers.

Source

This parameter should contain the phone number of the source phone.

Destination

Enter the destination number that is denied or allowed by this rule.

For example, if the destination contains 00, all phone numbers beginning with 00 are allowed or denied depending on the policy parameter

Policy

The policy parameter can only contain the values allow or deny. If this parameter is set to deny, the call from the source number to the destination number is blocked.

Multiline Hunt Group

Multiline Hunt Groups have become standard features with many PBXs and they are widely used. Examples include hotlines, support groups, etc.

The concept of such a hunt group is relatively simple. A hunt group is a telephony concept that refers to the methodology of distributing phone calls from a single destination number to a group of several phone lines. Multiline Hunting (sometimes MLHG or line hunting) is a feature that allows multiple telephone lines going into a business to act as a single group, called a hunt group.

This type of fallback is a somewhat more complex form of call forwarding. If the line called is busy, the call goes to the next available line. Only if no lines in the group are open, a busy signal is replayed to the calling party. In even more advanced scenarios, the call might be queued for the next available agent. Usually a wide variety of call distribution algorithms is supported such as linear hunting, circular hunting, and queuing.

The problem with a distributed IP-telephony system is, that the hunt group is a PBX driven feature. If the PBX is unavailable, the complete hunt group is usually taken out of service. The Mediatrix Boss can also provide Survivability for most of the Multiline Hunt Group functionalities available on common PBXs.

The Hunt Group functionality itself is configured in “Hunt Group Configuration” on page 116. This configuration here only becomes active in survivable mode, and only then the Mediatrix Boss will forward the called number to the desired hunt group.

Source Number

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This is the number the incoming call is made to. This can be a physically registered phone, or a virtual number without any registration. In any case, upon a match, this call will be sent to the configured hunt group in survivable mode.

Target Number

Enter the number of the configured hunt group you want to forward the call to in survivable mode.

Figure 46: Survivability and Features

SIP Settings


Determine how your Mediatrix Boss will handle SIP messages in Survivability mode. All available features and options are explained below in the order of their appearance in the WebGUI.

Enable B2BUA handling

This function is necessary for gateways that do not properly support SIP forking. Checking this box will make the Mediatrix Boss behave like a B2BUA (back-to-back user agent) towards the voice gateway in Survivability mode. This means that the Mediatrix Boss will assign each session from and to the gateway its own Call ID.

Towards the SIP endpoints, the Mediatrix Boss will continue to behave as a SIP Proxy. B2BUA handling is usually, not necessary if you have no forking. Forking on the other hand is necessary if you have multiple SIP endpoints with the same SIP user name, i. e. the same phone number. This is commonly described as keyset feature or multiline appearance. Incoming calls in such multiline scenarios sometimes pose difficulties for gateways because the SIP proxy in the Mediatrix Boss will fork the request. This will lead to the fact that multiple SIP responses are forwarded to the gateway which is a problem for some of them. With the B2BUA feature enabled, the Mediatrix Boss will terminate this call and only forward a single response to the gateway.

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Alive check via SIP instead of ICMP ping

There are two ways the Mediatrix Boss supports to check whether the SIP Server, for which it is providing the survivability functionality, is monitored. This monitoring can be applied at the network layer (via ICMP messages) or on the application layer with SIP-OPTION requests. The check with pings should work with all servers if the underlying network supports this. This is usually the case, only firewalls on the way, or a packet filter on the server itself could prevent the Ping Echo Reply messages from getting back to the Mediatrix Boss . Other than that, the pings will be sent in intervals of approx. one minute. This interval can be shorter depending on the amount of SIP messages flowing through the Mediatrix Boss . The interval can also vary if there are abnormal negative responses to requests forwarded to the server. The non-response to a single ICMP message will not lead to the Mediatrix Boss switching into Survivability mode.

If the SIP server supports SIP-OPTIONS requests (please refer to RFC 3261 for more information on this SIP request type), this should be used for doing the alive check. It will also enable the Mediatrix Boss to switch to survivability mode for situations in which the central SIP server is reachable, but not responding correctly. Whenever we receive 5xx or 6xx responses from the central server for an OPTIONS request, the Mediatrix Boss switches to Survivability mode. If no response is received, a single non-response will not lead to the Mediatrix Boss switching to survivability mode.

SIP Options User Part

Please insert your SIP User Name here. If left blank, it will be set to anonymous.

SIP Options From Host

Please insert the From-Host IP or Host name.

Registration Timeout

The Registration Timeout value is only used during Survivablility mode. In that mode, the Mediatrix Boss will acknowledge incoming Registers and Subscribes from the IP phones. These messages will be sent with a default "expires" value. If this value is very long, it is possible that the IP PBX will have lost the registration of many phones.

This parameter forces the phones to use a shorter registration timeout during survivability and as a result to register more often. Once the connectivity comes back, the IP PBX will get REREGISTERs from the phones in this branch office very rapidly.

Use short lr parameter in Record-Route

By default the record route header contains the parameter information "lr=on". If it is known that this causes problems in your system, you can check this box. If checked, the record route header will only contain "lr".

Turn off re-generation of Branch

Normally, the branch parameter of the VIA header is re-generated for each new request. The calculation of the parameter is derived from the transaction key, i.e. it is an md5 of From/To/CallID/CSeq. This is RFC-compliant and failsafe in the case of reboots, because messages lost due to a reboot will still be forwarded with the same branch parameter if they belong to the same transaction.

Some devices use different headers for CANCELS and INVITES (this is not RFC compliant!). If this is the case, and the initial INVITE and CANCEL are generated with different headers, you have to check this option to change the parameter calculation accordingly. The Mediatrix Boss will then store and reuse the VIA header.

Caution: This option enables a workaround making up for non-RFC compliant devices in your network and may cause problems in the event of a reboot.

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No Rerouting On

The Mediatrix Boss will perform rerouting on a number of replies from the central softswitch/PBX. This is necessary especially in conjunction with CAC. If the Mediatrix Boss receives a negative reply for a session initiation request, it will try to perform the specified rerouting. It will not do any rerouting on the responses selected here. The possible response codes are listed below. Select all by pressing SHIFT and selecting them with a cursor key. Alternatively, you can use CTRL + A to select all.

300 Multiple Choices

301 Moved Permanently

302 Moved Temporarily

305 Use Proxy

380 Alternative Service

400 Bad Request

401 Unauthorized

402 Payment Required

403 Forbidden

404 Not Found

405 Method Not Allowed

406 Not Acceptable

407 Proxy Authentication Required

408 Request Timeout

410 Gone

413 Request Entity Too Large

414 Request URI Too Large

415 Unsupported Media Type

416 Unsupported URI Scheme

420 Bad Extension

421 Extension Required

422 Session Interval Too Small Via

423 Interval Too Brief

480 Temporarily Unavailable

481 Call Leg/Transaction Does Not Exist

482 Loop Detected

483 Too Many Hops

484 Address Incomplete

485 Ambiguous

486 Busy Here

487 Request Terminated

488 Not Acceptable Here

491 Request Pending

493 Undecipherable

500 Internal Server Error

501 Not Implemented

502 Bad Gateway

503 Service Unavailable

504 Server Timeout

505 SIP Version not supported

513 Message Too Large

600 Busy Everywhere

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603 Decline

604 Does not exist anywhere

606 Not Acceptable

Force Return Route for NOTIFY

While some SIP servers support an outbound-proxy feature for addressing SIP messages to endpoints, they often do not fully implement the feature for additional messages beyond RFC3261. With this mode of operation, all messages specified in RFC 3261 would be addressed to the endpoint, but sent to the outbound proxy that then makes the final delivery.

While this works well for INVITE, ACK, BYE and REFER messages, other messages like SUBSCRIBE, NOTIFY, PUBLISH are often treated differently. This poses a special problem for NOTIFYs because they are sent depending on header fields in the SUBSCRIBE message. Therefore, similar to the Force Return Route feature, it is specifically possible to have the Mediatrix Boss change the contact information for SUBSCRIBE and NOTIFY requests. Notify requests will then always addressed to the Mediatrix Boss acting as a SIP Proxy. Since there is no such thing as a forking for these requests, it is easily possible to handle them correctly.

Mask Proxy Domain on Register

If this is selected, the Mediatrix Boss will replace the domain part of the SIP URI of each message flowing through it with the SIP Domain (IP address or DNS name) of the SIP server. Some SIP servers have problems accepting SIP messages for multiple domains or do not do so due to security reasons. Most systems however work effortlessly without such modifications.

Also check any options your server might offer to set domain aliases. This parameter should hence only be used if you encounter problems in relation to this topic.

Preset Record Route

The preset-record route parameter allows you to influence how the record route Header in the SIP messages will be written. The record route header in SIP makes sure that all messages belonging to a session go through all servers that were on the path of the original message.

In more detail, this means that the messages will go through all SIP entities on the way taken by the original message (according to the record route fields).

If the Mediatrix Boss is only used with a single interface and a single IP, you can completely disregard this field. But if the Mediatrix Boss is used with a WAN and a LAN address, it can be quite significant which IP address is set in this record route field.

By default, the Mediatrix Boss always selects the primary IP address of the interface the packet is leaving on. However, in some scenarios, especially in those involving VPNs and NAT, this can be the wrong one. It is therefore possible to explicitly configure the address and port to be used in the record-route field. This can also be interesting if the Mediatrix Boss is used behind another firewall that performs port forwarding. In that case, the external IP address and the forwarded port have to be entered here.

Note: By factory default all responses are applied. This is also the case after a factory reset. However, if you select one or more of the responses below, the following responses will be added automatically: 413, 414, 420, 421, 482, 483, 484, 485.

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Figure 47: Survivability Sip Settings

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Port Settings


Figure 48: Port Settings

Integrated GW Settings

These settings are specifically for the integrated gateway component of the Mediatrix Boss.

SIP Transmit Type

Select either UDP, TCP or TLS towards the gateway component. The latter serves for a secure connection and requires additional encryption keys. The section “SIP TLS Configuration” on page 98 and the Command Line Guide will provide more information regarding TLS certificates. The default setting here is UDP.

Gateway Port

Specify the port that is to be used towards the gateway component. Of course this port should not be in use already. Check whether the desired port is available in the SIP section of the GATEWAY tab. If left blank, the used default port is 5064.

Table 5: Port Settings

Receiving SIP Port 5060

Receiving TLS Port 5061

RTP Start Port 35000

RTP End Port 65000

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Strip digits

This parameter determines how many characters of the phone number must be removed in Survivability mode before passing the message on to the PSTN breakout gateway. Under normal operation, this will be done by the PBX. In survivability mode, you can have the Mediatrix Boss perform the necessary number manipulations with this option and the one below (Prefix Numbers).

Prefix numbers

After a certain amount of digits has been stripped (removed), it is possible to prefix any combination of digits before the call is forwarded by the Integrated Gateway. Note that the functionality here does of course not only apply to numbers, it could also be any characters allowed in URIs.

The most common case with a gateway will be phone numbers in which prefixes are commonly used. Prefixing something would hence be necessary if the gateway expects numbers in a certain format. As already explained before, this URI manipulation would normally be performed by the SIP server.

Since the Mediatrix Boss will continue the number manipulation that was formerly done by the SIP server, the gateway component will be unaware of the fact that it is operating in Survivability mode.

Figure 49: Integrated Gateway Settings

Domains

The Domain list parameter is one of the most important parameters in the Mediatrix Boss. The Mediatrix Boss will only apply the Survivability, Call Admission Control, and/or Session Border Controller logic to SIP messages directed at this domain.

The term 'domain' as used here refers to the rear part of a URI parameter. This means that if there are clients in your network which send a SIP message for [email protected] to the Mediatrix Boss it will only act upon it in the described way if foo.bar can be found in the domain list. Otherwise it will simply perform a DNS lookup and forward the message accordingly. In that case it will behave like a standard stateful SIP Proxy.

As a result messages will only be routed correctly in Survivability if all applicable domains are entered here. This must be at least the IP address of the IP PBX and, if used, also any DNS names as well as aliases. If you specify DNS names here, you should also specify the corresponding IP address (if possible) to avoid

Mediatrix Boss 71


unnecessary DNS lookups. If a SIP message arrives with the DNS name replaced by the IP address only a quick comparison has to be performed instead of a DNS reverse lookup. For domain names where no reverse lookups are possible, both the IP address as well as the domain name must be specified here to enable the correct operation.

Gateways

(See Figure 50 on page 73)

The gateway parameters are comparatively straightforward. The gateways will be used in the Survivability case. However, they will also be considered for Call Admission Control since a call to a local phone is of course not subject to admission control.

Other than that, the Mediatrix Boss will use the parameters configured here to perform a local breakout. The Mediatrix Boss theoretically supports up to 15 gateways in a hunting configuration. In practice, anything more than 3 is very uncommon.

IP address

The IP address here is that of the PSTN gateway. The Mediatrix Boss can interact with any PSTN gateway that supports SIP as specified in RFC3261. But some restrictions concerning features available in the phones etc. might apply.

It is also possible to specify DNS / DNS SRV names here. However, a WAN failure is the most common cause for the Mediatrix Boss to enter the Survivability mode. If your DNS server is unavailable in such cases, as is frequently the case, putting the DNS name of the gateway here will result in total failure. Therefore you should only use DNS names if you are sure the DNS server is available in all scenarios the Survivability function has to work in. A possibility to achieve this is to use the DNS-slave function of the Mediatrix Boss. It will then be a caching-DNS slave for some configurable master and hence answer DNS queries even if the master is offline.

Gateway Port

You may choose a custom port here if it is necessary in your company network. If you leave this field blank, the default port for the respective Transmit Type is taken.

Transmit Type

The transmit type can again be either udp, tcp and tls. The same restrictions and possibilities as for the IP PBX apply, so refer to “Survivability and Features” on page 63 for more information on this.

Strip digits

This parameter determines how many characters of the phone number must be removed in Survivability mode before passing the message on to the PSTN breakout gateway. Under normal operation, this will be done by the PBX. In Survivability mode, you can have the Mediatrix Boss perform the necessary number manipulations with this option and the one below (Prefix Numbers).

Prefix numbers

After a certain amount of digits has been stripped (removed), it is possible to prefix any combination of digits before the call is forwarded to the gateway. Note that the functionality here does of course not only apply to numbers, it could also be any characters allowed in URIs.

The most common case with a gateway will be phone numbers in which prefixes are commonly used. Prefixing something would hence be necessary if the gateway expects numbers in a certain format. As already explained before, this URI manipulation would normally be performed by the PBX.

Since the Mediatrix Boss will continue the number manipulation that was formerly done by the PBX, the gateway(s) will be unaware of the fact that it is operating in Survivability mode.

Access Codes

If a dialed number begins with this Access Code it will be forwarded to this particular gateway. If no Access Code is configured, dialed numbers are matched against the Prefix Number configured in the Basic Settings menu (refer to “Survivability Basic Settings” on page 60).

Access Codes can be used to group gateways for hunting. The Hunt Priority determines the hierarchy within a group of gateways (see below).

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Hunt Priority

To have the Mediatrix Boss perform gateway hunting Enable Hunting in the Survivability Basic Settings (explained in section “Survivability Basic Settings” on page 60) must be activated.

It is possible to group your configured gateways by giving them the same Access Code (see above). The Hunt Priority gives you the possibility to introduce a hierarchy within such a group of gateways. They will then be used as fallback gateways for each other according to this hierarchy, whenever the Mediatrix Boss encounters an error or a timeout with one of them.

The Survivability Basic Settings Registration Timeout and No Rerouting On have to be used to determine when to do a fallback.

Positive integers are legal values for the Hunt Priority field, where 0 is the highest priority possible. An empty field is interpreted as 0. This means that if no values are set for the Hunt Priority all gateways will be treated equally.

From Prefix

Depending on the gateway it might be necessary to add a prefix to the from-header of SIP messages, for example '00' or a '+' for international calls.

Rewrite Host

If you would like to rewrite the Request-URI, simply insert a value here and the Mediatrix Boss will do this for you.

CAC

It is possible to perform Call Admission Control in Survivability mode before the Call is being forwarded to a gateway. This is necessary since it cannot be assumed that the gateway is always locally connected to the LAN. It could be located off site. This could mean that in a backup case, it can only be reached via a backup link. There are many possible scenarios. The Call Admission Control parameters will allow limiting the number of calls going to the gateway. For more information on Call Admission Control refer to the more detailed explanation in the general section. The Redirect action will make the Call Overflow to the next possible gateway. If there is none, the Redirect will behave the same as a Deny. See section “CAC” on page 74 for more information.

Limit Class

Choose which class is to be limited to that particular gateway. The classes for limitation available here correspond of course to the classes you have configured at section “CAC” on page 74.

Action

Determine whether "Deny" or "Redirect" should be applied to the selected class.

Source IP and Destination IP

Specify the Source and Destination IP that is to be limited with this rule.

Figure 50: Gateways

Mediatrix Boss 73


CAC

The Mediatrix Boss can use its built-in bandwidth management capabilities to influence the SIP-signaling decisions. These SIP signaling decisions can both be influenced for packets coming from the phone as well as coming from the central SIP server. Real-time traffic can be classified and measured while it is flowing through the Mediatrix Boss (refer to Section “Bandwidth Management” on page 36 for more details). The limits for the different traffic bandwidth classes will be used as decision basis for either permitting or denying the establishment of new SIP sessions. Based on the information obtained from measuring the traffic, Call Admission checks based on predefined policies can now be made. These checks are enforced by the SIP Proxy. In case of a limit violation that would mean that a SIP 606 Not Acceptable message would be signaled to the central SIP server or the SIP endpoint. There are many variations to the parameters and also to the enforcement. Instead of signaling the Not Acceptable directly to the endpoint, the session can also be just marked (by modifying the request URI) to let the central SIP server do the enforcement. The central server can then decide to e.g. signal a busy line, or it can choose an alternative route (e.g. through a locally present PSTN gateway) to avoid a network congestion.

CAC

The CAC parameter indicates whether Call Admission Control shall be performed or not. Uncheck this in order to completely disable this feature.

CAC check if from IP network (strict checking)

This parameter should be checked if there is a local PSTN in the branch office. If this parameter is not checked, the Mediatrix Boss will assume that all calls to and from non-local destinations will go over the bottleneck link and hence require Call Admission Control (of course only if all other parameters like Link etc. match). This would however mean, that for a call coming in over the local gateway, going to the central PBX then coming back in to the Mediatrix Boss, Call Admission Control would be falsely performed although the Media Stream would stay in the local network. Strict checking will additionally check the SDP header to determine how the RTP streams would flow to correctly perform the Call Admission Control.

T38 Media Stream Measurement

Enabling T38 stream measurement will make sure that enough bandwidth is reserved, even though the actual bandwidth use can vary greatly. T38 media stream bandwidth can range anywhere from 10 kbit/s to 100 kbit/s depending on the nature of the page that is currently being faxed. A white page will be on the low end, a page with complex shapes on the high side. Therefore, a simple stream measurement as usually done for CAC is insufficient especially concerning small branch offices where statistical effects will not mask such a varying bandwidth. The smaller the branch, and the fewer the number of simultaneously possible connections, the more severe is the impact of such an inaccuracy in the stream measurement. It could lead to an additional connection being permitted, for which actually no bandwidth is available once the fax traffic picks up again. Therefore, this option should always be checked in small branch offices. It works in conjunction with the FAX-URI option that is explained below.

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Figure 51: Survivability and Call Admission Control (CAC Parameters continued)

FAX URIs

This function is especially important in conjunction with the Bandwidth Reservation, and the T38 fax protocol. If you are not using T38, you will not have to specify anything here. The particular difficulty lies in the nature of how fax handling with T38 is done in SIP networks. Faxing is done by first setting up a normal media stream with G.711, G.729 or a similar speech codec. Once the gateway or ATA has detected that it is dealing with fax traffic, it will initiate a codec renegotiation to the T38 codec. This renegotiation is done via INVITES, and hence sets up an additional media stream for an existing session. Once the T38 connection is established, the speech connection will not carry any data until the fax transmission is finished. This very special behavior needs to be handled specifically in CAC scenarios. If this box is checked, the bandwidth required for T38 will already be reserved for the initial call setup, even if a compressed codec is used there. If there is only insufficient bandwidth for the T38 connection, the original call will already be rejected.

CAC Bandwidth Check

Each SIP session uses a certain average bandwidth depending on the codec being used. For G.711 for example the bandwidth required for each session is around 80 - 85 kbit/s including the overhead packets. The bandwidth check parameter here, identifies how much bandwidth must still be available in order to permit another session. Note that you have to enter the parameter manually, since codices can be renegotiated during a session. This means that this parameter has to reflect the highest possible bandwidth that two endpoints can use after completing their codec negotiation. In pure VoIP scenarios that would be 11000 bytes/second for G.711 scenarios and 4000 bytes/second for G.729 scenarios. In the latter case you should make sure that G.729 is used by all phones. If not, the fallback codec would typically be G.711. Specifying the maximum bandwidth used here is important since otherwise it would mean that your session is permitted first, then renegotiated with another codec and denied there. This kind of behaviour would be very user unfriendly. For each Call Admission decision, as to whether a call should be permitted or not, the worst case is hence assumed. This parameter can also be used for deliberately leaving some room to the bandwidth limit by checking for a higher bandwidth than required. The difference would then remain unused which can be desirable under certain conditions. Note that this field is entered in "bytes/second", not "bits/s".

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Bandwidth Reservation

Bandwidth reservation determines the kind of call admission control done by the Mediatrix Boss. There are two types. If Bandwidth Reservation is disabled, the CAC will only count actually established calls. That way, the maximum number of possible calls can be fitted, but there is the danger of overbooking. Overbooking in those cases can happen if you have a large number of potential sessions. Those are sessions which are for example in a ringing state.

These sessions have not been established yet, and hence they are not counted. Moreover, there is the chance that at least some of these sessions are never completed (nobody picks up the phone). This means that the CAC check only runs over the actually established connections. Until a limit is reached, additional sessions will be permitted. In this window you can configure a certain number of sessions in a transient state (Session Progress, Ringing, etc.). Once permitted, all these sessions can be established and overbook the link by the number of sessions in a transient state that get established minus one. On a large site, with very many calls, this windows will be very small so that overbooking should not be an issue.

The other option is that of reserving bandwidth as soon as the call is setup. This means that a call is counted as established as soon as the number is dialed. This also means that sessions in a transient state are fully counted, even though they might never lead to an established session. This very conservative setup will completely prevent overbooking, but might waste bandwidth and thus prevent calls from being established that would theoretically have space especially in situations when calls are not picked up. However, the Mediatrix Boss does detect SIP-forking mechanisms which are for example used for keyset operation in VoIP scenarios. A forked request will only be interpreted as a single session.

CAC Table

We have already explained the basic approach of how the Call Admission Control is handled. In order to configure Call Admission control, we first need to define a link for which we want to do Call Admission Control. The properties of that link, as for example how much bandwidth is allowed to be used for voice, is defined inside a traffic class. See “Bandwidth Management” on page 36 section for more information about Bandwidth Management. The link definition here is done by entering the source and destination IP ranges. In the bandwidth management, multiple voice traffic classes can be defined for each link. After defining the link, we now need to configure a policy, since a reaction in case of an overload must be configured. These actions can also be configured in the Mediatrix Boss Series. The enforcement of the policy can be any action configurable in SIP. Each CAC policy consists of a Limit Class, an Action, and a Link. Define the three parameters, then press Add to add them to the list of Call Admission rules. In the following, we will define the different parameters in more detail:

Limit Class

The Limit Class definitions make the connection between the SIP and the IP layer. All traffic-class definitions are used for making assessments as to whether additional calls can be accommodated. For the Limit definitions, the Bandwidth Management Configuration is used where a traffic class with certain limits is defined. The limit is enforced by the Call Admission Control functionality. For more information on the definition of the traffic rules refer to the section “Bandwidth Management” on page 36 of this manual.

As for the configuration, simply select the class representing the real-time traffic you want to limit with this rule definition.

Action

For each Call Admission rule definition, the Mediatrix Boss permits the specification of an action in the case of a limit violation. As we have already mentioned above, simply denying calls is the simplest of all actions that can be taken. Sometimes however it is inappropriate. Imagine for example a case where we can route a call over multiple links. Since the Mediatrix Boss is operating at the edge of one or more bottleneck links, there might be an alternative route to the destination via a second link. The other possibility is to use the TDM network via a gateway realizing a local breakout. This second possible action is Redirect. Choose one of the two from the drop-down list. Future versions will come with additional actions.

Link

This includes: Source IP and Netmask, Destination IP and Netmask

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The notion of a link is fundamental to the way the Mediatrix Boss does Call Admission Control. The simplest case of a link is a physical network connection. The Mediatrix Boss however does not stop there. A link is closely associated with the routing functionality offered by the Mediatrix Boss. Typically source and destination addresses are used for classifying packets as belonging to a certain link (matches the standard-IP networking principles). The source address could for example be used to configure groups for the phones. Group 1 could get a range of IP addresses, Group 2 another.

Simply enter the Source and Destination-IP addresses or network addresses which you would like to have the Limit and Action applied to. The mentioned IP addresses pertain to the SIP-signaling traffic. The source would for example be the local branch network; the destination would most likely be the IP PBX. Type a "0" for any IP out of this range.

Ignore Prefix

All prefixes specified here will be ignored for any CAC decision. This means that if you specify a prefix xx here, the CAC checks will be based on the dialed number with the specified prefix being removed first. This feature will only be used in telephony scenarios where special prefixes can be dialed to e.g. enable the transmission of the Caller ID. Functions like these are often used by prefixing an access code to the actual number you want to dial. This prefix must of course not change the CAC assessment of where the call is going. In order to achieve this, simply add all possible prefixes that should be ignored to this list. Regular Expressions are allowed here for specifying several prefixes at once. Note however that the special telephony prefixes "*" or "#" are interpreted normally when in the beginning of the regular expressions.

Pickup Groups/CAC excluded Numbers

All prefixes specified here will not be considered at all for a CAC decision. Regular expressions can be specified. This can be used for excluding certain number ranges (if you know that there will always be enough bandwidth, or because you maybe want to exclude emergency calls). Excluded calls can cause an overload on a link if you are not careful in configuring your bandwidth rules.

In Voice scenarios with a central IP PBX as a SIP server, there is a commonly offered feature named Pickup group. This permits using any telephone to dial a certain special extension like for example "*9" in order to pick up the call of another ringing telephone inside this branch office. In such a case, the Mediatrix Boss must be told that the "*9" is not a regular number but rather the special extension for that. No Admission Control will be performed on the extensions entered here. Note that you can enter a regular expression here and the above described functionality will apply to all extensions matching this expression. This also means that characters having a predefined meaning in a regular expression must be escaped to match the real character. Escaping can be done by adding a backslash before the character in question. In order to match the "*9" used in the example above, you hence have to enter a "\*9" into the field. Refer to the Command Line Reference for more information on regular expressions.

Redirect to PBX

If there is no more bandwidth for permitting an extra session, the call can be either denied, or it can be redirected to a gateway. These actions will be discussed later in more detail. The field here specifies another possible action. The SIP Proxy can modify the dialed number by prefixing something to let the IP PBX know that the usual call routing path does not apply. In this case, the rerouting to e.g. a local gateway can be performed by the central IP PBX. This does not require any Call Admission Control awareness of the central PBX, simple call routing features with multiple alternative routes, as for example found in least cost routing features, do suffice. The central IP PBX can then route the call over the local gateway in the branch office, thus not generating traffic load on the bottleneck link.

Prefix

As already just described, this prefix will be added to the dialed number to inform the central SIP server that an alternative routing needs to be performed. This prefix can be any combination of alphanumeric characters and special characters like "*" or "#". In telephony applications, the prefix will most likely be numeric.

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Chapter 7 - VOICE Branch Session Border Controller (SBC) Functionality

Redirect External Call only

Just check the box to activate this self-explanatory item.

Diagnostics


Debug

This switch will make the SIP proxy write debug messages to the system log. These can either be seen in CLI mode on the Mediatrix Boss, or on the specified logging server. See the logging parameters in this document or refer to the Command Line Guide for more documentation on how to have a look at the logging.

Figure 52: Survivability Diagnostics

Branch Session Border Controller (SBC) Functionality

This template is very similar to the Survivability scenario, and, in fact, it will also act as a Survivability component in the network. Therefore, this paragraph provides a general introduction, but for the description of the actual features we will often refer to the Survivability section. Differences will of course be pointed out.

The main difference between the Branch SBC and the Survivability Template is the fact that with the former, the Mediatrix Boss will always do media handling in addition to the handling of SIP-signaling. The Branch SBC scenario will normally be used if there is a branch office which you would like to equip with real-time communication. A possible reason for this could be for example if the users in these branch offices are to be connected to a centralized softswitch (PBX). Especially in larger networks it is, however, often impossible to

Note: This is a dangerous function since it can break CAC altogether, especially when using regular expressions. You have to be aware that the expression specified here only matches the actual calls that you want to have filtered and no more.

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provide a flatly routed network across all locations which have to be connected. This implicates special challenges, because the peer-to-peer type communication of SIP endpoints concerning media handling actually requires such a flatly routed network.

Figure 53: Branch SBC: Basic Settings

Figure 54: Branch SBC: SBC Settings

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Branch SBC Basic Settings

Own number

See “Survivability Basic Settings” on page 60 for a more detailed description.

Prefix Number


Timeout


PBX Addresses

PBX Address

See “PBX Addresses” on page 61 for a more detailed description.

PBX Port


Trust endpoint URI

If this option is checked, the SBC will be forced to send outgoing requests to the URI of the request instead of sending it to the configured PBX.

PBX Alias


Failover PBX

If you have a failover system with a failover PBX(s), insert up to three IP addresses or host names here. Separate them with commas.

Failover Convergence IP

Just like the Failover PBX : if you have a second Mediatrix Boss available for a failover scenario just enter its IP address or host name here.

Survivability and Features

Survivability

Activate or deactivate this feature with the check box.

Call Forwarding

See “Survivability and Features” on page 63 for a more detailed description.

Class of Services


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Multiline Hunt Group


SIP Settings

All of the items available here are described in “SIP Settings” on page 65 in detail.

Port Settings

For default ports and restrictions, please refer to “Port Settings” on page 70.

Integrated Gateway Settings

Refer to “Integrated GW Settings” on page 70 for detailed information.

SBC Settings

Internal IP

Since the Branch SBC will do media handling if a SIP session leaves the boundaries of the local branch office, the Mediatrix Boss needs to know the internal network address for which it should not be doing media handling. For all sessions where both the source and the target are within this network, the media streams will continue to flow directly between the endpoints.

External IP

All Mediatrix Boss products support multiple virtual and non virtual interfaces. However, when acting as a branch session border controller, there must be a unique uplink interface through which outbound media streams should be routed. When looking at the SIP messages that are leaving the branch SBC, all SDP bodies will contain this external IP address, and hence all endpoints having a connection with another endpoint inside the Branch SBC will be sending their media streams to this external IP address.

Netmask

This parameter describes the netmask for the above configured network IP address. This netmask enables the Mediatrix Boss to check whether two endpoints are actually local or not.

Multiport Mapping

(Caution: Do not use unless explicitly needed)

Multiport mapping makes the Mediatrix Boss assign a unique port for each device connected behind it. It will always use this unique port number when communicating with the SIP server. The Mediatrix Boss will also change all related fields in SIP messages. The SIP server (PBX) will hence only see the contact information of the Mediatrix Boss with an individual port for each device. This mode of operation is usually not required and is only necessary for special scenarios involving SIP forking (multiple devices with the same user name part of the URI), where the requests already arrive forked from the server. If the Force Return Route feature is used in such scenarios to change the contact parts of the SIP messages, these would be forked again by the Mediatrix Boss.It is then necessary to utilize this feature to enable the Mediatrix Boss to uniquely determine where to send the already forked request. The destination port information where the SIP server sent the message is then used to correctly direct the message to the right SIP endpoint. It should again be reiterated that this type of message routing somewhat bends the conventions of SIP, but it is required for the described scenario, although there is one notable exception. If the PBX is able to support an outbound-proxy feature properly, addressing the forked request to the endpoint's URI but sending to the Mediatrix Boss, neither the Force Return Route nor the multiport feature are required. This is the most desirable configuration and should be used whenever possible.

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Special SUBSCRIBE record route handling

If special treatment for SUBSCRIBES is needed, you should check this option.

Topology Hiding (informational headers)

Check this box to hide internal information contained in the header (IP addresses and the like).


See “SIP Settings” on page 65 for a more detailed description.

Do not modify Gateway "Contact"

If this option is enabled, the contact-header of REGISTER messages coming from a gateway will not be modified.

Local Gateway

This is another Branch SBC specific parameter. You need to configure whether media handling should be done for calls to and from gateway calls. If checked, the gateway will be assumed to be local, and media handling will be disabled for all calls between local endpoints and the local gateway. If not checked, the gateway is assumed to be external, e.g. located at the central softswitch. This will enable media handling for all external calls.

Gateway DNS Alias

If you would like to use a DNS Alias for your gateway, simply check the box and fill out the field that opens automatically after the box has been checked.

Domains

See “Domains” on page 71 for detailed information.

Gateways

See “Gateways” on page 72 for more detailed information.

CAC

The Mediatrix Boss is able to do Call Admission Control in addition to Branch SBC functions. By default, this functionality is however not enabled in this mode. To enable it, simply activate the checkbox on top.

Find descriptions of all available items here: “CAC” on page 74.

Diagnostics

“Diagnostics” on page 78 will give you an overview on this item.

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Figure 55: Branch SBC CAC Parameter

SBC (Session Border Controller) Template

A Session Border Controller (SBC) is a VoIP session-aware device that controls call admission to a network at the border of that network. The central SBC function is not available in all Mediatrix Boss products. Contrary to the Branch SBC, the SBC function is not only an infrastructure function, but a security component providing a separation between networks. The Mediatrix Boss currently performs, NAT Traversal, SRTP termination as well as other functions in that mode. The Mediatrix Boss is still primarily built to fulfill these infrastructure functions. It is not built to perform certain other functions usually attributed to SBCs, such as protocol translation (e.g. to MGCP). While the functionality will increase in the future, the Mediatrix Boss is still positioned to be a pure SIP device. A typical centralized Session Border Controller breaks down into two logically distinct pieces.

Signaling

The Signaling-SBC function controls access of VoIP-signaling messages to the core of the network, and manipulates the contents of these messages.

Media

The Media-SBC function controls access of media packets to the network, provides differentiated services and QoS for different media streams. It can also be used to provide security functions like SRTP termination to differentiate between different security zones, and avoid things like service theft.

This template will act as a centralized session border controller, handling all media streams emanating from clients within the WAN network and being behind a firewall. The most important function of a centralized SBC is hence the ability to enable clients behind a NAT firewall to do IP communication. The challenge is to construct this to work behind arbitrary firewall types including those typically found in public Internet Hotspots. The goal is enable ubiquitous IP communication while keeping the network inside the SBC to remain secure. In order to properly understand this setup, we first need to do some explanation of the supported NAT networks.

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Explanation of NAT

NATs are active network devices placed in the data path. NATs are similar to firewalls, and different from routers, i. e. they are topologically sensitive. They have an "inside" and an "outside," and execute different operations on intercepted packets depending on whether the packet is going from inside to outside, or vice versa.

NATs are IP-header translators, and, in particular, NATs are IP-address translators. The header of an IP packet contains the source and destination IP addresses. If the packet is being passed from the inside to the outside, a NAT rewrites the source address in the packet header to a different value, and alters the IP and TCP header checksums in the packet at the same time to reflect the change of the address field. When a packet is received from the outside destined to the inside, the destination address is rewritten to a different value, and again the IP and TCP header checksums are recalculated. The "inside" does not necessarily use globally unique addresses to number every device within the network served by the NAT. The inside (or "local") network may use addresses from private address blocks, implying that the uniqueness of the address holds only for the site.

NATs are configured with a pool of public addresses, and when an "inside" host first sends an outbound packet, an address is drawn from this pool and mapped as a temporary alias to the inside host's local address. This mapped address is used as the new source address for the outgoing packet, and a local session state is set up in the NAT unit for the mapping between the private and the public addresses. This type of pure NAT has become very uncommon nowadays, but it is still mentioned for completeness here.

The variant of the NAT commonly used today is the Port-Translating NAT, or NAPT (today the term NAT is also frequently used to refer to this type of NAT mapping). This form of NAT is used in the context of TCP and User Datagram Protocol (UDP) sessions, where the NAT maps the local source address and source port number to a public source address and a public-side port number for outgoing packets. Incoming packets addressed to this public address and port pair are translated to the corresponding local address and port.

Again the NAPT is attempting to be transparent in terms of providing a consistent view of the session to each end, using a symmetric binding of a local address and port pair to an external address and port pair. NAPTs allow concurrent outgoing sessions to be distinguished by the combination of the mapped address and mapped port value. In this way each unique external pool address may be used for up to 65,535 concurrent mapped sessions.

For a while the terminology distinction between NATs and NAPTs was considered important, but this has faded over time.

It should be noted that a large number of mapped sessions in a firewall can pose both a security as well as a performance problem. Each NAT mapping is a pinhole in the firewall that can be used to sneak data into the protected network. In addition, with a large number of sessions, the firewall might have to do a lot of comparing for each arriving packet to check for the connection this packet belongs to. These restrictions represent problems especially in VoIP scenarios, since each active SIP client in a call state necessitates at least 2 active NAT mappings.

Signaling

The Mediatrix Boss will handle signaling and provide a Session Border Controller function between an inside and an outside network, typically the Internet. In addition, it will do far-end NAT traversal. After the client has registered, the remote-NAT mapping is learned by the Mediatrix Boss enabling it to send subsequent requests to this client. A SIP OPTIONS message is sent periodically (typically every 30 seconds) to keep the remote NAT mapping alive and thus keep the remote client operational. The SBC will also perform validation of received SIP messages before forwarding them to the internal SIP server. With that approach, malformed SIP messages will be detected and dropped. In addition to that, additional semantic checks will be made on the SIP messages based on the available information before forwarding them to the internal SIP server. The connection to the media handling is done by modifying the SDP bodies of the SIP messages. With that modification, the SBC will be in the middle of every media stream from a client to the inside network, as well as in every media stream between two clients behind two distinct NATs.

Warning: If SIP endpoints are to be used across the WAN port, you will need to manually change security settings. If your messages arrive on the WAN interface, the firewall will block those by default. It is however relatively easy to enable SIP and RTP traffic on the WAN interface. Please refer to the “Firewall” on page 56 section for more information about this topic.

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Media Handling

The media handler will create its own session for each connection setup done via the signaling handling. Similarly to the SIP approach to handling remote NATs, the remote NAT mapping is learned by the SBC, enabling the sending of packets to the client behind the remote NAT. Once the media stream has been initiated in both directions, no redirection of the media streams are permitted, except what is negotiated via SIP messages. This protects the users from session hijacking. Additionally, the source of the media stream packets is validated against the stored session parameters to prevent somebody from sending dummy packets to the respective port of the SBC and thus disrupting the session. It also features a dead peer detection to clean up invalid sessions. The Mediatrix Boss can also perform SRTP termination. With this function, the Mediatrix Boss will terminate all sessions emanating from clients on the outside network and send plain RTP to the SIP server or any other connected equipment, for example gateways. In the reverse direction, those devices will send plain RTP to the SBC which in turn will encrypt everything before sending it to the clients on the outside network.

The following will explain the parameters that can be configured for the SBC template.

Basic Settings

PBX Mode

There are two options for this drop-down box: Single and Multi PBX. The setting configured here is reflected in the next submenu PBX Addresses. If Single PBX is selected, only one PBX can be configured in the PBX Addresses menu. If the PBX Mode is set to Multi PBX, several virtual PBXs can be configured.

Presence Server IP

If you have a separate presence server, you can specify the IP address here. This would force the SBC to handle presence requests, as for example the SIP PUBLISH method separately from the standard SIP server. As such, you could have a separate SIP server and Presence server and let the SBC correctly separate the requests for these. The advantage of this is that your SIP server does not have to be aware of any presence functionality, because the SBC is already separating the requests.

Voicemail Server IP

You can specify the IP address of your voicemail server here. This is for special scenarios where the voicemail server is also integrated via SIP and where it should be ensured that in-dialog requests also flow correctly through the SIP server. This should only be relevant if you are using a SIP proxy type of server as your central SIP server. In such cases it can happen that in-dialog requests such as ACKs and BYEs are routed through the SIP server instead of directly to the SBC. Therefore, the Route headers of the messages must be modified accordingly.

PSTN Gateway IP

This field is similar to the one for the Voicemal-Server IP. If specified, it is used to aid the correct SIP-message modification. Again, if a back-to-back user agent is used instead of a SIP Proxy as the central server, this will be impossible. All signaling to the gateway will then be handled by the B2BUA and the SBC will be completely unaware of this. The media stream can of course still be directly passed to the voice gateway.

PBX Addresses

The PBX Mode setting as described above determines the template for the PBX Addresses menu.

1. Single PBX Template

a. PBX Address

Please enter the address of the central PBX here. It can either be entered in the form of an IP address, a host name or even as a DNS SRV address. Please note that entering a DNS or DNS SRV name can introduce more latency in forwarding messages to the IP PBX since the hostname has to be looked up first.

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b. SIP Transmit Type

The SIP Transmit Type parameter is the transmission type the Mediatrix Boss uses to contact the SIP server. The Mediatrix Boss supports 3 different SIP transport modes, UDP, TCP, and the encrypted TLS variant. The type to use depends first and foremost on the used SIP server. The TCP and UDP types are simple and just specify the used protocol type.

The only type requiring further explanation is TLS. For TLS additional items like the encryption keys are required. Since this is a very advanced topic, this cannot be configured via the WebGUI as of yet. The Mediatrix Boss will generate TLS keys automatically which can be downloaded for the use in Servers/Phones via SCP. Please consult the Command Line Reference for more information about how to do that.

c. PBX Port

Please define here which port the Mediatrix Boss is to use for contacting the SIP server. If left empty, port 5060 will be assumed for the transmit types UDP and TCP, 5061 for the transmit type TLS.

2. Multi PBX Template Virtual Interface to PBX mapping

The virtual interface to PBX mapping feature enables you to forward SIP-messages to specific PBXs.

Everything that the Mediatrix Boss receives at the Interface IP in the left column, will be forwarded to the PBX IP in the second column from the left. The check box Active gives you the possibility to disable a rule without deleting it.

The settings PBX Port and SIP Transmit Type are exactly the same as in the Single PBX Template.

Port Settings


Integrated Gateway Settings

Refer to “Integrated GW Settings” on page 70 for detailed information.

SBC Settings

Internal IP

The Mediatrix Boss can have multiple interfaces as well as multiple IP addresses per interface. It is therefore necessary to specify which IP address should be used for communicating with the SIP server. This IP address will then be used both for signaling as well as media handling.

External IP

The Mediatrix Boss can have multiple interfaces as well as multiple IP addresses per interface. The external IP address is the one the Mediatrix Boss will use when communicating with the clients. Clients in that respect are the SIP endpoints registered from the outside of the network boundary protected by the Session Border Controller.

Warning: INTERDEPENDENCIES

Please note:

• Each Interface IP can only be used for one rule here.

• The SBC Settings External and Internal IP must be empty! The rules which can be set with Virtual Interface to PBX mapping will be ignored if External and Internal IP are set!

• Interface IPs can be the WAN, LAN Interface 1 etc., as well as virtual interfaces.

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Internal URI match

The Internal URI match is a parameter used to classify URIs, which helps to identify unwanted SIP messages. This field can be a regular expression and should match all those requests that can occur when connected to this SIP server. The simplest form is certainly to use something like @foo which would match all URIs in messages with foo in the host part of their URI. You can use more sophisticated expressions, however, which could for example match legal phone number ranges. It is safe to leave this field blank.

SBC Mode Settings

Here you select the SBC Mode Setting. There are different modes that the SBC can operate in. The mode setting is focused on the type of message routing the unit should conduct. In SIP there are various header fields that determine the routing of a message. In detail those are the URI, the Via, the Record Route, and the Contact header. On the receiving end, the Contact header determines where requests (e.g. a BYE) into the reverse direction are routed. This information will end up in the URI of this message determining the ultimate destination of this message. The VIA and Record Route headers are used by proxies to determine the next hop of a message. In all three modes, the Mediatrix Boss does media handling. The difference is hence solely on the message routing level. The three possible values for the SBC Mode are: multiport, contact and proxy which are explained in the following.

multiport (mapping)

Multiport mapping fully maps all routing relevant headers. This means that all routing relevant fields in the SIP messages will be overwritten and will show the IP address of the Mediatrix Boss system. Every user behind the SBC in this mode will be mapped to a unique "virtual" port towards the PBX. Therefore the first user e.g. is assigned port 10000 the next user port 10001 etc. This is done for uniquely mapping all messages destined to a certain user and is especially relevant when using Multi Line appearances, such as huntgroups. In those cases the "Overwrite Contact" (which is the next mode explained below) must not be used.

In the Multiport SBC mode, the PBX is never really aware that an SBC is connected. It should support, however, one of the following three features, since the ports mentioned above are not actual network ports (although this would be possible with some additional configuration):

• Statically defining an SBC - Supporting the Path Header as specified in RFC 3327 - Supporting NAT handling (returning a message to the origination port)

Any of the three guarantees that all messages towards the SBC, although addressed to the specified port, are physically sent to the standard SIP port where the SBC is listening.

Reasons for applying this mode:

• Use this feature if you want to hide the IP addresses of the disjoint networks from each other

• Use this feature if your PBX has problems supporting proxies

• Use this feature in a cascaded scenarios where you want to connect proxies behind the SBC

• In order to use topology hiding, you have to use this feature

(overwrite) contact

This is the simplified variant of the multiport mapping. It influences the message routing in the same way as described above but it rewrites everything to the same port. It hence cannot be used for Multi Line appearances (same user name with different endpoints), as these routing associations will get lost through this mapping. However, it does not require any PBX support since the PBX will believe it is talking to a standard endpoint. It can be used for simple scenarios such as Internet Users, etc.

proxy

In this mode the SBC behaves like a standard proxy defined in RFC3261 for everything pertaining to message routing. As said above, the media handling is still done, as well as NAT handling. It hence has no impact on the core features of the SBC. The Contact header is not modified at all. This mode is most appropriate in a cascading scenario where you have other SBCs connected on the WAN side of this SBC. It also requires one of the following three features to be supported in the PBX:

• Statically defining an SBC

• Supporting the Path Header as specified in RFC 3327

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• Supporting NAT handling (returning a message to the originating port)

The advantage of this mode is that it is completely in line with the SIP standards, and hence has the highest compatibility even if proprietary enhancements have been made in the SIP signalling that might break with the above mentioned modes.

Multiport: Don't alter request URI user part

The SBC Mode (above) setting "multiport" can be modified with this checkbox. If it is checked the Mediatrix Boss will rewrite the host name and port, but it will leave the username untouched.

Add Path header to REGISTER

The Path Header (your Mediatrix Boss's IP) will be added to REGISTERS if you check this option.

Topology Hiding (informal headers)

Check this box to hide internal information contained in the header (IP addresses and the like).

Topology Hiding (Via headers)

You also have the possibility to hide information concerning the destination by checking this box.


For detailed information refer to “SIP Settings” on page 65.

Optimize Media Path

Enable this option and the SBC component of the Mediatrix Boss will check if a direct media connection is possible. If this check leads to a positive result, the Mediatrix Boss will alter the SDP entries for the particular conversation to enable this direct media connection.

Registration Check

If this is enabled the Mediatrix Boss will only forward SIP messages which:

are REGISTER or BYE messages or which have been sent

from a registered IP address,

from an IP address that belongs to your configured Domains (see next menu description below),

or from an IP address that belongs to your configured PBX Aliases

Domains

See “Domains” on page 717 for more information.

Security:

Secure Media (SRTP)

If this is enabled, the SBC will perform SRTP termination for all incoming sessions, and will only initiate encrypted sessions to the outside world. Internally, RTP will still be used.

Diagnostics

For more information refer to “Diagnostics” on page 78

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SIP Trunking Template

SIP Trunking is another possibility of using the Mediatrix Boss , making it perform as a Topology Hiding Session Border Controller. SIP Trunking will mostly be used for telephony applications, but the Mediatrix Boss also supports alternative media formats like video conferencing with an identical configuration. For the explanation of the parameters, we will heavily rely on pure VoIP terminology.

If SIP Trunking is used, the Mediatrix Boss will usually be placed at the edge of an office network.

Figure 56: Trunking Parameter

Note: The SIP-Trunk template is only available in version 6719.18 and lower!.

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Figure 57: Trunking Parameter (continued)

Basic Settings

Some of the settings in this template are completely identical to the already mentioned templates. We will hence often refer to the Survivability section, since it contains descriptions in further detail. Possible difference will, however, be pointed out.

Own Number (without direct dial)

Please refer to “Survivability Basic Settings” on page 60

Trunk Address

The trunk address needs to be set to the IP address, DNS name, or DNS SRV name of the SIP server of the trunking connection provider. If a DNS SRV name is specified, the “Basic Settings” on page 90 must be enabled.

SIP Transmit Type

The SIP Transmit Type parameter is the transmission type the Mediatrix Boss uses to contact the central SIP server. The Mediatrix Boss supports 3 different SIP-transport modes, UDP, TCP, and the encrypted TLS variant. The appropriate type to use depends exclusively on the used SIP server. The TCP and UDP types are straightforward and simply specify the used protocol type.

The only type requiring further explanation is TLS. For TLS additional things like the encryption keys are required. Since this is a very advanced topic, this cannot be configured via the WebGUI as of yet. The Mediatrix Boss will generate TLS keys automatically that can be downloaded for the use in Servers/Phones via SCP. Please consult the Command Line Reference for more information about how to do that.

This setting here will specify the way, the SIP messages are transmitted to your trunking provider. The choices are UDP, TCP, and TLS again, as specified in RFC3261. If your provider supports TLS, you should try to use it for security reasons. If not, the only other option to achieve a certain level of security is by using tunneling techniques likes VPNs to implement a secure channel across the

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Internet. However, this security only applies to the signaling level. The Mediatrix Boss supports SRTP pass-through by default and can also support SRTP termination. This means that if your PBX supports SRTP, the encrypted media can be passed through. In cases where this is not supported, the Mediatrix Boss can do the termination for your PBX and return plain RTP while speaking SRTP towards the provider.

Trunk Port

This field specifies the port that should be used to contact the trunking provider. As always, the default SIP ports (5060 for TCP and UDP, 5061 for TLS) will be used if nothing is specified. If non-standard ports are used by your provider, please enter them here.

Enable DNS SRV

If DNS SRV is enabled, the Mediatrix Boss can use it to discover a secondary server if the primary one is offline. DNS SRV is a protocol that allows you to define service classes for your domain. The product will look for the SIP service class, and then will lookup the IP address for the discovered host. DNS SRV is supported by all common DNS-server products. Also see the explanations for the Retransmisson Timers as well as the DNS SRV failover penalty for more detailed information on this.

PBX Addresses

Please configure the settings for your PBX with the following specifications.

PBX Address

Please enter the IP address or the DNS name of the PBX for which you would like to implement SIP Trunking. All requests coming from the trunking provider will be forwarded to this PBX. There are two different ways a PBX can behave. It register with the Trunking provider, or use a static registration with the provider. Simply set the address of the internal interface of the Mediatrix Boss as the SIP-trunking address. You do not have to have your PBX connected to any outside network, merely the routing / switching between the PBX and the Mediatrix Boss must be possible.


SIP Transmit Type

The transmit type parameter has the same effect as mentioned above towards the trunking provider.

PBX Port


SIP Settings

Retransmission Timer

The various retransmission timer settings are all highly related and therefore will be explained together. Please note that the retransmission logic only applies to the UDP variant of SIP. Both the TCP and TLS variant use the connection oriented Transport Control Protocol for the transmission of the messages. This transport layer protocol ensures that all sent messages reach their destination. This means that there are error-correction and retransmission mechanisms already built into the transport layer meaning that the SIP application layer does not have to handle retransmissions any more.

The situation is quite different with UDP, where messages can get lost. The retransmission timers refer to SIP requests. All SIP requests mandate a response from the endpoint (in our case either the PBX or the trunking provider). Since the PBX will usually be connected through a local ethernet interface, there should be no

Warning: If DNS SRV is enabled, this information will be ignored since the discovered values from the DNS server are used.

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problem with lost packets on this side. The trunking provider connection, however, usually goes over the Internet where messages can easily be lost or delayed. These timers set the interval during which the Mediatrix Boss waits for a response to a request before sending it again.

Retransmission Timer 1

This is the first retransmission timer and must be set in milliseconds. If no response (both provisional 1xx responses as well as final ones will be accepted here) to a request is received within this time, a retransmission is initiated. The accuracy of the retransmission will be approx. 50 msec. This means that if this timer is set to 500 msec, which is also the default value, the retransmission will occur in the interval between 450 msec and 550 msec after the first request has been sent without an arriving response in return.


See above for a more detailed explanation. The timer must also be set in milliseconds with the default being 1000 msec.


See above for a more detailed explanation. The timer must also be set in milliseconds with the default being 2000 msec.

Total Retransmission Time

After the final retransmission timer hits, the server will be considered down or unreachable, and no further attempts will be made. If DNS SRV is configured, the failover mode will be entered and a DNS lookup will be initiated to discover the secondary server. Once that server has been detected, messages will be sent there instead of the primary server. The primary server will only be contacted after the penalty box has expired. Once that has happened, an attempt is made to contact the primary server. If it still does not respond even to the retransmission sequence, it will again be put into the penalty box. If it is reachable, all subsequent requests will go there again.

Port Settings


SBC Settings

Internal IP

Since the Branch SBC will do media handling if a SIP session leaves the boundaries of the local branch office, the Mediatrix Boss needs to know the internal network address for which it should not be doing media handling. For all session where both the source and the target are within this network, the media streams will continue to flow directly between the endpoints.

External IP

All Mediatrix Boss products support multiple virtual and non virtual interfaces. However, when acting as a branch session border controller, there must be a unique uplink interface through which outgoing media streams should be routed. When looking at the SIP messages that are leaving the branch SBC, all SDP bodies will contain this external IP address, and hence all endpoints having a connection with another endpoint inside the Branch SBC will be sending their media streams to this external IP address.

Quality of Service - TOS value for SIP/RTP traffic

Set the value for the TOS byte here for determining the standard quality.

Diagnostics

Refer to “Diagnostics” on page 78, it will give you an overview of this item.

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Standalone Template

The standalone template is the simplest template explained here. The Mediatrix Boss can act as a mini-SIP server, thus providing basic call functionality and other basic functionalities supported by the SIP devices. PSTN gateways can be used for making external calls. This template can also be used in conjunction with the B2BUA to use SIP carrier accounts as gateways.

Basic Settings

Figure 58: Standalone Basic Settings

Own Number

Under normal operations, the phones can be reached via two different numbers, their complete PSTN number and internally by simply dialing an extension. An example will illustrate this. Let us assume that the branch office can be reached via the "+49 89 4711" from the outside. Internally there are the extensions "100" and "110". This means that from the PSTN you can reach the phones by dialling "+49 89 4711 100". Internally, the phones can be reached via their full number or via their extension only. The registration with the SIP server and the SIP Proxy in our case here is performed with the full number. By knowing the Own Number of the branch, the Proxy upon receiving a call for an extension in Survivability mode can now prepend the number of the branch and hence find the phone in the Registration database. This means that extension dialing still works in Survivability mode. This can be used to avoid having to do number manipulation in the gateway for incoming calls. If you do not want to use this feature, simply leave the field blank.

Prefix number

The Prefix number describes the prefix the number has if it is an outgoing call. Usually this is just a simple digit as for example a "0" for getting an outside line. Expressions for this can however be more complex than that. You can enter arbitrary regular expressions here. Only numbers having the correct prefix will be routed to the gateway. This prevents unavailable internal numbers from ending up in the PSTN and going somewhere unintentionally.

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Figure 59: Standalone SIP Settings

Timeout

The timeout value is the maximum time the proxy will wait for a response from a gateway upon a request before falling into hunting mode. If a negative response is received earlier, the hunting will of course commence sooner.

Enable Hunting

This feature only applies to scenarios where there are multiple gateways present. Then it might be desirable to perform a gateway hunting between these multiple gateways. This can be used for a variety of scenarios as for example stacking gateways or having gateways in multiple locations and applying Call Admission Control parameters to them. The decisive parameters for the gateway hunting are configured with the gateways. If this parameter is not set, only the first gateway will be considered for outbound calls. The others might however still be used for incoming calls since the SIP Proxy will of course handle their SIP requests just like he would do with those of any phone.

Always proxy RTP traffic

Check this box to force RTP traffic through your Mediatrix Boss. This box has to be checked if NAT handling is done in your network.

PBX Addresses/ Failover Convergence IP

If you have a second Mediatrix Boss available for a failover scenario just enter its IP address or host name here.

Port Settings


Domains

See “Domains” on page 71 for detailed information.

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Gateways

See “Gateways” on page 72 for more detailed information.

Figure 60: Standalone Diagnostics

Debug

This switch will make the SIP proxy write debug messages to the system log. These can either be seen in CLI mode on the Mediatrix Boss, or on the specified logging server. See the logging parameters in this document or refer to the Command Line Guide for more documentation on how to have a look at the logging.

ENUM Template

The ENUM (E.164 NUmber Mapping) template is relatively straightforward to explain. In the typical usage scenario the Mediatrix Boss is externally connected to the Internet and internally to a SIP Server / IP PBX . The Mediatrix Boss acts as an ENUM lookup server and as a Session Border Controller for incoming calls.

The IP PBX must be configured to forward all calls to the Mediatrix Boss appliance. The Mediatrix Boss will try to convert the dialed phone number into an E.164 number to perform an ENUM lookup. If it successfully finds a SIP URI registered under this number, it will forward the call via IP. Otherwise it will signal back a trunk busy to the IP PBX that should in turn try the next available trunk meaning it will route the call as before via PSTN. In that scenario, the Mediatrix Boss can be integrated into the existing Voice infrastructure. The Mediatrix Boss must of course also be configured to permit incoming calls, by opening the firewall etc. Also see the firewall section for more details on this.

There might be other usage scenarios, where the ENUM template can be useful. An extensive discussion would however be beyond the scope of this manual.

Basic Settings

The following parameters are all used to convert a telephone number into a full E.164 number.

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Enable Hunting

City Code

Enter the plain city code 3-digits in the US and other countries. It can also be a number of variable length in many countries.

Country Code

Enter the Country Code for your country without any preceding digits. This would be a "1" for the USA, the "49" for Germany, etc.

Long Distance Prefix

Enter the number you have to dial on your phone before making a long distance call. In the US this is a "1", in most European countries a "0".

International Prefix

Enter the number you have to dial on your phone before making an international call. In the US this is the "011" in most European countries the "00".

Figure 61: ENUM Basic Configuration

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Figure 62: ENUM SBC Configuration

PBX Addresses

PBX Address

This parameter is for entering the address of the connected IP PBX. It can either be entered in the form of an IP address, a host name or even as a DNS SRV address. Note that entering a DNS or DNS SRV name can introduce more latency in forwarding messages to the IP PBX since the hostname has to be looked up first.

SIP Transmit Type

The SIP Transmit Type parameter is the transmission type the proxy uses to contact the IP PBX. The Mediatrix Boss supports 3 different SIP transport modes, UDP, TCP, and the encrypted TLS variant. The type to use depends first and foremost on the used IP PBX. The TCP and UDP types are straightforward and just specify the used protocol type.

The only type requiring further explanation is TLS. For TLS additional things like the encryption keys are required. Since this is a very advanced topic, this cannot be configured via the WebGUI as of yet. The Mediatrix Boss will automatically generate TLS keys which can be downloaded for the use in the IP PBX via SCP. Please consult the Command Line Reference for more information.

PBX Port


Port Settings


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SBC Settings

Internal IP

Since the Branch SBC will do media handling if a SIP session leaves the boundaries of the local branch office, the Mediatrix Boss needs to know the internal network address for which it should not be doing media handling. For all session where both the source and the target are within this network, the media streams will continue to flow directly between the endpoints.

External IP

All Mediatrix Boss products support multiple virtual and non virtual interfaces. However, when acting as a branch session border controller, there must be a unique uplink interface through which outgoing media streams should be routed. When looking at the SIP messages that are leaving the branch SBC, all SDP bodies will contain this external IP address, and hence all endpoints having a connection with another endpoint inside the Branch SBC will be sending their media streams to this external IP address.

Debug

This switch will make the SIP proxy write debug messages to the system log. These can either be seen in CLI mode on the Mediatrix Boss, or on the specified logging server. See the logging parameters in this document or refer to the Command Line Guide Section for more documentation about how to check the logging output.

Custom Template

The custom template allows you to use your own configuration script for the SIP Proxy. Please refer to the Command Line Reference on more information on how to create your custom SIP Proxy configuration script. It is also possible to use one of the above defined scenarios, and then modify the script according to your needs.

Registered Users

This menu item lists any currently registered SIP proxy / Session Border Controller users. A user is represented by its URI and name, where the name may be an alphanumeric string or a telephone number extension.

More details on registered users are provided by the SER server command serctl ul show. Please refer to the Command Line Guide, in particular “Control Script: serctlc” on page 176, for further information about the usage of serctl.

Settings for SIP proxy templates are described extensively above.

SIP TLS Configuration

The SIP TLS configuration page is a simple tool to upload a TLS-private key and a TLS certificate used to communicate with the client in TLS mode. Naturally this is only relevant if you have a client that is able to support SIP in TLS mode. Otherwise you can ignore this section.

Note: The relevance of this information varies quite significantly. In Survivability and Branch SBC mode, this information is only utilized for Survivability functionality and plays no role in normal mode. For the Cascaded SBC and SIP Trunking topics this information is not used at all. However. for the SBC functionality this information is quite important, because it is used for the NAT detection.

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The Mediatrix Boss generates a new private key and a certificate automatically while booting if there is no existing private key or certificate. Therefore, if you do not want to setup authentication based on certificates, there is no need to do any configuration here.

Figure 63: SIP Proxy Users

Figure 64: SIP TLS configuration

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Upload Private Key and Certificate

To upload a new TLS-Private Key press the Browse button and select the new private key from your file system (Figure: Figure 64 on page 99).

In the next step press the Upload button and the new TLS private key will be saved onto the Mediatrix Boss. The procedure for the TLS certificate is essentially the same.

Create default private key and certificate

If you do not have any keys ready to use for TLS communication, you can also use the Mediatrix Bossto create a key set. Simply press Generate and the Mediatrix Bossgenerates a new default TLS private key and TLS certificate. These will be stored on the Mediatrix Boss and activated after pressing Apply Configuration.

B2BUA

The Mediatrix Boss has a built-in SIP B2BUA (Back-to-Back User Agent) which can be used for implementing a number of SIP scenarios. These scenarios include SIP-Trunk Termination, connecting to other SIP equipment (IP to IP gateway), Internet User Integration and others. There are also some implicit uses of the B2BUA, as for example in conjunction with the Survivability scenario where the B2BUA is used for providing feature interworking for not fully RFC3261 compliant SIP gateways, that cannot coexist with the direct configuration of the B2BUA. Another example would be the Autoattendant and the Conferencing.

The standard uses of the B2BUA are scenarios involving SIP carriers. SIP carriers in the sense used here are SIP-Trunking scenarios that require or do not require a SIP registration. As described in this manual there is another possibility to support this, by configuring SIP Trunking in the menu item Survivability and SBC Settings. The SBC should be used if a PBX or SIP server is used that natively actually supports SIP Trunks and just requires an intermediary for Security and NAT handling. The B2BUA should be used for scenarios where the SIP Trunk is embedded into a larger scenario as for example a Survivability scenario, where the B2BUA makes the SIP Trunk appear like a standard-SIP gateway.

As already mentioned, contrary to the SIP Trunk SBC scenario, the B2BUA can actively register with the SIP Carrier. Especially the smaller SIP Trunks require registration because they usually expect a UA (User Agent like a SIP phone or a Softclient) to be on the other side. The B2BUA can emulate a phone being permanently connected to the SIP carrier. Incoming calls for a SIP carrier can then be mapped to an arbitrary number of users or even to the SIP Proxy and hence an IP PBX. As such, it is possible to have the B2BUA also act like a virtual gateway and handle the necessary number conversions.

The B2BUA also provides a way for integrating Internet users into a larger scenario as for example a Branch SBC or Survivability scenario. In such a setup, employees can be allowed to connect via e.g. a Softphone over the Internet. The B2BUA will do the NAT handling in case the phones are connected from behind a firewall. The B2BUA will terminate the call coming from the outside and setup a new internal call. This can be desirable for security purposes if you are having e.g. mobile users or home workers that you do not want to connect via

Warning: The TLS certificate must match the TLS private key, otherwise no TLS communication will be possible.

Note: ABOUT THE B2BUA

• The B2BUA cannot be used in conjunction with the configuration of B2BUA mode in Survivability mode.

• The B2BUA cannot be used in conjunction with the Autoattendant.

• Contrary to the SIP Proxy, the B2BUA does not only handle the SIP signaling, it also handles the media sessions. It is therefore a lot more relevant which types of sessions are being initiated. The B2BUA must also support the media codecs. Due to these reasons, the B2BUA can only be used for Voice scenarios at the moment.

• The B2BUA can also be used to implement branch connection scenarios. In those cases it is for example possible to implement an office to office trunk between two Mediatrix Boss systems. In order to accomplish this, you have to configure one side as a SIP peer, the other as a carrier.

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e.g. VPN. This however will also restrict the number of features that can be used, since the B2BUA will not pass proprietary signaling to the other side. The B2BUA therefore should not be confused with a full SBC. If you want to configure full Internet access for several users, please refer section “SBC (Session Border Controller) Template” on page 83 for a more detailed description of this topic. The B2BUA is actually intended for more complicated scenarios in which you might want to connect one or maybe a few Internet users.

Configuration Concept Introduction

As you can see in “B2BUA” on page 100, there are three different sections to configure, the Carrier Accounts, the SIP Peers, and the Call Routing rules. SIP Carrier accounts are usually SIP trunks either with or without configuration. Calls coming in over these SIP Trunks, are routed using the Call Routing rules. These calls will be routed to a SIP peer, which can be either a SIP endpoint such as a phone or a gateway, or it can also be a PBX or SIP Server. The call routing however also allows you to configure calls coming from the associated PBX or endpoint to be routed through the SIP carrier account. As such, the call routing rules define a mapping between users and the SIP carrier accounts (see “Call Routing Rules” on page 104 for more information on the configuration). The following should give an overview over the configuration options.

Figure 65: The Back 2 Back User Agent

General Options

Active

This box must be checked for the B2BUA to be active. You need to perform the further configuration as described below in order for the B2BUA to actually handle calls. Note that the B2BUA runs on port 5070. If direct outside access is to be supported, please make sure to open the correct ports for the respective interfaces on the firewall, or use the Layer 7 settings for generally allowing SIP and RTP on the WAN interface.

Note: The default port for the B2BUA is always port 5070! Once set active, it will listen on this port. If nothing more is configured all calls will be rejected on the SIP layer.

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SIP Carrier Accounts

On this page it is possible to define a SIP carrier account. The account can be defined with or without active registration. SIP carriers vary in their terminology, i. e. how they call the various required settings, so we will try our best to explain which setting needs to be put where. As a first step, you should collect all information you have about your SIP carrier.

It is possible to have a whole list of SIP carrier accounts. Simply press Add, proceed by filling out all necessary fields, and then save your changes. In the following, we will explain the meaning of the different fields that are available.

Phone Number

The phone number is the number given to the user by the SIP carrier usually without any prefixes. This should be the number the user is available on within the SIP network of the carrier. On a SIP layer, this setting will be used in the From header of all call requests. Typically on a carrier requiring registration, this should match the Registration Number field.

Registration Number

This setting only applies, if you are configuring a SIP carrier with registration.

The Registration Number is usually the phone number assigned by the SIP carrier without any prefixes. As such, it should be identical with the Phone Number setting in virtually all practical scenario. In some rare cases, however, the registration must be set to a different value. You should therefore understand that his setting only applies to the registrations. It does not apply to any other SIP messages. Similarly to the Phone Number setting, this parameter will be used in the From header of the SIP Registration messages.

Username

If no authentication is done, this field can be left empty.

If authentication takes place, enter the Username for your SIP carrier account here. This Username parameter is used for authenticating you with your SIP carrier. Every SIP carrier usually requires authentication which means that you must enter a value here. Not entering a value will cause the system to malfunction. This username is utilized in the SIP-digest authentication and will determine the challenge sent by the server. This means that you will usually also have to set the password if setting a username is required. With many SIP carriers the username used for authentication is identical to the Phone Number setting.

Password

Enter the Password for your SIP carrier account here. Please also refer to the Username explanation for details as to whether this field is required. If digest authentication is required by your SIP carrier either during the registration as well as for making calls, you need to set this value.

Note: Even though this is called a number here, you should note that this can actually be any combination of alphanumeric characters, so a username would also work. Most carriers however use just plain numbers for this.

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Host

The host is the SIP server provided by the carrier. Usually a hostname will be provided, but it can of course also be an IP address. The SIP server is also often called Registrar, Registry, or Proxy Server.

Port

By default all SIP traffic is sent to the default 5060 port of the configured host. However, some SIP carriers use non-standard ports (mostly due to security). In those cases you need to specify the port to send all requests to. If left blank, the mentioned default port is used.

Send Registration

If your SIP carrier requires an active registration, you need to check this option. This will enable the periodic sending of a REGISTER request towards the SIP carrier. If the registration fails, the SIP carrier will also usually prevent making any calls. Incoming calls also depend on a working registration. The easiest way to debug registration problems is either an access portal by your SIP carrier to help you determine if the Registration was successful, or a network trace.

SIP Peers

SIP-user accounts (also called SIP Peers here) are very similar to SIP-carrier accounts. The difference is that the Mediatrix Boss usually acts as a client towards the SIP Carrier, while it acts as a Server towards a SIP Peer. This means a SIP Peer can be a Gateway, a User Agent, or another B2BUA behaving like a client in those cases. The information configured here must be used as well for configuring your SIP UA (User Agent) device, Softclient, or Gateway. As with SIP carriers, it is possible to manage a practically arbitrary number of SIP Peers in this section. Click Add then fill in the desired parameters and save your changes. In the following, we will explain the meaning of the different parameters:

Phone Number

The Phone Number specifies the number you want to allow a device to register with. It is common to use the same number for registration that will be assigned to the device in the number plan. It is also legal to use an alphanumeric string here, but the specification of a simple number is much more customary.

Username

Enter the Username the phone has to use for authentication. If you leave it blank, any device will be allowed to register without being challenged. Please note that you also should set a password, otherwise anybody knowing the username will be able to register.

Note: HOST ADDRESS

The host address is used for sending both the Registration requests as well as the INVITE requests for initiating a call.

This is a mandatory field. It has to be configured in order for the SIP Trunk to work. When using hostnames also make sure that the DNS is correctly configured to resolve this name. You can check this by entering the hostname or FQDN (fully qualified domain name) on the Diagnostics page and try to ping it. While the ping might fail, the Mediatrix Boss should at least be able to resolve the IP address which you can see in the output.

Note: If problems appear, please check the Host, Registration Number, Username, and Password fields carefully for completeness and correctness. Correct registration is highly depending on these values.

Note: You can leave this field and also the field in your UA blank. Your UA will then be able to connect without authentication.

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Password

Enter the Username the phone has to use for authentication here. You can leave this field and also the field in your UA blank. Your UA will then be able to connect without authentication by simply specifying the correct Username.

Host

Enter an IP address or hostname if your UA is using a fixed-IP address and does not dynamically register. When entering this IP address, the B2BUA will always try to contact your UA under the IP address specified. It will completely ignore the data sent in the Registration request. If your UA is not sending SIP REGISTER methods, you have to fill in this parameter.

Call Routing Rules

Having configured the SIP Peers and the SIP carrier accounts, call routing rules have to be defined now for the handling of calls. These rules will be used for making the appropriate connections. Besides routing calls among the configured Carrier Accounts and SIP Peers, you can also route calls towards the SIP Proxy / SBC component. When doing this, the B2BUA starts acting as an IP-to-IP gateway when considered from the SIP Proxy. It will receive calls from the B2BUA component very similarly to receiving calls from a SIP gateway.

You have to understand that each rule defined here only applies to a single direction. For instance: if we define a rule that all numbers starting with a "0" should be routed to the SIP carrier account, this does not define what to do with calls coming through the SIP Trunk. In order to accept calls from the SIP carrier account, we need to define a second rule. The structure of the rules is very simple. You simply define a Source and a Target for each rule.

Target Definition through Extension

The basis for each call routing rule is an Extension. The extension is essentially the called number and what you are defining here can most aptly be described as a number plan. First a match on the extension is done. If it matches, the source is checked. If both match, the call is forwarded to its destination.

Extension names are not limited to single specific extension "numbers". A single extension can also match patterns. In the source and target patterns, an extension name is a pattern if it starts with the underscore symbol (_). In an extension pattern, the following characters have special meanings:

Special Characters for Pattern Matching

X Matches any digit from 0-9

Z Matches any digit from 1-9

N Matches any digit from 2-9

1237-9 Matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9)

. Wildcard, matches one or more characters

Source

Enter the number the call is coming from. This can be either a SIP carrier account, or an Extension. Looking beyond the B2BUA, there are two more potential sources, the SIP Proxy, and Everywhere. If the latter parameter is selected, the Source will simply not be checked.

Warning: ABOUT HOST CONFIGURATION

You should not configure this field if your peer registers dynamically.

When using hostnames also make sure that DNS is correctly configured to resolve this name. You can check this by entering the hostname or FQDN (fully qualified domain name) on the Diagnostics page and try to ping it. While the ping might fail, the Mediatrix Boss should at least be able to resolve the IP address which you can see in the output.

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To

Choose either a SIP Peer or a SIP carrier account. Another option is to route the call to the SIP Proxy. In that case, you have to specify the number of the extension connected to the SIP Proxy that the phone should be routed to. For that, it is of no interest in which mode the SIP Proxy is configured. If it is for example configured in Survivability mode, the call will be passed on to the central SIP server, then from there forwards to the phone with the number entered in the Number field.

Number (only for SIP Proxy as target)

The Number to be dialed if a call is forwarded to the SIP Proxy. See the description of the To field for more information.

Prefix

The number specified here will be prefixed before the call is forwarded to its destination. This parameter is optional. If left empty no prefix will be added and hence only stripping will be done (if configured) before the number is forwarded to the destination.

Strip

Specify the number of digits to be stripped before routing the call. This field is optional and will be applied before the prefixing.

Diagnostics

By activating this check box you can have your Mediatrix Boss write additional debug messages concerning its B2BUA component into its syslog file.

Autoattendant Basic Settings

There is a small media server built into the Mediatrix Boss which offers the Autoattendant functionality. This allows you to create fallback telephony announcements and conference rooms to enable multiparty conferences, and even hunt groups.

In addition to the announcements, voice menus with prompts and DTMF selection menus can be created. As such, it is also possible to create an announcement to ask for an extension, then dial the read extension towards the PBX just like a regular gateway. There are numerous possibilities to catch erroneous inputs, catch no inputs, repeat announcements, etc.

The Basic Settings introduced in the following affect all submenus for the above mentioned topics.

Basic Settings

On the Basic Settings Page for the Autoattendant (see Figure 66 on page 106), it is possible to activate or deactivate the Autoattendant and configure the basic settings. These include the following parameters:

Activate

This item should be checked to activate the Autoattendant. If unchecked, the Autoattendant will be completely turned off. Please note as well that the conference rooms will be available if the Autoattendant is turned off.

All changes made while the Autoattendant is running and if APPLY CONFIGURATION has been executed once, new changes made will be applied by just clicking the save button.

IP PBX

The Autoattendant behaves like a virtual gateway. This means that if it is used for implementing e.g. extension dialing, the Autoattendent must forward the call to the PBX after having read the extension via DTMF. The Autoattendent will always do this via UDP SIP. If connected behind the Proxy, the signaling towards the PBX can be determined by setting up the SIP Proxy part of the Mediatrix Boss appropriately.

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Port

Enter the port number of the central IP PBX here. The SIP message will be addressed to this port. Please note that this setting will be overridden by the settings in the SIP Proxy if the "Use Local SIP Proxy" option is checked.

Use Proxy

In a typical scenario you will select "LOCAL" here. This forces all SIP messages to be sent through the Mediatrix Bosss own SIP Proxy component. Note that the Mediatrix Boss must be configured accordingly.

If "EXTERNAL" is selected here, all SIP messages will be forced through the Outbound Proxy and Port which you can configure in below this menu item.

Outbound Proxy and Port

If "EXTERNAL" is selected for the "Use Proxy" option (see description above), you have to enter its IP address and port here. If "LOCAL" or "NONE" is selected for the "Use Proxy" option, settings made here will remain without effect.

Figure 66: Autoattendant: Basic Settings

Announcements

The Announcement page allows you to configure announcements, voice menus, as well as more complex voice menus including user interaction. Refer to Figure 67 on page 107 for more information. As you can see, this component consists of different sections:

Announcement Extensions

The entire Autoattendant is based on the notion of extensions. Every announcement is uniquely identified by a virtual extension that you can jump to. In this menu the announcement to be played back if a certain extension is called, can be configured.

Note: The Autoattendant must be configured to Use Local Proxy if the Survivability functionality has to work in conjunction with the Autoattendant. This is the only thing that has to be configured, however. If checked, incoming calls will be answered by the Autoattendent in both normal and Survivability mode. By forwarding these calls through the SIP Proxy / Survivability component, these will also be handled in both modes of operation.

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Configure Prompts for Announcement Extensions

After playing back an announcement, actions can be set to follow it. This means that the user can be prompted to enter something. This interaction concept can be defined in this section.

Configure Active Times for Announcement Extensions

It is possible to configure a timer control for the announcements, e. g. different announcements depending on the time of the day, etc.

To add something, click Add under a table.

Announcement Extensions

Name

The first text-field Name is for the announcement name, it may not be empty and every name may be assigned only once. The Name is the unique identifier for the configuration and used as the reference to add the right voice menu to the defined announcement and servers.

Extension

The second field Extension contains the phone number, of the announcement. As mentioned previously, the Autoattendant is based on the notion of virtual extensions. Whenever a certain extensions is being called, the associated announcement is played back. It may not be empty and accepts only numbers.

Announcement

Select an announcement from the drop-down box Announcement. Additional voice announcements can be uploaded in the submenu /textitUpload Announcements.

Figure 67: Autoattendant: Announcement, Announcement menu and Active Time Configuration

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Delay Until Action

The field Delay Until Action... contains the time in seconds the is to wait server after an announcement before the following action (Hangup or Repeat) is executed. If the field is empty, the default value of 5 seconds will be valid.

Action

There are two different actions that can be configured after an announcement was played back:

Hangup

The call will be hung up after the announcement and the above described timeout unless a prompt is configured and the user enters an appropriate extension within this period of time.

Repeat

If the timeout configured in Delay Until Action expires, the announcement will be repeated. This cycle repeats indefinitely until the user hangs up.

Recurrences

This field is invisible if Action has been configured to Hangup. If the field Delay Until Action equals Repeat you can define the number of recurrences of the announcement in this field. The default value is one recurrence.

Configure Prompts for Announcement Extensions

This menu enables you to configure responses to user action.

It is possible to configure voice prompts in this table (Configure Prompts for Announcement Extensions) to initiate an action if a user presses a digit on his phone. The values you can use are either pressed digits and error conditions.

Announcement Name

This allows you to select the announcement you want to configure the user interaction for. It will give you a list of all your configured announcements.

Warning: The user's interaction has to be done with DTMF tones. These tones in turn are expected in RFC2833 format. This is the default setting of our Integrated Voice Gateway products. If you use external Voice Gateways or phones, please make sure they are properly set up.

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Figure 68: Autoattendant: Add an announcement menu

Action

a. Jump To

Jump To will connect the user to another announcement or conferencing extension. Refer to the Transfer Target for more information on which announcement the system will go to. You should also note that you can define a condition (via User Entered Number.)

b. Read Conference Number

This will read and validate a conference number. If the number is configured, you will be placed into the selected conference room. If this function is used the values in the drop-down box Read Target Number and Transfer Phone Number are ignored.

c. Read Telephone Number

This setting is for configuring the extension dialing. All other fields will be disabled and the Autoattendant will send the read extension to the PBX. If this function is used the values in the drop-down box Read Target Number and Transfer Phone Number are ignored.

User Entered Number

You need to select a condition here to which user input to respond. You can only select a single digit here. In addition to that you can specify error conditions.

Transfer Target

You can specify a fixed connection target here. There are two types of Transfer targets:

a. Target Phone Number

This will use the phone number specified in the Target Phone Number to be dialed on the PBX.

Warning: For a well functioning Autoattendant you should always configure the error condition because the default is to hang up on error. This is not the most user friendly approach. You should try to configure an announcement if an error occurs.

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b. Autoattendant Extensions

There will be a list of all configured announcements. You can select any and the Autoattendant will jump to this extension.

Transfer Phone Number

This parameter enables you to configure a fixed number to be dialed on the PBX. This number will only be dialed if the Transfer Target and Action parameters are configured to Target Phone Number and Jump To respectively.

Active Times

Figure 69: Autoattendant: Time circuit for an announcement

This functionality enables you to activate and deactivate announcements controlled by a timer. This is useful for the implementation of features such as replaying a special announcement about the business hours at night etc. The following settings are possible:

Announcement

Select the announcement the timer should work for. The drop-down box will only list the already configured announcements.

Start Hour

Enter the hour when the announcement should be activated. Please note that you have to enter the number in 24h format, i. e. a number between 0 and 23.

Start Minute

Enter the minute when the announcement should become active. Enter a number between 0 and 59.

End Hour

Enter the hour when the announcement should deactivated. Please note that you again have to enter the number in 24h format, i. e. a number between 0 and 23.

End Minute

Enter the minute when the announcement should be deactivated. Enter a number between 0 and 59.

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Start Day

Set the day, when the announcement should be activated. If you leave this setting blank, the announcement will be active everyday. If you do decide to set this parameter, you also need to set the End Day parameter.

End Day

Set the day for the deactivation of the announcement. Please note that just like the Start Day parameter, this setting is optional. Leaving it blank will lead to the timer being executed on a daily basis. Please note as well that this parameter only makes sense if the start parameter has also been set.

Figure 70: Autoattendant: Conference

Warning: TIMERS

There may be several timers configured for a single announcement. However, if there are several timers configured for a single announcement, they may not overlap! Overlapping timers will lead to unpredictable behavior. The start and end days for a timer must be in the same week and in the right order. Fri to Tue for example would be invalid!

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Chapter 7 - VOICE Conferencing

Conferencing

The feature Conference room allows you to create local conference rooms on the Mediatrix Boss (compare Figure 71 on page 112). This conferencing works independently from the phone and acts just like a media server. For the PBX, the Autoattendant Conferencing can behave like a gateway or a local user.

The first table Conference rooms contains statically defined conference rooms. In the following table Moderator you can create a moderator for a conference room.

Create a conference room

Figure 71: Autoattendant: Create a conferenceroom

Room Extension

The text-field Number contains the phone number of the conference room. It may not be empty and accepts only numbers. This is a virtual phone number that is assigned to a conference and can be called from remote. It can also be used to make connections from the Autoattendant.

Direct Inward Call

If direct inward dialing is allowed, you can e.g. configure your gateway to terminate certain calls on the Autoattendant. When configuring the correct extension (also see the Room Extension, you will be placed into the Conference room right away. If this is not checked, it is not possible to offer this directly. Entering a Conference Room is then only possible via the Autoattendant.

Default Member Type

In the Member type drop-down box you have the possibility to select "Speaker" to actively participate in a conference, or "Listener" to be just a passive member in the conference. In order to use the latter feature only, you need to select "Listener" as default type. Consequently, all active participants must be configured as moderators in this conference for a member is automatically just a listener.

Authentication

If authentication is desired for the members, the check box Authenticate must be activated. Everyone wanting to participate in the conference will then be asked to enter a password.

Authentication PIN

Every participant of the conference will be asked for a PIN number in advance. To enable is, the Authentication feature has to be activated.

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Allow Moderator

In order to add moderator support to the conference the check box Moderator must be activated. A conference will not begin unless the moderator joins. Until that time, all members will hear Hold Music. Similarly, the conference ends whenever the operator closes the call or even drops out of the conference. Also refer to the Default Member Type features for more information.

Create a conference moderator

To see the options available in the Conference Moderator configuration please refer to Figure 72 on page 113.

The moderator configuration is for differentiating moderators from ordinary participants in a conference. The following options can be configured here.

Selected Conference

Select a configured conference room for conducting the moderator configuration.

Figure 72: Autoattendant: Conference room with moderator

Direct Inward Call

This enables the direct access to the conference room via extension dialing e.g. from a Voice gateway. Please note that the PIN query will be made even if this feature is activated.

Authentication PIN

This authentication PIN should be different from the normal users' PIN. Only then, the system is able to differentiate ordinary participants from moderators.

Query Conference Participants if Moderator

The drop-down box Moderator query allows to select an announcement to be replayed before entering the conference, e. g. (Would you like to enter the conference as moderator or as member?). This announcement is only used if the moderator feature is enabled.

Note: Conference rooms can be used as follows. If a password is configured, the user will be prompted for the password when entering the room. If the password was correct, press the pound key or you can wait few seconds to continue.

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Chapter 7 - VOICE Voicemail

Voicemail

The Mediatrix Boss can be used as a fallback-voicemail server (Survivability) or the standard voicemail server (Branch SBC). Voicemails are limited to 5 minutes.

Figure 73: Configure Voicemail Accounts on the Mediatrix Boss

Centralized Voicemails

1. Voicemail Number: enter the number that leads to the recording of voicemails. This number must be configured as a PBX Extension Registration (next menu page) and it must be configured as the voicemail diversion on the phone.

2. Listen Number: the number that a user has to dial in order to listen to a recorded message has to be entered here. This also has to be a configured PBX Extension Registration in next menu page.

3. Compare last digits: Enter the number of digits (starting from the last) to check the calling parties number with the Configured Voicemails.

4. Use Password? Check this box if authentication by password is required. Please note that a password has to be entered in the next field. If the next field is empty, this setting will be treated as "No".

Configure(d) Voicemails

1. User Name: this name only serves descriptive purposes. Any name can be used.

2. Password: enter the password that has to be used for authentication here. Please note that the value you enter here, must only consist of standard phone keys (DTMF).

3. Phone Extension: The internal phone number of the user.

Note: Please note that this menu is not available on all products of this series.

Warning: This functionality depends on the Autoattendant to be configured and active. It may be used for the Survivability and the Branch SBC template.

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PBX Extension Registration

On the PBX Extension page you can configure mappings between local extensions and PBX registrations. As a requirement you must configure the PBX IP and Port on the Autoattendant Basic Settings page in order to be able to use the PBX Extension Registrations.

If activated, the Autoattendant will register the specified extensions as standard endpoints on the PBX. If called from the PBX, the associated extension, either an announcement or a conferencing room, would pick up. While it is possible with most PBXs to define static registrations, this mechanism provides a very simple way of e. g. entering a conferencing room. The conferencing room will thus simply receive an internal extension of a regular phone and can hence be called like one.

Figure 74: Autoattendant: PBX Extension Registration

You need to configure the following fields:

Local Extension

The Local Extension is the list of locally available extensions. Choose one to associate with a PBX registration.

PBX Number

The PBX Number is the URI used to register with the PBX. In most cases this will simply be a phone number. Please note that you do not have to enter the host part here, this information is taken from the basic settings.

User Name

The username configured on the PBX for authentication purposes. On many PBXs, this username is equal to the PBX Number. If you are not sure what to use here, try the PBX Number and see if the SIP registration succeeds.

Password

The password configured on the PBX for this user in order to digest authentication.

Realm

Enter the realm for which these credentials (User Name and Password) shall have validity. The Realm setting is mandatory, because it is possible to reuse User Names for different Realms.

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Chapter 7 - VOICE Hunt Group Configuration

Hunt Group Configuration

The field of application of Multiline Hunt Groups has already been explained for the Survivability functionary under “Survivability and Call Admission Control Template” on page 60. In this Hunt Group menu you have the possibility to configure hunt groups which will be responsive during normal operation. The submenu in Survivability and Features can be used to redirect calls to hunt groups usually configured on your PBX. Please keep in mind: In Survivability your PBX is not available and therefore cannot provide this call-forwarding functionality.

Conclusion:

The Hunt Group functionality can be used to create constantly active MLHGs, as well as to create fallback MLHGs, by redirecting calls made to MLHGs on your PBX if your PBX is currently unavailable. As already mentioned, for the latter case the submenu in Survivability and Features has to be configured accordingly!

The following should give a brief introduction into the configuration options of a hunt group.

Figure 75: Hunt group

In order to configure the hunt group functionality, there are two types of configuration options to consider. You can choose between configuring Simultaneous Ringing groups and Queuing Groups.

Simultaneous Ringing

This section enables you to configure standard hunt groups with simultaneous ringing, serial ringing and any combination of the two.

Name

Each Hunt Group must be given a descriptive name. This name is not actually used in the configuration, it is for distinguishing the different groups during configuration.

Number

Configure the Number for your hunt group. All of the Telephones that you configure to be a part of this group, will then be available on the number configured here.

Call Type

The Call Type determines the algorithm with which an incoming call should be distributed to the telephones. The parameter Ring All will ring all configured phones exactly once in the configured order. The type Round Robin will go from the first to the last, and then start over. If parallel ringing

Warning: Since the hunt group uses the same functionality as the Autoattendant and the Conferencing, each number assigned here must be unique. This means that you must not have e.g. a conference room with this number.

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is configured (in the Telephone field) for several stations, this will of course be respected. The algorithms Round Robin 1 and Round Robin 2 will repeat the round robin behavior once and twice respectively.

Telephones

This is the most complicated part of the configuration. In a Survivability scenario, this will simply be the numbers the stations register with on the proxy.

There are two different separators, / and - . A slash: / indicates that the phones it separates are to be rung simultaneously, whereas a dash: - is used to separated serial ringing groups, i.e. phones ringing after one another.

The following example should better illustrate this.

100/200/300 - 400 - 500 - 600/700

This will first ring the phones 100/200/300 simultaneously. If those do not pick up, it will ring the 400, then the 500, then the 600 and 700 simultaneously. Anybody picking up the phone will immediately stop this process.

Timeout Group

After the timeout (seconds) has passed a call will either be rejected, or it will be forwarded to the next call group (in a serial hunting scenario). Unless Round Robin is selected, the last call group will be tried for as many seconds as the Timeout allows. After that, the calling party will hear a busy signal and the call will be terminated if no more call groups are available.

Call Queuing

The configuration Call Queuing Groups is done exactly like the groups for Simultaneous Ringing. There is one additional setting called Timeout member which specifies the call-forwarding interval (seconds) from group member to group member.

The difference is that all callers will first end up in a call queue. Only then, they will be distributed to the various Telephones through the configured algorithms. While in the call queue, all users will hear hold music. If a phone becomes available, the hold music is immediately stopped and the call is sent to this Telephone. If more than one station is available, it will follow the configured call distribution algorithm.

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Chapter 7 - VOICE Upload and Deletion of hold music

Upload and Deletion of hold music

Figure 76: Autoattendant: Upload for music-on-hold title

This page is for Uploading any Music you would like to have played back if a user is put on Hold, is being transferred, etc. There are multiple components in the Mediatrix Boss making use of this feature. The Autoattendent will play back the music while callers wait.

The button Browse shows the local file system on the local host. To upload select the file you want to upload. To upload the selected title simply press Upload (Figure 76 on page 118).

The gateway component will also use this music and play it back whenever a user is put on hold and will forward Hold signaling to the PBX. Please note that some PBXs provide music on a central media server, in this case, you do not need to configure the Hold Music here.

Warning: FORMAT

Uploaded MP3 files must have a special format if you want them to be played back on the PSTN side of the Voice Gateway. In those cases, the MP3s need to be encoded with

8KHz and mono. Towards the IP side, any MP3 can be played back, although quality might be affected if encoded differently. The recommendation therefore is to encode all uploaded MP3s in the above mentioned way.

Warning: ENCODING

The files must be encoded in one of the following formats:

• PCMA

• PCMU

• AU

• WAV

• MP3

• GSM

For ideal results, your announcements should be sampled at 8KHz, mono. If they are sampled in a different way, the Mediatrix Boss can still handle them but will need to do a transcoding on the fly which will consume unnecessary CPU cycles. From a performance point of view, the PCM codecs are ideal. They do however take up quite a lot of space.

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Upload and Deletion of Announcements Software Configuration Guide

Upload and Deletion of Announcements

With the Load Announcements page you have a simple tool to upload or delete announcements (Figure 77 on page 119). The first two items from the Load Announcements page are similar to the ones on the Load Music-On-Hold page.

Figure 77: Autoattendant: Upload announcement page

The four following items to overwrite an announcement file are only relevant for conferencing rooms with authentication. The Upload and the Browse buttons work like the upper buttons. The name of the new file is irrelevant because the name will be changed to the correct name automatically. These files cannot be deleted.

The files are used for the following purpose:

The sound file get password represents the question for the PIN when entering the conference room.

The thankyou sound file is played after entering the correct password.

If you submit a wrong password you will hear the incorrect number sound file.

After the third wrong password goodbye is played and the Mediatrix Boss hangs up the call.

Note: For ideal results, your announcements should be sampled at 8KHz, mono. If they are sampled differently the Mediatrix Boss can still handle them but will need to do an on-the-fly transcoding which will consume unnecessary CPU speed. From a performance point of view the PCM codecs are ideal. They do, however, take up quite a lot of space.

Warning: Please note that these fixed files have to be encoded either in GSM or PCMU/PCMA codec. By default English announcements are contained for this.

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Chapter 7 - VOICE Upload and Deletion of Announcements

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C H A P T E R

8 DIAGNOSTICS

Syslog-File

This page displays the last lines of the Syslog output the Mediatrix Boss produces. The output of the Syslog File file is explained in more detail in the Command Line Guide. The output also depends on the Syslog settings and the mode that is configured there (see “Logging” on page 122 for a more detailed description of the possible settings). If Debug is used there, a lot of information will be displayed. The output can be refreshed by pressing the Reload button. It is also possible to adapt the number of lines displayed, by entering the desired number into the Number of Lines field.

Figure 78: Syslog File

Warning: By default, the Syslog file is stored in a RAM disk on the device. This means that the information is lost after a reboot. In order to implement a persistent storage, an external Syslog Server is recommended. However, it is possible to move the log file to the flash from the CLI interface.

Note: The Mediatrix Boss performs log rotation on a regular basis to avoid filling up the storage space. This log rotation will compress the current syslog file. All files are stored in the /var/log/ directory and can be accessed e.g. via WinSCP.

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Chapter 8 - DIAGNOSTICS Logging

Logging

Figure 79: Logging

Specify an external syslog server here, on which all important messages of the Mediatrix Boss are logged.

Syslog is a standard for logging system messages under Unix and Linux and meanwhile also many other platforms including Windows. It supports both local and remote (i.e. over the network) logging.

The configuration of a Syslog server and appropriate log level are especially recommended if the Mediatrix Boss is operated as a firewall!

How to configure Syslog

1. Specify the IP address of your Syslog server

2. Then choose one of the following log levels from the drop-down box:

• Debug: All messages are logged.

• Info: All messages, but real debugging messages are logged.

• Notice: All messages important for the user are logged.

• Warning: All messages describing failures are logged.

• Error: All error messages are logged.

• Critical: All critical failure messages are logged.

• Alert: All messages preventing a service from functioning properly are logged

• Emergency: All messages preventing a service from functioning are logged

This log level determines which messages shall be logged.

3. Check the box "Local logfile" if you wish to save logs on your Mediatrix Boss (additionally to or instead of an remote Syslog server).

4. Check the box "Locally buffer CDR data" to activate the collection of call detail records (CDRs) on your Mediatrix Boss in survivability mode. A detailed description is available in the Command Line chapter, please refer to section “Call Detail Records (CDRs)” on page 179.

If there is no local computer available as a Syslog Server for saving the selected messages, you can either install a free syslog server available in the Internet, or you can use the local logging on the Mediatrix Boss. This logging is, however, limited to the space available there. In order to enable the local logging, simply enable the Local Logfile checkbox. For more information on the evaluation of such logs, please refer to the Command Line Guide for additional information.

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Debugging Tools Software Configuration Guide

Debugging Tools

The debugging tools allow you to easily track down the reasons for network connectivity problems over the web interface without knowing any CLI commands.

Figure 80: Debugging tools

The following tools are available:

Ping

Ping sends network packets to a computer and waits for the response while measuring the elapsed time until a response arrives.

This program is able to examine if a computer is reachable or not. For this reason it is applicable for instance to check whether the connection to a certain host is working. You can easily trace down the reason for connectivity problems by trying to ping the first gateway on the way, then the second and so forth. Once that fails, you will know exactly which link is the problem.

Traceroute

This program traces the path packets take from the Mediatrix Boss to a certain host. It can complement and expedite certain debugging approaches taken with ping. Please note that when trying to trace routes over the internet, some routers will block this attempt. You will then be unable to get the desired information for a traceroute although the network connection is working properly. Usually, traceroute is used after you have detected a specific network problem with the ping tool to further narrow down the reason.

The output is shown in real time, line after line. This means: if ping does not respond for several seconds, the probability that the address is not reachable is very high.

Note: If you assume that an internet connection is inoperable, you have to define the IP address of your Mediatrix Boss instead of your system name, because in this case the DNS service is likely to be unable to resolve system names.

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Chapter 8 - DIAGNOSTICS SNMP

SNMP

This menu item contains configuration of the SNMP Daemon.

The SNMP Daemon offers basic system information and a service to monitor the Mediatrix Boss system status. If you would like to manage the Mediatrix Boss remotely via SNMP, a read-write access has to be configured accordingly. This includes the surveillance of running processes, diskspace usage and the CPU's load average. The following parameters can be set up (compare Figure 81 on page 125):

SNMP Basic Settings

In this menu section you can switch the SNMP service on-/ off and configure the basic setup.

SNMP read community string

Specify a SNMPv1 or SNMPv2c community that will be allowed read-only access.

SNMP Read-/Write community string

Specify a SNMPv1 or SNMPv2c community that will be allowed read-write access.

Enable SNMP trap sending

This checkbox enables active monitoring. The SNMP Daemon will send traps if a process is not running or if diskspace is running low. Additionally a trap is sent when the SNMPD Daemon is started and stopped.

SNMP trap community name

Specify the community name which will be used in traps.

Trap destination

You can specify one or more ip-addresses here as a destination for SNMP traps.

Proxy

This option enables the SNMP Daemon to act as a proxy, passing certain requests to a SNMP service running on another machine. The following configuration settings are available:

IP address

The IP-address of the machine to which the proxy should forward.

Port

Optional: The port on which the target machine is listening for SNMP. Default is port 161

Password

The SNMP community string on the target machine.

SNMP Version

Choose between SNMP version 1 or 2c

Object-ID (OID)

Optional: Part of the OID tree which should be passed through to the target machine. Default is the whole OID tree .1.3

Setting up the SNMP daemon

1. Check Active to enable SNMP daemon

2. Provide configuration details for the basic setup as described above. Click Save to save your changes. Your changes will be applied after selecting Apply configuration.

3. In addition to the Basic settings you can specify one or more Trap destinations and Proxies to be used by the SNMP daemon.

4. Use Add to add an entry and fill out the form. Use to edit, or to delete an existing entry.

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Figure 81: Figure 8.4: SNMP Configuration

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Chapter 8 - DIAGNOSTICS Firewall Report

Firewall Report

This section shows how many packets have been dropped and which firewall rule they matched. You can see the overall number and size of dropped packets and additional information about the matched rule (protocol, input/output interface, source and destination). Depending on the direction of packets, rules are joined in three chains INPUT, OUTPUT, and FORWARD. Information about dropped packets may be helpful for testing connections or improving your firewall rules.

The information shown on this page is a summary of packets dropped by the firewall and will be refreshed every time you choose this menu item or click Refresh. You can also reset firewall packet counters by clicking Reset .

Figure 82: Firewall report

Network Trace

The Network Trace enables you to collect network information for error analysis and solution. This graphical interface for tracing uses tcpdump in the background. It is therefore possible to further modify trace parameters via the command line interface (CLI).

To improve data exchange between customers and our support team, the menu Support Trace has been designed. Please refer to “Support Trace” on page 128 if you are looking for this option.

Creating a new trace

1. From the drop-down box select a network interface to listen on. In order to gather information from all interfaces, choose ANY.

2. Additionally, you can input certain other options for tcpdump via the Command Line.

You can specify port numbers, protocols, the number of packets to be traced and many other options. A trace file created by tcpdump can then be viewed by a network packet analyzer such as Ethereal.

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Network Trace Software Configuration Guide

An introduction to the network monitoring using these tools would be far beyond the scope of this manual, but there is a lot of documentation available on the web. Some briefly explained parameters:

a. -s0

Usually only the first 64 bytes of a packet will be traced. This reduces especially the size of long running traces. Please note that traces especially on the LAN interface can get very large rather quickly if no filters are specified. The -s0 parameter will remove the 64 Byte limitation. The entire packet including the payload will be visible in the trace file.

b. port 5060

This will introduce a filter into the trace only for the packets with source or destination

c. src host 10.10.10.10

This will restrict the tracing to all packets coming from the specified host. The IP 10.10.10.10 is of course just an example.

d. dst host 10.10.10.10

This will restrict the tracing to all packets going to the specified host.

e. proto tcp

You could also use protocols ip, icmp, udp here. This will trace only the specified packets.

An example combining several of the above parameters would be for example -s0 port 5060. This would trace only SIP traffic, or to be more accurate all traffic on Port 5060.

3. Click on Start to start the trace. In order to stop and download the trace file, simply click on Stop and confirm saving the file to your disk. If you just want to check, whether the trace is already completed (e.g. in case you have specified a maximum size for files or a number of packets to be traced) click on Status . Your trace will not be interrupted, if it is still running.

Figure 83: Starting trace

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Chapter 8 - DIAGNOSTICS Support Trace

Figure 84: Downloading trace file

Support Trace

If there is a recurring error on your system and you need support, please record a Support Trace and let it run until the error has occurred again. This functionality will collect overall runtime information about your system and is very helpful in finding errors fast and reliably.

After starting the Support Trace, please wait approx. 15 seconds before making the first text call.

As soon as you click on stop, there will be an option to download the collected information.

If the memory is about to run out the information gathering will be stopped automatically. So if the error has not occurred and the memory is full, i.e. the support trace recording has stopped, please download the support trace (to eliminate it on the Mediatrix Boss) and conduct another support trace to capture the crucial information.

Figure 85: This is where the Support Trace is started and stopped

Warning: If the SIP Proxy is active, it will be restarted when a support trace is recorded. This can be checked on the Status page. The SIP Proxy is usually active unless your Mediatrix Boss runs in slave mode.

As a result of this restart you might have to make your phones re-register. Most IP phones, however, register by themselves in periodic intervals.

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C H A P T E R

9 Command Line Interface

The Mediatrix Boss does not only offer configuration via the Web interface but further provides access through a CLI (Command Line Interface).

You can access this CLI in three ways:

Using a SSH client

Using the serial interface (COM1, 9600-8-N-1)

Using the console (i.e. by connecting a VGA monitor and a PS/2 keyboard)

In every case the username is root and the password is identical to the password set in the Web interface.

You will get a Linux prompt that can be used like any other Linux computer.

The configuration via the CLI is primarily used for special tasks as resetting to factory defaults, adding custom scripts and for extended debugging.

Before introducing the possible direct access methods in the following paragraph, we would like to inform you about the boundaries of the CLI.

Restrictions using the CLI

While the command line environment is a plain bash shell there are nevertheless limitations on what can or should be done here. These limitations result from the approach of the Mediatrix Boss to

1. remain upgradeable, even when certain changes are done on the command line

2. remain compatible with the web interface

• Your modifications may disappear if you select Apply config in the web interface.

• Your modifications may disappear if you reboot your Mediatrix Boss.

• Your modifications may disappear if you upgrade to a newer version of the system.

• Your Mediatrix Boss may suddenly lack functionalities or functionalities stop working as desired.

• The security of your Mediatrix Boss might be compromised.

In any of these cases, the manufacturer does not take any responsibility for damage that occurs (especially security breaches) because of improper use of this command line interface.

So the restrictions are (unless explicitly permitted by manufacturer's support):

Don't modify configuration files, unless for special program packets that are not meant to be configured from the graphical user interface.

Don't modify startup scripts.

Don't modify parts of software programs.

Don't install your own software packages.

Don't change passwords (even if you find out how to do that).

Because almost all features of the Mediatrix Boss products can be configured via the browser GUI (and this is what you should normally do) all that is left that can safely be done on the command line is:

Modify configuration files for special program packets that cannot be configured in the WebGUI (e.g. traffic shaping).

Check error conditions more thoroughly.

Debug network connections with the provided tools.

Modify the central configuration file /etc/config/baseconfig, which contains all configurable options of the web interface. See chapter 9.4 for details on this file.

To avoid accidental editing of managed configuration file running vi or vim on such a file will result in a warning and ask you whether to continue1. If you find the need to edit a managed configuration file repeatedly you can avoid the warning by passing -n as first argument, e.g.:

vi -n /etc/openser/openser.cfg

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Chapter 9 - Command Line Interface Access Methods

Access Methods

The Mediatrix Boss is preconfigured for the IP address 10.0.0.205/24. In addition to the suggested way of configuring the box by adding a PC to this network and configuring the Mediatrix Boss this way, there are additional, advanced methods of configuration. These methods are especially useful for immediate and proper configuration of the primary Ethernet interface. There are basically two methods one for local and one for remote command line configuration (of course, for initial setup of interface addresses only the local methods are applicable).

CLI Console

There are 12 console windows available when accessing Mediatrix Boss products with a keyboard and VGA monitor2. The windows can be reached by pressing the Alt key and one of the function keys F1 or F12 simultaneously. Some of these consoles are reserved for special purposes though, so only some of them are usable for an administrator's command line console:

Serial Line Console

It is also possible to access Mediatrix Boss products via RS232 (aka COM1) and a terminal emulator program like MS Windows' Hyperterminal. The connection parameters for the serial line are: 9600 baud, 8 data bits, no parity, 1 stop bit, VT100 emulation.

SSH Access

The third possibility to get a command line terminal is to use a SSH (i.e. secure shell) connection using any SSH client (e.g. OpenSSH on Unix/Linux, PuTTY on Windows).

Mediatrix Boss products allow SSH version 2 connections from all configured interfaces, unless set differently in the security settings of the graphical user interface.

1. This feature is available since release 4675.18 and 6060.2 respectively.

2. This applies only to Mediatrix Boss products with VGA connector. As the VGA connector is not accessible from outside of the product, this way to connect the Mediatrix Boss is reserved for service purposes. Opening the product and using the VGA connector should be done by the service engineering staff only.

Table 6: Function Keys

F1 Reserved for the start up screen

F2 Usable as command line console





F7 Reserved for future use



F10 Reserved for informational messages

F11 Reserved for error messages (severity warning and higher)

F12 Reserved for critical error messages

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Initial Setup using the Command Line Interface Software Configuration Guide

Initial Setup using the Command Line Interface

If you log in via one of the local methods (VGA console or RS232 console) you get a login menu.

Upon entering the shell you will see the following login prompt:

login:

The only available user to log on is root, with the same password as the admin's password in the browser interface, initially sesam. You should change this password via the browser as soon as possible! If you choose to separate CLI and Web-GUI administration (refer to “Change Password” on page 15), the CLI password can only be changed using the following command:

changepw <root> <new password>.norun

The shell used is a normal bash, as known from almost all Linux distributions and a number of commercial Unices. The actual numbers may vary, as they show the current version of the software at the time of this writing. Also the name Akira is just the initial name of the Mediatrix Boss and can be altered using the Web interface (system name).

After having logged the quick configuration menu will be the first view; by pressing s you can leave this menu and go to the shell.

The menu pages do the same as the analog Web-GUI pages, but with reduced functionality:

=======================[ Main menu ]========================

1: Configure first LAN interface

2: Configure second LAN interface

3: Configure WAN interface

4: Configure SSH and HTTPS ports

5: Configure default route (LAN)

6: Firewall/NAT

L: Change language

A: Apply configuration

S: Run shell

E: Exit

------------------------------------------------------------

Please enter your choice:

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Chapter 9 - Command Line Interface Initial Setup using the Command Line Interface

The setup of the follow-menus is rather self-explanatory. For example, by pressing 1 and RETURN, you will see the following:

Settings made in certain files here, have to be done again in the web interface (or in the basconfig directly, as shown below) to become permanent. All of these files will give out the following warning when you enter them in a shell:

Physical Network Settings

An additional submenu "Physical Network Settings" is available. This configuration is necessary to set up full duplex, half duplex or auto negotiation (default is auto negotiation).

=============[ Configure first LAN interface ]==============

Current values:

IP address: 192.168.140.1

IP netmask: 255.255.255.0

------------------------------------------------------------

1: Enable DHCP

2: Change IP address

3: Change IP netmask

S: Save and back to previous menu

B: Back to previous menu

------------------------------------------------------------

Please enter your choice:

WARNING: You are about to edit file(s) that are generated by the will be overwritten

the next time you call applyconfig or the "Apply configuration" link in the web GUI !

/etc/openser/openser.cfg.template

Are you sure you want to continue ? [Y/N]

Warning: Be careful if you intend to configure the Mediatrix Boss remotely: Initially there are no firewall rules in place and everyone who knows the IP address may log in and modify the box if the standard password has not been changed. Also pay attention that you don't change the IP address of the interface with which you're currently logged in if you're connected via SSH!

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Configuration utilities Software Configuration Guide

Configuration utilities

Configuring Mediatrix Boss products is normally done by editing one central configuration file called the baseconfig file which location is /etc/config/baseconfig. This file controls how the necessary configuration files for almost all services are generated by the applyconfig utility. Because the reference for the baseconfig file is quite extensive the details of this file are described in section “The "baseconfig" configuration file” on page 135 after explaining the utilities which work with this file first.

applyconfig

applyconfig processes the baseconfig file and generates the configuration files for all daemons and other services that can be configured via the WebUI. There are two modi in which applyconfig operates: with or without command line options. When called without any command line options all configuration files are generated, overwriting already existing ones. If arguments are passed only configuration files for those services are generated.

For example:

causes only the configuration file for the syslog daemon to be generated, overwriting the previous one.

Because old configuration files are overwritten you should not edit the controlled configuration files directly as the next you're pressing Apply configuration in the WebGUI all your changes will get overwritten by applyconfig.

restartservices

Now that applyconfig has written the configuration files we need to restart one or more (maybe even all) services. This is handled by the restartservices utility, which even allows groups of services to be restarted.

Like applyconfig this utility operates on all services when called without arguments. If arguments are given only the specified services are restarted. To get a list of known services and service group enter:

Another very useful command line argument is -n which causes restartservices to just show what it would do without actually doing something so you can be sure that it really will do what you want it to do, e.g.:

applyconfig syslog

restartservices -l

root@Akira/~ # restartservices -n logging

/usr/bin/setsid /etc/rc.d/rc3.d/K89klogd stop

/usr/bin/setsid /etc/rc.d/rc3.d/K90syslogd stop

/sbin/stophttpd

/sbin/stopsshd

/usr/bin/setsid /etc/rc.d/rc3.d/S10syslogd start

/usr/bin/setsid /etc/rc.d/rc3.d/S11klogd start

/sbin/starthttpd

/sbin/startsshd

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Chapter 9 - Command Line Interface Configuration utilities

chconfig

chkconfig is used for controlling services' configuration (checking configuration). Services can be enabled and disabled depending on whether they should be started when running restartservices or not. If a service is disabled it won't be started and will be simply ignored. When starting services the system looks up whether there is a locking file for a particular service in /etc/config. The locking file is an empty file named <service>.norun, e.g. /etc/config/dyndns.norun.

There are two modi in which chkconfig operates: with or without command line options. When called without any command line options all services that support chkconfig and their current status (enabled/disabled ) are listed. Typing chkconfig followed by the name of a particular service lists the current status of this service only.

For example:

Particular service can be enabled (or disabled ) by typing chkconfig followed by the name of the service and on (or off ).

For example:

enables DynDNS

disables it.

Furthermore, chkconfig can be run with -q (quiet mode) which will suppress output, e.g. chkconfig -q dyndns. This can be used for scripting purposes, since chkconfig return value is 0 for enabled services, and >0 otherwise.

lockconfig and unlockconfig

Sometimes it is desirable to edit a configuration file and to keep it from being overwritten by the web interface or applyconfig. The commands lockconfig and unlockconfig can be used to protect and unprotect these configuration files. The two commands basically determine for which templates applyconfig shall and shall not generate configuration files.

Both lockconfig and unlockconfig take the following parameters:

h (-help): Show help page

l (-list): Show the list of templates and the configuration files they're responsible for

All other parameters are interpreted as being template names. If you would like to edit /etc/d- hcpd.conf and prevent it from being overwritten by applyconfig or via the web interface, first, enter lockconfig -list and look for the line containing /etc/dhcpd.conf. In this line you can see that the template dhcpd is responsible for /etc/dhcpd.conf. Enter "lockconfig dhcpd" and the dhcpd template will not be executed and /etc/dhcpd.conf will not be overwritten.

If you would like to have a configuration file managed by the Mediatrix Boss again you have to use the unlockconfig command which is used in the same way as lockconfig, but has the opposite effect.

apply_flush_firewall

Mediatrix Boss products don't create Linux IPTables firewall rules directly but use an intermediate format called CIG which helps simplifying the creation on firewall setups.

root@Akira/~ # chkconfig dyndns

dyndns on

chkconfig dyndns on

chkconfig dyndns off

Note: If settings are edited and applyconfig is executed, a message will be displayed in the WebGUI if applyconfig runs into a locked configuration file.

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The "baseconfig" configuration file Software Configuration Guide

Custom IPTables rules may be added in /etc/sysconfig/firewall.custom.

The applyfirewall utility flushes the current firewall rules, compiles the CIG script (located in /etc/sysconfig/firewall.wig) into IPTables rules and applies them.

The flushfirewall tool just flushes the firewall, thus temporarily disabling it, which could be done for testing purposes. This, however will not flush NAT related rules. If you want to disable NAT too, call flushfirewall with the command line option -n.

saveconfiguration / restoreconfiguration

saveconfiguration saves the most important configuration files The files currently are:

/etc/passwd, /etc/htpasswd, /etc/config/baseconfig, /etc/ssl/*, /etc/ssh/*, /etc/

openvpn/*

into a file

/tmp/configuration-<date>.cpio.gz (or /root/configuration-<date>.cpio.gz

in older releases)

To restore a configuration that was saved with saveconfiguration use restoreconfiguration and supply the appropriate file as argument. Note that restoreconfiguration automatically calls applyconfig and restartservices to regenerate the configuration files for all services

and to restart them.

The "baseconfig" configuration file

This chapter serves as a reference to the parameters of /etc/config/baseconfig, not as a conceptual guide to techniques behind the products used. If you are interested in the basic concepts (e.g. on VPNs or iptables) please consult the products' web sites, mailing lists etc.

The purpose of the baseconfig file is to have a central configuration file that controls the generation of configuration files for all included services. The generation of configuration files is handled by the applyconfig script which is described in section “Configuration utilities” on page 133.

A complete reference to all baseconfig keys, their explanations and their possible values is provided in section “Parameter Reference for baseconfig” on page 136.

Syntax

The file baseconfig contains all configuration parameters that can be set using the web interface. Whenever data from the web interface is saved the entire file is read, the according entries are added/modified/deleted and then the entire file is rewritten in alphabetical order.

This behaviour has two consequences:

You may add as many additional parameters as you want (they just have no effect).

You cannot use comments in this file the normal way. If you want to add a comment to a parameter, just create a new parameter with the same name as the original parameter, followed e.g. by _comment.

Example:

You already could see the overall syntax of entries in this file:

flushfirewall -n

system_name=gatekeeper

system_name_comment=This is the name of my box

<Variable>=<Value>

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Chapter 9 - Command Line Interface Parameter Reference for baseconfig

<Variable> is the name of the parameter. A complete list of names used in the Mediatrix Boss can be found in section “Parameter Reference for baseconfig” on page 136. <Value> can be any entry, its interpretation depends on <Variable> and the service to be configured.

Parameter Reference for baseconfig

The following chapters describe all keys and their meanings used in the /etc/config/baseconfig file. See “The "baseconfig" configuration file” on page 135 for an overview of this file. We will adhere to their order of appearance in the WebGUI

System (system_)

These settings configure aspects of the box itself.

Port numbers (sshd_, https_)

Use these settings to customize port numbers for ssh and https.

Network

In the following the settings of the interfaces, which can also be configured via the WebGUI. However, via the Command Line more customized configuration is possible to a certain extent. Of course this is a field for the advanced administrator and should not be done by the inexperienced.

LAN Interface 1 (lana_)

These settings configure the first LAN interface eth0.

Name system_name

Description Computer name of the Mediatrix Boss

Example system_name=gatekeeper

Name system_dns

Description DNS server for name resolution of the Mediatrix Boss, e.g. for ping, dyndns, etc.

Example system_dns=217.145.104.11

Name sshd_port

Description Custom port number for ssh; valid number is any integer from range 0-65535. Default value is set to "22".

Example sshd_port=22000

Name httpd_port

Description Custom port number for https; valid number is any integer from range 0-65535. Default value is set to "443".

Example httpd_port=44300

Name lana_ip

Description IP address of first LAN interface or "dhcp" for dynamic IP support

Example lana_ip=10.0.0.205

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Parameter Reference for baseconfig Software Configuration Guide

LAN Interface 2 (lanb_)

These settings configure the second LAN interface eth2.

Name lana_nm

Description Net mask of first LAN interface

Example lana_nm=255.255.255.0

Name lana_dhcp_from

Description Start of IP range for DHCP on first LAN interface; Has to be in the same subnet as lana_ip/lana_nm and has to be smaller than lana_dhcp_to.

Example lana_dhcp_from=10.0.0.220

Name lana_dhcp_to

Description End of IP range for DHCP on first LAN interface; Has to be in the same subnet as lana_ip/lana_nm and has to be higher than lana_dhcp_from.

Example lana_dhcp_to=10.0.0.230

Name lana_dhcp_gw_ip

Description Specifies optional standard gateway other than the Mediatrix Boss itself.

Example lana_dhcp_gw_ip=10.0.0.223

Name lana_dhcp_dns1, lana_dhcp_dns2

Description Specifies optional DNS servers forwarded to a DHCP client along with its assigned address. Not to be confused with system_dns, which is used by the Mediatrix Boss itself.

Example lana_dhcp_dns1=217.145.104.11

Name lana_dhcp_ntp1, lana_dhcp_ntp2, lana_dhcp_ntp3

Description Specifies optional NTP time servers forwarded to a DHCP client along with its assigned address. Not to be confused with system_dns, which is used by the Mediatrix Boss itself.

Example lana_dhcp_ntp1=217.145.104.11

Name lana_nat

Description Value "0" (no) or "1" (yes); specifies whether NAT should be used between the first LAN interface (eth0) and WAN (eth1).

Example lana_nat=1

Name lanb_type

Description Specifies usage of second LAN interface (eth2).Valid values are:

"inactive": second LAN interface is not used at all; all values referencing lanb_ are not used

"dmz": second LAN is used as DMZ; NAT is not supported on this interface and the interface cannot be opened unlimited against the first LAN interface.

"internal": second LAN is a normal second internal network and treated like the first LAN

Example lanb_type=inactive

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Name lanb_ip

Description IP address of second LAN interface (eth2) or "dhcp" for dynamic IP support

Example lanb_ip=192.168.1.1

Name lanb_nm

Description Net mask of second LAN interface (eth2)

Example lanb_nm=255.255.255.0

Name lanb_dhcp_from

Description Start of IP range for DHCP on second LAN interface;

Has to be in the same subnet as lanb_ip/lanb_nm and has to be smaller than lanb_dhcp_to.

Example lanb_dhcp_from=192.168.1.10

Name lanb_dhcp_to

Description End of IP range for DHCP on second LAN interface;

Has to be in the same subnet as lanb_ip/lanb_nm and has to be higher than lanb_dhcp_from.

Example lanb_dhcp_to=192.168.1.20

Name lanb_dhcp_gw_ip

Description Specifies optional standard gateway other than the Mediatrix Boss itself.

Example lanb_dhcp_gw_ip=192.168.1.13

Name lanb_dhcp_dns1, lanb_dhcp_dns2

Description Specifies optional DNS servers forwarded to a DHCP client along with its assigned address. Not to be confused with system_dns, which is used by the Mediatrix Boss itself.

Example lanb_dhcp_dns1=217.145.104.11

Name lanb_dhcp_ntp1, lanb_dhcp_ntp2, lanb_dhcp_ntp3

Description Specifies optional NTP time servers forwarded to a DHCP client along with its assigned address. Not to be confused with system_dns, which is used by the Mediatrix Boss itself.

Example lanb_dhcp_ntp1=217.145.104.11

Name lanb_nat

Description Value "0" (no) or "1" (yes); specifies whether NAT should be used between the second LAN interface (eth2) and WAN (eth1).

Example lanb_nat=1

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WAN Interface (wan_)

These settings configure the WAN interface eth1 and its PPP connection, if required.

Virtual Interfaces (vif_intf)

Use these settings to configure virtual interfaces on your box (so called IP Aliasing). These settings are also there for another reason. A virtual interface can also be configured as a VLAN.

The acronym VLAN expands to Virtual Local Area Network. A VLAN is a logical local area network (or LAN) that extends beyond a single traditional LAN to a group of LAN segments, given specific configurations.

A LAN is a local area network and is defined as all devices in the same broadcast domain, routers stop broadcasts, switches just forward them. VLAN is a virtual LAN. In technical terms, a VLAN is a broadcast domain created by switches. Normally, it is a router creating that broadcast domain. With VLANs, a switch can create the broadcast domain.

This works by, you, the administrator, putting some switch ports in a VLAN other than 1, the default VLAN. All ports in a single VLAN are in a single broadcast domain. Since switches can talk to each other, some ports on switch A can be in VLAN 10 and other ports on switch B can be in VLAN 10. Broadcasts between these devices

Name wan_type

Description Type of WAN connection; one of:

"ip": static Internet address used; implies wan_ip, wan_nm

"pppoe": ISP connection using PPPoE is used for Internet access; implies wan_user, wan_pwd

"pptp": ISP connection using PPTP is used for Internet access; implies wan_user, wan_pwd, wan_ip, wan_gw

Example wan_type=pppoe

Name wan_ip

Description • Static IP address of WAN interface (eth1), if a static Internet address is used for connection

• IP address of Mediatrix Boss in transfer net to DSL modem, if PPTP is used

Example wan_ip=62.34.178.34

Name wan_gw

Description IP address of PPTP DSL modem

Example wan_gw=62.34.178.35

Name wan_nm

Description Netmask of WAN interface (eth1)

Example wan_nm=255.255.255.248

Name wan_user

Description User name for PPPoE and PPTP connections

Example wan_user=123567.213523@my_provider

Name wan_pwd

Description Password for PPPoE and PPTP connections

Example wan_pwd=kgvfasdhl

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will not be seen on any other port in any other VLAN, other than 10. However, these devices can all communicate because they are on the same VLAN. Without additional configuration, they would not be able to communicate with any other devices, not in their VLAN.

Similarly to physical LANs, you need a router to exchange packets between two VLANs. In order to achieve that, you can create a Virtual Interface for each VLAN the Mediatrix Boss is in. You can then define routing policies as well as security policies between the virtual interfaces to route packets any way you like.

Network related Topics

The following information might be of interest in the context of the interface configuration:

MTU Settings

This entry specifies the maximum size a package may have without being fragmented. By default this is set to 1500. The MTU size is set for each interface. We will only list two examples here, but there is one entry for each Interface (LAN Interface 2, DMZ etc.).

For virtual interfaces this is done differently since there is only one key that specifies all of its settings:

Warning: The x must be a consecutive number!

Name vif_intfx =[native interface;name;ip;netmask]

Description The option value must be enclosed in square brackets. The option value is a semicolon separated list which must contain the following information:

1. Interface name (such as lana,wan) for which the IP alias should be created2. Number of the virtual Interface Every Interface for aliasing needs to have a number

to uniquely identify it. The name of the interface on the device will then be "physical interface name":"number", only separated by a colon. For VLANs, this number is the same of the VLAN tag you want to be using.

3. IP Address of the network to be configured.4. Netmask of the network to be configured on this virtual interface.5. Activate VLAN If this switch is set to on, a VLAN interface will be created with the

number as tag.

Example vif_intf0=[lana;gw1;192.168.165.254;255.255.255.0] or

for VLAN vif_intf0=[lana;123;192.168.7.67;255.255.255.0;on]

Warning: Note that you cannot configure two interfaces with the same network and netmask. Especially overlapping netmasks will not be caught by the sanity check in the configuration management, but can cause very unwanted routing behavior.

Name lana_mtu

Description Determines the maximum size for packages leaving through LAN interface

Example lana_mtu=1500

Name wan_mtu

Description Determines the package size for the WAN interface

Example wan_mtu=1500

Name vif_intf0

Description Contains all information about the virtual interface

Example vif_intf0=[lana;0;1.2.3.4;255.255.255.0;on;1500]

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Dynamic DNS (dyndns_)

These settings configure an dynamic DNS service, which is especially useful, when no static address is available for the WAN interface.

For most services the same username/password may be used for a number of hostnames. If you use this feature, a malicious user on one Mediatrix Boss might be able to modify registered addresses from other boxes, which is a potential security risk!

Name Server (bind_)

These settings configure the Name Server function.

Name dyndns_type

Description Dynamic DNS service to be used; this Mediatrix Boss supports several suppliers of dynamic DNS services - have a look on their web sites for additional information and registration:

• "dhs" www.dhs.org

• "dyndns" www.dyndns.org

• "dyns" www.dyns.cx

• "easydns" www.easydns.com

• "eznet" www.ez-ip.net

• "hnorg" www.hn.org

• "justlinux" www.justlinux.com

• "ods" www.ods.org

• "tzocom" www.tzo.com

• "zoneedit" www.zoneedit.com

Example dyndns_type=dyndns

Name dyndns_user

Description Username you are registered with at the selected dyn-DNS service

Example dyndns_user=mediatrix_test

Name dyndns_pwd

Description Password for your registered dyn-DNS username

Example dyndns_pwd=test_pwd

Name dyndns_host

Description Registered hostname for this Mediatrix Boss. It will be reachable from the Internet using this name

Example dyndns_host=gatekeeper.testdom.com

Name bind_mode

Description Intended use of the Name Server; one of:

"on": Name Server enabled; implies bind_dnsx

"off": Name Server disabled, no query forwarding

"custom": Custom configuration of Name Server

Example bind_mode=custom

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http://www.dhs.org

http://www.dyndns.org

http://www.dyns.cx

http://www.easydns.com

http://www.ez-ip.net

http://www.hn.org

http://www.justlinux.com

http://www.ods.org

http://www.tzo.com

http://www.zoneedit.com


Routing (routex )

These settings contain static routes, one per parameter. The parameters need to be numbered consecutively; if a number is missing, processing of routing information will stop at that gap.

Bandwidth (htb_)

Name bind_dnsx

Description IP address of a DNS server (forwarder ) that should be asked if a DNS query could not be resolved

Example bind_dns0=10.42.1.10

Name bind_slaveZonex

Description IP address of a master server for the given zone of type slave

Example bind_slaveZone0=[zoneName;10.42.1.11;zoneFileName]

Name bind_forwardZonex

Description IP addresses of one or more forwarder servers for the given zone of type forward. IP addresses have to be separated by blank

Example bind_forwardZone0=[zoneName; 10.42.1.12 10.42.1.14]

Name routex

Description Static route with numerical identifier <x>; parameter contains all information in a semicolon separated array:

[<Network to be routed to>; <netmask of network to be routed to>; <gateway>; <interface>]

Example route0=[10.1.0.0;255.255.255.0;10.0.0.254;LANA]

route1=[10.2.0.0;255.255.255.0;10.0.0.254;LANA]

Name proxy_sip_transparent

Description Enables transparent sip proxy functionality

Example proxy_sip_transparent=on

Name htb_classx

Description

Example

Name htb_virtifx

Description

Example

Name htb_ifx _default

Description

Example

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NTP (ntp_)

These settings define NTP servers that can be used to synchronize the system time.

Settings for Bandwidth Daemon(bw_)

These settings configure the bandwidth daemon that monitors the network traffic and calculates the current average bandwidth. The bandwidth daemon is queried by the SIP proxy for Call Admission Control (CAC).

Security (including VPN)

Below you will find all security related possibilities available via the CLI:

VPN (vpnx _)

These settings contain VPN definitions, one tunnel per parameter set vpnx_. The tunnels have to be numbered consecutively; if a number is missing, processing of tunnels will stop at that gap.

These parameters describe IPsec and OpenVPN tunnels. While most parameters are common for both types, there are some specific ones:

IPsec only: vpnx_local_if, vpnx_remote

OpenVPN only: vpnx_local_ip, vpnx_remote_ip, vpnx_port, vpnx_tls

Name ntp_on

Description Turning on/off NTP; one of:

"on": NTP feature enabled; implies ntp_namex, ntp_prefx

"off": NTP feature disabled

Example ntp_on=on

Name ntp_namex

Description IP address of an NTP server

Example ntp_name0=10.42.1.10

Name ntp_prefx

Description Preferring answers sent by a particular NTP server; one of: "on", "off"

Example ntp_pref0=on

Name bw_rate

Description Sample rate, in tenth of second; the default value is "1"

Example 1

Name bw_size

Description Number of samples; combined with bw_rate this value determines the size of a time window the bandwidth daemon uses; the default value is "30"

Example 30

Name bw_log

Description Log level; possible values are: "none", "error", "warning", "info", "debug"

Example info

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Name vpnx _name

Description A (preferably not too long) unique name describing the tunnel connection.

Example pn0_name=munich-branch

Name vpnx _active

Description A possibility to activate ("1") or deactivate ("0") a tunnel temporarily.

Example vpn0_active=1

Name vpnx _type

Description Sets the type of the tunnel. It has either the value "openvpn" or "ipsec". If the type is "openvpn", a tun device with the VPN's number will be used (e.g. for vpn0_... a device /dev/tun0 will be created)

Example vpn0_type=ipsec

Name vpnx _partner_net

Description Remote side's network to be secured by the tunnel.

Example vpn0_partner_net=10.5.0.0

Name vpnx _partner_nm

Description Netmask of remote side's network be secured by the tunnel.

Example vpn0_partner_net=255.255.255.0

Name vpnx _partner_ip

Description Public IP address of remote VPN gateway.

If this value is given, vpnx_partner_name is not evaluated.

If neither this parameter nor vpnx_partner_name are set, a road warrior with dynamic IP address is assumed.

Example vpn0_partner_ip=219.23.132.67

Name vpnx _partner_name

Description Public DNS name of remote VPN gateway.

A dynamic DNS name may only be used with OpenVPN tunnels, as IPsec and dynamic DNS names result in synchronisation problems as soon as the dynamic DNS host's IP address changes.

If neither this parameter nor vpnx_partner_ip are set, a road warrior with dynamic IP address is assumed.

Example vpn0_partner_name=vpngate.mycompany.com

Name vpnx _local_if

Description Interface connected to the IPsec tunnel; alternatively an IP address on a local network, that will be connected to the tunnel then.

Example vpn0_local_if=LAN1

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Name vpnx _local_ip

Description Local side's IP address for OpenVPN transfer network. This address and vpnx_remote_ip must be unique (especially different OpenVPN connections may not use the same addresses).

This address must be used as remote IP on the remote gateway.

Example vpn0_local_ip=10.254.254.1

Name vpnx _remote_ip

Description Remote side's IP address for OpenVPN transfer network. This address and vpnx_local_ip must be unique (especially different OpenVPN connections may not use the same addresses).

This address must be used as local IP on the remote gateway.

Example vpn0_remote_ip=10.254.254.2

Name vpnx _port

Description Port number to be used for the OpenVPN connection. The port must be unique (especially different OpenVPN connections may not use the same port). Normally port numbers start from 5000 and are incremented for additional connections.

This port must be the same as configured on the remote gateway.

Example vpn0_port=5001

Name vpnx _ss

Description Defines:

• a shared secret for a password based setup of an IPsec connection

• a shared secret file for a shared secret based setup of an OpenVPN connection. The file /etc/openvpn/secrets/<ss>.key must exist.

This parameter and vpnx_cert are mutually exclusive.

Example vpn0_ss=test_secret

Name vpnx _cert

Description Defines a certificate for a certificate based setup of the IPsec/OpenVPN connection. This parameter and vpnx_ss are mutually exclusive. The following files must exist locally for IPsec connections:

• /etc/ipsec.d/certs/<certificate>-cert.pem containing a X.509 certificate

• /etc/ipsec.d/private/<certificate>-key.pem containing the RSA key

The following files must exist locally for OpenVPN connections:

• /etc/ssl/certs/<certificate>-cert.pem containing a X.509 certificate

• /etc/ssl/private/<certificate>-key.pem containing the RSA key

• /etc/ssl/CA/certs/<CA_certificate>-cert.pem containing the CA certificate the certificate was signed with

All these files are normally managed by the web interface.

Example vpn0_cert=roadwarrior

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Generic L2TP Server Settings (l2tp_)

These settings contain all information to set up a L2PT server on the Mediatrix Boss.

Name vpnx _remote

Description Describes the X.509 compatible subject of the remote certificate.

If the remote certificate was created on the Mediatrix Boss and exported to the remote PC as PKCS12 certificate, the subject can be checked in the web interface or with the following command on the Mediatrix Boss:

openssl x509 -in / etc/ssl/certs/<remotecertificate>.cert.pem -

text | grep Subject:

This information is only necessary for IPsec connections, for OpenVPN connections the CA cert information is processed automatically (TLS connection).

Example vpn0_remote=CN=L2TP-Mobile,OU=Mobile,O=Test-Org,C=DE

Name vpnx _tls

Description Certificate based OpenVPN connections use TLS; one of the connection partners has to be defined as TLS server, the other one as TLS client.

Example vpn0_tls=server

Name l2tp_active

Description This parameter allows a temporary switch off of the l2tp server without the need to delete all parameters. Values are "1" (active) and "0" (not active).

Example l2tp_active=1

Name l2tp_local_ip

Description Sets the local IP address within the network allocated to the L2TP transfer network. At the same time, together with "l2tp_netmask", defines the L2TP transfer network.

Example l2tp_localip=192.168.2.1

Name l2tp_netmask

Description Defines the size of the L2TP transfer network

Example l2tp_netmask=255.255.255.0

Name l2tp_rangestart

Description Within the L2TP transfer network, defines the first address to assign to a client that has not defined its own static IP address; the value has to be within the L2TP transfer network defined by l2tp_localip and l2tp_netmask.

Example l2tp_rangestart=192.168.2.100

Name l2tp_rangeend

Description Within the L2TP transfer network, defines the last address to be assigned to a client that has not defined its own static IP address; the value has to be within the L2TP transfer network defined by l2tp_localip and l2tp_netmask

Example l2tp_rangeend=192.168.2.200

Name l2tp_msdns

Description Defines a DNS server that will be assigned to the clients (optional)

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User Based L2TP Server Settings (l2tp_userx _)

These settings contain settings that apply to special users / computers upon their connection setup. The user specific parameters have to be numbered consecutively; if a number is missing, processing will stop at that gap.

When using L2TP to connect Windows XP and Windows 2000 computers, there is a number of protocol layers involved:

1. Mostly PPP to connect the remote computer to the Internet; authentication is done via connection settings.

2. An IPsec connection is set up to establish an encrypted communication channel; the authentication can be done via passwords (shared secrets) or certificates; both possibilities can be configured on a per-user basis.

3. An L2TP (Layer 2 tunnel protocol) connection is established on top of IPsec. No authentication is done on this layer.

4. On top there is another PPP session that finally establishes an IP session between the remote computer and the Mediatrix Boss. This session has to be authenticated again, using username and password. Within this session, the remote computer also receives its IP address.

Example l2tp_msdns=217.145.104.11

Name l2tp_mswins

Description Defines a WINS server that will be assigned to the clients (optional) to allow them WINS name resolution and browsing.

Example l2tp_mswins=10.0.0.200

Name l2tp_userx _name

Description PPP login name

Example l2tp_user0_name=testuser

Name l2tp_userx _pwd

Description PPP login password

Example l2tp_user0_pwd=testpwd

Name l2tp_userx _ip

Description Optional static IP address; if this value is set, the user always receives the same IP address, otherwise he gets one from the pool l2tp_rangestart - l2tp_rangeend. The value has to be within the L2TP transfer network defined by l2tp_localip and l2tp_netmask, but outside the pool defined by l2tp_rangestart and l2tp_rangeend.

This option might be interesting for setting special firewall rules for particular users.

Example l2tp_user0_ip=192.168.1.30

Name l2tp_userx _ss

Description Defines a shared secret for a password based setup of the IPsec connection. This parameter and l2tp_userx_cert / l2tp_userx_remote are mutually exclusive.

Example l2tp_user0_ss=test_secret

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General Firewall Information)(fw_)

These settings contain all information to configure iptables. In general firewall rules are executed in the following order:

Allow rules for active clients (DNS, dynamic DNS, syslog, etc.)

Allow rules for admin access (ssh and https)

Allow rules for IPsec and OpenVPN

Allow rules for base services (see fw_lana_services, fw_lanb_services)

Include file /etc/sysconfig/firewall.custom (except rules added with "iptables -I")

User defined rules (with fw_rulex_..., see below)

Deny all

Firewall Meta Rules

There are a number of "meta" rules that cause a whole range of rules to be generated.

Name l2tp_userx _cert

Description Defines a certificate for a certificate based setup of the IPsec connection. This parameter and l2tp_userx_ss are mutually exclusive. The following files must exist locally:

• /etc/ipsec.d/certs/<certificate>-cert.pem containing a X.509 certificate

• /etc/ipsec.d/private/<certificate>-key.pem containing the RSA key.

These files are normally managed by the web interface.

Example l2tp_user0_cert=roadwarrior

Name l2tp_userx _remote

Description Describes the X.509 compatible subject of the remote certificate.

If the remote certificate was created on the Mediatrix Boss and exported to the remote PC as PKCS12 certificate, the subject can be checked in the web interface or with the following command on the Mediatrix Boss: openssl x509 -in /etc/ssl/certs/<remotecertificate>.cert.pem -text | grep Subject:

Example l2tp_user0_remote=CN=L2TP-Mobile,OU=Mobile, O=Test-Org,C=DE

Name fw_seclevel

Description This parameter allows to disable all access configuration and set the Mediatrix Boss to a special secure mode. Valid values:

• "secure": All local Internet access is switched off. Only access between local networks and remote sites via tunnel connections are possible.

• "custom": normal operation

Example fw_seclevel=custom

Name fw_loglevel

Description Level indicating which rules generate messages; value is an integer between "0" (no logging at all) and "9" (very verbose logging). Default level is "4" (log all denied packets).

This parameter just states how many iptables LOG rules are created.

Example fw_loglevel=0

Name fw_ssh_via_lana

Description Permit ("1") or deny ("0") SSH access to the Mediatrix Boss via the first LAN interface (eth0)

Example fw_ssh_via_lana=0

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Name fw_ssh_via_lanb

Description Permit ("1") or deny ("0") SSH access to the Mediatrix Boss via the second LAN interface (eth2)

Example fw_ssh_via_lanb=0

Name fw_ssh_via_wan

Description Permit ("1") or deny ("0") SSH access to the Mediatrix Boss via the WAN interface (eth1)

Example fw_ssh_via_wan=0

Name fw_https_via_lana

Description Permit ("1") or deny ("0") HTTPS access to the Mediatrix Boss via the first LAN interface (eth0)

Example fw_https_via_lana=0

Name fw_https_via_lanb

Description Permit ("1") or deny ("0") HTTPS access to the Mediatrix Boss via the second LAN interface (eth2)

Example fw_https_via_lanb=0

Name fw_https_via_wan

Description Permit ("1") or deny ("0") HTTPS access to the Mediatrix Boss via the WAN interface (eth1)

Example fw_https_via_wan=0

Name fw_lana_all_open

Description Permit ("1") or deny ("0") unlimited access from the first LAN interface (eth0) to the internet.

This only applies to outgoing connections.

Example fw_lana_all_open=1

Name fw_lana_services

Description List of outgoing services, seperated by whitespaces, that are valid through the first LAN interface (eth0). Valid services are:

"DNS", "FTP", "SSH ", "TELNET ", "POP3", "SMTP", "HTTPS", "HTTP", "NTP", "PING", "SNMP"

Example fw_lana_services=FTP SSH HTTP HTTPS

Name fw_lanb_all_open

Description Permit ("1") or deny ("0") unlimited access from the second LAN interface (eth2) to the internet.

This only applies to outgoing connections.

Example fw_lanb_all_open=0

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Firewall Rules (fw_rulex _)

These settings contain rules, one per parameter set fw_rulex_.

The rules have to be numbered consecutively (substitute x with that number); if a number is missing, processing of rules will stop at that gap.

Almost all parameters apart from fw_rulex_in_int, fw_rulex_out_int (which may be ANY), fw_rulex_policy and fw_rulex_type are optional; generally not giving a parameter means a wildcard (all possible values).

If the security level (fw_seclevel) is switched from custom to secure, all existing firewall rule parameter are renamed to no_fw_rulex_ and therefore not applicable. If the security level is switched back, all these rules are activated again by renaming them back to their normal names.

Name fw_lanb_services

Description List of outgoing services, seperated by whitespaces, that are valid through the second LAN interface (eth2). Valid services are:

"DNS", "FTP", "SSH ", "TELNET ", "POP3", "SMTP", "HTTPS", "HTTP", "NTP", "PING", "SNMP"

Example fw_lanb_services=FTP SSH HTTP HTTPS

Name fw_lana_lanb_open

Description Permit ("1") or deny ("0") unlimited access between the two LANs (Only supported, if second LAN is internal)

Example fw_lana_lanb_open=1

Name fw_rulex _in_int

Description Incoming interface for rule x; one of "ANY ", "WAN ", "LAN1", "LAN2", a tunnel name (generating forward rule) or "LOCAL" (generating incoming rule)

Example see “Example for fw_rulex _” on page 151

Name fw_rulex _out_int

Description Incoming interface for rule x; one of "ANY ", "WAN ", "LAN1", "LAN2", a tunnel name (generating forward rule) or "LOCAL" (generating outgoing rule)


Name fw_rulex _s_ip

Description Source IP address; may also specify a network address.

In fact, if fw_rulex_s_nm is not "255.255.255.255", it has to be a valid network address.


Name fw_rulex _s_nm

Description Source IP address netmask.

If fw_rulex_s_nm is not "255.255.255.255", fw_rulex_s_ip has to be a valid network address.


Name fw_rulex _s_port

Description Source port for packet; only used if fw_rulex_type is "TCP" or "UDP"


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Example for fw_rulex _

This allows access from any internal network SSH (i.e. port 22) access to the internet computer with address 16.1.2.120.

Name fw_rulex _d_ip

Description Destination IP address; may also specify a network address.

In fact, if fw_rulex_d_nm is not "255.255.255.255", it has to be a valid network address.


Name fw_rulex _d_nm

Description Destination IP address netmask.

If fw_rulex_d_nm is not "255.255.255.255", fw_rulex_d_ip has to be a valid network address.


Name fw_rulex _d_port

Description Destination port for packet; only used if fw_rulex_type is "TCP" or "UDP"


Name fw_rulex _policy

Description ACCEPT " or "DENY " (which is useless in most cases, because the "DENY ALL" rules are next to these rules.


Name fw_rulex _type

Description Protocol type for this rule; is either "TCP", "UDP" or "ICMP" or a protocol number (e.g. from /etc/protocols)


Name fw_rulex _icmp_type

Description ICMP type to which the rule shall apply; only used if fw_rulex_type is ICMP. Valid values are:

"address-mask-request", "address-mask-reply ", "destination-unreachable", "echo-reply ", "echo-request", "parameter-problem", "redirect", "router-advertisement", "router-solicitation", "source-quench", "timestampreply ", "timestamp-request", "time-exceeded"


fw_rule0_in_int=ANYfw_rule0_out_int=WAN

fw_rule0_d_ip=16.1.2.120

fw_rule0_d_nm=255.255.255.255fw_rule0_d_port=22

fw_rule0_policy=ACCEPT

fw_rule0_type=TCP

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Port Forwarding (fwd_)

These settings contain port forwardings to computers in NAT'ed local networks, one per parameter. The parameters have to be numbered consecutively; if a number is missing, processing will stop at that gap.

Diagnostics

Behind the GUI of the DIAGNOSTICS tab:

Settings for System Logging (syslog_)

These settings configure the system's logging behaviour.

SNMP (snmp_)

These settings configure the SNMP-Daemon.

Name fwdx

Description Forwarding rule; parameter contains all information in semicolon separated array:

Port on Mediatrix Boss target IP address, target port

Example fwd0=[2222;10.0.0.10;22]

Sets port forwarding from Mediatrix Boss´s WAN interface port 2222 to the SSH port on local box 10.0.0.10

Name syslog_server

Description Name or IP address of a remote syslog server that shall get all system and kernel messages of this Mediatrix Boss

Example syslog_server=10.0.0.10

Name syslog_level

Description Severity level for logging; everything from this level on will be logged; values are (in descending order of verbosity): "debug", "info", "notice", "warning", "error", "critical", "alert", "emergency "

Example syslog_level=debug

Name syslog_file

Description If set, log messages will be logged to this local file. Use with caution! There is only limited space available and there is no mechanism to prevent syslog from filling up the entire file system.

Warning: Use this option only if you are know what you do!

Example syslog_file=1

Name snmp_enable

Description Enable or Disable the SNMP daemon

Example snmp_enable=on

Name snmp_ropass

Description Specify the read-only community pass for SNMP version 1 and 2c

Example snmp_ropass=example

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Voice (ser_)

These settings configure the SIP proxy service. This service features multiple configuration templates that are explained in more detail in the handbook. The parameter values discussed here are sometimes used for multiple configuration templates. This is always explained together with the keys. Besides the parameter values as explained below, the keys for tls have to be setup if this is desired. All our products will autocreate tls keys for use with the SIP Proxy. If the interaction with phones is desired, there are two ways of setting this up. Either you download the preconfigured keys (they are generated during the first boot of the device, or you upload the desired keys to the Mediatrix Boss.

SIP Proxy Scenarios

The most important parameter is that for selecting the used template profile is the ser_type parameter. This one selects the scenario the product is to be used in. The following scenarios are possible.

Disabled

Disable SIP Proxy functionality.

Custom

Custom template that permits configuration via CLI. This is usually not recommended unless for specialized projects. This is mostly used in large projects with specialized configuration templates that are put into the Mediatrix Boss through mass deployment tools.

Standalone

Standalone template which will make SIP Proxy act as a standalone SIP server. This is a very simple template to setup simple configurations. The configuration of voice gateways to enable outbound calls is supported.

Survivability

Survivability template which will forward all messages to a central SIP server unless this one is unavailable. In this case, the SIP Proxy will take over the role of the central SIP server. This template also support CAC scenarios.

Name snmp_rwpass

Description Specify the read-write community pass for SNMP version 1 and 2c

Example snmp_rwpass=example

Name snmp_proxy(0..9)

Description Enable the snmp daemon to act as a proxy format= [ip- adress;port;password;version;OID]

Example snmp_proxy0=[127.0.0.1;22;password;2c;.1.3]

Name snmp_trap

Description Enables the trap-sending functionality of the SNMP daemon

Example snmp_trap=on

Name snmp_trap_community

Description Specify the SNMP community name for sending traps

Example snmp_trap_communitys=foo

Name snmp_trap_dest(0..9)

Description Specify the ip-adress destination for SNMP traps

Example snmp_trap_dest=127.0.0.1

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Session Border Controller

Session Border Controller template that makes the SIP Proxy act as an SBC in front of a SIP server / B2BUA. There is a great variety of configurations for this. Depending on the exact usage, the Survivability and Branch SBC configurations could also apply. This selection is the template for a typical centralized Session Border Controller used to connect endpoints coming from the Internet, usually behind NAT Routers. This template will hence perform NAT handling.

Branch Session Border Controller

Branch SBC which sits in front of a SIP server but at a decentral location instead of a central one as with the SBC template. As such, it will typically serve in Enterprise Branch Office scenarios, or in Hosted scenarios. A hosted service provider can use the product to terminate his service at the customer premise. In order to use that service, the Mediatrix Boss would receive one data center IP address as well an IP address in the local range. With that, the phones on the customer premise would only interact with the Mediatrix Boss through a local IP address. No changes to the network, routing or anything else would be necessary to support deploying a VoIP solution.

SIP Trunking Scenario

The Mediatrix Boss offers two different ways of handling SIP Trunking scenarios, via the SIP Proxy / SBC component and via the B2BUA. The selection here will make the Mediatrix Boss behave like a special SBC for SIP Trunking scenarios. As such, it will not perform a registration with a provider, the locally connected PBX / SIP server behind the SBC has to perform that. The SBC approach has the advantage that features between the local PBX and the Trunking Provider are transparently handed through. This is something a B2BUA approach cannot absolutely guarantee.

ENUM Scenario

With this template the SIP Proxy will act as an IP-to-IP gateway performing message forwarding based on ENUM lookups as well as SBC functionality for incoming calls.

Configuration Explanation

Note: The SIP-Trunk template is only available in version 6719.18 and lower!

Name ser_type

Description This value selects the profile type, the SIP Proxy is to be setup in.The Mediatrix Boss appliances support a number of predefined scenarios that are explained in more detail in the following.

"disabled" Disabled

"custom" Custom template

"standalone" Standalone

"survivability" Survivability

"sbc" Session Border Controller

"branch" Branch Session Border Controller

"enum"

Example ser_type=survivability

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Name ser_action

Description For each Call Admission rule definition, the SIP proxy permits the specification of an action in the case of a limit violation. As mentioned with the CAC parameters, simply denying calls is the simplest of all actions that can be taken. Sometimes however it is inappropriate. Imagine for example a case where we can route a call over multiple links. Since the Mediatrix Boss Series is operating at the edge of one or more bottleneck links, there might be an alternative route to the destination via a second link. The other possibility is to use the TDM network via a gateway realizing a local breakout. This second possible action is Redirect. Choose one of the two from the Drop-Down list. Future versions will come with additional actions. The possible action are "redirect" and "deny".

Example ser_action0=deny

Name ser_amt

Description This parameter is relevant for Survivability, Standalone, Branch SBC, and Enum scenarios. This parameter represents the digit combination a user has to dial for getting an outside line. In Standalone mode this will be directly used for determining the calls that have to go to the gateway. In the other modes, the parameter will only be used in Survivability mode. In theory, you can put any regular expression here, although in reality this will mostly be a single digit as for example a 0 or a 9.

Example ser_amt=0

Name ser_cac

Description This option applies to the Branch SBC as well as to the Survivability scenarios. All Call Admission control related settings for these two scenarios will only apply if this option is activated. This parameter however does not apply for any Call Admission Control settings towards a gateway. It only applies to to the Call Admission Control towards the central SIP server. The possible actions are "on" and "off".

Example ser_cac=on

Name ser_cacchecknet

Description This parameter should be checked if there is a local PSTN in the branch office. If this parameter is not checked, the SIP Proxy will assume that all Calls to and from non-local destinations will go over the bottleneck link and hence require Call Admission Control (of course only if all other parameters match). This would however mean, that for a Call coming in over the local gateway, going to the central SIP server then coming back in to the SIP Proxy, Call Admission Control would be falsely performed although the Media Stream would stay in the local network. Strict checking will additionally check the SDP header to determine how the RTP streams would flow to correctly perform the Call Admission Control. The possible actions are "on" and "off".

Example ser_cacchecknet=on

Name ser_callforward

Description This parameter is a list, meaning that you have to attach a number from 0 - ... . Therefore, it is of course possible to have multiple entries where the numbers must of course be unique. Also note that this parameter can only be used in conjunction with the ser_callforwardtarget parameter. This parameter is only relevant in the Survivability scenario and will only become active in survivability mode. The parameter will identify an incoming call in survivability mode. If the user part of the request URI matches the this parameter, the call will immediately be redirected to the user given in the ser_callforwardtarget parameter with the matching index.

Example ser_callforward0=4711

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Name ser_callforwardtarget

Description This parameter is a list, meaning that you have to attach a number from 0 - ... . Therefore, it is of course possible to have multiple entries where the numbers must of course be unique. Also note that this parameter can only be used in conjunction with the ser_callforward parameter. This parameter is only relevant in the Survivability scenario and will only become active in survivability mode. A call in survivability mode directed to the user name given in the ser_callforward with the matching index will be directed to the username given here.

Example ser_callforwardtarget0=4712

Name ser_children

Description Determines the number of threads being forked off in the SIP proxy per interface. A higher number of threads will lead to a higher efficiency in using the available CPU resources at the expense of higher memory and resource usage. Recommended number for smaller Mediatrix Boss boxes is 8, for bigger ones 16 or 32.

Example ser_children=8

Name ser_cxc

Description Usually the Mediatrix Boss will try to send SIP messages coming from internal devices to the PBX. This however only applies if the domain this message is addressed to is either the PBX or a DNS alias of it. Another possibility is if the domain is specified in the domain field. If that is not the case, the messages will be forwarded to the domain they are addressed to. This behavior can be overridden by configuring the ser_cxc parameter. If set to true all messages will be forwarded to the PBX.

Example ser_cxc=on

Name ser_debug

Description This option will enable debug support for the SIP proxy. The debugging messages will be printed out in the system log. See "syslog" documentation for more information on how to read and use this debug output. Note that some output will be generated by default. Among these is CDR information in survivability mode that will always be generated. The debug mode should only be switched on for testing purposes since depending on the traffic conditions, a lot of log information can be generated. The possible values are "on" and "off".

Example ser_debug=off

Name ser_domain

Description The Domain list parameter is one of the most important parameters for the SIP Proxy. The SIP Proxy will only apply the Survivability / Call Admission Control / SBC logic to SIP messages directed at specified domains. Domain as used here refers to the rear part of a URI parameter. This means that if you have some client send a SIP message for [email protected] to the Mediatrix Boss appliance, it will only act upon it in the way described here if foo.bar can be found in the domain list. Otherwise it will simply perform a DNS lookup and forward the message appropriately. Possible values are either foo.bar, or an ip address.

Example ser_domain0=foo.bar

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Name ser_dst_ip

Description The notion of a link is fundamental to the way the Call Admission Control is done. The simplest case of a link is a physical network connection. The Mediatrix Boss however does not stop there. A link is closely associated with the routing functionality offered by the Mediatrix Boss. Typically source and destination addresses are used for classifying packets as belonging to a certain link (matches the standard IP networking principles). The source address could for example be used to configure groups for the phones. Group 1 could get a range of IP addresses, Group 2 another. Simply enter the Source and Destination IP addresses or network addresses that you want have the limit and action applied to. The mentioned IP addresses pertain to the SIP signalling traffic. The source would for example be the local branch network; the destination would most likely be the IP PBX. Use a "0" for each byte of the IP address where you want to specify network ranges.

Example ser_dst_ip0=192.168.1.1

Name ser_dualreg

Description This enables dual registration functionality inside the SIP proxy for supporting Survivability scenarios with Cisco or Siemens phones. In principal, this is not necessary since our survivability functionality is completely transparent to the UAs. Some UAs however support displaying something to the user to inform him about survivability mode. One of these options lets the phones register to a primary server and a backup server. As soon as the primary server stops responding, they will register to the backup server for continuous operation. Sometimes, it is desirable to use this phone functionality. The Mediatrix Boss supports this by running on port 5080 in addition to the standard port. If enabled, the SIP proxy will stop responding on port 5060 and only respond on port 5080 thus emulating the backup server functionality. This option is very rarely used. The possible values are either "on" or empty for "off".

Example ser_dualreg=on

Name ser_excludegw

Description This parameter has to be considered in conjunction with the ser_return parameter. It applies to all scenarios involving the explicit definition of a gateway. If the return route for SIP messages is forced by the proxy, it might be necessary to exclude the local gateway from this, since your SIP server might want to stay in direct touch with the local gateway. This can have multiple reasons, the most likely reason would be features. The possible values are either "on" or "off".

Example ser_excludegw=on

Name ser_failover

Description This option enables you to turn on or off failover server support. Failover server support is usually realized in conjunction with DNS SRV or VRRP to have a backup take over the IP address of the primary SIP proxy. The option here only refers to the SIP part of this construct which requires to have a backup server be in the same conditions concerning states and registrations as the primary server. If this option is set, all state information will be replicated to the failover server that can be specified with the ser_failover_server parameter. The possible values are either "on" or "off".

Example ser_failover=on

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Name ser_failover_server

Description For more information on this option please also refer to ser_failover_.

This option will set the IP address of the failover server that all stated information is to be forwarded to. Note that the backup server should again have the primary server set as failover pendant, since the roles are bound to change depending on configuration of course. Although setting this up in such a manner will by no means result in messages bouncing back and forth since this is automatically detected by the software. The entry has to be an IP address.

Example ser_failover_server=10.42.50.91

Name ser_faxuri

Description This parameter must be seen in conjunction with the ser_t38redirection parameter. This means that all streams for T38 will be treated differently than the normal voice streams. In contrast to normal voice streams, T38 streams will exhibit a great deal of variance when it comes to a call in progress. T38 streams can vary from anywhere between 10 kbit/s and 100 kbit/s in throughput, meaning that a measurement is not something you want to rely on. Therefore, we do a reservation for these cases, where the bandwidth for the fax call remains reserved for the entire duration of the call. In this case, all calls matching this URI will have the maximum bandwidth reserved for the entire duration of the call.

Example [email protected]

Name ser_forwarding

Description This function activates server based call forwarding in Survivability mode, or as a Standalone server. It does not have an effect in any other usage scenario. Even in the Survivability scenario, it will only have an effect if you are currently in survivability mode. In that mode, this flag instructs the SIP Proxy to catch any SIP 302 response sent by an UA and creates a new INVITE request from that. This also works in conjunction with SIP forking. If the original request was forked and one UA responds with 302, all other branches will be Cancelled. The possible values are "on" or "off".

Example ser_forwarding=off

Name ser_fr_inv_timer

Description Once the proxy receives an INVITE it waits the number of seconds given in this option for a final response like OK or CANCEL. If no such final response is received within the given time frame the proxy sends a 408 Request Timeout back to the phone that sent the INVITE. The default value is 200 seconds.

Example ser_fr_inv_timer=300

Name ser_gateway_ip

Description This parameter is a list, meaning that you have to attach a number from 0 - ... . Therefore, it is of course possible to have multiple entries where the numbers must of course be unique. As the name already indicates, this parameter denotes the gateway IP address. Gateways can be defined for most scenarios except ENUM and SBC. For all remaining scenarios, it is possible to define multiple gateways. The gateways will be addressed in a hunting mode, meaning that the SIP proxy will first always address the first gateway. If it gets an error from there, it will proceed to the second one and so on. For the exact behavior in this case, please see the centryser_hunting parameter.

Example ser_gateway_ip0=192.168.1.2

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Name ser_gateway_prefix

Description The prefix determines what digits or characters to attach to the front of the URI before forwarding to the gateway. The prefixing will only apply in Survivability mode. The prefixing will always be executed after stripping the URI. Also refer to ser_gateway_strip for more information on this.

Example ser_gateway_prefix0=0

Name ser_gateway_strip

Description Determines how many digits should be stripped from the front of the number before forwarding the request to the gateway. Note that this only applies to Survivability scenarios. In normal mode, the SIP Proxy will never do changes to the URI when forwarding requests to a gateway. Note that this parameter does not specify what to remove from the front of the number, it rather specifies the number of digits to remove. It should be noted that this not only works for digits, but for all characters permitted in SIP URIs.

Example ser_gateway_strip0=1

Name ser_gateway_transmit

Description This determines the SIP transmit type to be used towards the gateway.

RFC3261 currently defines three transmit types, udp, tcp, and tls. Most systems currently use udp, so this will most likely be your protocol of choice. There are reasons however for using tcp, which is also supported by a lot of systems. If you need security, there is no other way but to use tls. Beware however that you cannot do traces of tls encrypted signalling to track down errors. It is therefore highly recommended to set up a system with an unencrypted protocol and only switch to tls once everything works. The possible values are either "udp", "tcp", or "tls".

Example ser_gateway_transmit0=tcp

Name ser_hunting

Description This parameter determines the behavior of the SIP Proxy if multiple gateways are being set up. If set to on, a gateway hunting will be done meaning that the SIP Proxy will automatically try the next gateway if the previous one did not respond or responded with an error. Mostly this error will be a "486 Busy". Note that the hunting usually only applies in Survivability mode, since in normal mode, the central SIP server can correctly address the SIP messages to go to the correct gateway. In these cases the central SIP server would perform the gateway hunting. There are however special scenarios where the SIP Proxy itself is specified as a gateway (e.g. Branch SBC functions). In that case, the Proxy will of course also perform the hunting functionality. The possible values are either "on" or "off".

Example ser_hunting=on

Name ser_limitcall

Description Each SIP session uses a certain average bandwidth depending on the codec being used. For G.711 for example the bandwidth required for each session is around 75 - 80 kbit/s including the overhead packets. The bandwidth check parameter here, identifies how much bandwidth must still be available in order to permit another session. Note that you have to enter the parameter manually, since codices can be renegotiated during a session. This means that this parameter has to reflect the worst case that can occur in your network. Otherwise this would mean that your session is permitted first, then renegotiated with another codec and denied there. This kind of behaviour would be very user unfriendly. For each Call Admission decision, as to whether a call should be permitted or not, the worse case is hence assumed. This field should therefore reflect this worst case. It can also be used for deliberately leaving some room to the bandwidth limit by checking for a higher bandwidth than required. The difference would then remain unused which can be desirable under certain conditions. Note that this field is entered in "bytes/second", not "bits/s".

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Example ser_limitcall=8000

Name ser_limitclass

Description The Limit Class definitions make the connection between the SIP and the IP layer in Call Admission Control scenarios. All traffic class definitions are used for making assessments as to whether additional calls can be accommodated. For the Limit definitions, the Bandwidth Management Configuration is used where a traffic class with certain limits is defined. The limit is enforced by the Call Admission Control functionality. For more information on the definition of the traffic rules refer to the Bandwidth section of this handbook or the User Guide.

Example ser_limitclass0=eth0-2:4

Name ser_multiport

Description Only use this parameter if you know what you are doing. This option has a severe impact on the SIP request routing behavior. Setting this option will make the SIP proxy modify all SIP requests in such a way, that every UA connected behind the SIP proxy will be mapped to a separate port on the SIP proxy. This means that the central SIP server will think that all phones have the IP address if the SIP proxy, but each one with a separate port. This enables supporting features in e.g. NAT scenarios where the return route also has to pass through the SIP proxy. Please also refer to ser_return for more information. This option is not required if your SIP server features an option to route requests through a SIP proxy. The possible values are either "on" or "off".

Example ser_multiport=on

Name ser_notifyreturn

Description This parameter enables or disabled the function of sending Survivability Notifies. If enabled, the SIP proxy will send a SIP NOTIFY message to all registered devices each time the Survivability mode changes. This means that a message will be sent, if the Survivability mode becomes active, another one will be sent if normal operations is being restored. The possible values are either "on" or "off".

Example ser_notifyreturn=on

Name ser_ownnumber

Description This parameter is being used for short dialing in Survivability and Standalone mode. As such, it of course also applies to the Branch SBC mode. If you have a telephone with the phone number e.g. 49893333567, where 567 is the short number of the phone, you want to be able to use both the long and the short number in survivability mode where the PBX normally doing the translation is offline. The phone registers (SIP registration) with the full number. To make it reachable now, the Mediatrix Boss must know the prefix, hence the 49893333 which it will attach to the short number to locate the phone. If you do not set this up, it will only have an impact on this feature (Short numbers in Survivability mode). It is not used for anything else.

Example ser_ownnumber=49893333

Name ser_pbx

Description This entry is the IP address of the softswitch, SIP server, or SIP PBX. In Survivability, Enum, SBC, and Branch SBC scenarios, this will be the target of all messages received from any UAs. In the Standalone scenario, this entry does not have any effect, since the Mediatrix Boss will be the main SIP server. It is possible to specify both IP addresses and domain names. The domain names can also be DNS SRV entries. Also see the 9.5.6 parameter that determines the protocol to use for communication with the SIP server. You can specify either IP addresses or domain names.

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Example ser_pbx=192.168.10.3

Name ser_pbxalias

Description This parameter is a list, meaning that you have to attach a number from 0 - ... . Therefore, it is of course possible to have multiple entries where the numbers must of course be unique.

This parameter will be used for identifying downstream messages as coming from the PBX. This is important for e.g. performing the CAC functionality where incoming calls have to be handled. Messages can come from phones or the PBX. The PBX could be a load-balanced or cluster system meaning that messages could come from multiple IP addresses. The PBX Alias is the way to tell the Mediatrix Boss that all these messages are coming from the PBX. If the PBX only has a single IP address it is not necessary to set this parameter, you can just leave it blank. You can specify either IP addresses or domain names.

Example ser_pbxalias0=192.168.10.3

Name ser_pickup

Description This parameter is a list, meaning that you have to attach a number from 0 - ... . Therefore, it is of course possible to have multiple entries where the numbers must of course be unique.

This parameters allows you to specify extensions / URIs that will be excluded from Call Admission Control. Usually this is only relevant for pure telephony scenarios. There it is more than common to use certain extensions for doing control functionalities. This means that you make a call to such a special extensions for e.g. activate a call forwarding. Although a call is made, there is no real media that is ever sent in any case. Another example for this are pickup group extensions. Enter a list of all extensions that should be included from CAC. Note that it is possible to specify arbitrary regular expressions. Therefore also beware to escape (prefix by \) all special characters used in regular expressions as for example .,?,* . Also see section “Regular expressions” on page 182 for more details.

Example ser_pickup0=\*5

Name ser_prefixpbx

Description As described in the ser_redirectpbx parameter, this prefix will be added to the dialled number to inform the central SIP server that an alternative routing needs to be performed. This prefix can be any combination of alphanumeric characters and special characters like "*" or the hash sign. In telephony applications, the prefix will most likely be numeric.

Example ser_prefixpbx=*0

Name ser_proxy

Description This parameter will have an effect on the response routing. This parameter must be seen in conjunction with the centryser_return parameter since it will only have an effect if this one is set. The central SIP server will return these messages to the IP address specified here. This applies only to SIP requests. Response routing is determined by the Record-Route parameter which is explained in ser_rrpreset.

Example ser_proxy=192.168.6.7

Name ser_port

Description This specifies the port for the return routing of the messages. Similar to centryser_proxy parameter, this has only an effect in conjunction with the centryser_proxy parameter. Please see these two for more information on this option.

Example ser_port=5060

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Name ser_redirectpbx

Description If there is no more bandwidth for permitting an extra session, the Call can be either denied, or it can be redirected to a gateway. These actions are described with the ser_action in more detail. The field here specifies another possible action. The SIP Proxy can modify the dialled number by prefixing something to let the IP PBX know that the usual call routing path does not apply. In this case, the rerouting to e.g. a local gateway can be performed by the central IP PBX. This does not require any Call Admission Control awareness of the central PBX, simple call routing features with multiple alternative routes, as for example found in least cost routing features, do suffice. The central IP PBX can then route the call over the local gateway in the branch office, thus not generating traffic load on the bottleneck link. Also refer to the ser_prefixpbx and ser_redirectpbxamtonly parameters for more information. The possible values are either "on" or "off".

Example ser_redirectpbx=on

Name ser_redirectamtonly

Description This parameter only has an effect in conjunction with the ser_redirectpbx parameter. If enabled, the SIP Proxy will only permit the redirection of outgoing calls. All other non-internal calls will not get redirected but rather denied with a "606 Not Acceptable" message.

Example ser_redirectamtonly=on

Name ser_reserve

Description Enables or disables the bandwidth reservation. If enabled, the SIP proxy will reserve the bandwidth for all initiated SIP requests that have not been finally answered. This means that there could be unused bandwidth in your line, since reserved bandwidth might never be claimed because a Session is denied. In that case, other Sessions could have been accommodated. The higher the bandwidth available on your link, the more significant this effect gets. On the other hand, if this option is enabled, I can never overbook a link. The possible values are either "on" or "off".

Example ser_reserve=on

Name ser_sbc_pbx_mode

Description This key refers to the PBX Mode setting of the SBC template (refer to “Basic Settings” on page 85).

Example ser_sbc_pbx_mode

Name ser_sbc_registration_check

Description This is the basic configuration key for the checkbox "Registration Check" in the SBC Settings menu (refer to “SBC Settings” on page 86).

Example ser_sbc_registration_check=off

Name ser_src_ip

Description This is a Call Admission Control parameter that has to be seen in conjunction with the ser_limitclass and the ser_dst_ip parameter. This specifies the destination of the SIP signalling for which this CAC rule defined in the ser_limitclass should apply. This parameter is again a list parameter. The matching up with the ser_limitclass is done through the index number. See also the ser_dst_ip parameter for a more in depth explanation of the meaning.

Example ser_src_ip0=10.10.1.1

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Name ser_survcachetimeout

Description This is the time the last alive message from the central PBX will assumed to be valid, even if checks fail in the meantime. This lazy timeout is intended to reduce the number of alive messages sent, and to avoid a frequent switching due to e.g. a flapping line. This time must be given in seconds.

Example ser_survcachetimeout=30

Name ser_reroutecodes

Description This option only applies to survivability scenarios. By default, the SIP Proxy will reroute for all error conditions, meaning that for every non successful reply sent for a SIP request, a rerouting to the gateway will be induced. For this parameter just specify all codes where no rerouting should apply. It is possible to specify arbitrary response codes, not only the ones defined in the RFC. The most frequently used however are:

"400" 400 Bad Request

"401" 401 Unauthorized

"402" 402 Payment Required

"403" 403 Forbidden

"404" 404 Not Found

"405" 405 Method Not Allowed

"406" 406 Not Acceptable

"410" 410 Gone

"480" 480 Temporarily Unavailable

"486" 486 Busy

"487" 487 Request Terminated

"603" 603 Decline

"606" 606 Not Acceptable list separated by blanks.

Example ser_reroutecodes=400 401 402

Name ser_rewrite

Description This option is only relevant for all scenarios that involve a separate SIP server. Some SIP servers react very sensitively to the SIP URI. If a UA does not support outbound proxy functionality, the SIP domain when routing messages through the SIP proxy will be that of the SIP proxy, unless of course DNS / DNS SRV is being used. As mentioned, some SIP servers will then refuse to handle the sent requests. Therefore, the SIP proxy allows you to rewrite the SIP domain on all messages sent to the SIP server. The domain that the requests will get rewritten with is that of the ser_pbx parameter. The possible values are either "on" or "off".

Example ser_rewrite=off

Name ser_rrpreset

Description This parameter enables presetting the record route header that will be inserted by the SIP proxy in the SIP requests that are forwarded to the central SIP server. This parameter will only be applied if its use has been enabled by setting the ser_rrpreseton parameter. By default, the SIP proxy will insert the correct Record Route header. The ability to do this automatically is however hampered in NAT and multiple interface scenarios. In such circumstances, it is necessary to statically fix this parameter.

Example ser_rrpreset=off

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Name ser_rrpreseton

Description This parameter enables or disables the use of the ser_rrpreset parameter. Also refer to this one for more information. The possible values are either "on" or "off".

Example ser_rrpreseton=off

Name ser_rtp

Description This parameter determines if the SIP proxy also handles the media streams. In Branch SBC and SBC scenarios, the SIP proxy will always do this. In all other scenarios, this is optional and can be modified with this parameter. If set active, the SDP information in the SIP messaging will be modified in a way so that the media stream runs through the SIP proxy. This is a requirement in NAT scenarios and for SRTP termination (optionally available). The possible values are either "on" or "off".

Example ser_rtp=off

Name ser_return

Description This parameter determines if the SIP proxy will change all upstream messages in such a way, that all requests coming from the main server will also flow through the SIP proxy under all circumstances. This applies to all scenarios that involve another SIP server such as SBC, Survivability and ENUM scenarios. This will make the SIP proxy modify all contact fields that determine the SIP routing such as the Contact, Route, Record-Route header fields. If your SIP server and UAs support Outbound Proxy functionalities, it is not necessary to set this field. Be very careful using this field since it can break some functions of your SIP server. In will for example have adverse effects on any features involving SIP forking. The possible values are either "on" or "off".

Example ser_return=off

Name ser_t38redirection

Description If enabled, the SIP proxy will use bandwidth reservation to handle the greatly varying bandwidth requirements of a T38 media stream instead of measuring it. In order to let the SIP proxy know where he can expect T38 to be negotiated you can specify special fax URIs. See the ser_faxuri parameter for more information on that.

Example ser_t38redirection=on

Name ser_rereg

Description Timeout value for SIP Proxy to confirm registrations in survivability mode.

This means that independent of the registration timeout value given in the UA, the SIP proxy will only OK them with the timeout given here in survivability mode. This is to expedite the reregistration with the SIP server once the connectivity is restored.

Example ser_rereg=720

Name ser_transmit

Description This determines the SIP transmit type to be used towards the SIP server.

RFC3261 currently defines three transmit types, udp, tcp, and tls. Most systems currently use udp, so this will most likely be your protocol of choice. There are reasons however for using tcp, which is also supported by a lot of systems. If you need security, there is no other way but to use tls. Beware however that you cannot do traces of tls encrypted signalling to track down errors. It is therefore highly recommended to set up a system with an unencrypted protocol and only switch to tls once everything works. The possible values are either "udp", "tcp", or "tls".

Example ser_transmit=udp

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Name ser_timeout

Description Timeout value for SIP Proxy to take action itself. This parameter is important for all scenarios, although a save default is in place for e.g. the SBC template. For survivability, this parameter is absolutely critical, since it defines the timeout for the Survivability function. If there is no response provisional or final within this time period, the SIP Proxy will assume Survivability mode.

Example ser_timeout=2

Name ser_b2buasurv

Description This function is necessary for gateways that do not properly support SIP forking. Checking this box will make the Mediatrix Boss behave like a B2BUA towards the voice gateway in survivability mode. This means that for each session from and to the gateway, its own Call ID will be assigned. Towards the SIP endpoints, the Mediatrix Boss will continue to behave as a SIP Proxy. Usually, this function will not be necessary if you have no forking. Forking is necessary if you have multiple SIP endpoints with the same SIP username. This will normally be the case in voice scenarios where you have multiple SIP endpoints with the same phone number. This is commonly described as keyset feature or multiline appearance. Incoming calls in such multiline scenarios sometimes pose difficulties for gateways because the SIP proxy in the Mediatrix Boss will fork the request. This will lead to the fact that multiple SIP responses are forwarded to the gateway which is a problem for some of them. With the B2BUA feature enabled, the Mediatrix Boss will terminate this call and only forward a single response to the gateway.

This parameter applies only to the Survivability and Branch SBC templates.

Example ser_b2buasurv=on

Name ser_city_code

Description This is used to convert numbers routed to the Mediatrix Boss to full E.164 numbers for doing proper ENUM lookups. Enter the plain city code (3-digits area code in the US). It can also be a number of variable length in many other countries.

This parameter applies only to the ENUM template.

Example ser_city_code=89

Name ser_country_code

Description This parameter applies only to the ENUM template. Enter the Country Code for your country without any preceding digits. This would be a "1" for the USA, the "49" for Germany, etc. This parameter will be used to properly convert any incoming numbers to E.164 format. You have to enter this parameter correctly. Otherwise outbound calls will not work.

Example ser_country_code=49]

Name ser_dest_ip_mask

Description This applies only to the Branch SBC and Survivability Templates. This parameter is part of the link definition for doing CAC. CAC is done on a per link basis. This link definition is done via the Source and Destination IP address. The parameter here is the netmask for the destination IP. It will usually be the PBX IP address for which you do CAC. In that case, the netmask will only point to a single host instead of a network.

Example ser_dest_ip_mask=255.255.255.255

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Name ser_distance_prefix

Description Enter the number you have to dial on your phone before making a long distance call. In the US this is a "1", in most European countries a "0". Do not confuse this with the prefix you have to dial on your phone to get an outside line on the PBX. This value is needed to correctly convert incoming and outgoing numbers into E.164 format. Assume for example that you want to call somebody with the area code 221. In order to do that, you have to dial a number in front of that, e.g. 0221 xxxx. In that case enter the 0 for the the parameter here.

This parameter applies only to the ENUM template.

Example ser_distance_prefix=0

Name ser_external_ip

Description All Mediatrix Boss products support multiple virtual and non virtual interfaces. When acting as a session border controller, there however must be a unique uplink interface through which outgoing media streams should be routed. When looking at the SIP messages that are leaving the LAN network of the SBC towards the outside of the SBC, all SDP bodies will contain this external IP address. Thus, all endpoints having a connection with some endpoint inside the SBC will be sending their media streams to this external IP address. The external IP address however has a different meaning in the different templates. In a branch session border controller scenario it denominates the side towards the PBX, in most cases the WAN interface or a VPN connection. In the central SBC scenario, the external IP address will point to the endpoints that can register from behind NAT firewalls.

This parameter only applies to the various SBC templates.

Example ser_external_ip=10.10.10.10

Name ser_fix_looseroute

Description This only applies to the Survivability and Branch SBC templates. Loose routing is defined in RFC3261 and will make the Mediatrix Boss route all in dialog messages the same way that the dialog setup was routed. This however assumes properly behaving SIP endpoints and a properly behaving SIP servers. In simple scenarios, where the endpoints are registered to a server, this works for most endpoints. With a survivability scenario, where the messages are routed onwards, and where the proxy is on the return route in addition to that, the correct behavior of loose routing is not guaranteed with all devices. In order to avoid that, loose routing can be disabled in normal operation (non survivability). If you see in dialog messages like ACK and BYEs not flowing as expected. A typical sign is ACKs or BYEs coming from the softswitch being directly returned to it.

Example ser_fix_looseroute=on

Name ser_fromhost

Description Please refer to the ser_usesipping parameter for more information on that.

When using the SIP OPTIONS messages to check whether a server is still available, you can configure the URI host part the Mediatrix Boss should be sending. Simply specify the host part of the URI with this parameter. The switch must respond to the URI with a non 5xx and 6xx response. You can also refer to the ser_userpart for setting the user part of the URI.

This only applies to the Survivability and Branch SBC templates.

Example ser_fromhost=sipserver.foo.bar

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Name ser_gateway_access

Description The access code is a way of forcing calls onto a certain gateway in Survivability mode. This only applies to Survivability mode! Under normal operation, it is assumed that the PBX is responsible for directing the calls to the appropriate gateway. Any call received in Survivability mode that starts with an appropriate prefix will be forwarded to the best matching gateway. If you put a 0 there for example, any number starting with a 0 would be sent to this gateway. The check is made sequentially, not in a best match manner! This means the gateway with the first access code match will be selected. If there is no match, the first gateway will be selected. If hunting is enabled and the gateway with the access code match is busy, the call will be forwarded to the next gateway in the list.

This only applies to the Branch SBC and Survivability templates.

Example ser_gateway_access=*9

Name ser_gateway_action

Description The Redirect action will make the Call Overflow to the next possible gateway. If there is none, the Redirect will behave the same as a Deny. The Deny action will abort the call with a "606 Not Acceptable" response. The CAC has to be activated for this parameter to have an effect.

This only applies to the Survivability template.

Example ser_gateway_action=deny

Name ser_gateway_cac

Description It is possible to perform Call Admission Control in Survivability mode before the Call is being forwarded to a gateway. This is necessary since it cannot be assumed that the gateway is always connected to the LAN. It could be located off site. This could mean that in a backup case, it can only be reached via some backup link. There are many scenarios that are conceivable. The Call Admission Control parameters will allow limiting the number of calls going to the gateway. For more information on Call Admission Control refer to the more detailed explanation in the general section. This only applies to the Survivability template.

Example ser_gateway_cac=on

Name ser_gateway_dest_ip

Description This parameter applies only for CAC scenarios towards the gateway. In order to configure Call Admission control, we first need to define a link. The link definition here is done by entering the source and destination IP ranges. After defining the link, you can configure a policy, since a reaction in case of an overload must be configured. Here you can define the source IP address for which the CAC rule towards the gateway would apply. Usually this will be the IP range of your SIP phones. Also refer to the ser_gateway_cac parameter for more information.


Example ser_gateway_dest_ip=10.10.10.10

Name ser_gateway_limitclass

Description In order to define a CAC action, you will need to associate a limit class with your CAC link. Also refer to the ser_cac and ser_gateway_cac sections for more information on this.


Example ser_gateway_limitclass=eth0-1:3]

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Name ser_gateway_src_ip

Description This parameter applies only for CAC scenarios towards the gateway. In order to configure Call Admission control, we first need to define a link. The link definition here is done by entering the source and destination IP ranges. After defining the link, you can configure a policy, since a reaction in case of an overload must be configured. Here you can define the destination IP address for which the CAC rule towards the gateway would apply. Usually this will be the IP address of the gateway. Also refer to the ser_gateway_cac parameter for more information.


Example ser_gateway_src_ip=10.23.34.10

Name ser_gwdnsalias

Description This parameter only has an effect if ser_gwdnsaliascheck is set. In Branch SBC mode, the Mediatrix Boss can act as a virtual gateway for the PBX. If a media gateway is placed inside the branch SBC, there is no way for the PBX to directly address the media gateway on the internal private network of the branch SBC. In those cases, the PBX can send the messages directly to the branch SBC IP address. The branch SBC will always change the contact header of any message or response coming from the gateway to the external IP address of itself. Therefore, the PBX will also send all subsequent requests through the SBC. That way, the media gateway is completely hidden from the PBX. It could be the case however, that the gateway is provisioned via a DNS name in the PBX. In that case, you will also want to have the DNS name in the contact header of any SIP message coming from the gateway. You can set this DNS name here.

This only applies to the Branch SBC template.

Example ser_gwdnsalias=gw.foo.bar

Name ser_gwdnsaliascheck

Description This item will activate the setting of the ser_gwdnsalias parameter. Please see the ser_gwdnsaliascheck key for more information.

This only applies to the Branch SBC template.

Example ser_gwdnsaliascheck=on

Name ser_ignoreprefix

Description All prefixes specified here will be ignored for any CAC decision. This means that if you specify a prefix of xx here, the CAC checks will be based on the dialed number with the specified prefix being removed first. This feature will only be used in telephony scenarios where special prefixes can be dialed to e.g. enable the transmission of the Caller ID. Function like these are often used by prefixing an access code to the actual number you want to dial. This prefix must of course not change the CAC assessment of where the call is going. This only applies to the Survivability and Branch SBC operation modes.

Example ser_ignoreprefix=on

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Name ser_internal_ip

Description Whenever it is operating as an SBC, the Mediatrix Boss is basically forming the bridge between two different networks. All Mediatrix Boss products however support multiple virtual and non virtual interfaces. When acting as a session border controller, there must be a unique interface through which local endpoints communicate should be routed. When looking at the SIP messages that are coming from the outside network, all SDP bodies will be rewritten with this internal IP address. Hence, all endpoints having a connection with some endpoint outside the SBC will be sending their media streams to this internal IP address. At least one interface of the Mediatrix Boss must have this internal IP address assigned. For more information on this topic also refer to ser_external_ip. This only applies to the different SBC templates.

Example ser_internal_ip=192.168.1.1

Name ser_internal_mask

Description Please also refer to ser_internal_ip for more information on this topic.

This parameter defines the internal network that is associated with the SBC. This can actually be a superset of the netmask associated with the actual interface of the Mediatrix Boss. It defines what IP addresses are to be treated as being SBC internal. All other IP addresses will be treated to be external ones no matter from which interface of the SBC they are This only applies to the different SBC templates.

Example ser_internal_mask=255.255.255.0

Name ser_international_prefix

Description Enter the number you have to dial on your phone before making an international call. In the US this is the "011" in most European countries the "00". This parameter will be used to properly convert dialed numbers to E.164 format.

This only applies to the ENUM mode of operation.

Example ser_international_prefix=00

Name ser_local_gw

Description You need to configure whether media handling should be done for calls to and from gateway calls. If turned on, the gateway will be assumed to be local, and media handling will be disabled for all calls between local endpoints and the local gateway. If not checked, the gateway is assumed to be external, e.g. located at the central softswitch. This will enable media handling for all external calls.

This parameter only applies to the Branch SBC operational mode.

Example ser_local_gw=on

Name ser_looseroute

Description Loose routing is defined in RFC3261 and will make the Mediatrix Boss route all in dialog messages the same way that the dialog setup was routed. This however assumes properly behaving SIP endpoints and a properly behaving SIP servers. In simple scenarios, where the endpoints are registered to a server, this works for most endpoints. With a survivability scenario, where the messages are routed onwards, and where the proxy is on the return route in addition to that, the correct behavior of loose routing is not guaranteed with all devices. In order to avoid that, loose routing can be disabled in normal operation (non survivability). If you see in dialog messages like ACK and BYEs not flowing as expected. A typical sign is ACKs or BYEs coming from the softswitch being directly returned to it.

Setting this value to on will turn on this feature. In order to have standard loose routing, simply leave this blank.

This only applies to the Survivability and Branch SBC operational modes.

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Example ser_looseroute=on

Name ser_pbx_port

Description With this parameter, you can configure to which port the Mediatrix Boss will be sending the SIP requests towards the PBX. If this is not defined, the standard ports for the selected transmit types will be chosen. This is port 5060 for UDP and TCP and port 5061 for SIP TLS.

Example ser_pbx_port=5060

Name ser_pbxnode

Description Some IP PBXs / Softswitches support clustering or redundancy through a second host. There are two ways of implementing this clustering, either transparent, in which case we do not have to handle anything explicitly in the branch office, or with two IP addresses. This field is meant for the latter concept which is frequently being used if the 2 nodes of a cluster have been spatially separated. In that case they are available through different WAN links thus increasing the robustness of the overall solution. If such redundancy precautions have been taken in the data center, the branch equipment must also support this. If this field is set, the Survivability function of the Mediatrix Boss will only take over if both the PBX as well as the 2. Node have failed. If only the first node has failed, messages will automatically be forwarded to the second node.

This parameter only applies to the Survivability and Branch SBC modes of operation.

Example ser_pbxnode=62.222.111.5

Name ser_pickup

Description All prefixes specified here will not be considered at all for a CAC decision. Regular expressions can be specified here. This can be used for excluding certain number ranges because you know that there will always be enough bandwidth, or because you maybe want to exclude emergency calls. Excluded calls can cause an overload on a link if you are not careful in configuring your bandwidth rules.

In Voice scenarios with a central IP PBX as SIP server, there is a commonly offered feature named Pickup group. This permits using any telephone and dial a certain special extension like for example "*9" in order to pick up the call of another ringing telephone inside this branch office. In such a case, the Mediatrix Boss must be told that the "*9" is not a regular number but rather the special extension for that. No Admission Control will be performed on the extensions entered here. Note that you can enter a regular expression here and the above described functionality will apply to all extensions matching this expression. This also means that characters having a predefined meaning in a regular expression must be escaped to match the real character. Escaping can be done by adding a backslash before the character in question. In order to match the "*9" used in the example above, you hence have to enter a "\*9" into the field.

Caution: This is a dangerous function since it can break CAC altogether, especially when using regular expressions. Be very careful that the expression specified here only matches the actual calls that you want to have filtered and no more.

Example ser_pickup=9

Name ser_presence

Description This is a very rarely needed parameter. You only need to consider it, if your presence server is to directly communicate with the SBC without interaction of your PBX. It is being used for doing proper internal classification of messages and separating out all presence requests. If you are unsure about this, simply leave it blank which should work okay for almost all scenarios. In that case, the presence messages would be sent to your SIP server that should in turn forward them correctly.

This applies only to the SBC mode of operation.

Example ser_presence=10.11.12.13

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Name ser_pstn_gw

Description This parameter specifies the IP address of the used SIP gateway. It usually does not have to be specified, but the Mediatrix Boss can optimize some media streams if it e.g. knows the IP address of the VoiceMail and the Media gateway. This applies to the SBC mode of operation only.

Example ser_pstn_gw=10.10.10.5

Name ser_retrans_final

Description This parameter specifies the time after which the Mediatrix Boss will do the final retransmission of a message to the trunking side if no response (provisional or final) was received in this period. The time is specified in milliseconds.

This only applies to the SIP Trunking template.

Example ser_retrans_final=8000

Name ser_retrans1

Description This parameter specifies the time after which the Mediatrix Boss will do the first retransmission of a message to the trunking side if no response (provisional or final) was received in this period. The time is specified in milliseconds.


Example ser_retrans1=2000

Name ser_retrans2

Description This parameter specifies the time after which the Mediatrix Boss will do the second retransmission of a message to the trunking side if no response (provisional or final) was received in this period. The time is specified in milliseconds.



Name ser_retrans3

Description This parameter specifies the time after which the Mediatrix Boss will do the third (and usually final retransmission) of a message to the trunking side if no response (provisional or final) was received in this period. The time is specified in milliseconds.



Name ser_rtp_proxy_tos

Description You can define the TOS bit that will be set for all outgoing media streams.

Note that this parameter only applies to the SBC templates and to those calls where the SBC component is doing actual media handling. If you want to do TOS tagging beyond this, please refer to the firewall section.

Example ser_rtp_proxy_tos=0x184

Name ser_sipport

Description This defines the port where the Proxy / SBC component will be listening for incoming SIP requests. If left blank, the default 5060 port will be used. Note that setting this port only applies to the UDP and TCP variants. Refer to the ser_tlsport for setting the TLS port.

Example ser_sipport=5064

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Name ser_src_ip_mask

Description This parameter is used for CAC settings only, so you should also refer to the ser_cac section for more information. This parameter defines the netmask for the ser_src_ip parameter. This parameter applies only to the Survivability and Branch SBC operational modes.

Example ser_src_ip_mask=255.255.255.0

Name ser_srtp

Description Whenever turned on, the Mediatrix Boss will start terminating SRTP calls that are coming from clients and passing to the internal network of the SBC. The Mediatrix Boss will then rewrite all SDP bodies of incoming messages. There, SAVP offerings will be replaced with AVP offerings. Once the call is established, the media stream will be decrypted. To the SIP endpoint side, the media stream will be encrypted if SAVP was offered by the client. This parameter only applies to the SBC mode of operation.

Example ser_srtp=on

Name ser_surv

Description In the Branch SBC mode of operation, Survivability functionality must be separately enabled. Set this value to on to activate survivability. In this case, the Mediatrix Boss will behave exactly the same way as in Survivability mode, but it will do the media handling in addition to that.

This parameter only applies to the Branch SBC mode of operation.

Example ser_surv=on

Name ser_tlsmethod

Description Sets the TLS protocol method which can be:

• TLSv1 - means the Mediatrix Boss will accept only TLSv1 connections (rfc3261 conformant).

• SSLv3 - means Mediatrix Boss will accept only SSLv3 connections

• SSLv2 - means Mediatrix Boss will accept only SSLv2 connections (almost all old clients support this).

• SSLv23 - means Mediatrix Boss will accept any of the above methods, but the initial SSL hello must be v2 (in the initial hello all the supported protocols are advertised enabling switching to a higher and more secure version). The initial v2 hello means it will not accept connections from SSLv3 or TLSv1 only clients.

If you want RFC3261 conformance and all your clients support TLSv1 (or you are planning to use encrypted tunnels only between different Mediatrix Boss proxies) use TLSv1. If you want to support older clients use SSLv23 (in fact most of the applications with SSL support use the SSLv23 method). This applies to the central SBC mode of operation.

Example ser_tlsmethod=[VALUE]

Name ser_tlsport

Description This defines the port where the Proxy / SBC component will be listening for incoming SIP TLS requests. If left blank, the default 5061 port will be used. Note that setting this port only applies to the TLS SIP variant. Refer to the ser_sipport for setting TCP and UDP ports.

Example ser_tlsport=5071

Name ser_tos

Description The TOS (Type Of Service) to be used for the sent SIP packages. The values apply to both UDP and TCP packages.

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Example ser_tos=0x180

Name ser_trunk_dnssrv

Description If this parameter is set activated, any hostname entered in ser_trunk_host will be interpreted as a DNS SRV name. Thus a DNS SRV lookup will be performed at first. The message will be forwarded to the server with the highest priority. If multiple servers have the same priority, the messages will be distributed according to the weight assigned to the DNS SRV host. If the first host fails, the backup host is contacted. This parameter only applies to the SIP Trunking mode of operation.

Example ser_trunk_dnssrv=on

Name ser_trunk_port

Description This parameter sets the port to which outgoing SIP messages are addressed on the Trunking provider side. Note that this parameter only needs to be set if DNS SRV is not being used. With DNS SRV enabled, the target host port will be queried from the DNS server. If left blank, the default of 5060 for UDP and TCP and 5061 for TLS respectively will be used. This parameter only applies to the SIP Trunking mode of operation.

Example ser_trunk_port=5060

Name ser_trunk_srvpenalty

Description This specifies the time period for the penalty box. If a host obtained via a DNS SRV lookup is not reachable for a certain amount of time, it will be placed into the penalty box. This means that no further attempts will be placed to that server for the specified time. Only after that time, another request is tried to that server. This parameter only applies to the SIP Trunking mode of operation.

Example ser_trunk_srvpenalty=30

Name ser_trunk_transmit

Description The SIP Transmit Type parameter is the transmission type the Mediatrix Boss uses to contact the central local PBX. The Mediatrix Boss supports 3 different SIP transport modes, UDP, TCP, and the encrypted TLS variant. The type to use depends first and foremost on the used IP PBX. The TCP and UDP types are straightforward and just specify the used protocol type.

The only type requiring further explanation is TLS. For TLS additional things like the encryption keys are required. Since this is a very advanced topic, this cannot be configured via the WebGUI as of yet. The Mediatrix Boss will generate TLS keys automatically that can be downloaded for the use in Servers/Phones via SCP. Please consult the Command Line Reference for more information on how to do that.

This parameter applies only to the SIP Trunking template.

Example ser_trunk_transmit=udp

Name ser_userpart

Description Please refer to the ser_usesipping parameter for more information on that.

When using the SIP OPTIONS messages to check whether a server is still available, you can configure the URI user part the Mediatrix Boss should be sending. Simply specify the user part of the URI with this parameter. The switch must respond to the URI with a non 5xx and 6xx response. You can also refer to ser_fromhost for setting the host part of the URI.


Example ser_userpart=user

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Name ser_usesipping

Description There are two ways the Mediatrix Boss supports to check whether the SIP Server for which we are providing the survivability functionality is monitored. This monitoring can be applied at the network layer (via ICMP messages) or on the application layer with SIP OPTION requests. The check with pings should work with all servers as long as the underlying network supports this. This is usually the case except firewalls on the way, or a packet filter on the server itself prevents the Ping Echo Reply messages from getting back to the Mediatrix Boss. Other than that, the pings will be sent in intervals of around a minute. This interval can be shorter depending on the amount of SIP messages flowing through the Mediatrix Boss. The interval can also vary if there are abnormal negative responses to requests forwarded to the server. The non-response to a single ICMP message will not lead to the Mediatrix Boss switching into survivability mode.

Whenever the SIP server supports SIP OPTIONS (please refer to RFC 3261 for more information on this SIP request type) requests, this should be used for doing the alive check. It will also enable the Mediatrix Boss to switch into survivability mode for situations in which the central SIP server is reachable, but not responding correctly. Whenever we receive 5xx or 6xx responses from the central server for an OPTIONS request, the Mediatrix Boss switches to survivability mode. If no response is received, a single non-response will not lead to the Mediatrix Boss switching to survivability mode.


Example ser_usesipping=on

Name ser_voicemail

Description This is a very rarely needed parameter. You only need to consider it, if your voicemail system is to directly communicate with the SBC without interaction of your PBX. It is being used for doing proper internal classification of messages. If you are unsure about this, simply leave it blank which should work okay for almost all scenarios.

This applies only to the SBC mode of operation.

Example ser_voicemail=10.10.10.10

Name ser_multiport

Description This activates multiport mapping towards the SIP server. If activated the Mediatrix Boss will always change the contact header of SIP requests that are being forwarded to the SIP server. As such, the SIP server will think that the SBC is the actual endpoint. The same holds true in the other direction where all headers are modified that the endpoint returns the messages to the SBC. Since now all endpoints connected behind the SBC would seem to come from the same IP and port combination, namely the SBC. In order to avoid confusion especially with multiple registrations of different phones, each registered phone will be mapped to a unique port. Any message arriving from the PBX on that port will hence be forwarded to the correct phone. The phones will continue to send to the configured SIP port, so the port mapping only applies to the direction towards the SIP server.

This applies only to the SBC and Branch SBC modes of operation.

Example ser_multiport=on

Name ser_topohide

Description This base configuration key activates the topology hiding feature. If it is active "To:" and "From:" headers are rewritten. By just using this option, the normal RFC compliant mechanics of routing SIP messages are not affected. However, this means that some headers as for example VIA cannot be modified. This key applies only to the SBC and Branch SBC modes of operation.

Example ser_topohide=on

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Factory and Password Reset Software Configuration Guide

Factory and Password Reset

The Mediatrix Boss supports both a factory reset and a password reset. The latter just resets the password while the factory reset restores the complete configuration to factory default.

There are two ways of doing a factory reset or a password reset. If you still have network access to the device, you can use SSH to do the reset. If you really lost your password, you must have physical access to the device in order to do the reset.

Factory Reset and Password Reset via SSH

In order to to a factory reset via SSH simply enter the following command.

/etc/rc.d/init.d/checkfactorydefault force

Password Reset via RS232

After connecting a serial console or a keyboard and monitor you need to power up or reboot you box (press Ctrl-Alt-Delete). Then wait until you see the following prompt:

Enter passwordreset and press enter. Your Mediatrix Boss will boot normally but with the password reset to the default password.

Factory reset via RS232

After connecting a serial console or a keyboard and monitor you need to power up or reboot you box (press Ctrl-Alt-Delete). Then wait until you see the following prompt:

Name ser_topohide_vias

Description Refer also to the ser_topohide parameter. In addition to the general header fields, the header fields concerning the message routing such as VIA and Record Route headers will be modified. In a simple configuration, with a server, an SBC and a client, this configuration does not pose a problem. The SBC will remove the VIA headers in both directions meaning that any routings behind or before the SBC will not work properly if they rely on the presence of VIA or record route headers. This could for example impact the use of SIP proxies outside the SBC since the SBC will always send outbound messages to the registered user. This feature is hence a security feature that hides all headers, but endangers proper RFC3261 message routing. If you want to preserve proper RFC3261 compliant message routing, you should not activate this feature. This applies only to the SBC mode of operation.

Example ser_topohide_vias=on

Note: Note that the factory reset has no influence on the installed software. The factory reset will only restore the complete configuration of the device (all contained in the /etc directory) to safe factory defaults for the currently installed software version. In order to change the software version, you need to upgrade or downgrade the software of the device.

SYSLINUX 1.62 0x3e713b13 Copyright (C) 1994-2001 H. Peter Anvin boot:

Note: The default password is always sesam, the default CLI user root for the WebGUI admin.

SYSLINUX 1.62 0x3e713b13 Copyright (C) 1994-2001 H. Peter Anvin boot:

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Chapter 9 - Command Line Interface Additional Information

Enter factoryreset and press enter. Your Mediatrix Boss box will boot, perform a factory reset and the reboot again. After that your box will boot up normally but of course needs to be reconfigured as all settings are reset to the factory defaults.

Additional Information

Traffic Control / Quality of Service

The Mediatrix Boss Series products feature sophisticated traffic control and bandwidth management support. Since traffic control is an immensely complex topic, you should refer to http://lartc.org/howto/ to get a more detailed introduction as to how Linux handles this. Here we will only describe where to find the respective configuration files. There are two ways for configuring traffic control in the Mediatrix Boss Series. The Mediatrix Boss series has a config file where the kind of configuration can be selected. It can be found under /etc/sysconfig/trafficshaping. If configuration via the shaper script is selected, all of the features of Linux traffic control can be utilized. The downside is a very complex configuration process, which must be done in /usr/sbin/shaper.sh. A very simple but frequently used configuration can be found there which can be adapted to your liking.

The preferred way of configuring traffic control is via htb.init. HTB stands for Hierarchical Token n Bucket which can be configured to work in conjunction with SFQ (Stochastic Fairness Queuing) and Priority Queuing. An example configuration is available from our website which documents the way this is configured. Online help for this topic is available on the box itself by displaying the README file in the htb directory.

Testing Behavior with force_send_notifies

The following commands given in the shell have the effect that the SIP Proxy of the Mediatrix Boss will send a NOTIFY (of the desired type, i.e. normal or survivable) to all registered phones as soon as it receives the next SIP package after the command has been given.

force_send_notifies survivable and force_send_notifies backup have the same effect: They will make the registered phones behave as they would in Survivability Mode. After the command force_send_notifies normal they will go back to normal behavior; they have no effect on the SIP Proxy itself!

Use

These commands can be used for testing the behavior of the phones in survivable and in normal mode. As a result of using for example force_send_notifies survivable the behavior of the registered phones in survivable can be simulated, or the phones can be forced to behave normal although the SIP Proxy is in Survivability Mode.

Control Script: serctlc

There is a script openserctl to control the SIP Proxy server process. It can be used to manage users, access control lists, in memory contacts, and to monitor server health.

There are a lot of possibilities offered by openserctl. However, we will narrow it down to four commands which are probably the most interesting in our context. If you are interested in further information, simply browse the web. There are a lot of websites providing in-depth descriptions about the topic.

Examining server status

openserctl monitor shows the server version, uptime, pending and completed transactions, and the number of major category responses the server has sent.

For example:

[cycle #: 1; if constant make sure server lives]

Server: OpenSer (1.1.1mediatrix-tls (i386/linux))

Note: Note that you have to be quick enough here, otherwise your box will reboot normally. Once you entered the first letter, you can wait as long as you like.

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http://lartc.org/howto/

Additional Information Software Configuration Guide

Now: Sun May 18 18:44:32 2008

Up Since: Fri May 16 07:34:32 2008Up time: 213000 [sec]

Transaction Statistics:Module name = tm; statistics=11

tm:received_replies = 8465

tm:relayed_replies = 8355 tm:local_replies = 71

tm:5xx_transactions = 0

tm:6xx_transactions = 6 tm:inuse_transactions = 2

Stateless Server Statistics: Module name = sl;

statistics=9 sl:1xx_replies = 0

sl:2xx_replies = 0 sl:3xx_replies = 0

sl:4xx_replies = 0

sl:5xx_replies = 0 sl:6xx_replies = 0

sl:sent_replies = 0

sl:sent_err_replies = 0 sl:received_ACKs = 0

UsrLoc Stats:Module name = usrloc;

statistics=6 usrloc:aliases-users = 0

usrloc:aliases-contacts = 0 usrloc:aliases-expires = 0

usrloc:location-users = 1

usrloc:location-contacts = 1 usrloc:location-expires = 16

Show User List

The command openserctl ul show will list any currently registered clients. The output will look like this:

===Domain list===

---Domain---

name : 'location'size : 512

table: 0x402ee6d0

d_ll {n : 2

first: 0x402f1a74

last : 0x402f089c}

lock : 0

...Record(0x402f1a74)... domain: 'location'

aor: 'test'

~~~Contact(0x402f708c)~~~domain : 'location'

aor: 'test'

Contact: 'sip:[email protected]:5060' Expires: 2501q :0.00

Call-ID: '[email protected]' CSeq : 101

State: CS_SYNC next: (nil) prev: (nil)~~~/Contact~~~~

.../Record...

...Record(0x402f089c)... domain: 'location'

aor: 'joe'

~~~Contact(0x402f0924)~~~

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domain : 'location'

aor: 'joe'Contact: 'sip:192.168.0.101:14354'

Expires: 432

q :0.00Call-ID: '[email protected]' CSeq : 11

State: CS_SYNC

next: (nil) prev: (nil)

~~~/Contact~~~~

.../Record...

---/Domain---

===/Domain list===

Show online Users

With openserctl online a list of the users who are currently online will be displayed (you will see their usernames).

Ping a user

openserctl ping <uri> allows you to check whether a certain client is available or not. This is especially useful if there is a routing problem with a certain user account. The result lets you see if the problem lies between the client and the server (SBC component), or if the cause for the problem has to be located elsewhere.

Before using openserctl ping <uri> execute a serctl ul show which contains the URIs of all configured users as the contact. Simply look for the user name you would like to ping.

Example

openserctl ping sip:192.126.24.8:5060

The answer will be the corresponding user name if it is available. If the ping is unsuccessful, i.e. no answer is received from the client, the server will do exactly the same, and will therefore not give out any response.

Improving VRRP

Using the Command Line will enable you to smoothen the handling of SIP messages in the VRRP scenario.

Background Information

SIP messages always carry their source and destination IP addresses. The currently active master router of the setup assumes the VRRP IP address only on the network layer but on the IP-level SIP messages always originate from the master of slave address instead of the VRRP address. If the master sends a SIP message to the PBX, some PBXs will respond to the IP address from which they received the SIP message instead of honoring the address mentioned in the SIP message. As a consequence SIP messages from the PBX will be lost if the master fails in the meanwhile and the slave takes over. There is, however, a possibility to to force the response to the VRRP IP address.

Steps

Please do the following on both Mediatrix Boss units:

Log in on your Mediatrix Boss

Add the following file: /etc/sysconfig/firewall.custom

Then enter these three lines into above mentioned file and replace the IP addresses in italic letters with the corresponding IP addresses of your company network:

iptables -t mangle -A OUTPUT -p udp -destination-port 5060 -d PBX IP -j MARK -set-

mark 0x51C

iptables -t mangle -A FORWARD -p udp -destination-port 5060 -d PBX IP -j MARK -set-mark 0x51C

iptables -t nat -A POSTROUTING -m mark -mark 0x51C -j SNAT -to-source VRRP IP

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Save your changes

Go back to the WebGUI and select Apply Configuration

Make a test run

Call Detail Records (CDRs)

You might already be very well aware of the Survivability functionality of your Mediatrix Boss. What you might not be aware of is that your Mediatrix Boss offers you the option to store Call Detail Records (CDRs) during Survivability mode.

The collection of CDRs can be enabled in the Logging menu under DIAGNOSTICS (please refer to “Logging” on page 122).

Q How does the collection of CDRs work?

While operating normally, your IP PBX might, besides its other tasks, be collecting information for a billing system. As long as your PBX is running smoothly, your Mediatrix Boss will not interfere or execute identical services simultaneously.

However, should your PBX fail, your Mediatrix Boss will take over the following:

• it will handle all SIP messages

• and record automatically all successful as well as all attempted calls in the form of CDRs if the before mentioned box in the Logging menu has been checked.

Q How is the format of the collected CDRs?

These records do not directly provide information about the connection or the duration of a call. To read as much information in them as possible it is very useful to know how they are formatted.

The following are example CDR records for one internal call in chronological order:

a. Initializing record, or INVITE:3

Every SIP call starts with an INVITE. If the dialed number is correct, not busy and no Call Forwarding is set, this will lead to a ringing phone.

Dec 5 21:05:39 localhost /usr/sbin/sip[1226]: ACC: transaction answered:call_id=f0db37765bcdc44e5ec73ce81f7fa516 @ 10.11.250.1, method=INVITE, from=phone

<sip:499 @ 10.11.250.1>;tag=2e93f5d134d4072;epid=SC200BCC, fromuri=sip:499 @

10.11.250.1, to="p"<sip:498 @ 10.11.250.1>;tag=112097237522090, touri=sip:498 @ 10.11.250.1, code=200

b. Connection established, or ACK:

The call is accepted and subsequently established with ACK for "acknowledged".

Dec 5 21:05:39 localhost /usr/sbin/sip[1227]: ACC: request acknowledged:call_id=f0db37765bcdc44e5ec73ce81f7fa516 @ 10.11.250.1, method=ACK, from=phone

<sip:499 @ 10.11.250.1>;tag=2e93f5d134d4072;epid=SC200BCC, fromuri=sip:499 @

10.11.250.1, to=sip:498 @ 10.11.250.1;tag=112097237522090, touri=sip:498 @

10.11.250.1, code=200

c. Connection is terminated, or BYE:Dec 5 21:05:45 localhost /usr/sbin/sip[1223]: ACC: transaction answered:call_id=f0db37765bcdc44e5ec73ce81f7fa516 @ 10.11.250.1, method=BYE,

from=<sip:498 @ 10.11.250.1>;tag=112097237522090, fromuri=sip:498 @ 10.11.250.1,

to=<sip:499 @ 10.11.250.1>;tag=2e93f5d134d4072, touri=sip:499 @ 10.11.250.1, code=200

Q How do I read this information?

3. Please note that spaces before and after the sign have been added to all samples of SIP messages to improve readability.

Field Description

Date and Time Date and Time when the message was recorded. Important for calculating the duration of the call.

Host name Hostname the request was recorded on. Only important if logged to a central log server.

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Q What are the specifications for this format?

The Date, Host name, Process, and Information parameters are required by Syslog which also dictates the format. They are either separated by a blank, or by a colon. The parameters reflecting the actual CDR information are in the "key=value" pair format. The key indicates the type of the parameter. The value part starts after the "=" sign and is always terminated by a comma.

Q Where are the CDRs stored?

Stored CDR files can be found in the directory /var/cdr of your Mediatrix Boss.

Since this is located in the RAM, any file here will be lost if a reboot is executed.

The files are named according to their date of creation, e. g. 081112.cdr for a file recorded November 12, 2008. So every day at midnight the file is closed and saved using the current date.

Q How can I download the CDRs?

As already mentioned, the files will be saved to the directory /var/cdr of your Mediatrix Boss if the logging of CDRs is activated (please refer to 8.2). There they can be downloaded (and deleted) periodically by your accounting software via SFTP/SCP. Please note that your Mediatrix Boss deletes CDRs automatically after two days.

Another possibility to save it would be to send the syslog output which also contains these CDRs over your network to another computer with a syslog server.

Using Radius

Starting with r887, our products have limited support for authenticating local users via RADIUS. The following describes how to set up an user and set up RADIUS support.

Q How can I create a user?

A problem with RADIUS is that while it can tell the Mediatrix Boss whether a username and password combination is valid, it can't tell the Mediatrix Boss which user and group IDs to use. But these are vital information which must be present.

This means we need to create a local user with no password to provide this informations.

This step requires manually editing the file /etc/passwd, e.g. using the editor vi.

The new user will get a user ID of 100 or above (it must be unique) and a group ID of

100 (this one is fixed). So the first user will get a UID (user ID) of 100 and a GID (group

ID) of 100, the second user a UID of 101 and a GID of 100, and so on and so forth.//

To actually create the user, a new line in the file /etc/passwd must be added which looks like this:

username:!:100:100User name or description:/tmp:/bin/bash

Process Process the request was generated on.

Information Information as to what type of CDR record you are dealing with.

Call ID String uniquely identifying the call MethodRequest the record was generated for.

From Header Indicates who initiated the call (caller).

From URI URI of the user who initiated the call (caller).

To Header Indicates the callee.

To URI URI of the callee.

Response Indicates the response code for a SIP transaction.

Field Description

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Explanation:

• The first field contains the user name.

• The second field normally contains the password but contains only an exclamation mark in in this case since the password is done against a RADIUS server.

• The UID is in the third field

• and the GID in the fourth field.

• A description or full user name can be given in the fifth field (it can also be left empty).

• The sixth field contains the home directory, which we set to /tmp for all users.

• The last field contains the shell to use (this is always /bin/bash).

Example

To add two users, e. g. Guybrush and Elaine, Guybrush will be given the UID of 100, Elaine a UID of 101. The resulting entries would be:

guybrush:!:100:100:Guybrush Threepwood:/tmp:/bin/bash elaine:!:101:100:Elaine

Marley:/tmp:/bin/bash

Q How do I set up a RADIUS Server File?

The first file to edit is /etc/raddb/server. It contains the information about the RADIUS server to authenticate against. This is what it looks like:

server:port sharedsecret timeout

Explanation:

• Server is the name or IP address of your RADIUS server,

• port the port on which the server accepts RADIUS connections (normally 1812).

• Sharedsecret is the shared secret configured on the RADIUS server for your client or client network.

• Timeout is the time in seconds to wait for an answer. In most cases 1 is enough, higher values can be problematic in the web GUI if the RADIUS server cannot be reached.

Example

Assuming your RADIUS server has the IP 10.10.5.42 and the shared secret configured on the server is BigWhoop, your /etc/raddb/server file would be:

10.10.5.42:1812 BigWhoop 1

Q How can I set up PAM?

To have the Mediatrix Boss try to contact the RADIUS server for password checks, there are three services that are able to use RADIUS:

• login (local console login)

• sshd (remote SSH login)

• and mini_httpd (web GUI).

It is recommended to only change one of them at first and to try whether it works as expected and then change the other.

If you have easy direct access to the console, you could log in via console, change the SSH PAM file to add RADIUS and then test with SSH whether it works as expected.//

Q Where are the files located and what is their content?

The files are located in the /etc/pam.d directory, and by default they have the following content (at r887):

/etc/pam.d/login:

auth required pam_unix.so

account required pam_access.so account required pam_unix.so

password required pam_unix.so shadow use_authtok

session required pam_unix.sosession required pam_env.so

session required

pam_motd.so

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/etc/pam.d/mini\_httpd:auth required pam_deny.so

/etc/pam.d/sshd:auth required pam_unix.so

account required pam_access.so

account required pam_unix.sopassword required pam_unix.so shadow use_authtok

session required pam_unix.so

session required pam_env.so session required pam_motd.so

session required pam_limits.so

pam_motd.so

In all cases, to enable RADIUS support all you need to do is insert a new line in front of the first line with the following content:

auth sufficient pam_radius_auth.so

Example

To enable RADIUS support for the web GUI, /etc/pam.d/mini_httpd should have the following content:

auth sufficient pam_radius_auth.so auth required

For local login and SSH, both /etc/pam.d/login and /etc/pam.d/sshd should contain:

auth sufficient

pam_radius_auth.so auth required

pam_unix.soaccount required pam_access.so

account required pam_unix.so

password required pam_unix.so shadow use_authtok session required pam_unix.so

session required pam_env.so

session required pam_motd.sosession required pam_limits.so change_uid

Regular expressions

Various options for the SER accept regular expressions. A regular expression, or regexp, is a way of describing a class of strings. It matches every input record whose text belongs to that class. Thus they are used in comparison expressions to match input strings.

The simplest regular expression is a sequence of letters, numbers, or both. Such a regexp matches any string that contains that sequence. Thus, the regexp 'foo' matches any string containing 'foo'. Other kinds of regexps let you specify more complicated classes of strings.

Regular Expression Operators

You can combine regular expressions with the following characters, called regular expression operators, or metacharacters, to increase the power and versatility of regular expressions.

Here is a table of metacharacters. All characters not listed in the table stand for themselves.

Operator Description Example

'ˆ' This matches the beginning of the string or the beginning of a line within the string.

'^00' matches two zero digits at the beginning of a string, and may be used to identify international calls.

'$' This is similar to 'ˆ', but it matches the end of a string or the end of a line within the string.

'p$' matches a record that ends with a 'p'.

182 Mediatrix Boss


In regular expressions, the '*', '+', and '?' operators have the highest precedence, followed by concatenation, and finally by '|'. As in arithmetic, parentheses can change how operators are grouped.

Case is significant in regular expressions, both when matching ordinary characters (i.e., not metacharacters), and inside character sets. Thus a 'w' in a regular expression matches only a lower case 'w' and not an upper case 'W'.

'.' This matches any single character except a newline.

'.P' matches any single character followed by a 'P' in a string. Using concatenation we can make regular expressions like 'U.A', which matches any three-character sequence that begins with 'U' and ends with 'A'.

'[...]' This is called a character set. It matches any one of the characters that are enclosed in the square brackets.

Ranges of characters are indicated by using a hyphen between the beginning and ending characters, and enclosing the whole thing in brackets.

'[MVX]' matches any one of the characters 'M', 'V', or 'X' in a string.

'[0-9]' matches any digit.

'[^...]' This is a complemented character set.

The first character after the '[' must be a 'ˆ'. It matches any characters except those in the square brackets (or newline).

'[^0-9]' matches any character that is not a digit.

‘|’ This is the alternation operator and it is used to specify alternatives. The alternation applies to the largest possible regexps on either side.

'^P|[0-9]' matches any string that matches either '^P' or '[0-9]'. This means it matches any string that contains a digit or starts with 'P'.

'(...)' Parentheses are used for grouping in regular expressions as in arithmetic. They can be used to concatenate regular expressions containing the alternation operator, '|'.

'*' This symbol means that the preceding regular expression is to be matched zero or as many times as possible to find a match.

The '*' repeats the smallest possible preceding expression. Use parentheses if you wish to repeat a larger expression. It finds as many repetitions as possible.

'ph*' applies the '*' symbol to the preceding 'p' and looks for matches to one 'p' followed by any number of 'h's. This will also match just 'p' if no 'h's are present.

'+' This symbol is similar to '*', but the preceding expression it to be matched at least once.

'wh+y' would match 'why' and 'whhy' but not 'wy', whereas 'wh*y' would match all three of these strings.

'?' This symbol is similar to '*', but the preceding expression can be matched once or not at all.

'fe?d' will match 'fed' and 'fd', but nothing else.

'\' This is used to suppress the special meaning of a character when matching.

'\$' matches the character '$'.

Operator Description Example

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10 Sample scenarios

Survivability

In the Survivability scenario, the Mediatrix Boss is simply used as a SIP Proxy that can take over the functionality of a central softswitch (compare Figure 86 on page 186). All you need to do to configure this is use the Mediatrix Boss as an outbound proxy for your local phones and gateways. The Mediatrix Boss will constantly monitor the connection to the softswitch. Once the connection breaks, the Mediatrix Boss will instantly take over.

Using the Mediatrix Boss in your branch office can also help address the QoS problems typically associated with converged networks. Maintaining voice quality is a requirement that cannot be tackled with standard quality of service means. Too many voice calls on a WAN link will degrade the voice quality of all connections. In order to avoid that, the Mediatrix Boss can enforce hard limits on the number of possible calls based on predefined bandwidth constraints. This functionality is commonly referred to as Call Admission Control.

The Mediatrix Boss will continuously monitor the VoIP traffic streams. If the maximum allowable number of calls or the maximum allowable user bandwidth is reached, all additional calls flowing over that WAN link would be denied. This is accomplished via standard SIP mechanisms, so that neither special software in the data center nor special phone hardware is necessary. The advantages of this decentralized over centralized server-based ones are numerous:

Works with all SIP compliant servers

Works also if NATs and Topology Hiding is used in the network

Works for multiple SIP servers delivering different applications

Works for a mix of Video and VoIP

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Chapter 10 - Sample scenarios Branch Connectivity and Branch Session Border Controller

Branch Connectivity and Branch Session Border Controller

The more generic Branch Connectivity Solution is very similar to the Standard Survivability scenario. For an illustration, please compare to Figure 86 on page 186. There is one major difference in the handling of network. In the Branch Connectivity Scenario, the Mediatrix Boss has two network connections, one internal, one external one. The SIP signalling and VoIP media handling

Figure 86: Survivability Scenario

is translated between these two networks. As such, the device serves as a decentralized Session Border Controller. Contrary to a far end NAT traversal solution like a central SBC, the decentral SBC has a number of advantages especially in Enterprise scenarios.

Clean Network Separation with Clean Routing

Enhanced stability because there is no need for firewall pinholing for potentially thousands of phones

Increased Security and Simplification of Firewall rules due to Server-Server only Communication

Single Device Demarcation line between Headquarter and Branch Office

Improved bandwidth utilization because branch traffic stays in branch office

Enables the use of local PSTN gateways even behind NATs in branch offices

Additional Functionality can be implemented in the Branch

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Hosted Communication

The Hosted scenario is a slight variation of the Branch Connectivity Scenario. The difference is probably the ownership of the device as well as the security policy. In Hosted Scenarios, the Mediatrix Boss is a customer premise termination equipment managed by the Hosted Operator. This means that the Mediatrix Boss is used to deliver a service to the customer premises. Security and Survivability are important topics there. Figure 87 on page 187 should give you an overview.

Scenarios with Integrated Voice Gateways

If you have a Mediatrix Boss with integrated Voice Gateway cards, you can support the same scenarios as described here, with having the Voice Gateway built into the device. Figure 88 on page 188 should give an overview of how this fits together.

SIP Trunking

When used for SIP Trunking, the Mediatrix Boss will support such functionalities as:

Securing Network Boundaries and providing Connectivity between your PBX and the external network

NAT handling for both signalling and payload

Figure 87: Hosted Scenario

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Chapter 10 - Sample scenarios SIP Trunking

Figure 88: BiaB Scenario

Signalling control of calls through the network, and gracefully reject calls when necessary, protecting the network from Denial of Service attacks and sudden spikes in congestion.

Support for both Full Topology Hiding as well as SIP Proxy mode for improving interoperability and Feature interworking

Advanced Security Features such as TLS and SRTP Termination for implementing secure connectivity over insecure network links without having to upgrade all your equipment. You can also use VPN based encryption for communication towards the SIP Trunking provider

DNS SRV handling for supporting redundancy configurations with your SIP Trunking Provider

Figure 89 on page 188 shows a SIP trunking scenario.

The Mediatrix Boss supports two different types of SIP Trunking scenarios. They can act as a transparent Session Border Controller for PBXs that natively support SIP trunks or for Carriers not needing a Registration with their service. The other way is acting as a true Back-to-Back User Agent supporting Registrations, Transcoding, etc. If that is selected proprietary SIP signalling will however not be supported. The following gives an overview of the different modes of operation:

Figure 89: SIP Trunking Scenario

SBC Mode

• Full Termination of Media

• Malformed Packet Protection and Topology hiding, but supports transparent pass-through of signalling

• DNS SRV Solution and Support for failover and load balancing solutions

B2BUA Mode

• Can Actively Register

• Can handle various Individual Accounts (as common with some smaller SIP providers)

• Full Termination of both Signalling and Media

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The choice which way to use will depend on whether your carrier requires SIP registration and whether your PBX connected to this trunk can perform the registration or not. If it can, the transparent SBC approach is the suitable one because it can preserve the features across the SIP Trunk. With this approach even proprietary signalling such as SIP-Q is possible.

If that is not the case, you will need to use the Back-to-Back User Agent based approach. This can be configured in the B2BUA section.

Note: Products with Integrated Voice Gateway and Autoattendent functionality only support the SIP Trunking without Registration. They cannot handle individual user accounts with digest authentication with the SIP Provider.

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Gateway Component

C H A P T E R

11 System Overview

This chapter provides an overview of the Mediatrix devices supported:

Introduction to the Mediatrix devices and the models available.

Description of the Bypass feature for models that support it.

How to use the DEFAULT/RESET button (partial reset and factory reset procedures).

Introduction

Several Mediatrix devices are available. They offer a low total cost of ownership as they reduce installation and maintenance costs.

Moreover, the Mediatrix unit integrates features such as TLS, SRTP, and HTTPS designed to bring enhanced security for network management, SIP signalling and media transmission aspects.

The following describes the devices that the application supports.

Mediatrix 3000 Series

The Mediatrix 3000 series ISDN VoIP Digital Gateways allow enterprises to lower communications costs over any IP link. Designed specifically for enterprise applications, the Mediatrix 3000 digital gateways make use of existing broadband access equipment to connect to any standards-based VoIP network.

The Mediatrix 3000 digital gateways meet the requirements of enterprises that want to connect their ISDN equipment, such as PBXs, through a BRI (Basic Rate Interface) or PRI (Primary Rate Interface) interface to an IP network or as a gateway to the PSTN.

Mediatrix 3000 digital gateways provide transparent ISDN port extensions over an IP network. The remote ISDN terminals can be managed centrally and benefit from PBX services such as calling groups and voice mail.

The following are the current Mediatrix 3000 Series models:


The Mediatrix 3200 Series offers a mix of FXS and FXO interfaces. Its FXO interfaces make it the ideal solution to deploy private or hosted toll bypass networks. It provides a simple, transparent and cost-effective way of maintaining a connection to the PSTN.

The FXO interfaces provide a gateway to the PSTN for IP-based PBX and Key Systems. Thereby, Mediatrix 3200 gateways allow the deployment of VoIP Remote Line Extension and Branch Office Connectivity solutions without sacrificing any local PSTN access points.

By connecting CO lines from a selected site to VoIP networks, the Mediatrix 3200 series enable service providers and enterprises to use VoIP connections between pre-determined local networks.

Table 8: Mediatrix 3000 Models

Model Interfaces Service Slot Used

Mediatrix 3404 5 BRI ports Slot 2

Mediatrix 3408 10 BRI ports Slot 2 and Slot 3

Mediatrix 3631 1xT1/E1-PRI interface Slot 2

Mediatrix 3632 2xT1/E1-PRI interface Slots 2 and 3

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Chapter 11 - System Overview Introduction


See the Hardware Configuration Guide for more details.


The Mediatrix 3300 is an essential component of any enterprise network, for voice, data, and security needs. It connects analog phones, faxes, and lines to an IP network to allow cost-effective communication scenarios for immediate savings and enhanced productivity.

It also offers an FXO port that can provide SCN access for various VoIP endpoints such as IP phones, FXS devices, softphones and IP-based PBX and Key Systems (see “FXO Configuration” on page 296 for more details).

The following are the current 3300 Series models:

See the Hardware Configuration Guide for more details.


The Mediatrix 3700 series models offer different combinations of T1/E1, BRI and FXS/FXO interfaces in a unique analog/digital offering to allow enterprises to lower communications costs over any IP link. Designed specifically for enterprise applications, the Mediatrix 3700 series models make use of existing broadband access equipment to connect to any standards-based VoIP network.

The Mediatrix 3700 series models meet the requirements of enterprises that want to connect their ISDN equipment, such as PBXs, through a BRI (Basic Rate Interface) or PRI (Primary Rate Interface) interface to an IP network or as a gateway to the PSTN.

They also offer to connect analog phones, faxes, and lines to an IP network to allow cost-effective communication scenarios for immediate savings and enhanced productivity.




Mediatrix 3208 6 FXO ports + 2 FXS ports Slot 2

Mediatrix 3216 12 FXO ports + 4 FXS ports Slot 2 and Slot 3



Mediatrix 3308 7 FXS Lines

1 FXO Line

Slot 2

Mediatrix 3316 14 FXS Lines

2 FXO Lines

Slots 2 and 3


Model Interfaces VoIP Calls Capacity Service Slot Used

Mediatrix 3731 1xT1/E1 PRI interface

7 FXS ports

up to 37 Slots 2 and 3

Mediatrix 3732 6 FXO ports + 2 FXS ports

1xT1/E1 PRI interface

Up to 38 Slots 2 and 3

Mediatrix 3734 1xT1/E1 PRI interface

5 BRI ports


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The Mediatrix 3731, 3732, 3741 and 3742 models offer true FXO interfaces, which makes them the ideal solution to deploy private or hosted toll bypass networks. They provide a simple, transparent and cost-effective way of maintaining a connection to the PSTN.

The FXO interfaces provide a gateway to the PSTN for IP-based PBX and Key Systems. Thereby, Mediatrix 3700 gateways with FXO interfaces allow the deployment of VoIP Remote Line Extension and Branch Office Connectivity solutions without sacrificing any local PSTN access points.

By connecting CO lines from a selected site to VoIP networks, the Mediatrix 3700 series enable service providers and enterprises to use VoIP connections between pre-determined local networks.

DSP Limitation

Some of the Mediatrix Boss models currently suffer from a limitation of their DSPs. When using a codec other than G.711, enabling Secure RTP (SRTP) and/or using conferences has an impact on the Mediatrix Boss’s overall performance as SRTP and conferences require CPU power. This means there is a limitation on the lines that can be used simultaneously, depending on the codecs enabled and SRTP. This could mean that a user picking up a telephone on these models may not have a dial tone due to lack of resources in order to not affect the quality of ongoing calls. See “Security” on page 391 for more details on SRTP limitations.

The DSPs offer channels as resources to the Mediatrix Boss. The Mediatrix Boss is limited to two conferences per DSP. See “Conference” on page 429 for more details on Conference limitations.

Please note that:

One FXS line requires one channel.

There is a maximum of 2 conferences per DSP

Each conference requires one additional channel

In the following tables, compressed RTP refers to codecs other than G.711. Numbers in Bold indicate a possible under-capacity.


Table 16 describes the Mediatrix 3200 series processing capacity.

Mediatrix 3741 7 FXS ports

5 BRI ports


Mediatrix 3742 6 FXO ports + 2 FXS ports

5 BRI ports

Up to 15 Slots 2 and 3

Table 14: Mediatrix 3700 Models (Continued)

Model Interfaces VoIP Calls Capacity Service Slot Used

Table 15: Mediatrix 3200 Offered Channels vs. Processing Capacity

Model

Offered Chanels Processing Capacity

Phys. Channels

3-way Conf. Channels

G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

Mediatrix 3208 8 2 10 8 8 6

Mediatrix 3216 16 4a

a. Two conferences available for lines 1-7 and two conferences available for lines 8-14.

20 16 16 12

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Chapter 11 - System Overview DSP Limitation








Model


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

Mediatrix 3308 8 2 10 8 8 6

Mediatrix 3316 16 4a

a. Two conferences available for lines 1-7 and two conferences available for lines 8-14.

20 16 16 12


Model


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

Mediatrix 3404 (1 x 5 BRI Card)

8 0 8 8 8 6

Mediatrix 3408 (2 x 5 BRI Card)

16 0 16 16 16 12


Model


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

Mediatrix 3621

(1 PRI Card)

15 0 15 15 15 15

Mediatrix 3631 (1 PRI Card)

30 0 30 30 30 24

Mediatrix 3632 (2 PRI Cards)

60 0 60 60 60 48

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Mediatrix 3731

Table 19 describes the Mediatrix 3731 processing capacity.

Mediatrix 3732


Mediatrix 3734


Mediatrix 3741



Card


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

7 FXS-1 FXO 8 2 10 8 8 6

PRI 30 0 30 30 30 24


Card


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

2 FXS-6 FXO 8 2 10 8 8 6

PRI 30 0 30 30 30 24


Card


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

5 BRI 8 0 8 8 8 6

PRI 30 0 30 30 30 24


Card


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

5 BRI 8 0 8 8 8 6

7 FXS-1 FXO 8 2 10 8 8 6

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Chapter 11 - System Overview RESET/DEFAULT Button

Mediatrix 3742


RESET/DEFAULT Button

The RESET/DEFAULT button allows you to:

Cancel an action that was started.

Revert to known factory settings if the Mediatrix Boss refuses to work properly for any reason or the connection to the network is lost.

Reconfigure a unit.

At Run-Time

You can use the RESET/DEFAULT button at run-time – you can press the button while the Mediatrix Boss is running without powering the unit off. Table 24 describes the actions you can perform in this case.


Card


Phys. Channels


G.711 RTP Channels

Compr. RTP Channels

G.711 SRTP Channels

Compr. SRTP

Channels

5 BRI 8 0 8 8 8 6

2 FXS-6 FXO 8 2 10 8 8 6

Table 24: RESET/DEFAULT Button Interaction

RESET/DEFAULT Button

Pressed for:Action Comments LEDs Pattern

2 to 6 seconds Restarts the Mediatrix Boss

No changes are made to the Mediatrix Boss settings.

Power LED:

• blinkinga, 1Hz, 50% duty

All other LEDs:

• OFF

a. The Power LEDs of the Mediatrix 3000 Series models blink green.

7 to 11 seconds Sets the Mediatrix Boss in Partial Reset Mode

Sets some of the Mediatrix Boss configuration to pre-determined values.

All LEDs

• blinking, 1Hz, 50% duty

12 to 16 seconds Restarts the Mediatrix Boss in Factory Reset

Deletes the persistent configuration values, creates a new configuration file with the default factory values, and then restarts the unit.

All LEDs

• steady ON

17 seconds and more

No action is taken The RESET/DEFAULT button pressed event is ignored.

N/A

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At Start-Time

You can use the RESET/DEFAULT button at start-time – you power the unit off, and then depress the button until the LEDs stop blinking and remain ON. This applies the “Factory Reset” procedure (see “Factory Reset” on page 201). This feature reverts the Mediatrix Boss back to its default factory settings.

Mediatrix Boss 199


Partial Reset

The Partial reset provides a way to contact the Mediatrix Boss in a known and static state while keeping most of the configuration unchanged.

Following a partial reset, the Mediatrix Boss management interface is set to the Rescue interface. The default address for this interface is 192.168.0.1/24. Any existing network interface that conflicts with the Rescue interface address is disabled.

You can contact the Mediatrix Boss at this address to access its configuration parameters. It is not advised to access the unit on a regular basis through the Rescue network interface. You should reconfigure the unit’s network interfaces as soon as possible in order to access it through another interface. See “After a Partial Reset” on page 200 for more details.

In a partial reset, the following services and parameters are also affected:

AAA service: Sets the “admin” password to administrator and the “public” password to an empty string.

SNMP service: Resets the enableSnmpV1, enableSnmpV2, and snmpPort values to their default values.

WEB service: Resets the serverPort to its default value.

CLI service: The CLI variables revert back to their default value.

LFW service: When a partial reset is triggered and the firewall is enabled, the configuration is rolled back if it was being modified. A new rule is then automatically applied in the firewall to allow access to the 'Rescue' interface. However, if the firewall is disabled, the configuration is rolled back but no rule is added.

HOC service: The Management Interface reverts back to its default value.

To trigger the Partial Reset:

1. Insert a small, unbent paper clip into the RESET/DEFAULT hole located at the rear of the Mediatrix Boss. While pressing the RESET/DEFAULT button, restart the unit.

Do not depress before all the LEDs start blinking (between 7-11 seconds).

2. Release the paper clip.

This procedure can also be performed at run-time.

After a Partial Reset

Following a partial reset, you should:

1. Create or activate network interfaces as described in “Interfaces Configuration” on page 82. Do not disable the Rescue interface!

2. Change the Mediatrix Boss system management network interface to something other than Rescue as described in “Management Interface Configuration” on page 471. Note that you must be able to contact the interface you select in order to continue with the following steps.

3. Contact the Mediatrix Boss through the new system management network interface.

4. Disable the Rescue network interface as described in “Interfaces Configuration” on page 82.

Disabling the Partial Reset

This section describes configuration that is available only in the MIB parameters of the Mediatrix Boss. You can configure these parameters by creating a configuration script containing the configuration variables.

You can disable the partial reset procedure, even if users depress the Reset/Default button. The following parameters are supported:

Table 25: Partial Reset Parameters

Parameter Description

All All the actions are allowed: reset, partial reset and factory reset.

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The reset action restarts the unit.

The partial reset action provides a way to contact the unit in a known and static state while keeping most of the configuration unchanged.

The factory reset action reverts the unit back to its default factory settings.

To change the partial reset behaviour:

1. Set the password associated with a specific user name by putting the following line in the configuration script:MIB.ResetButtonManagement="Value"

where:

• MIB: The MIB module where to locate the variable, which is MbAgw24.

• Value may be as follows:

Factory Reset

The Factory reset reverts the Mediatrix Boss back to its default factory settings. It deletes the persistent MIB values of the unit, including:

The firmware pack download configuration files.

The SNMP configuration.

The PPPoE configuration, including the PPP user names and passwords.

The Factory reset creates a new configuration file with the default factory values. It should be performed with the Mediatrix Boss connected to a network with access to a DHCP server. If the unit cannot find a DHCP server, it sends requests indefinitely.

To trigger the Factory Reset:

1. Power the Mediatrix Boss off.

2. Insert a small, unbent paper clip into the RESET/DEFAULT hole located at the rear of the Mediatrix Boss. While pressing the RESET/DEFAULT button, restart the unit.

Do not depress before the LEDs stop blinking and are steadily ON. This could take up to 30 seconds.

3. Release the paper clip.

The Mediatrix Boss restarts.

This procedure resets all variables in the MIB modules to their default value.

When the Mediatrix Boss has finished its provisioning sequence, it is ready to be used with a DHCP-provided IP address and MIB parameters.

This procedure can also be performed at run-time.

DisablePartialReset All actions are allowed except the partial reset.

Table 25: Partial Reset Parameters (Continued)


Table 26: Partial Reset Values

Value Meaning

100 All

200 DisablePartialReset

Note: The Factory reset alters any persistent configuration data of the Mediatrix Boss.

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C H A P T E R

12 Web Interface Configuration

The Mediatrix Boss contains an embedded web server to set parameters by using the HTTP protocol.

This chapter describes the following:

How to access the web interface and description of the various menus available.

How to submit changes.

Introduction

To configure the web-based configuration service:

1. Set the TCP port on which the web service listens for HTTP requests by putting the following line in the configuration script:web.serverPort="Value"

2. Set the TCP port on which the web service listens for HTTP requests by putting the following line in the configuration script:

web.secureServerPort="Value"

Standards Supported • RFC 2616: Hypertext Transfer protocol - HTTP/1.1.

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Chapter 12 - Web Interface Configuration Using the Web Interface

Using the Web Interface

Media5 recommends that you use the latest version of the Microsoft® Internet Explorer web browser to properly access the web interface.

To use the web interface configuration:

1. Click the Gateway link.

The Information web page displays. It stays accessible for as long as the Internet browser used to access the Mediatrix Boss web interface is opened.

Figure 90: Information Web Page

The Installed Hardware section (Mediatrix 3000 Series only) lists the cards installed in the Mediatrix Boss:

Table 27: Hardware Codes Description

Card Number Quantity Description Products

Mediatrix 3301-010

1 FXS/FXO card

Mediatrix 3308, Mediatrix 3731, or Mediatrix 3741

2 FXS/FXO card

Mediatrix 3316

Mediatrix 3301-020

1 E1/T1 card Mediatrix 3621, Mediatrix 3631, Mediatrix 3731, or Mediatrix 3734

2 E1/T1 card Mediatrix 3632

Mediatrix 3301-060

1 BRI card Mediatrix 3404, Mediatrix 3734, or Mediatrix 3741

2 BRI card Mediatrix 3408

Mediatrix 3301-080

1 2 FXS/6 FXO card

Mediatrix 3208, Mediatrix 3732, or Mediatrix 3742

2 2 FXS/6 FXO card

Mediatrix 3216

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Menu Items

The Menu frame is displayed at the top of the browser window. It contains management links that allow you to display web pages in the Content frame. The management links available vary depending on the Mediatrix unit you are using.

Table 28: Menu Frame Links

Link Description

System

Information: Displays, in read-only format, the status of the Mediatrix Boss.

Services: Allows you to start/stop the services running on the Mediatrix Boss. See “Chapter 13 - Services” on page 211 for more details.

Hardware: Allows you to configure the hardware installed on the Mediatrix Boss. See “Chapter 14 - Hardware Parameters” on page 215 for more details.

Syslog: Allows you to configure the Mediatrix Boss to properly handle syslog messages and notification messages. See “Chapter 15 - Syslog Configuration” on page 219 for more details.

Network QoS: Allows you to configure packets tagging sent from the Mediatrix Boss. See “Chapter 16 - Local QoS (Quality of Service) Configuration” on page 225 for more details.

ISDN

Status: Allows you to view the status of the Mediatrix Boss ISDN parameters. See “Chapter 17 - ISDN Configuration” on page 229 for more details.

Primary Rate Interface: Allows you to configure the PRI ISDN parameters of the Mediatrix Boss. See “PRI Configuration” on page 232 for more details.

Basic Rate Interface: Allows you to configure the BRI ISDN parameters of the Mediatrix Boss. See “BRI Configuration” on page 239 for more details.

Interop: Allows you to set parameters to properly work, communicate, or connect with specific ISDN devices. See “Interop Parameters Configuration” on page 247 for more details.

Timer: Allows you to set timer parameters. See “ISDN Timers Configuration” on page 250 for more details.

Services: Allows you to set the ISDN optional services. See “Services Configuration” on page 251 for more details.

R2

Status: Allows you to view the status of the Mediatrix Boss R2 parameters. See “Chapter 18 - R2 CAS Configuration” on page 259 for more details.

R2: Allows you to configure the parameters related to R2. See “R2 Channel Associated Signaling” on page 261 for more details.

Signaling: Allows you to configure the R2 signaling parameters of the Mediatrix Boss. See “R2 Signaling Variants” on page 264 for more details.

Timers: Allows you to configure the R2 timers of the Mediatrix Boss. See “R2 Timers Variants” on page 268 for more details.

Digit Timers: Allows you to configure the R2 digit timers of the Mediatrix Boss. See “R2 Digit Timers Variants” on page 272 for more details.

Link Timers: Allows you to configure the R2 link timers of the Mediatrix Boss. See “R2 Link Timers Variants” on page 274 for more details.

Tones: Allows you to configure the R2 CAS tones of the Mediatrix Boss. See “R2 Tones Variants” on page 275 for more details.

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POTS

Status: Allows you to view the status of the Mediatrix Boss POTS parameters. See “Chapter 19 - POTS Configuration” on page 287 for more details.

Config: Allows you to configure the POTS parameters of the Mediatrix Boss. See “Chapter 19 - POTS Configuration” on page 287 for more details.

SIP

Gateways: Allows you to add and remove SIP gateways in the Mediatrix Boss. See “Chapter 20 - SIP Gateways” on page 305 for more details.

Servers: Allows you to configure the SIP server and SIP user agent parameters of the Mediatrix Boss. See “Chapter 21 - SIP Servers” on page 307 for more details.

Registrations: Allows you to configure the registration parameters of the Mediatrix Boss. See “Chapter 22 - SIP Registration” on page 315 for more details.

Endpoints: Allows you to configure the administrative state of the Mediatrix Boss’s ports. See “Chapter 23 - Endpoints State Configuration” on page 323 for more details.

Authentication: Allows you to configure authentication parameters of the Mediatrix Boss. See “Chapter 24 - SIP Authentication” on page 327 for more details.

Transport: Allows you to configure the SIP transport parameters of the Mediatrix Boss. See “Chapter 25 - SIP Transport Parameters” on page 331 for more details.

Interop: Allows you to configure the SIP interop parameters of the Mediatrix Boss. See “Chapter 26 - Interop Parameters” on page 335 for more details.

Misc: Allows you to configure interoperability features of the Mediatrix Boss. See “Chapter 27 - Miscellaneous SIP Parameters” on page 351 for more details.

Telephony

DTMF Maps: Allows you to configure the various DTMF maps of the Mediatrix Boss. See “Chapter 28 - DTMF Maps Configuration” on page 365 for more details.

CODECS: Allows you to configure the voice and data codec related parameters of the Mediatrix Boss. See “Chapter 29 - Voice & Fax Codecs Configuration” on page 373 for more details.

Call Forward: Allows you to configure three types of Call Forward. See “Chapter 30 - Call Forward Configuration” on page 411 for more details.

Services: Allows you to configure the Mediatrix Boss subscriber services. See “Chapter 31 - Telephony Services Configuration” on page 419 for more details.

Tone Customization: Allows you to override the pattern for a specific tone defined for the selected country. See “Chapter 32 - Tone Customization Parameters Configuration” on page 437 for more details.

Music on Hold: Allows you to configure the Music on Hold service of the Mediatrix Boss. See “Chapter 33 - Music on Hold Parameters Configuration” on page 441 for more details.

RTP stats: Allows you to read and configure the RTP statistics collected by the Mediatrix Boss. See “Chapter 34 - RTP Statistics Configuration” on page 443 for more details.

Misc: Allows you to configure the country in which the Mediatrix Boss is located. See “Chapter 35 - Country Parameters Configuration” on page 447 for more details.

Table 28: Menu Frame Links (Continued)

Link Description

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Submitting Changes

When you perform changes in the web interface and click the Submit button, the Mediatrix Boss validates the changes. A message is displayed next to any invalid value. A message is also displayed if a service must be restarted and a link is displayed at the top of the page. This link brings you to the Services page. In this page, each service that requires to be restarted has a “*” beside its name. See “Chapter 13 - Services” on page 211 for more details.

If you are not able to restart one or more services, click the Reboot link in the top menu. The Reboot page then opens. You must click Reboot. This restarts the Mediatrix Boss. If the unit is in use when you click Reboot, all calls are terminated.

Where to Go From Here?

If you want to configure the Mediatrix Boss to perform a basic call, this usually involves the following:

Call Router

Status: Allows you to view the current status of the call routing service. See “Chapter 36 - Call Router Configuration” on page 453 for more details.

Route Configuration: Allows you to configure the call routing service of the Mediatrix Boss. See “Chapter 36 - Call Router Configuration” on page 453 for more details.

Auto-routing: Allows you to configure the auto-routing feature of the Mediatrix Boss. See “Chapter 37 - Auto-Routing Configuration” on page 507 for more details.

Management

Configuration Scripts: Allows you to configure the various configuration scripts parameters of the Mediatrix Boss. See “Chapter 38 - Configuration Script” on page 513 for more details.

Certificates: Allows you to add and delete security certificates in the Mediatrix Boss. See “Chapter 39 - Certificates Management” on page 527 for more details.

Table 28: Menu Frame Links (Continued)

Link Description

Table 29: Basic Call Configuration Steps

Action Description Where to?

Performing a Firmware Upgrade

The first step to do upon receiving a new Mediatrix Boss is to upgrade its firmware to the latest version available.

“Chapter 35 - Firmware Download” on page 453

Configuring the POTS parameters (Mediatrix 3208 / 3216 / 3308 / 3316 / 3731 / 3732 / 3741 / 3742)

You must minimally configure the FXS interfaces so that they can send and receive calls.

“Chapter 19 - POTS Configuration” on page 287

Configuring the ISDN Interface

You must minimally configure the ISDN interface so that it can send and receive calls. This means setting the Endpoint Type and Connection Type (BRI models) parameters.

“Chapter 17 - ISDN Configuration” on page 229

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Chapter 12 - Web Interface Configuration Where to Go From Here?

Using Secure Communication

The Mediatrix Boss allows you to use a secure communication whenever required. You must set the Mediatrix Boss with security parameters:

Configuring the SIP Endpoint

Configuring the SIP endpoint allows you to register your ISDN telephone or FXS interfaces to a SIP server. This includes setting the following parameters:

• Registrar Server Host

• Proxy Home Domain Host

• User Name

• Friendly Name

• Gateway Name

“Chapter 21 - SIP Servers” on page 307

“Chapter 22 - SIP Registration” on page 315

“Chapter 20 - SIP Gateways” on page 305

Configuring the Call Router with Routes

You must create routes that will route calls from ISDN / FXS to SIP and from SIP to ISDN / FXS.

“Routes” on page 470

Configuration of the Call Router: Mapping

You must create mappings that will allow you to properly communicate from ISDN / FXS to SIP and from SIP to ISDN / FXS.

“Mappings” on page 476

Table 29: Basic Call Configuration Steps (Continued)

Action Description Where to?

Table 30: Secure Communication Steps

Step Where to?

1. Properly synchronize the time on the Mediatrix Boss by configuring a valid SNTP server and time zone.

“SNTP Configuration” on page 80

“Time Configuration” on page 81

2. Transfer a valid CA certificate into the Mediatrix Boss.

“Chapter 39 - Certificates Management” on page 527

3. Use secure signalling by enabling the TLS transport protocol.

“Chapter 25 - SIP Transport Parameters” on page 331

4. Use secure media by:• Defining the SRTP/ SRTCP

base port.

• Setting the RTP secure mode to “Secure” or “Secure with fallback”.

“Miscellaneous Parameters” on page 380

“Security” on page 391

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13 Services

This chapter describes how to view and start/stop system and network parameters of the Mediatrix Boss.

Services Table

The Mediatrix Boss uses many services grouped in two classes: system and user. You can perform service commands on user services, but not the system services.

Whenever you perform changes in the various sections of the web interfaces, this usually means that you must restart a service for the changes to take effect. When a service needs to be restarted, it is displayed in bold and the message Restart needed is displayed in the Comment column.

If you are not able to restart a service because it is a system service, click the Reboot link in the top menu. The Reboot page then opens. You must click Reboot. This restarts the Mediatrix Boss. If the unit is in use when you click Reboot, all calls are terminated.

To manage the Mediatrix Boss services:

1. In the web interface, click the System link, then the Services sub-link.

Figure 91: System – Services Web Page

2

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The following are the services available. Note that the services available may differ depending on the Mediatrix unit you are using.

Table 31: Mediatrix Boss Services

Service Description

System Services

Authentication, Authorization and Accounting (AAA)

Authenticates a user and grants rights to perform specific tasks on the system.

Certificate Manager (CERT) Manages certificate files and provides access to these certificates.

Configuration Manager (CONF)

Responsible of configuration scripts transfers, as well as configuration image upload/download for backup/restore of the unit configuration.

Device Control Manager (DCM)

Auto-detects and identifies the hardware components of the unit.

Local Quality Of Service (LQOS)

Configures the packets tagging sent from the Mediatrix Boss.

Process Control Manager (PCM)

Responsible to boot and restart the unit.

Service Controller Manager (SCM)

Responsible to:

• Manage services information.

• Offer proxy functionality for service interoperation.

User Services

Call Routing (CROUT) Routes calls between interfaces.

DHCP Server (Dhcp) Allows the user to lease IP addresses and send network configuration to hosts located on any network.

Endpoint Administration (EpAdm)

Holds basic administration and status at endpoint and unit level.

Endpoint Services (EpServ) Manages endpoint behaviour and holds configuration parameters related to endpoints (such as DTMF maps, telephony services, etc.).

IP Routing (IpRouting) Allows the user to configure the unit's routing table.

Integrated Services Digital Network (ISDN)

Configures the Integrated Services Digital Network (ISDN) Basic Rate Interfaces (BRI) or Primary Rate Interfaces (PRI) parameters of the Mediatrix Boss.

Media IP Transport (MIPT) Holds basic configuration parameters (such as voice/data codec) and implements basic functionality related to media stream.

Music on Hold (MOH) Allows you to configure the Music on Hold parameters.

Network Address Translation (Nat)

Allows the user to change the source or destination address/port of a packet.

Network Firewall (Nfw) Allows the user to filter forwarded packets.

Notifications and Logging Manager (NLM)

Handles syslog messages and notification messages.

Network Traffic Control (Ntc) Controls the bandwidth limitation applied to physical network interfaces.

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2. In the User Service section, select the service startup type of a service in the Startup Type column.

You can put only user services in manual startup type. Proceed with caution when setting services to manual because this could prevent you from successfully contacting the unit.

3. Select if you want to perform service commands on one or more services in the Action column.

When a service needs to be restarted to apply new configuration you have set elsewhere in the web interface, it is displayed in bold and the message Restart needed is displayed in the Comment column.

If you stop, start or restart a service, any dependent services are also affected. The tabs of the services that have been stopped or have never been started because their startup type is manual are greyed out. Upon clicking these tabs, a list of services that must be restarted is displayed.

4. Click the Restart Required Services button at the bottom of the page.

Plain Old Telephony System Lines service (POTS)

Holds basic configuration parameters (such as DTMF dialing delays) and implements basic functionality related to POTS lines (such as enabling/disabling individual lines).

SIP Endpoint (SipEp) Manages the behaviour of the system regarding SIP.

Telephony Interface (TELIF) Configures the basic specification of each telephony interface.

Web (WEB) Allows accessing the unit through web pages, using HTTP.

Table 32: Startup Types

Type Description

Auto The service is automatically started when the system starts.

Manual The administrator must manually start the service.

Table 33: Actions

Action Description

Starts the service.

Stops the service.

Restarts the service.

Table 31: Mediatrix Boss Services (Continued)

Service Description

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14 Hardware Parameters

This chapter describes the hardware installed on your Mediatrix Boss.

Hardware Card Configuration

The Hardware page will differ depending on the product and model you have. However, this page will allow you to set a few parameters.

BRI Hardware Configuration

This section applies to the Mediatrix 3404 / 3408 / 3734 / 3741 / 3742 models.

To configure the Mediatrix Boss BRI hardware:

1. In the web interface, click the System link, then the Hardware sub-link.

Figure 92: System – Hardware Web PageMediatrix

2. In the BRI Cards Configuration section, select the reference of the clock source in the Clock Reference drop-down menu.

A clock reference is needed to package the ISDN voice frames. It can be generated internally or obtained from an external source (e.g., public ISDN). If you do have an interface connected to a public ISDN, select the clock reference from this interface. The clock reference provides error-free ISDN data and fax transmission.

The clock reference may either be provided by the network or by one of the BRI ports of the Mediatrix Boss.

Note: This web page does not apply to the Mediatrix 3200, and Mediatrix 3300 Series.

2

Caution: If the clock reference is not configured according to the network environment, clock slips and related voice quality degradations may occur.

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Chapter 14 - Hardware Parameters Hardware Card Configuration

If you want to configure the clock reference of a specific interface, you must set the Endpoint Type of this interface to NT. See “BRI Configuration” on page 239 for more details.

If the selected source does not provide a signal, this setting automatically falls back to None (no synchronization).

3. Click Submit if you do not need to set other parameters.

PRI Hardware Configuration

This section applies to the Mediatrix 3600 Series model.

To configure the Mediatrix Boss PRI hardware:


Figure 93: System – Hardware Web Page

Table 34: BRI Clock Reference

Reference Description

None The internal clock does not synchronize with any other source.

Other Card

The internal clock synchronizes with the other BRI card of the Mediatrix Boss. This interface must be configured in TE mode (Endpoint drop-down menu of the Interface Configuration section) to provide the clock reference to the other interfaces.

Note: This choice is not available on the Mediatrix 3404 model.

Bri0 to Bri4 The internal clock synchronizes with the signal provided by the corresponding interface. This interface must be configured in TE mode (Endpoint drop-down menu of the Interface Configuration section) to provide the clock reference to the other interfaces.

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2. In the PRI Cards Configuration section, select the reference of the clock source in the Clock Reference drop-down menu.

A clock reference is needed to package the ISDN or R2 voice frames. It can be generated internally or obtained from an external source (e.g., public ISDN or R2). If you do have an interface connected to a public ISDN or R2, select the clock reference from this interface. The clock reference provides error-free ISDN/R2 data and fax transmission.

The clock reference may either be provided by the network or by one of the PRI ports of the Mediatrix Boss.

If you want to configure the clock reference of a specific ISDN interface, you must set the Endpoint Type drop-down menu to NT. See “PRI Configuration” on page 232 for more details.

If you want to configure the clock reference of a specific R2 interface, you must set the Clock Mode drop-down menu to Master. See “R2 Channel Associated Signaling” on page 261 for more details.

3. Select whether the line uses T1 or E1 in the Line Type drop-down menu.

You must restart the unit if you change this setting.

Currently, R2 works only on the E1 line type.

4. Select whether the unit uses the ISDN or R2 signaling in the Signaling drop down menu.

If you are using ISDN signalling, refer to “Chapter 17 - ISDN Configuration” on page 229 for more information on the ISDN parameters to configure.

If you are using R2 signalling, refer to “Chapter 18 - R2 CAS Configuration” on page 259 for more information on the R2 parameters to configure.

When changing from R2 to ISDN or ISDN to R2, you must change your routes accordingly. For instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to R2, you must change the route to r2-Slot2/E1T1.


Caution: If the clock reference is not configured according to the network environment, clock slips and related voice quality degradations may occur.

Table 35: ISDN PRI Clock Reference



Other Card

The internal clock synchronizes with the other PRI interface of the Mediatrix Boss. This interface must be configured in TE mode (Endpoint drop-down menu of the Interface Configuration section) to provide the clock reference to the other interfaces.

Note: This choice is not available on the Mediatrix 3621 and 3631 models.

Table 36: R2 Clock Reference



Other Card

The internal clock synchronizes with the other R2 interface of the Mediatrix Boss. This interface must be configured in Slave mode (Clock Mode-down menu of the R2 Channel Associated Signaling section) to provide the clock reference to the other interfaces.


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15 Syslog Configuration

This chapter describes how the Mediatrix Boss handles syslog messages and notification messages.

For a list and description of all syslog messages and notification messages that the Mediatrix Boss may send, refer to the Notification Reference Guide.

Syslog Daemon Configuration

The Syslog daemon is a general purpose utility for monitoring applications and network devices with the TCP/IP protocol. With this software, you can monitor useful messages coming from the Mediatrix Boss unit. If no Syslog daemon address is provided by a DHCP server or specified by the administrator, no messages are sent.

For instance, if you want to download a new firmware into the Mediatrix Boss, you can monitor each step of the firmware download phase. Furthermore, if the unit encounters an abnormal behaviour, you may see accurate messages that will help you troubleshoot the problem.

The Mediatrix Boss supports RFC 3164 as a “device” only (see definition of device in section 3 of the RFC).

Standards Supported • RFC 3164: The BSD Syslog Protocol

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To configure the Mediatrix Boss syslog client:

1. In the web interface, click the System link, then the Syslog sub-link.

Figure 94: System – Syslog Web Page

2. In the Service Severity section, select the minimal severity to issue a notification message for the various services in the corresponding drop-down menus.

Any syslog message with a severity value greater than the selected value is ignored. Available values are:

Table 37: Severity Values

Severity Description Notification Messages Issued

Disable N/A No notification is issued.

Debug Message describing in detail the unit's operations.

All notification messages are issued.

Info Message indicating a significant event for the unit's normal operations.

Notification messages with severity “Informational” and higher are issued.

Warning Message indicating an abnormal event or situation that could be potentially risky. The unit may not be fully operational.

Notification messages with severity “Warning” and higher are issued.

Error Message indicating an abnormal event or situation, the system's operation is affected. The unit may not be operational.

Notification messages with severity “Error” and higher are issued.

Critical Message indicating a critical event or situation that requires immediate attention. The unit is not operational.

Notification messages with severity “Critical” are issued.

2

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A higher level mask includes lower level masks, e.g., Warning includes Error and Critical. The default value is Warning.

3. In the Technical Assistance Centre section, enable diagnostic traces by setting the Diagnostic Traces drop-down menu to Enable.

At the request of Media5’s Technical Support personnel, enabling these traces will allow Media5 to further assist you in resolving some issues. However, be advised that enabling this feature issues a lot of messages to the syslog host. These messages may be filtered by using the Diagnostic Traces Filter field.

4. If applicable, define the filter applied to diagnostic traces by clicking the Edit button in the Filter field.

The following opens:

Figure 95: Diagnostic Traces Window

You can use the filter to narrow down the number of traces sent at the request of Media5’s Technical Support personnel.


Configuring PCM Capture


Media5 Technical Support personnel may ask you to enable the PCM traces. PCM traces are an efficient tool to identify problems with:

Echo in your network

DTMF signals

Caller ID signals

Fax signals (or false Fax detection)

Message Waiting Indicator signals

Any other analog or digital signal

The PCM traces are two different RTP streams made specifically to record all analog signals that are either sent or received on the analog side of the Mediatrix Boss. Only the configured port, port #1 and/or #2, are sending the PCM traces for a maximum of four simultaneous RTP streams.

Note: Enabling all the traces could affect the performance of the Mediatrix Boss.

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Chapter 15 - Syslog Configuration Syslog Daemon Configuration

The RTP streams are sent to a configurable IP address, normally an IP address on your network where it can be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP streams of the VoIP call.

PCM capture supports sending streams with a ptime higher than 20 ms. The Mediatrix Boss does not support sending streams with a ptime of 10 ms.

To enable PCM capture:

1. Enable the PCM capture by putting the following line in the configuration script:

mipt.pcmCaptureEnable="1"

2. Set the unit's endpoint on which the PCM capture must be performed by putting the following line in the configuration script:

mipt.pcmCaptureEndpoint="Value"

The format is InterfaceName-Channel#. For digital interfaces (such as ISDN), you must append a -Channel# for the requested channel.

The list of endpoints is available under EpAdm.EndpointTable. Valid examples (depending of the platform) are:

• PCM capture is to be done on channel #3 of a PRI interface located in slot #2: Slot2/E1T1-3

• PCM capture is to be done on channel #2 of a BRI interface: Bri1-2

• PCM capture is to be done on the 16th FXS port: Port16

3. Set the IP address where the captured PCM packets should be sent by putting the following line in the configuration script:

mipt.pcmCaptureIpAddr="Value"

The PCM traces destination must be set so it can be recorded in a Wireshark capture on your network, normally sent to the PC doing the capture.

Configuring the Syslog Daemon Application

You must configure the Syslog daemon server to capture those messages. Refer to your Syslog daemon’s documentation to learn how to properly configure it to capture messages.

Note: Note that PCM capture does not support capturing on multiple endpoints simultaneously.

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16 Local QoS (Quality of Service)Configuration

This chapter describes how to configure the local QoS parameters. The local QoS tags packets sent from the Mediatrix Boss. It does not process nor classify packets coming from the network.

Introduction

QoS (Quality of Service) features enable network managers to decide on packet priority queuing. The Mediatrix Boss supports the Differentiated Services (DS) field.

The Mediatrix Boss supports the Real Time Control Protocol (RTCP), which is used to send packets to convey feedback on quality of data delivery.

The Mediatrix Boss does not currently support the Voice Band Data service class. It also does not support RSVP (Resource Reservation Protocol).

Specific Service Class Configuration

You can override the default value for each service class of the Mediatrix Boss:

Signalling

Voice

T.38

What are Differentiated Services?

Differentiated Services avoids simple priority tagging and depends on more complex policy or rule statements to determine how to forward a given network packet. An analogy is made to travel services, in which a person can choose among different modes of travel – train, bus, airplane – degree of comfort, the number of stops on the route, standby status, the time of day or period of year for the trip, and so forth.

For a given set of packet travel rules, a packet is given one of 64 possible forwarding behaviors – known as per hop behaviors (PHBs). A six-bit field, known as the Differentiated Services Code Point (DSCP), in the Internet Protocol header specifies the per hop behavior for a given flow of packets. The DS field structure is presented below:

0 1 2 3 4 5 6 7+---+---+---+---+---+---+---+---+

| DSCP | CU |

+---+---+---+---+---+---+---+---+MSB LSB

• DSCP: Differentiated Services CodePoint.

• CU: Currently Unused. The CU bits should always be set to 0.

For both signalling and media packets, the DSCP field is configurable independently. The entire DS field (TOS byte) is currently configurable.

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Chapter 16 - Local QoS (Quality of Service) Configuration Specific Service Class Configuration

To set specific service class values:

1. In the web interface, click the Network link, then the QoS sub-link.

Figure 96: Network – QoS Web Page

2. In the Service Class Configuration section of the QoS page, set a specific DiffServ value for each class in the DiffServ column.


2

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17 ISDN Configuration

This chapter describes how to configure the Integrated Services Digital Network (ISDN) Basic Rate Interfaces (BRI) and/or Primary Rate Interfaces (PRI) parameters of the Mediatrix Boss.

Introduction

ISDN is a set of digital transmission protocols defined by a few international standards body for telecommunications, such as the ITU-T. One or another of these protocols are accepted as standards by virtually every telecommunications carrier all over the world.

ISDN replaces the traditional telephone system so that one or two pairs of telephone wires can carry voice and data simultaneously. It is a fully digital network where all devices and applications present themselves in a digital form.

ISDN is a User-Network Interface (UNI) signalling protocol with a user and a network side. The user side is implemented in ISDN terminals (phones, terminal adapters, etc.) while the network side is implemented in the exchange switches of the network operator. Both sides have different signaling states and messages. The Mediatrix Boss ISDN interfaces can be configured to work as user (TE) or network (NT) interfaces.

Depending on your product, you can configure two types of ISDN interfaces:

The ISDN Basic Rate Interface (BRI) – Mediatrix 3404, 3408, 3734, 3741, and 3742 models.

The ISDN Primary Rate Interface (PRI) – Mediatrix 3621, 3631, 3632, 3731, 3732, and 3734 models.

ISDN Reference Points

ISDN specifies a number of reference points that define logical interfaces between the various equipment types on an ISDN access line. The Mediatrix Boss supports the following ISDN reference points:

S: The reference point between user terminals and the NT2. This is used in point-to-multipoint BRI connections.

T: The reference point between NT1 (Modem) and NT2 (PBX) devices. This is used in point-to-point PRI/BRI connections.

All other ISDN reference points are not supported.

Figure 97: ISDN Reference Points

S

BRI point-to-multipoint (S-bus)

BRI point-to-point

PRI

TE NT1

TETE

TE

NT2

NT1

NT1NT2

T

S/T

S T

ISDN

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Inband Tones Generation

In an ISDN network, most of the call setup tones are played locally by the TE equipment (i.e., telephone handset), although some require that the tones be played inband by the NT.

When interworking with other networks occurs, the need for the tones to be played inband is more likely to arise.

The Mediatrix Boss may enable inband tones to be played locally, on a per-interface basis. This option is present when the unit is acting as both the NT and the TE UNI-side. However, in TE mode, only the ringback tone is played.

The Call Setup tones (dial tone, ringback tone, etc.) are played in the direction where the call has been initiated. The call disconnection tones are played in both directions, but of course will not arrive to the peer who disconnected the call.

When an inband tone is played, a Progress Indicator information element (IE) #8 “Inband information or appropriate pattern available” is added to the ISDN message corresponding to the call state change, and in a PROGRESS ISDN message if no state change is occurring.

When an interface is acting as the TE, as soon as the NT advertises that it plays inband tones through a Progress Indicator IE #8 or #1, the local inband tones generation is disabled for the rest of the call.

Whenever a tone is played inband locally or the ISDN peer advertises that inband informations are available, the Mediatrix Boss is notified. The IP media path can then be opened earlier in the call, and can be closed with some delay after the call disconnection initiation.

The following table summarizes the inband tones generation behaviour for the NT mode.

The following table summarizes the inband tones generation behaviour for the TE mode.

Note that when the PRI/BRI interface Signalling Protocol drop-down is set to QSIG, the Signal IE does not exist so it has no effect on the inband tones generation. In QSIG, inband tones are played when the inband tones generation is activated on the incoming side of the call. See “PRI Configuration” on page 232 or “BRI Configuration” on page 239 for more details.

Signal Handling

The Signal IE is used by the NT ISDN side to tell its TE peers that they must generate an inband tone locally. Thus, the Signal IEs are sent by the NT only.

When the Signal IE handling is enabled on a given ISDN interface acting as a TE, inband tones are played towards the IP side when a Signal IE is received. On a NT interface, a Signal IE is inserted in the ISDN messages sent to the TE when appropriate.

Table 38: Inband Tones Generation Behaviour - NT

Signal IE Handling EnabledInband Tones Generation

EnabledInband Tone Played

No No No

No Yes Yes

Yes Don’t care No

Table 39: Inband Tones Generation Behaviour - TE

Signal IE Handling Enabled

Signal IE Received

Inband Tones Generation

Enabled

NT Peer Advertised

Inband Tones

Inband Tone Played

No Don’t care No Don’t care No

Yes Yes No Don’t care Yes

Yes No Don’t care Don’t care No

No Don’t care Yes Yes No

230 Mediatrix Boss

Setting PRI Hardware Parameters Software Configuration Guide

Setting PRI Hardware Parameters

You must set hardware-related parameters. You can do so in the System / Hardware page. The Hardware page differs depending on the product and model you have.

To configure the Mediatrix Boss hardware:




If you want to configure the clock reference of a specific interface, you must set the Endpoint Type drop-down menu to NT. See “PRI Configuration” on page 232 for more details.


You must restart the unit if you change this setting.

4. Select the ISDN signaling in the Signaling drop down menu.

When changing from R2 to ISDN or ISDN to R2, you must change your routes accordingly. For instance, if you are in R2 with a route r2-Slot2/E1T1, then change to ISDN, you must change the route to isdn-Slot2/E1T1.

This parameter is available only for the Mediatrix 3621/3631/3632 models. Other models support only the ISDN protocol.


Table 40: Clock Reference



Other Card

The internal clock synchronizes with the other PRI interface of the Mediatrix Boss. This interface must be configured in TE mode (Endpoint drop-down menu of the Interface Configuration section) to provide the clock reference to the other interfaces.


2 43

Mediatrix Boss 231

Chapter 17 - ISDN Configuration PRI Configuration

PRI Configuration

This section applies to the following models:

Mediatrix 3621

Mediatrix 3631

Mediatrix 3632

Mediatrix 3731

Mediatrix 3732

Mediatrix 3734

The Primary Rate Interface (PRI) port supports 30 x 64 kbit/s B-channels, 1 x 64 kbit/s D-channel and 1 x synchronization timeslot on a standard E1 (G.704) physical layer. In its T1 version, a PRI interface supports 23 x B-channels and 1 x D-channel, all at a 64 kbit/s rate. E1 is mostly deployed in Europe, while T1 is more present in North America.

Standards Supported • ITU-T Recommendation G.703: Physical/Electrical Characteristics of Hierarchical Digital Interfaces

• ITU-T Recommendation G.704: Synchronous Frame Structures Used at Primary and Secondary Hierarchy Levels

• ITU-T Recommendation I.431: Primary rate user-network interface - Layer 1 specification

• ITU-T Recommendation Q.921: ISDN user-network interface - Data link layer specification

• ITU-T Recommendation Q.931: ISDN user-network interface layer 3 specification for basic call control

Caution: You can configure ISDN ports while they are active. However they are internally disabled to modify the configuration and then re-enabled. All active calls on the port are dropped during this process. Configuration changes should only be performed during planned down times. Most of the ISDN parameters change require a restart of the ISDN service to be applied.

Caution: The Mediatrix Boss PRI ports can be used as a T reference point, but not as U reference points (2-wire). Never connect a U SCN line or a U TE into the Mediatrix Boss PRI ports.

232 Mediatrix Boss

PRI Configuration Software Configuration Guide

To configure the PRI parameters:

1. In the web interface, click the ISDN link, then the Primary Rate Interface sub-link.

Figure 99: ISDN – Interface Configuration Section

2. Select to which interface you want to apply the changes in the Select Interface drop-down menu at the top of the window.

You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix Boss in the Apply to the Following Interfaces section at the bottom of the page. You can select specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.

The Mediatrix 3632 model has two interfaces.

3. If applicable, use the Line Type Configure link to set the line type, as described in “Setting PRI Hardware Parameters” on page 231.

4. Select the endpoint type in the Endpoint Type drop-down menu.

Table 41: Endpoint Type

Type Description

TE Terminal Equipment.The endpoint emulates the subscriber (terminal) side of the digital connection. You can connect the SCN to the endpoint.

45

67

8

10

12

14

9

11

13

16

18

15

17

2019

2122

2

2324

25

2

27

29

26

28

Mediatrix Boss 233


The setting used for the Mediatrix Boss must be opposite to the setting used in the PBX. For instance, if the PBX is set to TE, then the Mediatrix Boss must be set to NT.

When the PRI interface Signalling Protocol drop-down is set to QSIG (see Step 9), the endpoint type is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.

5. Select the clock mode of the interface in the Clock Mode drop-down menu.

The interface can either generate the clocking for the line or accept the clock from the line.

The clock mode is used to give the user the possibility to set an endpoint in TE mode and still generate the clock by specifying the clock mode to master. The clock source can then be selected from the Clock Reference drop-down menu (see “Setting PRI Hardware Parameters” on page 231 for more details). The clock mode could be used, for instance, to synchronize several units in NT mode via a T1 or E1 line.

6. Select the port pinout in the Port Pinout drop-down menu.

7. Set the Monitor Link State drop-down menu with the physical link state of the ISDN interface.

Note that if the Monitor Link State parameter is enabled and the Ignore SIP OPTONS on no usable endpoints parameter is also enabled in the SIP / Interop page, this will influence how the SIP options are answered. See “SIP Interop” on page 336 for more details.

NT Network Termination. The endpoint emulates the central office (network) side of the digital connection. You can connect a PBX to the endpoint.

Table 41: Endpoint Type (Continued)

Type Description

Note: If you want to use a specific interface as the reference clock, you must set it to TE.

Table 42: Clock Mode

Mode Description

auto The setting is derived from the endpoint type.

• NT: clock master

• TE: clock slave

Master The interface generates the clock.

Slave The interface accepts the clock from the line.

Table 43: Port Pinout

Mode Description

Auto The pinout is set according to the Endpoint Type parameter setting (Step 4).

Te Forces the pinout to TE regardless of the Endpoint Type value.

Nt Forces the pinout to NT regardless of the Endpoint Type value.

Table 44: Interface Link State


Enable The ISDN endpoint's operational state is affected by its interface physical link state. When the link state of the ISDN interface is down, the operational state of its matching endpoint becomes “disable”.

Disable The ISDN endpoint's operational state is not affected by its interface physical link state.

234 Mediatrix Boss


8. Select the transmission encoding of bits in the Line Coding drop-down menu.

Make sure that the transmission encoding matches with the remote system. For further information, see ITU-T Recommendation G.703.

9. Select the frame format in the Line Framing drop-down menu.

Line Framing is used to synchronize the channels on the frame relay circuit (when a frame starts and finishes). Without it, the sending and receiving equipment would not be able to synchronize their frames.

For further information, see ITU-T Recommendation G.704.

10. Select the protocol to use for the signalling channel in the Signalling Protocol drop-down menu.

This signalling must match the connected ISDN equipment or network.

The 5ESS and DMS100 protocols support basic call only and the model is derived directly from the DSS1 protocol.

11. Select the value of the network location in the progress indicator messages that the unit sends in the Network Location drop-down menu.

This defines the location code to be inserted in ISDN causes code information elements, i.e., the type of network to which the system belongs. The following values are available:

• User

• Private

• Public

• Transit

• International

Table 45: Transmission Encoding

Coding Description

B8ZS Bipolar with 8-Zeros Substitution (T1 lines).

HDB3 High-Density Bipolar with 3-zeros (E1 lines).

AMI Alternate Mark Inversion (E1 and T1 lines).

Table 46: Line Framing

Format Description

SF Super frame. Sometimes known as D4 (T1 lines).

ESF Extended super frame (T1 lines).

CRC4 Cyclic redundancy check 4 (E1 lines).

NO-CRC4 No Cyclic redundancy check 4 (E1 lines).

Table 47: Signalling Protocols

Protocol Description

DSS1 Digital Subscriber Signaling System No.1

DMS100 Digital Multiplex System 100

NI2 National ISDN No.2

5ESS 5 Electronic Switching System

QSIG ECMA's protocol for Private Integrated Services Networks

Mediatrix Boss 235


12. Set the Preferred Encoding Scheme drop-down menu with the data encoding scheme in the bearer capabilities (user information layer 1 protocol).

This encoding scheme is used when initiating a call on the ISDN side. The supported encoding schemes are G.711 u-Law and G.711 a-Law.

G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711 a-Law as preferred encoding protocol.

13. Set the Fallback Encoding Scheme drop-down menu with the fallback data encoding scheme in case the preferred encoding scheme is not available.

The supported encoding schemes are G.711 u-Law and G.711 a-Law.

If the proposed encoding scheme in the bearer capability received in the SETUP message is different than the preferred encoding scheme, then the fallback encoding scheme is used.

14. Define the range of active bearer channels in the Channel Range field.

15. Select the strategy for selecting bearer channels in the Channel Allocation Strategy drop-down menu.

16. Define the maximum number of active calls on the interface in the Maximum Active Calls field.

This limits the total number of concurrent calls on the interface. Entering 0 indicates no maximum number of active calls.

17. Select whether or not the signal information element is enabled in the Signal Information Element drop-down menu.

• When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT), the signal information element is sent to the User UNI-side.

• When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the tone indicated in the signal information element is played in the IP direction.

Note: The fallback encoding scheme is valid only when receiving a SETUP message. The user sending the SETUP message does not indicate alternative bearer capability.

Table 48: Channel Allocation Strategy

Allocation Description

Ascending Starting from the lowest-numbered non-busy bearer channel and going toward the highest-numbered non-busy bearer channel, the Mediatrix Boss selects the first bearer channel available.

Descending Starting from the highest-numbered non-busy bearer channel and going toward the lowest-numbered non-busy bearer channel, the Mediatrix Boss selects the first bearer channel available.

RoundRobinAscending The Mediatrix Boss starts from the bearer channel that follows the bearer channel used for the last call. For instance, if channel #1 was used in the last call, the unit starts with channel #2. Going toward the highest-numbered non-busy bearer channel, the unit selects the first channel available. If the highest channel is unavailable, the search continues from the lowest-numbered non-busy bearer channel.

RoundRobinDescending The Mediatrix Boss starts from the bearer channel that precedes the bearer channel used for the last call. For instance, if channel #3 was used in the last call, the unit starts with channel #2. Going toward the lowest-numbered non-busy bearer channel, the unit selects the first channel available. If the lowest channel is unavailable, the search continues from the highest-numbered non-busy bearer channel.

236 Mediatrix Boss


18. Select whether or not inband tone generation is enabled in the Inband Tone Generation drop-down menu.

When activated at the User UNI-side (Endpoint Type drop-down menu is set to TE), only the ringback tone is generated.

When the Signalling Protocol drop-down menu is set to QSIG (Step 9) and this variable is activated, the incoming side of the call plays the tones inband.

19. Select whether or not inband DTMF dialing is enabled in the Inband DTMF Dialing drop-down menu.

If you select Enable, the Mediatrix Boss accepts inband DTMF digits when Overlap Dialing occurs.

20. Select whether or not overlap dialing is enabled in the Overlap Dialing drop-down menu.

21. Define the maximum length of the calling party name for calls from SIP to ISDN in the Calling Name Max Length field.

Available values range from 0 to 82.

22. Select whether or not exclusive B-Channel selection is enabled for calls from SIP to ISDN in the Exclusive B-Channel Selection drop-down menu.

If you select Enable and initiate a call, only the requested B channel is accepted; if the requested B channel is not available, the call is cleared.

23. Select whether or not to enable the Sending Complete information element into SETUP messages for calls from SIP to ISDN in the Sending Complete drop-down menu.

Some ISDN switches may require that the Sending Complete information element be included in the outgoing SETUP message to indicate that the entire number is included and there are no further destination digits to be sent.

24. Select whether or not to enable sending the RESTART message upon a signalling channel “UP” event in the Send Restart On Startup drop-down menu.

The RESTART message requests a restart for the interface specified.

25. Set the link establishment strategy in the Link Establishment drop-down menu.

Table 49: Overlap Dialing Parameters


Enable The Mediatrix Boss transports the called-party number digit by digit, after the first SETUP message, which contains no called party information at all.

Disable The Mediatrix Boss transports the full called party information in the first SETUP message from the terminal. This means that the user must dial the number before going off-hook.

Table 50: Link Establishment Parameters


OnDemand When the data link is shut down, the unit establishes a new link only when required.

Permanent When the data link is shut down, the unit immediately attempts to establish a new link.

Mediatrix Boss 237


26. Set the STATUS causes that can be received without automatically clearing the call in the Accepted Status Causes field.

A STATUS message may be received indicating a compatible call state and containing one of the following causes:

The default action is to clear the call upon receiving a STATUS message. If a STATUS message is received indicating a compatible call state and containing the supplied STATUS causes, the clearing of the call is prevented.

The comma separated list must use the following syntax:

96, 98, 100

Note that any other cause value is ignored.

27. Set the range of PROGRESS causes accepted by the unit in the Accepted Progress Causes field.

Causes excluded from this range trigger call disconnections.

The string has the following syntax:

• ',': Separator between non-consecutive lists of causes or single cause.

• 'n': A single cause, where n is the cause number.

• 'm-n': List of causes where m is the start cause number and n is the end cause number.

Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.

28. Select the strategy for sending ISDN Progress messages in the Send Isdn Progress drop-down menu.

Table 51: STATUS Causes

Cause Description

96 Mandatory information element is missing.

97 Message type non-existent or not implemented.

98 Message not compatible with call state or message type non-existent or not implemented.

99 Information element/parameter non-existent or not implemented.

100 Invalid information element contents.

101 Message not compatible with call state.

Note: You must consider the following:

• The space character is not allowed.

• Causes must be specified in low to high order.

• Cause duplication is not allowed.

Table 52: Send ISDN Progress Parameters


Send All Send an ISDN Progress message in all situations where call progression is signaled.

Send Inband

Send an ISDN Progress message only when call progression contains an indication of in-band information.

Send Alerting

Send an ISDN Alerting message instead of ISDN Progress message when call progression contains an indication of in-band information. If call progression does not contain in-band information, no message is sent at this step.

238 Mediatrix Boss

BRI Configuration Software Configuration Guide

The strategy for sending Progress messages should be adapted to the configuration of the peer ISDN switch. Some switches may terminate calls when receiving one or many ISDN progress messages.

29. Select the strategy for sending the Progress Indicator Information Element in the Send Progress Indicator IE drop-down menu.

The strategy for the Progress Indicator IE should be adapted to the configuration of the peer ISDN switch.

This parameter controls sending of a Progress Indicator IE in ISDN messages where Progress Indicators are allowed. See the parameters in “Interop Parameters Configuration” on page 247 for a control over which ISDN message allows Progress Indicators.


BRI Configuration

This section applies to the following models:

Mediatrix 3404

Mediatrix 3408

Mediatrix 3734

Mediatrix 3741

Mediatrix 3742

A Basic Rate Interface (BRI) port supports 2 x 64 kbit/s B-channels for switched voice or data connections and 1 x 16 kbit/s D-channel for signalling.

Table 53: Send Progress Indicator IE Parameters


Send All Send the Progress Indicator IE in all situations.

Send Inband Only

Send the Progress Indicator only when the Progress Description contains an indication of in-band information.



• ITU-T Recommendation I.430: Basic user-network interface - Layer 1 specification

• ITU-T Recommendation Q.921: ISDN user-network interface - Data link layer specification

• ITU-T Recommendation Q.931: ISDN user-network interface layer 3 specification for basic call control

Caution: The Mediatrix Boss ISDN BRI ports are configurable to operate as network or terminal ports. The pin-out of the sockets is switched according to this configuration. Wrong port configurations, wrong cabling or wrong connections to neighbouring equipment can lead to short circuits in the BRI line powering. Refer to the Hardware Installation Guide to avoid misconfigurations.

Caution: The Mediatrix Boss BRI ports can be used as a S or T reference point, but not as U reference points (2-wire). Never connect a U SCN line or a U TE into the Mediatrix Boss BRI ports.

Mediatrix Boss 239

Chapter 17 - ISDN Configuration BRI Configuration

The Mediatrix 3404 / 3734 / 3741 / 3742 has 5 ISDN BRI ports. It supports up to 8 simultaneous ISDN voice/data channels over any IP connection.

The Mediatrix 3408 has 10 ISDN BRI ports. It supports up to 16 simultaneous ISDN voice/data channels over any IP connection.

To configure the BRI parameters:

1. In the web interface, click the ISDN link, then the Basic Rate Interface sub-link.

Figure 100: ISDN – Basic Rate Interface Web Page

2. In the Interface Configuration section of the Basic Rate Interface page, select to which interface you want to apply the changes in the Select Interface drop-down menu at the top of the window.


The Mediatrix 3404 model has 5 interfaces in Slot 1, while the Mediatrix 3408 model has 10 interfaces in Slots 1 and 2 (5 in each).

2

34

56

7

9

11

13

8

10

12

15

17

14

16

1918

20

2

2122

2324

2625

240 Mediatrix Boss


3. Select the endpoint type in the Endpoint Type drop-down menu.

The setting used for the Mediatrix Boss must be opposite to the setting used in the PBX. For instance, if the PBX is set to TE, then the Mediatrix Boss must be set to NT.

When the BRI interface Signalling Protocol drop-down is set to QSIG (see Step 7), the endpoint type is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.



The clock mode is used to give the user the possibility to set an endpoint in TE mode and still generate the clock by specifying the clock mode to master. The clock source can then be selected from the Clock Reference drop-down menu (see “Chapter 14 - Hardware Parameters” on page 215 for more details). The clock mode could be used, for instance, to synchronize several units in NT mode via a BRI line.

5. Set the Monitor Link State drop-down menu with the physical link state of the ISDN interface.


Table 54: Endpoint Type

Type Description

TE Terminal Equipment.The endpoint emulates the subscriber (terminal) side of the digital connection. You can connect the SCN to the endpoint.

NT Network Termination. The endpoint emulates the central office (network) side of the digital connection. You can connect a PBX or ISDN telephones to the endpoint.

Note: If you want to use a specific interface as the reference clock, you must set it to TE.

Table 55: Clock Mode

Mode Description

auto The setting is derived from the endpoint type.

• NT: clock master

• TE: clock slave



Note: In a BRI configuration, setting the clock mode to master for a TE endpoint is invalid. Slave mode is automatically applied in this case.



Enable The ISDN endpoint's operational state is affected by its interface physical link state. When the link state of the ISDN interface is down, the operational state of its matching endpoint becomes “disable”.

Disable The ISDN endpoint's operational state is not affected by its interface physical link state.

Mediatrix Boss 241


6. Select the connection type of the endpoint in the Connection Type drop-down menu.

The type of connection depends on the equipment to which the Mediatrix Boss port is connected and it must be the same for all interconnected pieces of equipment.

The Point to MultiPoint configuration is not available in QSIG.

7. Select the protocol to use for the signalling channel in the Signalling Protocol drop-down menu.

This signalling must match the connected ISDN equipment or network.

8. Select the value of the network location in the progress indicator messages that the unit sends in the Network Location drop-down menu.

This defines the location code to be inserted in ISDN causes code information elements, i.e., the type of network to which the system belongs. The following values are available:

• User

• Private

• Public

• Transit

• International

9. Set the Preferred Encoding Scheme drop-down menu with the data encoding scheme in the bearer capabilities (user information layer 1 protocol).

This encoding scheme is used when initiating a call on the ISDN side. The supported encoding schemes are G.711 u-Law and G.711 a-Law.

G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711 a-Law as preferred encoding protocol.

Table 57: Connection Type

Type Description

Point to Point Can only attach one device (for instance a PBX or SCN) and acts as a T reference point.

Point to MultiPoint

Can attach more than one ISDN device and acts as a S reference point. Up to 8 TEs and one NT can be connected to a S-bus.

Note: If you are using a Mediatrix Boss connected to a S-Bus in point-to-multipoint TE mode, you cannot currently connect any additional ISDN devices to the S-Bus.

Table 58: Signalling Protocols


DSS1 Digital Subscriber Signaling System No.1

DMS100 Digital Multiplex System 100

NI2 National ISDN No.2

5ESS 5 Electronic Switching System

QSIG ECMA's protocol for Private Integrated Services Networks

Note: The Mediatrix Boss currently supports only the DSS1 and QSIG signalling protocols.

242 Mediatrix Boss


10. Set the Fallback Encoding Scheme drop-down menu with the fallback data encoding scheme in case the preferred encoding scheme is not available.

The supported encoding schemes are G.711 u-Law and G.711 a-Law.

If the proposed encoding scheme in the bearer capability received in the SETUP message is different than the preferred encoding scheme, then the fallback encoding scheme is used.




For a Mediatrix 3404 / 3408, the maximum number of simultaneous calls is limited to 8, even if 5 BRI ports can physically support 10 calls.

13. Select whether or not the signal information element is enabled in the Signal Information Element drop-down menu.

• When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT), the signal information element is sent to the User UNI-side.

• When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the tone indicated in the signal information element is played in the IP direction.

14. Select whether or not inband tone generation is enabled in the Inband Tone Generation drop-down menu.

When activated at the User UNI-side (Endpoint Type drop-down menu is set to TE), only the ringback tone is generated.

When the BRI interface Signalling Protocol drop-down is set to QSIG (see Step 7) and this parameter is activated, the incoming side of the call plays the tones inband.

Note: The fallback encoding scheme is valid only when receiving a SETUP message. The user sending the SETUP message does not indicate alternative bearer capability.







Mediatrix Boss 243


15. Select whether or not inband DTMF dialing is enabled in the Inband DTMF Dialing drop-down menu.

If you select Enable, the Mediatrix Boss accepts inband DTMF digits when Overlap Dialing occurs.

16. Select whether or not overlap dialing is enabled in the Overlap Dialing drop-down menu.

17. Define the maximum length of the calling party name for calls from SIP to ISDN in the Calling Name Max Length field.

Available values range from 0 to 82.

18. Select whether or not exclusive B-Channel selection is enabled for calls from SIP to ISDN in the Exclusive B-Channel Selection drop-down menu.

If you select Enable and initiate a call, only the requested B channel is accepted; if the requested B channel is not available, the call is cleared.

19. Select whether or not to enable the Sending Complete information element into SETUP messages for calls from SIP to ISDN in the Sending Complete drop-down menu.

Some ISDN switches may require that the Sending Complete information element be included in the outgoing SETUP message to indicate that the entire number is included and there are no further destination digits to be sent.

20. Select whether or not to enable sending the RESTART message upon a signalling channel “UP” event in the Send Restart On Startup drop-down menu.

The RESTART message requests a restart for the interface specified.

21. Set the link establishment strategy in the Link Establishment drop-down menu.

22. Set the actual keypad string that is to be considered as a hook-flash in the Hook-Flash Keypad field.

An ISDN telephone may send INFORMATION messages that contain a “Keypad Facility”. You can thus trigger a supplementary service (Hold, Conference, etc.) by sending a keypad facility.

Since the keypads can be received via several INFORMATION messages, they are accumulated until they match or reset if the keypad reception timeout (second) has elapsed since the last keypad has been received. The keypad reception timeout can only be modified via a script. If the keypad reception timeout is set to 0, it disables the timeout, thus assuming that all keypads will be received in a single INFORMATION message.

Setting this variable to an empty string disables the hook-flash detection.

The permitted keypad must be made up of IA5 characters. See ITU-T Recommendation T.50.

Table 60: Overlap Dialing Parameters


Enable The Mediatrix Boss transports the called-party number digit by digit, after the first SETUP message, which contains no called party information at all.

Disable The Mediatrix Boss transports the full called party information in the first SETUP message from the terminal. This means that the user must dial the number before going off-hook.

Table 61: Link Establishment Parameters


OnDemand When the data link is shut down, the unit establishes a new link only when required.

Permanent When the data link is shut down, the unit immediately attempts to establish a new link.

244 Mediatrix Boss


23. Set the STATUS causes that can be received without automatically clearing the call in the Accepted Status Causes field.

A STATUS message may be received indicating a compatible call state and containing one of the following causes:

The default action is to clear the call upon receiving a STATUS message. If a STATUS message is received indicating a compatible call state and containing the supplied STATUS causes, the clearing of the call is prevented.

The comma separated list must use the following syntax:

96, 98, 100

Note that any other cause value is ignored.

24. Set the range of PROGRESS causes accepted by the unit in the Accepted Progress Causes field.

Causes excluded from this range trigger call disconnections.

The string has the following syntax:

• ',': Separator between non-consecutive lists of causes or single cause.

• 'n': A single cause, where n is the cause number.

• 'm-n': List of causes where m is the start cause number and n is the end cause number.

Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.

25. Select the strategy for sending ISDN Progress messages in the Send Isdn Progress drop-down menu.

Table 62: STATUS Causes

Cause Description

96 Mandatory information element is missing.

97 Message type non-existent or not implemented.

98 Message not compatible with call state or message type non-existent or not implemented.

99 Information element/parameter non-existent or not implemented.

100 Invalid information element contents.

101 Message not compatible with call state.

Note: You must consider the following:

• The space character is not allowed.

• Causes must be specified in low to high order.

• Cause duplication is not allowed.

Table 63: Send ISDN Progress Parameters


Send All Send an ISDN Progress message in all situations where call progression is signaled.

Send Inband

Send an ISDN Progress message only when call progression contains an indication of in-band information.

Send Alerting

Send an ISDN Alerting message instead of ISDN Progress message when call progression contains an indication of in-band information. If call progression does not contain in-band information, no message is sent at this step.

Mediatrix Boss 245


The strategy for sending Progress messages should be adapted to the configuration of the peer ISDN switch. Some switches may terminate calls when receiving one or many ISDN progress messages.

26. Select the strategy for sending the Progress Indicator Information Element in the Send Progress Indicator IE drop-down menu.

The strategy for the Progress Indicator IE should be adapted to the configuration of the peer ISDN switch.

This parameter controls sending of a Progress Indicator IE in ISDN messages where Progress Indicators are allowed. See the parameters in “Interop Parameters Configuration” on page 247 for a control over which ISDN message allows Progress Indicators.


Bypass Feature (Mediatrix 3404/3408/3734/3741/3742 Models)

In the event of a power or network failure, the bypass feature permits users to make and receive calls even when the Mediatrix Boss is not operating. The Mediatrix Boss BRI 3 and BRI 4 ports may either act as a SCN bypass. For instance, if you decide to connect a SCN line into the BRI 4 port, you can use a BRI telephone connected into the BRI 3 port to make calls.

Furthermore:

The port on which the SCN line is connected must be configured as a TE.

The other port must be configured as a NT.

Refer to “BRI Configuration” on page 239 for more details on how to configure the line type.

During normal operation, the direct connection between the BRI 3 and BRI 4 ports is switched out through commuting relays and both ports resume normal functions. When power is removed from the Mediatrix Boss, the relay setting is restored to a connected state and the SCN line can be used as an emergency line. Consequently, a BRI telephone used on the other port is directly connected to this SCN line. When the power is restored, this automatically removes the Bypass connection; this means that any ongoing call on the Bypass connection is terminated.

Table 64: Send Progress Indicator IE Parameters


Send All Send the Progress Indicator IE in all situations.

Send Inband Only

Send the Progress Indicator only when the Progress Description contains an indication of in-band information.

Note: If you are using a crossover Ethernet cable to connect the SCN line to the Mediatrix Boss and there is a power failure, the bypass feature does not work properly.

246 Mediatrix Boss

Interop Parameters Configuration Software Configuration Guide

Interop Parameters Configuration

The interop parameters allow the Mediatrix Boss to properly work, communicate, or connect with specific ISDN devices.

To set the interop parameters:

1. In the web interface, click the ISDN link, then the Interop sub-link.

Figure 101: ISDN – Interop Web Page




The Mediatrix 3734/3741/3742 models have 5 interfaces.


3. Define the behaviour of the Progress Indicator In Setup drop-down menu.

This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the SETUP message when acting as the originating side.

See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator IE (“PRI Configuration” on page 232 and “BRI Configuration” on page 239).

4. Define the behaviour of the Progress Indicator In Setup Ack drop-down menu.

This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the SETUP ACK when acting as the terminating side.

3

2

54

76

8

10

2

11

9

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Chapter 17 - ISDN Configuration Interop Parameters Configuration


5. Define the behaviour of the Progress Indicator In Call Proceeding drop-down menu.

Thie menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the CALL PROCEEDING message in response to a SETUP message when acting as the terminating side.


6. Define the behaviour of the Progress Indicator In Progress drop-down menu.

This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the PROGRESS message in response to a SETUP message when acting as the terminating side.


7. Define the behaviour of the Progress Indicator In Alerting drop-down menu.

This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the ALERTING message.


8. Define the behaviour of the Progress Indicator In Connect drop-down menu.

This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the CONNECT message.


9. Define a value, in milliseconds (ms), in the Maximum Facility Waiting Delay (ms) field.

This value defines the maximum amount of time to wait for a FACILITY message, after receiving a SETUP message, before going on with normal call processing.

After receiving a SETUP message, the system waits for this amount of time for a FACILITY message. As soon as it receives a FACILITY message or the delay expires, it goes on with normal call processing.

A FACILITY message can contain useful information for the call. For example, it can contain a Calling Name.

You must enable the Supplementary Services to use the delay. See “PRI Configuration” on page 232 or “BRI Configuration” on page 239 for more details.

Setting the value to 0 deactivates this feature.

10. Define whether or not a message with progress indicator No. 1 MUST be considered as offering inband information available in the Use Implicit Inband Info drop-down menu.

If so, the network must activate the B-channel connection.

The progress indicator No. 1 means that the call is not end-to-end ISDN; further call progress information may be available inband.

11. Define how the Calling Name Identity Presentation (CNIP) is delivered when DMS-100 is selected as the signaling protocol in the Dms-100 CNIP Delivery field.

Table 65: DMS-100 CNIP Delivery Parameters


DisplayIe Use a Display Information Element for delivering the CNIP.

FacilityIe Use a Facility Information Element for delivering the CNIP.

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Interop Parameters Configuration Software Configuration Guide

When receiving an incoming call, the three possible sources of Calling Party Name are checked in the following order: USER-USER, DISPLAY and FACILITY. The last found is used.

The Calling Party Name is accepted in a DISPLAY information element only when explicitly identified as a Calling Party Name (i.e. only when "Display Type" = "Calling Party Name" in the information element).

When initiating a call, the Calling Party Name is sent according to the method selected above. If the method selected is not supported for the protocol in use, the default method for this protocol is used. The following table shows which method is used vs. the configuration of CallingNameDelivery:

See “PRI Configuration” on page 232 and “BRI Configuration” on page 239 for more details on signaling protocols.

12. Define whether or not an ISDN message with a TEI broadcast needs to be interpreted as a TEI 0 when the connection type is 'PointToPoint' in the Allow TEI Broadcast in PTP drop-down menu.

See “BRI Configuration” on page 239 for more details on the connection type.

This parameter is available for the following BRI models:

• Mediatrix 3404

• Mediatrix 3408

• Mediatrix 3734

• Mediatrix 3741

• Mediatrix 3742


Table 66: Calling Name Delivery Method vs. Configuration

ProtocolCalling Name Delivery

eFacility eDisplay eUserUser eSignalingProtocol

DSS1 IE User-User IE User-User IE User-User IE User-User

Dms100 IE Facility IE Display IE Display IE Display

NI-2 IE Facility IE Facility IE Facility IE Facility

5ESS IE Facility IE Facility IE User-User IE Facility

QSIG IE Facility IE Facility IE Facility IE Facility

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Chapter 17 - ISDN Configuration ISDN Timers Configuration

ISDN Timers Configuration

This section allows you to set timer parameters.

To set the ISDN timers:

1. In the web interface, click the ISDN link, then the Timer sub-link.

Figure 102: ISDN – Timer Web Page






3. Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is automatically cancelled.

Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring until the party answers.

4. Set the value, in milliseconds (ms), of the Layer 1 Timer T3 in the L1 Timer T3 field.

Timer 3 (T3) is a supervisory timer that has to take into account the overall time to activate. This time includes the time it takes to activate both the TE-NT and the NT-TE portion of the customer access.

The expiry of Timer T3 is intended to provide an indication that the network side cannot complete the activation procedure, probably due to a failure condition or the terminal cannot detect INFO 4.


3

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Services Configuration Software Configuration Guide

Services Configuration

This section allows you to set the ISDN optional services.

To set the ISDN optional services:

1. In the web interface, click the ISDN link, then the Services sub-link.

Figure 103: ISDN – Services Web Page






3. Select whether or not supplementary services FACILITY messages should be accepted on the ISDN interface in the Facility Services drop-down menu.

This controls how to use the FACILITY information element.

Table 67: Supplementary Services


Disable When receiving a FACILITY message containing supplementary services information, the Mediatrix Boss replies with a STATUS message saying the FACILITY is not supported.

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7

9

6

8

13

10

2

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Generic procedures for the control of ISDN supplementary services are defined in the recommendation ITU-T Q.932.

The following supplementary services FACILITY message is supported:

• CNIP: Service offered to the called user and which provides the name of the calling user (calling party name) to the called user. It is defined in the standard ECMA-164.

You can define a waiting delay as described in the section “Interop Parameters Configuration” on page 247.

4. Select whether or not calling number digits information is sent on the ISDN in the Calling Line Information Presentation drop-down menu.

The Calling Line Information Presentation (CLIP) is an optional service that can be offered by the ISDN provider. When you call somebody, your telephone number is transmitted to the other phone.

When the PRI/BRI interface Signalling Protocol drop-down is set to QSIG, this parameter does not depend on the PRI/BRI Interface Endpoint Type drop-down menu (NT or TE).

See “PRI Configuration” on page 232 or “BRI Configuration” on page 239 for more details.

5. Select whether or not the calling number restriction at the originating ISDN side is enabled in the Calling Line Information Restriction drop-down menu.

The Calling Line Information Restriction (CLIR) is a service offered to the TE at the originating interface.

When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the Calling Number information element is marked with a “restricted” Presentation Indicator.

When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT), the Calling Number information element Presentation Indicator is always overridden with “restricted” by the Network.

Enable When receiving a FACILITY message containing supplementary services information, the Mediatrix Boss accepts and interprets the message, processing supported supplementary service messages and silently discarding unsupported supplementary service messages.

Table 67: Supplementary Services (Continued)


Note: To activate the ISDN hold feature, you must also set the Default Hook Flash Processing feature to Use Signaling Protocol in “General Configuration” on page 419.

Table 68: CLIP Parameters


UserOnly Sends a Calling Number IE that contains the calling number digits when acting as the User UNI-side (Endpoint Type drop-down menu set to TE) otherwise the Calling Number IE is not sent.

Enable Sends a Calling Number IE that contains the calling number digits.

Disable The Calling Number IE is never sent.

Table 69: CLIP Parameters (with QSIG)


UserOnly Sends a Calling Number IE that contains the calling number digits.

Enable Sends a Calling Number IE that contains the calling number digits.

Disable The Calling Number IE is never sent.

252 Mediatrix Boss




6. Select whether or not to activate the delivery of the Calling Number digits at the destination ISDN side even if the Calling Party information element's Presentation Indicator is set to “restricted” in the Calling Line Information Restriction Override drop-down menu.

When activated at the ISDN destination-side Network UNI-side (Endpoint Type drop-down menu set to NT), the Calling Number digits is sent to the User UNI-side (TE), but the Presentation Indicator is left untouched.



7. Select whether or not to send a Connected Number IE within the CONNECT message at the originating ISDN side in the Connected Line Identification Presentation drop-down menu.

The Connected Line Identification Presentation (COLP) is an optional service offered at the originating interface by the NT to the TE.

8. Select whether or not to set the Connected Number Information Element restriction at the destination ISDN side in the Connected Line Identification Restriction drop-down menu.

The Connected Line Identification Restriction (COLR) is a service offered to the TE at the destination interface.

Table 70: CLIR Parameters (with QSIG)


Enable The Presentation Indicator in the Calling Number IE is set to “restricted” in outgoing calls.

Disable There is no restriction on the calling user number.

Table 71: CLIR Override Parameters (with QSIG)


Enable In incoming calls, when the Calling Number IE is present and its Presentation Indicator is set to “restricted”, the restriction is overridden and the calling user number is presented to the called user.

Disable The restriction is not overridden.

Table 72: COLP Parameters


Enable Sends a Connected Number IE within the CONNECT message, which contains the connected number digits once the transformation of the routing table has been applied.

Disable The Connected Number IE is never sent.

Table 73: COLR Parameters


Enable When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), marks the Connected Number IE with a 'restricted' Presentation Indicator, which keeps privacy over the connected number digits. This option has no effect when activated at the Network UNI-side (Endpoint Type drop-down menu set to NT).

Mediatrix Boss 253


9. Select whether or not to set the Connected Number Information Element restriction override at the originating ISDN side in the Connected Line Identification Restriction Override drop-down menu.

10. Define whether or not NOTIFY messages can be sent in the Outgoing Notify drop-down menu.

The following NOTIFY messages are supported:

• REMOTE HOLD: Sent when the remote peer holds the call.

• REMOTE RETRIEVAL: Sent when the remote peer retrieves the call.

11. Set the Maintenance Service Call Termination drop-down menu with the call termination strategy after reception of a service message requesting a maintenance on the associated bearer channel.

12. Define the AOC-E Support drop-down menu how to send the total charge at the (E)nd of the call in AOC-E messages.

Disable No restriction is applied.

Table 74: COLR Override Parameters


Enable When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT), the connected number digits are delivered even if the Presentation Indicator is set to 'restricted'. This option has no effect when activated at the User UNI-side (Endpoint Type drop-down menu set to TE). This is a national option designed for emergency services.

Disable No restriction override is applied.

Table 75: Outgoing Notify Parameters


Enable NOTIFY messages can be sent.

Disable NOTIFY messages are never sent.

Table 76: Maintenance Service Parameters


Graceful The call proceeds normally until the user clears the call. The associated bearer is then set to maintenance. This is the default value.

Abrupt The call is terminated immediately and set to maintenance.

Table 77: AOC-E Support Parameters


no The AOC-E support is disabled. No information is forwarded to the peer interface.

transparent

On an NT interface, the information is sent as received from the network. No information is sent if the network does not provide information.

On a TE interface, the information is forwarded to the peer interface if AOC messages are received from the network.

automatic On an NT interface, always send the information. If the network does not provide information, 'noChargeAvailable' is sent.

On a TE interface, the information is forwarded to the peer interface if AOC messages are received from the network.

Table 73: COLR Parameters (Continued)


254 Mediatrix Boss


13. Enter a number in one or more of the MSN fields.

This enables the Multiple Subscriber Numbers (MSN) supplementary service with these numbers. A MSN is a telephone number associated with a line.

The MSN supplementary service enables each individual terminal on one access to have one or more identities. You can enter up to three MSNs. The value of this field must be digits.

If the Called E.164 received from a call does not match any MSN numbers, the call is silently discarded.

This supplementary service applies only on a BRI Interface configured in TE Point to Multi-Point. See “BRI Configuration” on page 239 for more details.


explicit On an NT interface, always send the information if the phone requests AOC on a per-call basis. 'noChargeAvailable' is sent if the network does not provide information. If the phone does not request AOC on a per-call basis, no information is sent.

On a TE interface, send an AOC request to the network. If the network rejects the request, no information is forwarded to the peer interface. Otherwise, the information is forwarded to the peer interface is AOC messages are received from the network.

Table 77: AOC-E Support Parameters (Continued)


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CAS Parameters

C H A P T E R

18 R2 CAS Configuration

This chapter describes how to configure the R2 CAS parameters of the Mediatrix Boss.

Introduction

CAS stands for Channel Associated Signaling. With this method of signaling, each traffic channel has a dedicated signaling channel. In other words, the signaling for a particular traffic circuit is permanently associated with that circuit. Channel-associated call-control is still widely used today, mostly in South America, Africa, Australia, and in Europe.

The Mediatrix Boss uses the MFC/R2 CAS protocol. This is a compelled sequence multi-frequency code signaling. MFC/R2 can be used on international as well as national connections.

In MFC/R2 signaling, the equipment units at the exchanges that send and receive digits, and the signaling between these units, are usually referred to as register and interregister signaling.

The terms forwards and backwards are heavily used in descriptions of MFC/R2. Forwards is the direction from the calling party to the called party. Backwards is the direction from the called party to the calling party.

You can configure Mediatrix Boss parameters for the E1 R2 CAS.

Line Signals for the Digital Version of MFC/R2

The MFC/R2 digital line signals (defined in ITU-T Q.421) are the ABCD bits of CAS in timeslot 16 of an E1. They represent the states of the line, and are similar to the states of an analog line. In general, only bits A and B are used. In most systems, bits C and D are set to fixed values and never change. There are some national variants where bit C or D may be used for metering pulses.

Interregister Signals

The interregister, or interswitch, signals in MFC/R2 signaling (defined in ITU-T Q.441) are encoded as the presence of 2, and only 2, out of 6 specific tones, spaced at 120 Hz intervals. Two sets of tones are defined – one for forward signals, and one for backward signals. There are 15 combinations of 2 out of 6 tones, so there are 10 signals for the digits 0 to 9, and 5 additional signals available for supervisory purposes.

MFC/R2 uses a separate set of frequencies for the forward and backwards directions.

The interregister signals are sent in-band. They may pass transparently through several nodes in the network between the two terminating switches. The signals are arranged in groups. When a call begins, the calling end uses group I signals, and the called end uses group A. The called end may tell the calling end to switch to using group II and group B signals, or to switch back to group A. In some countries, there are also groups III and C, used for caller number transfer. Groups III and C do not exist in the ITU specifications.



• ITU-T Recommendation Q.421: Digital line signalling code

• ITU-T Recommendation Q.441: Signalling code

Note: This chapter applies only to the Mediatrix 3621, Mediatrix 3631, and Mediatrix 3632 models.

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Chapter 18 - R2 CAS Configuration Selecting the R2 Signaling Protocol

MFC/R2 uses a system called compelled signaling. To ensure the sending end never sends signals too fast, each signal from the sending end results in an acknowledgement from the receiving end. The sending end is instructed signal by signal what it should send next – a dialed digit, a digit of caller ID, etc.

Selecting the R2 Signaling Protocol

You must set the unit to use the R2 signaling protocol. You can do so in the System / Hardware page. The Hardware page differs depending on the product and model you have.

To configure the Mediatrix Boss hardware:




If you want to configure the clock reference of a specific interface, you must set the Clock Mode drop-down menu to Master. See “R2 Channel Associated Signaling” on page 261 for more details.


Currently, R2 works only on the E1 line type.

Table 74: Clock Reference



Other Card

The internal clock synchronizes with the other R2 interface of the Mediatrix Boss. This interface must be configured in Slave mode (Clock Mode drop-down menu of the R2 Channel Associated Signaling section) to provide the clock reference to the other interfaces.


2 43

Note: Before version 1.1r8.76, the Isdn service needed to be restarted when modifying the Line Type. Since version 1.1r8.76, the Line Type variable has been moved to the Ex1Pri_1 service and now you rather need to restart the unit instead of restarting the service when the Line Type is modified.

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4. Select the R2 signaling in the Signaling drop down menu.

When changing from R2 to ISDN or ISDN to R2, you must change your routes accordingly. For instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to R2, you must change the route to r2-Slot2/E1T1.


R2 Channel Associated Signaling

The R2 Channel Associated Signaling section allows you to define the general parameters related to R2.

To configure the R2 CAS parameters:

1. In the web interface, click the R2 link, then the R2 Config sub-link.

Figure 105: R2 Channel Associated Signaling Web Page


The number of interfaces available vary depending on the Mediatrix Boss model you have.



Table 75: R2 Interface Clock Mode

Mode Description



3

2

4

7

5

9

11

8

6

10

1213

1415

16

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Chapter 18 - R2 CAS Configuration R2 Channel Associated Signaling

The clock source can be selected from the Clock Reference drop-down menu (see “Selecting the R2 Signaling Protocol” on page 260 for more details). The clock mode could be used, for instance, to synchronize several units in master clock mode via an E1 line.

4. Select the port pinout in the Port Pinout drop-down menu.

5. Set the Monitor Link State drop-down menu with the physical link state of the R2 interface.


6. Select the transmission encoding of bits in the Line Coding drop-down menu.

Make sure that the transmission encoding matches with the remote system. For further information, see ITU-T Recommendation G.703.

7. Select the frame format in the Line Framing drop-down menu.

Line Framing is used to synchronize the channels on the frame relay circuit (when a frame starts and finishes). Without it, the sending and receiving equipment would not be able to synchronize their frames.

For further information, see ITU-T Recommendation G.704.

8. Define the range of active bearer channels in the Channel Range field.

Table 76: Port Pinout

Mode Description

Auto The pinout is set according to the Clock Mode parameter setting (Step 3).

Te Forces the pinout to TE regardless of the Clock Mode value.

Nt Forces the pinout to NT regardless of the Clock Mode value.



Enable The R2 endpoint's operational state is affected by its interface physical link state. When the link state of the R2 interface is down, the operational state of its matching endpoint becomes "disable".

Disable The R2 endpoint's operational state is not affected by its interface physical link state.

Table 78: Transmission Encoding

Coding Description

B8ZS Bipolar with 8-Zeros Substitution (T1 lines). Currently not available.

HDB3 High-Density Bipolar with 3-zeros (E1 lines).

AMI Alternate Mark Inversion (E1 and T1 lines).

Table 79: Line Framing

Format Description

SF(D4) Super frame. Sometimes known as D4 (T1 lines). Currently not available.

ESF Extended super frame (T1 lines). Currently not available.

CRC4 Cyclic redundancy check 4 (E1 lines).

NO-CRC4 No Cyclic redundancy check 4 (E1 lines).

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11. Set the Encoding Scheme drop-down menu with the voice encoding scheme in the bearer capabilities.

This encoding scheme is used when initiating a call on the R2 side. The supported encoding schemes are G.711 u-Law and G.711 a-Law.

12. Select the protocol to use for the line signaling in the Line Signaling Protocol drop-down menu.

This signaling must match the connected equipment or network. The Mediatrix Boss currently supports only the Q421-2BitsSignaling signaling, which is the R2 line signaling type ITU-U Q.421. It is typically used for PCM systems.

13. Select the R2 incoming digit signaling method in the Incoming Digit Signaling drop-down menu.

Digit signaling is also known as Address Signaling, selection signals and register signaling. The digits are used primarily to indicate the called number, but can also have other meanings.







Table 81: Incoming Digit Signaling Parameters


MfcR2 Multi Frequency Compelled - R2.

DtmfR2 Dual Tone Multi Frequency - R2.

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Chapter 18 - R2 CAS Configuration R2 Signaling Variants

14. Select the R2 outgoing digit signaling method in the Outgoing Digit Signaling drop-down menu.

Digit signaling is also known as Address Signaling, selection signals and register signaling. The digits are used primarily to indicate the called number, but can also have other meanings.

15. Select the country in the Country Selection drop-down menu.

You have the following choices:

• BrazilR2

• MexicoR2

• VenezuelaR2

• PhilipinesR2

• ITU-TR2

16. Set the Digit Attenuation field with the additional attenuation, in dB, for MFR2/DTMF digits generation.

By default, MFR2/DTMF digits generation power is determined by country selection. This parameter provides a mean to reduce this power.


R2 Signaling Variants

This section allows you to decide whether or not you want to override the default R2 signaling parameters. The Mediatrix Boss uses the following default values:

Table 82: Outgoing Digit Signaling Parameters


MfcR2 Multi Frequency Compelled - R2.

DtmfR2 Dual Tone Multi Frequency - R2.

Table 85: R2 Signaling Parameters Default Values

ParameterDefault Value (ms)

Bra. Mex. Arg. Sau. Ven. Phi. ITU-T

Bits CD 1 1 1 1 1 1 1

ANI Length 0 (Variable ANI length)

0 (Variable ANI length)






DNIS Length 0 (Variable DNIS length)

0 (Variable DNIS length)






ANI Request Enable Disable Enable Enable Enable Enable Enable

Send ANI request after nth DNIS Digits

0 (Variable number of DNIS digits)



1 0 (Variable number of DNIS digits)



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Override Default Country Settings

You can override the default R2 signaling parameters. In that case, you will have access to the R2 Signaling Variants section to define the signaling you want.

To override the R2 signaling default settings:

1. In the web interface, click the R2 link, then the Signaling sub-link.

Figure 106: R2 Signaling Variants Web Page



3. Select whether or not you want to override the default setting of R2 signaling parameters in the Override Default Country Settings drop-down menu.

Collect Call Blocked Enabled

Enable Disable Disable Disable Disable Disable Disable

ANI Category NatSubscriberNoPrio

Nat Subscriber NoPrio






Line Free Category LineFreeNoCharge

Line Free NoCharge

Line Free Charge

Line Free Charge

Line Free Charge

Line Free Charge

Line Free Charge

ANI Restricted Enable Disable Disable Disable Disable Disable Disable

Table 85: R2 Signaling Parameters Default Values (Continued)



Table 86: R2 Signaling Override


Disable The interface uses the default country configuration.

3

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Chapter 18 - R2 CAS Configuration R2 Signaling Variants

Overriding the default settings is considered as advanced configuration. Media5 recommends not to modify the country variants unless you know exactly what you are doing.

R2 Signaling Variants

This section allows you to define R2 signaling parameters. You can click the Reset to Default button at any time to revert back to the default R2 signaling.

To set R2 signaling parameters:

1. In the R2 Signaling Variants section, set the C and D bits when the device transmits line signals in the Bits CD field.

The device ignores the C and D bits of received line signals.

Figure 107: R2 Signaling Variants Section

2. Set the ANI Length field with the length of Automatic Number Identification (ANI) to be requested or sent.

If a variable-length ANI is used, the End of ANI tone set in “R2 Tones Forward Groups” on page 278 is sent to indicate the end of the ANI digits. When fixed-length ANI is used and the available ANI digits are longer than the requested length, the last n digits are sent. If available ANI digits are shorter, then the End of ANI tone set in “R2 Tones Forward Groups” on page 278 is sent.

• 0: Variable ANI length.

• 1..20: Specific ANI length.

3. Set the DNIS Length field with the length of the Dialed Number Identification Service (DNIS) expected.

DNIS is the called party or the destination number. If a variable length is defined, then the I-15 digit is used to indicate the end of DNIS.

• 0: Variable DNIS length used.

• 1..20: Specific DNIS length expected.

4. Define whether or not ANI should be requested in the ANI Request field.

ANI is the calling party number. When ANI is requested, the calling party category followed by the actual ANI is sent. If this parameter is enabled, the ANI request is sent after the nth DNIS digit (defined in Step 5) is received.

5. Set the Send ANI Request after nth DNIS Digits field with the number of DNIS digits to be received before sending the ANI request (if the ANI Request field is set to Enable).

If a variable number is used, the ANI request is sent after all DNIS digits have been received.

• 0: Variable number of DNIS digits.

Enable The interface uses the specific country configuration as defined in the R2 Signaling Variants section. To retrieve the default configuration associated with the current country, click the Reset to Default button. Proceed to “R2 Signaling Variants” on page 266.

Table 86: R2 Signaling Override (Continued)


12

3

54

67

98

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• 1..10: Specific number of DNIS digits.

6. Define whether or not the Collect Call Blocked Option is used in the Collect Call Blocked Enabled drop-down menu.

For an incoming collect call, a signal of Group II-8 is sent forward from the caller to the called party. The called party implements the collect call blockage (when enabled) by sending backward to the caller a signal of Group B-7 indicating that the collect calls are not being accepted by the called party. Consequently, the originator of the call gets a busy tone and the local calling party circuit that has been used for the call is dropped when the originator puts the phone on hook.

7. Set the ANI Category drop-down menu with the group II forward signal to be sent upon receiving a calling party category request.

This tone indicates the category of the calling party.

8. Set the Line Free Category drop-down menu with the group B backward signal to be sent by the incoming R2 register to indicate line free condition of the destination party.

9. Set the ANI Restricted drop-down menu with the behaviour of the unit following the reception of a reject request after sending the Send next digit (ANI) request.

The request is generally rejected when the calling party is unable to send its identification.

Table 87: ANI Category Parameters


NatSubscriberNoPrio The call is set up from a national subscriber’s line and is non-priority.

NatSubscriberPrio The call is set up from a national subscriber’s line to which priority treatment of calls has been granted.

NatMaintenance The call comes from a national maintenance equipment.

NatSpare Spare.

NatOperator The call is set up from a national operator’s position.

NatData The call will be used for national data transmission.

IntSubscriberNoPrio The call is set up from an international subscriber’s line and is non-priority.

IntData The call will be used for international data transmission.

IntSubscriberPrio The call is set up from an international subscriber’s line to which priority treatment of calls has been granted.

IntOperator The call is set up from an international operator’s position.

CollectCall The call is set up for Call Collect.

Table 88: Line Free Category Parameters


LineFreeNoCharge The called party’s line is free but is not to be charged on answer.

LineFreeCharge The called party’s line is free but is to be charged on answer.

Table 89: ANI Restricted Parameters


Enable A congestion tone is sent in response to the reject request and the call MUST be dropped.

Disable The unit uses the same behaviour as the End of ANI Tone and the call WILL be completed.

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Chapter 18 - R2 CAS Configuration R2 Timers Variants


R2 Timers Variants

This section allows you to decide whether or not you want to override the default R2 timers parameters. The Mediatrix Boss uses the following default values:

Table 92: R2 Timers Default Values



Seizure Ack Timeout 2000 2000 2000 2000 2000 2000 2000

Fault Seizure Ack Timeout

60000 60000 60000 60000 60000 60000 60000

Double Seizure Timeout

100 100 100 100 100 100 100

Double Answer Timeout

1000 1000 1000 1000 1000 1000 1000

Answer Timeout 0 0 0 0 0 0 0

ReAnswerTimeout 1000 1000 1000 1000 1000 1000 1000

Release Guard Timeout

100 100 100 100 100 100 100

InterCall Guard Timeout

100 100 100 100 100 100 100

Congestion Tone Guard Timeout

1000 1000 1000 1000 1000 1000 1000

Unblocking Timeout 100 100 100 100 100 100 100

Address Complete Timeout

8000 8000 8000 8000 8000 8000 8000

Wait Answer Timeout 60000 60000 60000 60000 60000 60000 60000

Digit Complete Timeout

4000 4000 4000 4000 4000 4000 4000

Wait GroupB Response Complete Timeout

3000 3000 3000 3000 3000 3000 3000

Wait Immediate Response Complete Timeout

1000 1000 1000 1000 1000 1000 1000

Play Tone Guard Timeout

70 70 70 70 70 70 70

Accept Call Timeout 2000 2000 2000 2000 2000 2000 2000

Clear Forward Guard Timeout

1500 1500 1500 1500 1500 1500 1500

Clear Backward Guard Timeout

1500 1500 1500 1500 1500 1500 1500

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You can override the default R2 timers. In that case, you will have access to the R2 Timers Variants section to define the timers you want.

To override the R2 timers default settings:

1. In the web interface, click the R2 link, then the Timers sub-link.

Figure 108: R2 Timers Variants Web Page



3. Select whether or not you want to override the default setting of R2 timers in the Override Default Country Settings drop-down menu.

Fault On Answered Guard Timeout

250 250 250 250 250 250 250

Fault On Clear Backward Guard Timeout

250 250 250 250 250 250 250

Fault On Seize Ack Guard Timeout

250 250 250 250 250 250 250

Fault On Seize Guard Timeout

250 250 250 250 250 250 250

Table 92: R2 Timers Default Values (Continued)



Table 93: R2 Timers Override



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Chapter 18 - R2 CAS Configuration R2 Timers Variants


R2 Timers Variants

This section allows you to define R2 timers. You can click the Reset to Default button at any time to revert back to the default R2 timers.

To set R2 timers:

1. In the R2 Timers Variants section, set the Seizure ACK Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits for the seizure acknowledgement signal after sending a seizure signal.

Figure 109: R2 Timers Variants Section

2. Set the Fault Seizure Ack Timeout field with the maximum time, in milliseconds (ms), an incoming R2 register waits for a seizure acknowledge failure condition to clear.

3. Set the Double Seizure Timeout field with the minimum time, in milliseconds (ms), an outgoing R2 register waits after a double seizure is recognized before releasing the connection.

4. Set the Double Answer Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits after receiving a clear-backward signal before releasing the connection.

5. Set the Release Guard Timeout field with the maximum time, in milliseconds (ms), an incoming R2 register waits before sending an idle line signal when a clear forward line signal is received.

6. Set the InterCall Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits after receiving an idle line signal before attempting a new seizure of the line.

7. Set the Congestion Tone Guard Timeout field with the maximum time, in milliseconds (ms), an incoming R2 register waits after sending a congestion tone before sending a clear forward line signal and transit to the idle state.

Enable The interface uses the specific country configuration as defined in the R2 Timers Variants section. To retrieve the default configuration associated with the current country, click the Reset to Default button. Proceed to “R2 Timers Variants” on page 270.

Table 93: R2 Timers Override (Continued)


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R2 Timers Variants Software Configuration Guide

8. Set the Unblocking Timeout field with the maximum time, in milliseconds (ms), a both-way trunk waits before assuming an idle state when a blocking condition is removed.

This will prevent a too aggressive seizure of the trunk.

9. Set the Address Complete Timeout field with the maximum time, in milliseconds (ms), that the caller waits for the reception of an Address Complete Tone after sending all ANI or DNIS digits.

10. Set the Wait Answer Timeout field with the maximal time, in milliseconds (ms), that the caller waits for an Answer signal (ANSW) after receiving a Group B Line Free Signal Tone.

This timer is effective only when the line is free.

11. Set the Digit Complete Timeout field with the maximal time, in milliseconds (ms), that the caller waits for a Group I forward tone after sending either a Group A next DNIS digit, next ANI digit, or next calling category Tone.

12. Set the Wait GroupB Response Complete Timeout field with the maximal time, in milliseconds (ms), that the caller waits for the confirmation of the end of the compelled sequence after receiving a Group B Signal Tone.

The end of the compelled sequence is detected by a transition of the backward tone to off.

13. Set the Wait Immediate Response Complete Timeout field with the maximal time, in milliseconds (ms), that the caller waits for the confirmation of the end of the compelled sequence after receiving a Group B Signal Tone.

The end of the compelled sequence is detected by a transition of the backward tone to off. This timer is specific to the immediate accept Signal Tone.

14. Set the Play Tone Guard Timeout field with the maximum time, in milliseconds (ms), an incoming R2 register waits after receiving the confirmation of the reception of the Group B Signal Tone before playing one of the calling tones in the caller direction.

15. Set the Accept Call Timeout field with the time, in milliseconds (ms), that the unit waits to accept a R2 CAS call.

16. Set the Clear Forward Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits after sending a clear forward line signal before transiting to the idle state.

17. Set the Clear Backward Guard Timeout field with the maximum time, in milliseconds (ms), an incoming R2 register waits after sending a clear backward line signal before sending the idle line signal and transit in the idle state.

18. Set the Fault On Answered Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits for the fault to clear before sending the clear forward line signal.

In the case bb = 0 is recognized while in the answered state, no immediate action is taken. However, the clear forward signal is sent if bb = 1 is restored or the answered guard timeout is reached.

19. Set the Fault On Clear Backward Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits for the fault to clear before sending the clear forward line signal.

In the case bb = 0 is recognized while in the clear backward state, no immediate action is taken. However, the clear forward signal is sent if bb = 1 is restored or the clear backward guard timeout is reached.

20. Set the Fault On Seize Ack Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits for the fault to clear before sending the clear forward line signal.

In the case bb = 0 is recognized while in the seize acknowledge state prior to the answer signal, no immediate action is taken. However, the clear forward signal is sent if bb = 1 is restored or the seize acknowledge guard timeout is reached.

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Chapter 18 - R2 CAS Configuration R2 Digit Timers Variants

21. Set the Fault On Seize Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2 register waits for the fault to clear before sending the clear forward line signal.

In the case bb = 0 is recognized while in the seize state, no immediate action is taken. However, when the seize acknowledgement signal is recognized after the seize ack timeout period has elapsed or the seize guard timeout is reached, the clear forward signal is sent.


R2 Digit Timers Variants

This section allows you to decide whether or not you want to override the default R2 digit timers. The Mediatrix Boss uses the following default values:


You can override the default R2 digit timers. In that case, you will have access to the R2 Digit Timers Variants section to define the timers you want.

To override the R2 digit timers default settings:

1. In the web interface, click the R2 link, then the Digit Timers sub-link.

Figure 110: R2 Digit Timers Variants Web Page

Table 96: R2 Digit Timers Default Values



MFC Pulse Inter Digit Timeout

100 100 100 100 100 100 100

MFC Pulse Min On Timeout

150 150 150 150 150 150 150

MFC Max Sequence Timeout

10000 20000 20000 20000 20000 20000 20000

MFC Max On Timeout 5000 10000 10000 10000 10000 10000 10000

MFC Max Off Timeout 5000 10000 10000 10000 10000 10000 10000

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3. Select whether or not you want to override the default setting of R2 digit timers in the Override Default Country Settings drop-down menu.


R2 Digit Timers Variants

This section allows you to define R2 digit timers. You can click the Reset to Default button at any time to revert back to the default R2 digit timers.

To set R2 digit timers:

1. In the R2 Digit Timers Variants section, set the MFC Pulse Inter Digit Timeout field with the minimum delay, in milliseconds (ms), between the end of transmission of the last signal of the compelled cycle and the start of the next one.

Figure 111: R2 Digit Timers Variants Section

2. Set the MFC Pulse Min On Timeout field with the minimum time, in milliseconds (ms), a backward tone can be on from the backward perspective.

3. Set the MFC Max Sequence Timeout field with the maximum time, in milliseconds (ms), for a complete compelled signaling cycle from the forward perspective.

4. Set the MFC Max On Timeout field with the maximum time, in milliseconds (ms), a forward tone can be on from the forward perspective.

5. Set the MFC Max Off Timeout field with the maximum time, in milliseconds (ms), a forward tone can be off from the forward perspective.


Table 97: R2 Digit Timers Override



Enable The interface uses the specific country configuration as defined in the R2 Digit Timers Variants section. To retrieve the default configuration associated with the current country, click the Reset to Default button. Proceed to “R2 Digit Timers Variants” on page 273.

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Chapter 18 - R2 CAS Configuration R2 Link Timers Variants

R2 Link Timers Variants

This section allows you to decide whether or not you want to override the default R2 link timers. The Mediatrix Boss uses the following default values:


You can override the default R2 link timers. In that case, you will have access to the R2 Link Timers Variants section to define the timers you want.

To override the R2 link timers default settings:

1. In the web interface, click the R2 link, then the Link Timers sub-link.

Figure 112: R2 Link Timers Variants Web Page



3. Select whether or not you want to override the default setting of R2 link timers parameters in the Override Default Country Settings drop-down menu.

Table 100: R2 Link Timers Default Values



Link Activation Timeout

1000 1000 1000 1000 1000 1000 1000

Link Activation Retry Timeout

3000 3000 3000 3000 3000 3000 3000

Table 101: R2 Link Timers Override



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R2 Tones Variants Software Configuration Guide


R2 Link Timers Variants

This section allows you to define R2 link timers. You can click the Reset to Default button at any time to revert back to the default R2 link timers.

To set R2 link timers:

1. In the R2 Link Timers Variants section, set the Link Activation Timeout field with the maximum time, in milliseconds (ms), the unit waits for an activation indication coming from the physical link.

The activation indication is used to indicate that the physical layer connection has been activated.

Figure 113: R2 Link Timers Variants Section

2. Set the Link Activation Retry Timeout field with the maximum time, in milliseconds (ms), the unit waits before attempting to re-establish the physical link.

The attempt is made when the physical layer connection has been deactivated.


R2 Tones Variants

This section allows you to decide whether or not you want to override the default R2 tones parameters. The Mediatrix Boss uses the following default values:

Enable The interface uses the specific country configuration as defined in the R2 Link Timers Variants section. To retrieve the default configuration associated with the current country, click the Reset to Default button. Proceed to “R2 Link Timers Variants” on page 275.

Table 101: R2 Link Timers Override (Continued)


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Table 104: R2 Tones Default Values



R2 Tones Forward Groups

End of DNIS Tone I15 None I15 I15 None I15 I15

End of ANI Tone I15 I15 I15 I15 I15 I15 I15

Restricted ANI Tone I12 None None None I12 None None

R2 Tones Backward Groups

Send Next DNIS Digit Tone

A1 A1 A1 A1 A1 A1 A1

Send Previous DNIS Digit Tone

A9 None A9 A2 A2 A9 A9

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Switch to Group II Tone

A3 A3 A3 A3 A3 A3 A3

Network Congestion Tone

A4 A4 A4 A4 A4 A4 A4

Send Calling Party Category Tone


Immediate Accept Tone

None None A6 A6 A6 A6 A6

Send DNIS Digit N-2 Tone


Send DNIS Digit N-3 Tone


Repeat all DNIS Tone None None A10 None A10 A10 A10

Send Next ANI Digit Tone

A5 A2 A5 A5 A9 A5 A5

Send Calling Party Category Tone Switch to Group C

None A6 None None None None None

Send Special Information Tone

None None B2 B2 B2 B2 B2

User Busy Tone B3 B2 B3 B3 B3 B3 B3


B4 B4 B4 B4 B4 B4 B4

Unassigned Number Tone

B8 None B7 B5 B5 B5 B5

Line Free with Charge Tone

B1 B1 B6 B1 B6 B6 B6

Line Free with Charge Tone (Supplementary)

B6 None None None None None None

Line Free without Charge Tone

B7 B5 B7 B6 B7 B7 B7

Line Out of Order Tone

B8 None B8 B8 B8 B8 B8

Changed Number Tone

B3 None None None None None None

Send Next ANI Digit Tone

None C1 None None None None None

Repeat All DNIS Tone switch to Group A


Send Next DNIS Digit Tone switch to Group A


Table 104: R2 Tones Default Values (Continued)



276 Mediatrix Boss



You can override the default R2 tones. In that case, you will have access to the R2 Tones Forward Groups and R2 Tones Backward Groups sections to define the tone you want.

To override the R2 tones default settings:

1. In the web interface, click the R2 link, then the Tones sub-link.

Figure 114: R2 Tones Variants Web Page



3. Select whether or not you want to override the default setting of R2 tones parameters in the Override Default Country Settings drop-down menu.



Send Previous DNIS Digit Tone switch to Group A


Switch to Group II Tone


Table 104: R2 Tones Default Values (Continued)



Table 105: R2 Tones Override



Enable The interface uses the specific country configuration as defined in the R2 Tones Forward Groups and R2 Tones Backward Groups sections. To retrieve the default configuration associated with the current country, click the Reset to Default button. Proceed to “R2 Tones Forward Groups” on page 278.

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R2 Tones Forward Groups

These tones fall into Groups I and II. All Group I and II tones use the ITU-T R2 forward group tone frequencies. Most of the tones in Group I are used for addressing and identification. Group II tones give information about the origin of the call. Group II tones are also sent by an outgoing MFC/R2- or International MFC/R2-register in response to one of the following backward signals:

Change over to the reception of Group B signals.

Send calling party's category.

You can click the Reset to Default button at any time to revert back to the default R2 tones.

To set the R2 tones forward groups:

1. In the R2 Tones Forward Groups section, select the forward Group 1 tone used to send after the DNIS digits in the End of DNIS Tone drop-down menu.

You have the choice between none and I1 to I15 Group 1 forward signals MF Tone.

Figure 115: R2 Tones – Forward Groups Section

2. Select the forward Group 1 tone used to send after the ANI digits in the End of ANI Tone drop-down menu.


Table 106: R2 Group I Forward Tones

MF Tone Description

1 I-1 Digit 1 (Language: French, if first signal sent in intl. link)

2 I-2 Digit 2 (Language: English, if first signal sent intl. link)

3 I-3 Digit 3 (Language: German, if first signal sent in intl. link)

4 I-4 Digit 4 (Language: Russian, if first signal sent in intl. link)

5 I-5 Digit 5 (Language: Spanish, if first signal sent in intl. link)

6 I-6 Digit 6 (Language: Spare, if first signal sent in intl. link)



9 I-9 Digit 9 (Discriminating digit, if first signal sent in intl. link)

10 I-10 Digit 0 (Discriminating digit, if first signal sent in intl. link)

11 I-11 Country code indicator, outgoing half-echo suppressor required

12 I-12 Country code indicator, no echo suppressor required

13 I-13 Test call indicator (call by automatic test equipment)

14 I-14 Country code indicator, outgoing half-echo suppressor inserted

15 I-15 Signal is not used

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3. Select the forward Group 1 tone used to reject a query in the Restricted ANI Tone drop-down menu.

This tone is generally used in response to the identification request when the caller party is unable to send his identification to the called party. If no tone is defined, the End of ANI tone is sent to the caller.



R2 Tones Backward Groups

Backward tones fall into Groups A and B. The Group A tones acknowledge Group I forward signals. Under certain conditions, Group A tones also acknowledge Group II tones. Group B tones convey the following information to an outgoing MFC/R2 register:

The condition of the switch equipment in the incoming exchange.

The condition of the called subscriber's line.

Table 107: R2 Group A Backward Tones

MF Tone Description

1 A-1 Send next digit (n + 1)

2 A-2 Send last but one digit (n -1)

3 A-3 Address-complete, changeover to reception of Group B signals

4 A-4 Congestion in the national network

5 A-5 Send calling party’s category

6 A-6 Address-complete, charge, set-up speech conditions

7 A-7 Send last but two digit (n - 2)

8 A-8 Send last but three digit (n - 3)

9 A-9 Spare for national use

10 A-10 Spare for national use

11 A-11 Send country code indicator

12 A-12 Send language or discrimination digit

13 A-13 Send nature of circuit

14 A-14 Request for information on use of an echo suppressor

15 A-15 Congestion in an international exchange or at its output

Table 108: R2 Group B Backward Tones

MGF Tone Description

1 B-1 Spare for national use

2 B-2 Send special information tone

3 B-3 Subscriber’s line busy

4 B-4 Congestion

5 B-5 Unallocated number

6 B-6 Subscriber’s line free, charge

7 B-7 Subscriber’s line free, no charge

8 B-8 Subscriber’s line out of order

Mediatrix Boss 279


To set the R2 tones backward groups:

1. In the R2 Tones Backward Groups section, select the backward Group A tone used to request the next DNIS digit in the Send Next DNIS Digit Tone drop-down menu.

You have the choice between none and A1 to A15 Group A backward signals MF Tone.








Table 109: R2 Group C Backward Tones


1 C-1 Send next number of Subscriber A

2 C-2 Send first digit BNUM

3 C-3 Send Group II and Group B signals

4 C-4 Congestion

5 C-5 Send next digit BNUM

6 C-6 Repeat last digit BNUM

7 C-7 Spare for national use









Table 108: R2 Group B Backward Tones (Continued)


280 Mediatrix Boss


Figure 116: R2 Tones – Backward Groups Section

2. Select the backward Group A tone used to request the previous DNIS digit in the Send Previous DNIS Digit Tone drop-down menu.


3. Select the backward Group A tone used to request to the caller a switch of Group II signals in the Switch to Group II Tone drop-down menu.


4. Select the backward Group A tone to be sent when a congestion network is detected in the Network Congestion Tone drop-down menu.


5. Select the backward Group A tone sent when the backward group requests the calling party category in the Send Calling Party Category Tone drop-down menu.


6. Select the backward Group A tone sent when the backward group accepts the call immediately in the Immediate Accept Tone drop-down menu.


7. Select the backward Group A tone used to request the previous - 1 DNIS digit in the Send DNIS Digit N-2 Tone drop-down menu.


8. Select the backward Group A tone used to request the previous - 2 DNIS digit in the Send DNIS Digit N-3 Tone drop-down menu.


9. Select the backward Group A tone used to request all DNIS digits in the Repeat All DNIS Tone drop-down menu.


10. Select the backward Group A tone used to request the next ANI digit in the Send Next ANI Digit Tone drop-down menu.


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11. Select the backward Group A tone used to request the calling party category and then switch to Group C signals in the Send Calling Party Category Tone Switch to Group C drop-down menu.


12. Select the backward Group B tone used to send a special information tone in the Send Special Information Tone drop-down menu.

You have the choice between none and B1 to B15 Group B backward signals MF Tone.

13. Select the backward Group B tone used to signal a user busy in the User Busy Tone drop-down menu.


14. Select the backward Group B tone used to signal a network congestion in the Network Congestion Tone drop-down menu.


15. Select the backward Group B tone used to signal a unassigned number in the Unassigned Number Tone drop-down menu.


16. Select the backward Group B tone used to signal that the line is free and charge must be applied in the Line Free with Charge Tone drop-down menu.


17. Select a supplementary backward Group B tone used to detect that the line is free and charge must be applied in the Line Free with Charge Tone (Supplementary) drop-down menu.

This is a supplementary (optional) tone used on the reception only. You have the choice between none and B1 to B15 Group B backward signals MF Tone.

18. Select the backward Group B tone used to signal that the line is free and no charges must be applied in the Line Free without Charge Tone drop-down menu.


19. Select the backward Group B tone used to signal that the line is out of order in the Line Out of Order Tone drop-down menu.


20. Select the backward Group B tone used to signal that the subscriber has changed number in the Changed Number Tone field.


21. Select the backward Group C tone used to request the next ANI digit in the Send Next ANI Digit Tone drop-down menu.

You have the choice between none and C1 to C15 Group C backward signals MF Tone.

22. Select the backward Group C tone used to request all DNIS digits and then switch to Group A signals in the Repeat All DNIS Tone switch to Group A drop-down menu.


23. Select the backward Group C tone used to request the next DNIS digit and then switch to Group A signals in the Send Next DNIS Digit Tone switch to Group A drop-down menu.


24. Select the backward Group C tone used to signal a network congestion in the Network Congestion Tone drop-down menu.


282 Mediatrix Boss


25. Select the backward Group C tone used to request the previous DNIS digit and then switch to Group A signals in the Send Previous DNIS Digit Tone switch to Group A drop-down menu.


26. Select the backward Group A tone used to request to the caller a switch of Group II signals in the Switch to Group II Tone drop-down menu.



Mediatrix Boss 283


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POTS Parameters

C H A P T E R

19 POTS Configuration

This chapter describes how to configure the POTS (Plain Old Telephony System) line service, which allows you to configure the analog specification of each line, as well as gateways-specific parameters.

POTS Status

The POTS parameters are displayed in the POTS / Status page.

Line Status

The Line Status table lists the link state of the FXS lines.

Figure 117: POTS – Status Web Page

The State column may have one of the following values:

Idle: The line is available

In Use: The line is currently used

Disabled: The line is disabled

Bypass: The line is on bypass

Down: The power of the line is down

Note: This web page is available only on the following models:

• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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Chapter 19 - POTS Configuration General POTS Configuration

FXO Line Status

The FXO Line Status table lists the link state updated by the link state verification mechanism.

Figure 118: POTS – FXO Line Status Table

You can enable the line state verification in “FXO Line Verification” on page 299.

The unit does not automatically detect when a previously connected port has changed to the disconnected status. The unit only detects the change of status when it attempts to use the port or after a restart.

The unit automatically detects within seconds when a disconnected port becomes connected.

Unknown: The line fault detection is disabled.

Up: When last polled, the line was connected.

Down: When last polled, the line was disconnected.

General POTS Configuration

The General Configuration section allows you to select the detection/generation method of caller ID.

To configure the general POTS parameters:

1. In the web interface, click the POTS link, then the Config sub-link.

Figure 119: POTS Web Page


• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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2. Select the detection/generation method of caller ID in the Caller ID customization drop-down menu.

This allows selecting the detection/generation method of caller ID. See “Caller ID Information” on page 290 for more details.

3. Select the caller ID transmission method in the Caller ID Transmission drop-down menu.

It allows selecting the transmission type of the caller ID.

4. Determine the type of vocal information that can be obtained by dialing a pre-defined digit map in the Vocal Unit Information drop-down menu.

When entering special characters on your telephone pad, the Mediatrix Boss talks back to you with relevant information.

To access the vocal unit information:

a. Take one of the telephones connected to the Mediatrix Boss.

b. Dial one of the digits sequence on the keypad.

Table 110: Caller ID Parameters


Country Uses the default caller ID of the country defined in the Country section of the Telephony > Misc page (“Country Configuration” on page 447).

EtsiDtmf ETSI 300 659-1 (DTMF string sent between the first and second ring).

EtsiFsk ETSI 300 659-1 (FSK (V.21) sent between the first and second ring).

Table 111: Caller ID Transmission Parameters


Country Uses the default caller ID of the country defined in the Country section of the Telephony > Misc page (“Country Configuration” on page 447).

First Ring The caller ID is sent after the first ring.

Ring Pulse The caller ID is sent between a brief ring pulse and the first ring.

Line Reversal Ring Pulse

The caller ID is sent between a brief ring pulse and the first ring on an inverted polarity line.

DT-AS The aaller ID is sent after the dual tone alerting state tone.

Line Reversal DT-AS

The caller ID is sent after the dual tone alerting state tone on an inverted polarity line.

No Ring Pulse The caller ID is sent before the first ring.

Table 112: Caller ID Parameters


None The vocal information feature is disabled.

All Enable all vocal information digit maps.

Table 113: Vocal Unit Information

Digits to Dial Information Vocally Sent by the Mediatrix Boss

*#*0 List of IP addresses of the Mediatrix Boss (static or DHCP).

*#*1 MAC address of the Mediatrix Boss.

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Chapter 19 - POTS Configuration General POTS Configuration


Caller ID Information

The caller ID is a generic name for the service provided by telephone utilities that supply information such as the telephone number or the name of the calling party to the called subscriber at the start of a call. In call waiting, the caller ID service supplies information about a second incoming caller to a subscriber already busy with a phone call. However, note that caller ID on call waiting is not supported by all caller ID-capable telephone displays.

In typical caller ID systems, the coded calling number information is sent from the central exchange to the called telephone. This information can be shown on a display of the subscriber telephone set. In this case, the caller ID information is usually displayed before the subscriber decides to answer the incoming call. If the line is connected to a computer, caller information can be used to search in databases and additional services can be offered.

The following basic caller ID features are supported:

Date and Time

Calling Line Identity

Calling Party Name

Visual Indicator (MWI)

Caller ID Generation

There are two methods used for sending caller ID information depending on the application and country-specific requirements:

caller ID generation using DTMF signalling

caller ID generation using Frequency Shift Keying (FSK)

The displayed caller ID for all countries may be up to 20 digits for numbers and 50 digits for names.

DTMF Signalling

The data transmission using DTMF signalling is performed during or before ringing depending on the country settings or endpoint configuration. The Mediatrix Boss provides the calling line identity according to the following standards:

Europe: ETSI 300 659-1 January 2001 (Annex B): Access and Terminals (AT); Analogue access to the Public Switched Telephone Network (PSTN); Subscriber line protocol over the local loop for display (and related) services; Part 1: On-hook data transmission.

FSK Generation

Different countries use different standards to send caller ID information. The Mediatrix Boss is compatible with the following widely used standards:

ETSI 300 659-1

Continuous phase binary FSK modulation is used for coding that is compatible with:

BELL 202

ITU-T V.23

Note: The Mediatrix Boss does not support ASCII special characters higher than 127.

Note: The compatibility of the Mediatrix Boss is not limited to the above caller ID standards.

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FXS Configuration Software Configuration Guide

FXS Configuration

The FXS Configuration section allows you to define how a FXS endpoint behaves in certain conditions.

To configure the FXS parameters:

1. In the web interface, click the POTS link, then the FXS Config sub-link.

Figure 120: FXS Config Web Page

2. In the FXS Configuration section, set the Line Supervision Mode drop-down menu with the power drop and line polarity used to signal the state of a line.

Power drop and polarity reversal are also called battery drop and battery reversal.

3. Set the Disconnect Delay field with the value used to determine whether or not call clearing occurs as soon as the called user is the first to hang up a received call.

This parameter has no effect when you are acting as the calling party.

If you set the value to 0, the call is disconnected as soon as the called user hangs up the call.

Table 114: Line Supervision Mode Parameters


None Power drop or polarity reversal is not used to signal the state of the line.

DropOnDisconnect Activates the Power Drop on Disconnect feature. A short power drop is made at the end of a call when the call is disconnected by the remote party.

The drop duration can be configured in the FXS Power Drop on Disconnect Duration field (Step 5).

ReversalOnIdle Activates the Polarity Reversal on Idle feature. The polarity of the line is initially in reversed state. The polarity of the line returns to the positive state when the user seizes the line or when the line rings for an incoming call. The polarity of the line is reversed again when the call is disconnected.

ReversalOnEstablished Activates the Polarity Reversal on Established option. The polarity of the line is initially in the positive state. The polarity of the line is reversed when the call is established and returns to the positive state when the call is disconnected.

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Chapter 19 - POTS Configuration FXS Configuration

If the value is greater than 0, that value is the amount of time, in seconds, the unit waits after the called user hangs up before signalling the end of the call.

4. Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is automatically cancelled.

Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring until the party answers.

5. Set the Inband Ringback drop-down menu to define whether or not the FXS endpoint needs to generate a ringback for incoming ringing call.

6. Set the Shutdown Behavior drop-down menu with the FXS endpoint behavior when it becomes shut down.

A FXS endpoint becomes shut down when the operational state of the endpoint becomes Disabled and the Shutdown Endpoint When Operational State is 'Disable' And Its Usage State Is 'idle-unusable' parameter of the SIP > Endpoints page is set to Enable. See “Administration” on page 324 for more details.

This parameter is not used by FXS endpoints used for bypass when the Activation column of the FXS Bypass section is set to Endpoint Disabled. See “FXS Bypass” on page 295 for more details.

7. Set the Power Drop on Disconnect Duration field with the power drop duration, in milliseconds, that is made at the end of a call when the call is disconnected by the remote party.

This value only has an effect when the Line Supervision Mode drop-down menu is set to DropOnDisconnect.

8. Set the Service Activation drop-down menu with the method used by the user to activate supplementary services such as call hold, second call, call waiting, call transfer and conference call.

Table 115: Inband Ringback Parameters


Disable The FXS endpoint does not play local ringback to the remote party.

Enable The FXS endpoint plays local ringback to the remote party via the negotiated media stream. The local ringback is generated only when the telephone is on-hook. The FXS ports never play the local ringback for the call waiting.

Table 116: FXS Shutdown Behavior Parameters


Disabled Tone

A disabled tone is played when the user picks up the telephone and the FXS endpoint is shut down.

Power Drop

The loop current is interrupted when the FXS endpoint is shut down and no tone is played when the user picks up the telephone.

Table 117: Service Activation Parameters


Flash Hook Service activation is performed by flash hook or hanging up.

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FXS Country Customization

The FXS Country Customization section allows you to override the current default country parameters of certain features. Refer to “Appendix A - Country-Specific Parameters” on page 533 for the pre-defined values for a specific country.

To define the FXS country customization parameters:

1. In the FXS Country Configuration section, select whether or not you want to override the current country parameters in the Override Country Customization drop down menu.

This allows overriding FXS related default country settings for the loop current and flash hook detection features.

Figure 121: FXS Country Customization Section

Flash Hook And Digit

Service activation is performed by flash hook, flash hook followed by a digit or hanging up.

The digit dialed has a different behaviour depending on the current call context:

• One call active and one waiting call:

Flash hook then dial the digit 2: Answer the waiting call.

• One call active and one call on hold:

Flash hook then dial the digit 1: Terminate the active call and recover the call on hold.

Flash hook then dial the digit 2: Hold the active call and recover the call on hold.

Flash hook then dial the digit 3: Enter the conference mode.

Flash hook then dial the digit 4: Transfer the call on hold to the active call.

When hanging up in this context, the telephone rings to notify the user there is still a call on hold.

• In conference mode:

Flash hook then dial the digit 2: Return to one active call and one call on hold.

When hanging up in this context, all calls are finished.

Table 117: Service Activation Parameters (Continued)


Table 118: Line Supervision Mode Parameters


Disable The line uses the default country FXS settings.

Enable The line uses the FXS country configuration set in the following steps.

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2. Set the Country Override Loop Current field with the loop current generated by the FXS port in ma.

When a remote end-user goes on-hook, the Mediatrix Boss signals the far end disconnect by performing a current loop drop (< 1 mA) on the analog line. This current loop drop, also referred to as “Power Denial” mode, is typically used for disconnect supervision on analog lines. The Mediatrix Boss maintains a current drop for one second (this value cannot be configured), then a busy tone is generated to indicate the user to hang up. See the description for the FXS Line Supervision Mode drop-down menu in “FXS Configuration” on page 291 for more details.

When one of its analog lines goes off-hook, the Mediatrix Boss controls the endpoint in a fixed loop current mode. When selecting a country (see “Country Configuration” on page 447 for more details), each country has a default loop current value. However, you can override this value and define your own loop current.

3. Note that the actual measured current may be different than the value you set, because it varies depending on the DC impedance. Set the Country Override Flash Hook Detection Range field.

This is the range in which the hook switch must remain pressed to perform a flash hook.

When selecting a country (see “Country Configuration” on page 447 for more details), each country has a default minimum and maximum time value. However, you can override these values and define your own minimum and maximum time within which pressing and releasing the plunger is actually considered a flash hook.

The range consists of the minimal delay and maximal delay, in ms, separated by a “-”. The minimal value allowed is 10 ms and the maximum value allowed is 1200 ms. The space character is not allowed.

Flash hook can be described as quickly depressing and releasing the plunger in or the actual handset-cradle to create a signal indicating a change in the current telephone session. Services such as picking up a call waiting, second call, call on hold, and conference are triggered by the use of the flash hook.

A flash hook is detected when the hook switch is pressed for a shorter time than would be required to be interpreted as a hang-up.

Using the “flash” button that is present on many standard telephone handsets can also trigger a flash hook.


Calling Party Name of the Caller ID


This section describes configuration that is available only in the MIB parameters of the Mediatrix Boss. You can configure these parameters as follows:

by using a MIB browser

by using the CLI

by creating a configuration script containing the configuration variables

You can set the Calling Party Name of the caller ID (CLIP) when the calling party is tagged as private.

To set the Calling Party Name of the caller ID (CLIP):

1. Specify the Calling Party Name of the caller ID (CLIP) when the calling party is tagged as private by putting the following line in a configuration script:

pots.FxsCallerIdPrivateCallingPartyName="Value"

Value may be any string of characters up to 50 characters.

• When empty, no Calling Party Name parameter is sent.

Standards Supported • ETSI EN 300659-3a

a. CLIR section

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• When set to 'P', no Calling Party Name parameter is sent but a Reason for Absence or Caller Party Name parameter is sent with the value 0x50 (Private).

FXS Bypass

The FXS Bypass section allows you to define whether or not the bypass feature is enabled. This feature allows its users to maintain telephone services in the event of a power outage or network failure.

During normal operation, the SCN line connected to the Bypass connector is switched out of the circuit through commuting relays. The Bypass connector can be activated by three different conditions:

When power is removed from the Mediatrix Boss.

When the FXS endpoint’s operational state is disabled and the endpoint is not in use.

When the user signals the configured DTMF map.

If one of these conditions is met, a phone/fax used on a FXS connector that supports the bypass feature is directly connected to the SCN Bypass line. The FXS connector stays in Bypass connection until:

The error conditions have been cleared.

The device connected to it is on-hook and a delay has elapsed.

For more information on your model’s bypass feature and which FXS connectors are available for bypass, please refer to your model’s Hardware Configuration Guide.

To define the FXS bypass parameters:

1. In the FXS Bypass section, specify when the bypass needs to be activated in the corresponding Activation column’s drop-down menu.

Figure 122: FXS Bypass Section


• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3741

Note: For the Mediatrix 3308, Mediatrix 3316, Mediatrix 3731 and Mediatrix 3741 models: If an event that activates the FXS bypass occurs and the FXO port is in use, the bypass activation waits until the FXO port is no longer in use. See “FXO Configuration” on page 296 for more details.

Table 119: Activation Parameters


Power Off The bypass is activated only when the unit power is off or the POTS service is not started.

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2. Set the corresponding Activation DTMF Map field with the DTMFs to signal to enable the bypass.

This field is only used when the corresponding Activation drop-down menu is set to On Demand.

3. Set the corresponding Deactivation Timeout field with the delay, in seconds, to wait before deactivating the bypass after an on hook if the bypass is activated on demand.

The delay is restarted after each on hook. The bypass is not deactivated if the delay expires while the FXS is off hook.

This field is only used when the corresponding Activation drop-down menu is set to On Demand.


FXO Configuration

The FXO Config page allows you to configure gateways-specific parameters.

The FXO port’s use depends on the FXS bypass state (as defined in the FXS Bypass section of the FXS Configuration page – “FXS Bypass” on page 295):

The FXO port becomes disabled as IP-FXO gateway when the FXS bypass is active. In this case:

Endpoint Disabled The bypass is activated when:

• the operational state of the endpoint is Disable, and

• the Shutdown Endpoint When Operational State is 'Disable' And Its Usage State Is 'idle-unusable' parameter of the SIP > Endpoints page is set to Enable. See “Administration” on page 324 for more details.

The bypass is also activated for the same conditions as the ones defined in Power Off.

On Demand The bypass is activated when the user enters the DTMF map configured in the corresponding Activation DTMF Map field (Step 2).

The bypass is deactivated when the user on hooks the phone for at least the number of time configured in the corresponding Deactivation Timeout field (Step 3).

The bypass is also activated for the same conditions as the ones defined in Power Off.

Table 119: Activation Parameters (Continued)


Note: For the Mediatrix 3308, Mediatrix 3316, Mediatrix 3731 and Mediatrix 3741 models, the bypass is automatically disabled after the timeout, whether or not the bypass is in use. Furthermore, if the conditions that activate the bypass are no longer present, the bypass is disabled.


• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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• calls from the IP side to the FXO port are discarded

• calls coming from the SCN side on the FXO port are routed to FXS port #1.

The FXO port becomes enabled and available for calls when the FXS bypass is disabled.

If an event that activates the FXS bypass occurs and the FXO port is in use, the bypass activation waits until the FXO port is no longer in use.

FXO Dialing Configuration

The FXO Dialing Configuration section allows you to set dialing parameters.

To set FXO dialing parameters:

1. In the web interface, click the POTS link, then the FXO Config sub-link.

Figure 123: POTS FXO Config Web Page

2. Set the Pre Dial Delay field with the delay, in milliseconds, between the time the line is successfully seized, or dial tone detected, and the moment the destination phone number is dialed.

Some country specifications include a mandatory pre-dial delay. In that case, the highest value between that and the value set in this field is used.

3. Set the Dial Tone Detection Mode drop-down menu with the proper behaviour.

When dial tone detection is enabled, the unit waits for a dial tone on the FXO line before initiating the dialling sequence. If no dial tone is detected, the line is considered as busy with an incoming call. This mechanism helps avoid collisions between incoming and outgoing calls.

Table 120: Dial Tone Detection Modes


Disable Dial tone detection is disabled.

Country Tone The unit tries to detect the tone specified for this purpose in the current country's tone specification. Some country specifications omit this information. In that case, the unit behaves as if the parameter is set to Disable.

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The following table lists the default dial tone detection frequency and cadence for each supported country.

4. Set the Dial Tone Detection Timeout field with the value, in milliseconds, indicating how long the unit waits for a dial tone before considering the line is busy with an incoming FXO call.


FXO Answering Configuration

The FXO Answering Configuration section allows you to define how the FXO line must behave when answering calls.

To set FXO answering parameters:

1. In the FXO Answering Configuration section, set the corresponding Wait Before Answering Delay column’s FXO interface field with the waiting period, in milliseconds, before answering an incoming FXO call.

If this delay expires before the caller ID signal is decoded, the call proceeds without caller ID information.

If a minimal waiting period is required for the selected country, the highest of both values is used.

Figure 124: FXO Answering Configuration Section

2. Set the corresponding Answering On Caller ID Detection column’s drop-down menu with the proper behaviour.

This parameter enables answering upon caller ID detection instead of the waiting delay configured in Step 1.

When enabled, an incoming FXO call is answered on the first occurrence of either:

• The reception of the caller ID signal.

• The expiration of the delay configured by the Wait Before Answering Delay field and the country wait before answering delay.

Table 121: Default Dial Tone Detection

Country FrequencyCadence

ON (s) OFF (s)

Austria 450 Hz 0.5 0.0

Czech Republic 425 Hz 0.3 0.0

France 440 Hz 0.5 0.0

Germany1/2 425 Hz 0.15 0.0

Italy 425 Hz 0.15 0.0

NorthAmerica1 440 Hz 0.5 0.0

Spain 425 Hz 0.15 0.0

Switzerland 425 Hz 0.5 0.0

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3. Set the corresponding Wait For Callee To Answer column’s drop-down menu with the proper behaviour.

When the endpoint is set up for automatic call (see “Automatic Call” on page 421), enabling this variable makes the endpoint wait until the called the party connection is established before answering the incoming call.


FXO Incoming Call Configuration

This section allows you to define how each line behaves when there is an incoming call.

To set FXO incoming call parameters:

1. In the FXO Incoming Call Behavior section, set the corresponding Not Allowed Behavior column’s drop-down menu with the proper behaviour.

Under certain circumstances (locked port, configuration, etc.), incoming FXO calls are not allowed. When that is the case, the FXO endpoint behaves in one of the manners below.

Figure 125: FXO Incoming Call Behavior Section


FXO Line Verification

The line state verification mechanism allows the detection of defective or down lines based on the absence of current when closing the loop. You can view the line state in the FXO Line State table (“FXO Line Status” on page 288).

To set FXO line verification parameters:

1. In the FXO Line Verification section, set the Link State Verification drop-down menu with the proper behaviour.

Figure 126: FXO Answering Configuration Section

2. Set the Link State Verification Timeout field with the value, in milliseconds, indicating how long the unit waits to successfully take the line before considering the line is defective or down.


Table 122: Not Allowed Behavior Parameters


Do Not Answer The incoming call is left unanswered.

Play Congestion Tone The incoming call is answered, a congestion tone is played for 10 seconds, and then the call is terminated.

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FXO Force End of Call

FXO Force end of call, also known as Far End Disconnect, refers to methods for detecting that a remote party has hung up. If the Far End Disconnect signal is not sent to or properly detected by the Mediatrix Boss, the connection will not be released by the unit, thus freezing the FXO line in the off hook state.

The FXO Force End of Call section allows you to define various methods to detect a far end disconnect.

To set FXO force end of call parameters:

1. In the FXO Force End of Call section, set the Force End Of Call On Call Failure drop-down menu with the proper behaviour.

This parameter enables or disables forced-end-of-call on call failure.

The forced end of call on call failure occurs when a SCN caller tries to reach someone on the IP network and the SCN caller hangs up before reaching the IP callee.

If the connection is not established, the Mediatrix Boss cannot detect that the caller has hung up. You can configure the Mediatrix Boss to disconnect an unsuccessful communication after a specific number of seconds. This number of seconds is defined in a timeout that starts to count when the SCN caller contacts the Mediatrix Boss. If the connection is not successful, the line is closed when the call time reaches the defined timeout value.

This feature forcefully terminates a call that stayed in an error state for some time. When the line falls in an error state where a SIT, a ROH, a BUSY or any error tone is played outbound to the FXO line, the unit waits for the timeout specified in the Call Failure Timeout field and then hangs up.

Figure 127: FXO Force End of Call Section

2. Set the Call Failure Timeout field with the waiting period, in seconds, before terminating a call in an error state.

3. Set the Forced End Of Call On Silence Detection Mode drop-down menu with the proper behaviour.

This parameter enables or disables forced-end-of-call on silence detection.

The silence detection feature applies when two parties are in communication and one of them hangs up. It allows the Mediatrix Boss to close a line when no voice activity or silence is detected for a specified amount of time.

When silence is detected on the IP and/or SCN side for an amount of time specified in a timeout, the call is terminated. This feature is useful to free resources in the event of an IP network failure that prevents the end of call to be detected or when the SCN end of call tone was not detected.

The current implementation of silence detection relies on the power of the media signal. A silence is detected if the power level of the media signal is lower than -60 dBm.

This feature forcefully terminates a call that stayed silent for some time. When silence is detected on the inbound and/or outbound media for an amount of time specified in the Silence Detection Timeout field, the call is terminated.

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Note: The silence detection feature could inadvertently disconnect a communication when one party puts the other on hold. Using the hold tone can prevent detection of silence when the call is put on hold by the IP peer. See “Tone Override” on page 438 for more details.

Table 123: Forced End of Call on Silence Detection Mode Parameters


Disable Forced-end-of-call on silence detection is disabled.

300 Mediatrix Boss


This feature is useful to free resources in the event of a network failure preventing the end-of-call to be detected or when the FXO end-of-call tone was not detected.

4. Set the Silence Detection Timeout field with the maximum amount of time, in seconds, that a call can remain silent, before it is terminated and the line is released.

5. Set the Forced End Of Call On Tone Detection Mode drop-down menu with the proper behaviour.

This parameter enables or disables forced-end-of-call upon tone detection. It terminates a call upon detection of an end-of-call tone on the inbound from the FXO line.

The Mediatrix Boss can monitor special tones that indicate the remote SCN user has hung up. You can use the default value for a selected country or customize a tone. Table 125 lists the default frequency and cadence detected by supported country.

When customizing a tone, Media5 suggests to ask your Central Office about the tone it generates to indicate a call has been disconnected. You will thus be able to customize your tone according to this information.

6. If you have selected Custom Tone in Step 5, set the Tone Detection Custom Frequency field with the Frequency, in Hertz (Hz), to detect in the custom cadence.

You can set any value between 350 Hz and 620 Hz.

A customized tone detection can only detect a single frequency. To detect tones made of multiple frequencies, create the cadence for only one of the frequencies found in the tone.

Inbound And Outbound Silent The call is terminated if both inbound and outbound media are silent at the same time.

Table 124:


Disable Force-end-of-call upon tone detection is disabled.

Country Tone

The unit tries to detect the tone specified for this purpose in the current country's tone specification. Some country specifications omit this information. In that case, the unit behaves as if the parameter is Disable.

Custom Tone Terminates a call upon detection of a custom tone. See Steps 6-8.

Table 125: Default Frequency and Cadence Supported

Country FrequencyCadence

ON1 (s) OFF1 (s) ON2 (s) OFF2 (s) Repetition

Austria 450 Hz 8.0 0.0 0.0 0.0 1

Czech Republic 425 Hz 8.0 0.0 0.0 0.0 1

France 440 Hz 8.0 0.0 0.0 0.0 1

Germany 425 Hz 8.0 0.0 0.0 0.0 1

Italy 425 Hz 0.2 0.2 0.6 1.0 4

North America 440 Hz 8.0 0.0 0.0 0.0 1

Spain 425 Hz 8.0 0.0 0.0 0.0 1

Switzerland 425 Hz 8.0 0.0 0.0 0.0 1

Custom Tone Settings

Table 123: Forced End of Call on Silence Detection Mode Parameters (Continued)


Mediatrix Boss 301


7. Set the Tone Detection Custom Cadence field with the cadence to detect.

A cadence is a series of frequencies that are played for a specified time, making up a tone. The format for a cadence is:

on1,off1,on2,off2,on3,off3

In this string, “on” and “off” are numerical values representing the time, in milliseconds, that the frequency can and cannot be detected, respectively. For instance, “2000, 1000, 2000, 0” is a cadence in which the frequency plays for 2 seconds, stops for 1 second, and plays for 2 more seconds. This example is also equivalent to setting the string “2000, 1000, 2000”.

You can specify up to three “on,off” pairs. If you specify less than those six values, “0” values will be added as necessary. Specifying more than six will only use the six first values.

A cadence starting with a value of zero (0) is invalid. The first zero (0) found in the string signals the end of the cadence (i.e. “200, 0, 300” is the same as “200”).

• To detect a continuous tone, use a single “on” value, e.g., “200, 0” or “200” (the off time is 0 ms.). A continuous tone of 200 ms is used if the field is empty.

• To detect a tone in which two or more frequencies are used, for instance with a cadence of “200 ms, 500 ms, 300 ms, 400 ms”, in which the respective frequencies would be “400 Hz, 300 Hz, 400 Hz, 500 Hz”, detect a frequency that comes twice or more in the tone. In the above example, detect the 400 Hz frequency by using the cadence “200 ms, 400 ms, 300 ms, 500 ms”. In this example, the 400 ms and 500 ms off times represent the times that the 400 Hz frequency cannot be heard, even though another frequency may be playing.

The “on” and “off” values can be from 0 to 32767 ms.

8. Set the Detection Custom Repetition field with the number of times the custom cadence must be detected to consider the custom end-of-call tone has been detected.


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20 SIP Gateways

This chapter describes how to add and remove SIP gateways in the Mediatrix Boss.

SIP Gateways Configuration

Multiple SIP gateways may be used for a number of reasons, such as:

Redirecting ISDN calls to different SIP servers depending on the call.

Hunt calls across several gateways.

Adding a SIP gateway triggers a warning message if the total number of registrations configured reached the defined limit. See “Number of Registrations” on page 319 for more details.

To configure multiple SIP gateways:

1. In the web interface, click the SIP link, then the Gateways sub-link.

Figure 128: SIP – Gateways Web Page

You can add a new gateway by clicking the button. The Mediatrix Boss supports a maximum of 5 gateways.

You can delete an existing gateway by clicking the button.

2. If you are adding a new gateway, enter its name in the Name field.

The Mediatrix Boss supports only alphanumeric characters, “-”, and “_”.

3. Select the network interface on which the gateway listens for incoming SIP traffic in the Network Interface drop-down menu.

This value applies to all transports (e.g., UDP, TCP, etc.).

The LAN interface may be used as a SIP gateway to be bound on the LAN. However, there is no routing between the LAN and the uplink interface.

4. Set the SIP port on which the gateway listens for incoming unsecure SIP traffic in the Port field.

This is used only when the UDP and/or TCP transports are enabled.

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If two or more SIP gateways use the same port, only the first SIP gateway starts correctly. The others are in error and not started. The SIP gateway is also in error and not started if the port is already used.

The default value is 0. If you set the port to 0, the default SIP port 5060 is used.

5. Set the SIP port on which the gateway listens for incoming secure SIP traffic in the Secure Port field.

This is used only when the TLS transport is enabled.

The default value is 0. If you set the port to 0, the default secure SIP port 5061 is used.


The state of the SIP gateways is displayed in the SIP Gateway Status section.

Note: The port “0” is the equivalent to the “well known port”, which is 5060 in SIP. Using 0 and 5060 is not the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If you set the port to 5060, it will be present in the SIP packet. For example: “[email protected]” if the port is 0 and “[email protected]:5060” if the port is 5060.

Note: The port “0” is the equivalent to the “well known port”, which is 5061 in SIP for TLS. Using 0 and 5061 is not the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If you set the port to 5061, it will be present in the SIP packet. For example: “[email protected]” if the port is 0 and “[email protected]:5061” if the port is 5061.

Table 126: SIP Gateway States

State Description

Ready The gateway is ready to make and receive calls.

Cannot start, port already in use

The gateway cannot open its IP port because the port is already used by another service. This generally occurs when the administrator adds a new gateway but forgets to configure a different IP port.

Network down The SIP gateway is not started or the network interface on which the SIP gateway is associated does not have an IP address.

Restarting The SIP gateway cannot make or receive calls while it is restarting.

Waiting for time synchronization

The gateway is started but it cannot open its SIP TLS port because the real-time clock is not synchronized. This generally occurs when the SNTP server is not set or is unreachable.

Server unreachable The gateway is started but it cannot make and receive calls because the SIP server is unreachable. This state is only reported when a KeepAlive mechanism is used.

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21 SIP Servers

This chapter describes how to configure the SIP server parameters of the Mediatrix Boss.

It describes the following:

How to define the SIP servers IP information.

How to define the SIP gateways IP information.

Introduction

The Mediatrix Boss uses the following types of servers:

Standards Supported • RFC 2543: SIP: Session Initiation Protocol

• RFC 3261: The Session Initiation Protocol (SIP)

• RFC 3903: Session Initiation Protocol (SIP) Extension for Event State Publication

Table 127: SIP Servers

Server Description

Registrar Server Accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles.

Proxy Server An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is passed on to another entity that can further process the request. Proxies are also useful for enforcing policy and for firewall traversal. A proxy interprets, and, if necessary, rewrites parts of a request message before forwarding it.

Outbound Proxy Server An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. The outbound proxy receives all outbound traffic and forwards it. Incoming traffic may or may not go through the outbound proxy. The outbound proxy’s address is never used in the SIP packets, it is only used as a physical network destination for the packets.

When the outbound proxy is enabled, the proxy is still used to create the To and From headers, but the packets are physically sent to the outbound proxy.

SIP Outbound Proxy (From RFC 3261)

A proxy that receives requests from a client, even though it may not be the server resolved by the Request-URI. Typically, a user agent is manually configured with an outbound proxy.

When enabled, the initial route for all SIP requests contains the outbound proxy address, suffixed with the loose routing parameter “lr”. The Request-URI still contains the home domain proxy address. Requests are directed to the first route (the outbound proxy).

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Chapter 21 - SIP Servers TLS Persistent Connections Status

TLS Persistent Connections Status

The TLS Persistent Connections Status table allows you to browse the status of the TLS persistent connections of the Mediatrix Boss. These connections are associated with the SIP servers (outbound proxy, registrar and home domain proxy). Note that this section is not displayed if there is no information to show.

Figure 129: SIP – TLS Persistent Connections Status Section

The following information is available:

Table 128: TLS Persistent Connection Parameters


Gateway The SIP gateway used to register.

Local Port Local port used by the TLS persistent connection.

Remote Host The remote host used to establish the TLS persistent connection. The remote host can be a host name or an IP address of the proxy, outbound proxy or registrar.

Remote IP Address The resolved IP address of the remote host used to establish the TLS persistent connection.

Status The current state of the TLS persistent connection.

• Up: The TLS connection is established.

• Down: The TLS connection is not established.

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SIP Servers Configuration

This section describes how to configure the IP address and port number of the SIP servers.

If any of the SIP servers parameters corresponds to a domain name that is bound to a SRV record, the corresponding port must be set to 0 for the unit to perform DNS requests of type SRV (as per RFC 3263). Otherwise, the unit will not use DNS SRV requests, but will rather use only requests of type A because it does not need to be specified which port to use.

To set the SIP servers configuration:

1. In the web interface, click the SIP link, then the Configuration sub-link.

Figure 130: SIP – Servers Web Page

2. Enter the SIP registrar server static IP address or domain name and port number in the Registrar Host field.

You must enter the information as IP address:Port number. For instance:

192.168.0.5:5060

3. Enter the SIP Proxy server static IP address or domain name and port number in the Proxy Host field.


192.168.0.5:5060

4. Enter the SIP outbound proxy server static IP address or domain name and port number in the Outbound Proxy Host field.

The outbound proxy is enabled if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to 0.0.0.0:0 or leaving the field empty disables the outbound proxy.


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Chapter 21 - SIP Servers Multiple SIP Gateways

Multiple SIP Gateways

The Mediatrix Boss allows you to have multiple SIP gateways (interfaces). You can configure each SIP gateway to register to a specific registrar. You can also configure each SIP gateway to send all requests to an outbound proxy. See “Chapter 20 - SIP Gateways” on page 305 for more details.

SIP Gateway Specific Registrar Servers

This section allows you to define whether the available SIP gateways use the default registrar server or rather use a specific registrar server.

To set specific registrars servers information:

1. In the SIP Gateway Specific Registrar Servers section of the Servers page, select whether or not a SIP gateway uses a specific registrar server in the Gateway Specific drop-down menu.

If you select No, the SIP gateway uses the server information as set in the SIP Default Servers section.

Figure 131: SIP Servers – Specific Registrar Section

2. Enter the IP address or domain name and port number of the registrar server currently used by the registration in the Registrar Host field.


192.168.0.5:5060

3. If you do not need to set other parameters, do one of the following:

• To save your settings without refreshing the registration, click Submit.

• To save your settings and refresh the registration now, click Submit & Refresh Registration.

SIP Gateway Specific Messaging Servers

This section allows you to define whether the available SIP gateways use the default proxy and outbound proxy server or rather use specific servers.

To set specific proxy servers information:

1. In the SIP Gateway Specific Messaging Servers section of the Servers page, select whether or not a SIP gateway uses a specific proxy and outbound proxy server in the Gateway Specific drop-down menu.


Figure 132: SIP Servers – Specific Proxy Section

2. Enter the IP address or domain name and port number of the messaging server currently used by the registration in the Proxy Host field.


192.168.0.5:5060

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3. Enter the IP address or domain name and port number of the outbound proxy server currently used by the registration in the Outbound Proxy Host field.


192.168.0.5:5060





SIP Gateway Specific Proxy Servers

This section allows you to define whether the available SIP gateways use the default proxy and outbound proxy server or rather use specific servers.

To set specific proxy servers information:

1. In the SIP Gateway Specific Proxy Servers section of the Servers page, select whether or not a SIP gateway uses a specific proxy and outbound proxy server in the Gateway Specific drop-down menu.


Figure 133: SIP Servers – Specific Proxy Section

2. Enter the IP address or domain name and port number of the proxy server currently used by the registration in the Proxy Host field.


192.168.0.5:5060

3. Enter the IP address or domain name and port number of the outbound proxy server currently used by the registration in the Outbound Proxy Host field.


192.168.0.5:5060





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Outbound Proxy Loose Router Configuration


You can specify the type of routing of the outbound proxy configured in “SIP Servers Configuration” on page 309.

You can use two types of configuration:

Default configurations that apply to all the endpoints of the Mediatrix Boss.

Specific configurations that override the default configurations.

You can define specific configurations for each endpoint in your Mediatrix Boss. For instance, you could enable a codec for all the endpoints of the Mediatrix Boss and use the specific configuration parameters to disable this same codec on one specific endpoint.

Using one or more specific parameter usually requires that you enable an override variable and set the specific configuration you want to apply.

The following types are available:

Standards Supported • RFC 3261: SIP: Session Initiation Protocol, section 6

• RFC 2543: SIP: Session Initiation Protocol

Table 129: Outbound Proxy Router Status

Type Description

LooseRouter This is the most current method for SIP routing, as per RFC 3261, and will become the standard behaviour once RFC 3261 compliance is achieved. See “Introduction” on page 307 for details.

StrictRouter Pre-RFC 3261, RFC 2543 compatible SIP routing.

The initial route for all SIP requests contains the home domain proxy address (the Request-URI). Requests are directed to the outbound proxy.

In other words, the Request-URI is constructed as usual, using the home domain proxy and the user name, but is used in the route set. The Request-URI is filled with the outbound proxy address.

NoRouteHeader

Removes the route header from all SIP packets sent to an outbound proxy. This does not modify persistent TLS connection headers.

Note: The Router header will not be removed from the SIP packets if the unit is configured to use the TLS Fallback feature. This feature requires the information of the SIP Outbound Proxy in the SIP packet to work correctly.

Loose Router

A proxy is said to be loose routing if it follows the procedures defined in the RFC 3261 specification (section 6) for processing of the Route header field. These procedures separate the destination of the request (present in the Request-URI) from the set of proxies that need to be visited along the way (present in the Route header field). A proxy compliant to these mechanisms is also known as a loose router.

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To set the outbound proxy router status:

1. Select one of the available routing types by putting the following line in the configuration script:sipEp.defaultProxyOutboundType="Value"

where Value may be one of the following:

2. If you want to set a different routing type for one or more SIP gateways, put the following lines in the configuration script:

sipEp.gwSpecificProxy.EnableConfig[GatewayName="default"]="1"

sipEp.gwSpecificProxy.OutboundType[GatewayName="Specific_Gateway"]="Value"

where:

• Specific_Gateway is the name of the SIP gateway you want to configure.

• Value is the refresh router status as defined in Step 1.

Table 130: Outbound Proxy Router Values

Value Meaning

100 LooseRouter

200 StrictRouter

300 NoRouteHeader

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22 SIP Registration

This chapter describes how to configure the registration parameters of the Mediatrix Boss.

Endpoints Registration

Each endpoint of the Mediatrix Boss has its own registration information. You can set information for each endpoint such as its telephone number and friendly name.

Adding an endpoint registration triggers a warning message if the total number of registrations configured reached the defined limit. See “Number of Registrations” on page 319 for more details.

To set endpoints registration information:

1. In the web interface, click the SIP link, then the Registrations sub-link.

Figure 134: SIP – Registrations Web Page


• RFC 3261: The Session Initiation Protocol (SIP)

• RFC 3863: Presence Information Data Format (PIDF)

• RFC 3903: Session Initiation Protocol (SIP) Extension for Event State Publication

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2. In the Endpoints Registration section of the Registrations page, enter a user name for each endpoint in the User Name column.

The user name (such as a telephone number) uniquely identifies this endpoint in the domain. It is used to create the Contact and From headers. The From header carries the permanent location (IP address, home domain) where the endpoint is located. The Contact header carries the current location (IP address) where the endpoint can be reached.

Contacts are registered to the registrar. This enables callers to be redirected to the endpoint’s current location.

3. Enter another name for each endpoint in the Friendly Name column.

This is a friendly name for the endpoint. It contains a descriptive version of the URI and is intended to be displayed to a user interface.

4. Define whether or not the endpoint registration needs to register to the registrar in the Register column.

An endpoint configured to register (set to Enable) will become unavailable for calls from or to SIP when not registered.

You can define the behaviour of an endpoint when it becomes unavailable in the defaultRegistrationUnregisteredBehavior MIB variable.

5. Define whether or not the endpoint needs to subscribe to a messaging system in the Messaging drop-down menu.

The current state of the subscription is displayed in the Endpoints Messaging Subscription Status table.

You can enter the address of the Messaging server in “SIP Servers Configuration” on page 309.

6. Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down menu.

You must have SIP gateways already defined. See “Chapter 20 - SIP Gateways” on page 305 for more details. If you select all, the configuration applies to all gateways available.

Table 131: MWI Subscription State

State Description

Unsubscribed The unit/endpoint is not subscribed and never tries to subscribe. This case occurs if the network interface used by the SIP gateway is not up or the unit/ endpoint is locked.

Subscribing The subscription is currently trying to subscribe.

Subscribed The subscription is successfully subscribed.

Refreshing The subscription is trying to refresh.

Unreachable The last subscription attempt failed because the messaging server is unreachable.

AuthFailed The last subscription attempt failed because authentication was not successful.

Rejected The last subscription attempt failed because the messaging server rejects the subscription.

ConfigError The last subscription attempt failed because it was badly configured. Check if the username and the messaging host are not empty.

InvalidResponse The received 200 OK response contact does not match the contact of the messaging server, or the 200 OK response for an unsubscribe contains a contact.

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• To save your settings and refresh the registration now, click Submit & Refresh.

Unit Registration

Unit registration is used to register a contact not directly related to endpoints. This is generally used to indicate to a registrar the IP location of the Mediatrix Boss when it is used as a gateway.

Adding a unit registration triggers a warning message if the total number of registrations configured reached the defined limit. See “Number of Registrations” on page 319 for more details.

To set unit registration information:

1. In the Unit Registration section of the Registrations page, enter a user name in the User Name column.

Figure 135: SIP Registrations – Unit Registration Section

The user name (such as a telephone number) uniquely identifies this user in the domain.

You can add a new user by clicking the button.

You can delete an existing user by clicking the button.

2. Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down menu.

You must have SIP gateways already defined. See “Chapter 20 - SIP Gateways” on page 305 for more details. If you select all, the configuration applies to all gateways available.




Registration Configuration

This section allows you to define registration refresh parameters.

See “Registration Refresh Parameters” on page 320 for more registration parameters.

To set the registration configuration:

1. In the Registration Configuration section of the Registrations page, set the Default Registration Refresh Time field with the time, in seconds, at which a registered unit begins updating its registration before the registration expiration.

Figure 136: SIP Registrations – Registration Configuration Section

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In SIP, a registration is valid for a period of time defined by the registrar. Once a unit is registered, the SIP protocol requires the User Agent to refresh this registration before the registration expires. Typically, this re-registration must be completed before the ongoing registration expires, so that the User Agent's registration state does not change (i.e., remains 'registered').

For instance, if the parameter is set to 43 and the registration lasts one hour, the unit will send new REGISTER requests 59 minutes and 17 seconds after receiving the registration acknowledgement (43 seconds before the unit becomes unregistered).

Setting this parameter to 0 means that the User Agent will fall into the 'unregistered' state BEFORE sending the re-REGISTER requests.

This value MUST be lower than the value of the "expires" of the contact in the 200 OK response to the REGISTER, otherwise the unit rapidly sends REGISTER requests continuously.

You can also set a different registration refresh time for one or more SIP gateways by using the MIB parameters of the Mediatrix Boss. See “Registration Refresh” on page 320 for more details.

2. Set the Proposed Expiration Value In Registration field with the suggested expiration delay, in seconds, of a contact in the REGISTER request.

The SIP protocol allows an entity to specify the “expires” parameter of a contact in a REGISTER request. The server can return this “expires” parameter in the 200 OK response or select another “expires”. In the REGISTER request, the “expires” is a suggestion the entity makes.

The “expires” parameter indicates how long, in seconds, the user agent would like the binding to be valid.

Available values are from 1 s to 86,400 s (one day).

This value does not modify the delay before a re-REGISTER.

• The delay is the “expires” of the contact in the 200 OK response to the REGISTER request minus the value set in the Default Registration Refresh Time field.

• If the “expires” of the contact in the 200 OK response to the REGISTER is not present or not properly formatted, then the delay is the default registration proposed expiration value minus the value set in the Default Registration Refresh Time field.

Setting the parameter to 0 disables the expiration suggestion.

You can also set a different expiration delay for one or more SIP gateways by using the MIB parameters of the Mediatrix Boss. See “Registration Expiration” on page 320 for more details.




Note: Normally, the Mediatrix Boss cannot make or receive calls until the REGISTER has completed successfully. Because the timeout for a SIP transaction in UDP is 32 seconds, it is possible to have an ongoing re-REGISTER transaction at the same moment that the registration itself expires. This could happen if the Default Registration Refresh Time field is set to a value lower than 32.

In that case, the user agent becomes unregistered, and will become registered again only when the re-REGISTER request is answered with a positive response from the server. See “Unregistered Endpoint Behaviour” on page 321 for a workaround if the unit cannot make calls during that period.

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Number of Registrations

The Mediatrix Boss limits the total number of registrations to 64. The total number of registrations is the sum of all the endpoints and gateways (“SIP Gateways Configuration” on page 305) pairs. The Mediatrix Boss supports a maximum of 5 gateways. An endpoint configured with "All" gateways generates as many pairs as the number of gateways. In a setup with 3 gateways, one endpoint configured with "All" as the gateway name counts for 3 in the total number of registrations.

The registrations are enabled gateway by gateway until the limit is reached. Endpoints Registrations are used first, then Unit Registrations. The remaining registrations are not registered and do not appear in the status table. If you click the Submit And Refresh button and the configured number of registrations exceeds the defined limit, a warning is displayed on the web interface and a syslog notify (Level Error) is sent.

Adding a gateway or an endpoint triggers a warning message if the total number of registrations configured reached the defined limit.

Let’s suppose for instance that we have the current SIP Gateways configuration and the following SIP Registration configuration:

Figure 137: Example, Gateway Configuration

Figure 138: Example, Registrations Configuration

The following table describes how to compute the total number of registrations for this example:

Table 132: Number of Registrations Example

Parameter Setting Nb of Registrations

Endpoint Registration 1 in Figure 138

Gateway Name set to alla

a. When the Gateway Name is set to all, this must be multiplied by the number of gateways set in Figure 137. In this example, there are 3 gateways set.

3

Endpoint Registration 2 in Figure 138

Gateway Name set to gw2 1

Unit Registration 1 in Figure 138 Gateway Name set to all 3

Unit Registration 2 in Figure 138 Gateway Name set to all 3

Unit Registration 3 in Figure 138 Gateway Name set to gw1 1

Unit Registration 4 in Figure 138 Gateway Name set to default 1

Total Number of registrations 12

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Registration Refresh Parameters


Default vs. Specific Configurations






Registration Refresh

You can set the default registration refresh time in the web page (“Registration Configuration” on page 317), but you can also set a different registration refresh time for one or more SIP gateways.

To set registration refresh parameters:

4. If you want to set a different registration refresh time for one or more SIP gateways, put the following lines in the configuration script:

sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"sipEp.gwSpecificRegistration.RefreshTime[GatewayName="Specific_Gateway"]="Value"

where:


• Value is the refresh time value.

Registration Expiration

You can set the default registration proposed expiration value in the web page (“Registration Configuration” on page 317), but you can also set a different registration refresh time for one or more SIP gateways.

The SIP protocol allows an entity to specify the “expires” parameter of a contact in a REGISTER request. The server can return this “expires” parameter in the 200 OK response or select another “expires”. In the REGISTER request, the “expires” is a suggestion the entity makes.

The “expires” parameter indicates how long, in seconds, the user agent would like the binding to be valid.

You can configure the “expires” parameter the Mediatrix Boss sends.

To configure the registration expiration:

5. If you want to set a different registration refresh time for one or more SIP gateways, put the following lines in the configuration script:

sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"

sipEp.gwSpecificRegistration.ProposedExpirationValue[GatewayName="Specific_Gateway"]="Value"

where:


• Value is the expiration delay value.

This value does not modify the time before a re-REGISTER.

• The delay is the “expires” of the contact in the 200 OK response to the REGISTER request minus the value set in the gwSpecificRegistrationRefreshTime parameter.

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• If the “expires” of the contact in the 200 OK response to the REGISTER is not present or not properly formatted, then the delay is the default registration expires minus the value set in the gwSpecificRegistrationRefreshTime parameter.

Unregistered Endpoint Behaviour

You can specify whether an endpoint should remain enabled or not when not registered. This is useful if you want your users to be able to make calls even if the endpoint is not registered with a SIP server.

The following values are supported:

To specify unregistered endpoint behaviour:

1. Set the unregistered endpoint behaviour by putting the following line in the configuration script:

sipEp.defaultRegistrationUnregisteredBehavior="Value"

where Value may be as follows:.

2. If you want to set a different behaviour for one or more SIP gateways, put the following lines in the configuration script:

sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"sipEp.gwSpecificRegistration.UnregisteredBehavior[GatewayName="Specific_Gateway"

]="Value"

where:


• Value is one of the values described in Step 1.

Table 133: Unregistered Endpoint Behaviour Parameters

Value Description

disablePort When the endpoint is not registered, it is disabled. The user cannot make or receive calls. Picking up the handset yields a fast busy tone, and incoming INVITEs receive a “403 Forbidden” response.

enablePort When the endpoint is not registered, it is still enabled. The user can receive and initiate outgoing calls. Note that because the endpoint is not registered with a registrar, its public address is not available to the outside world; the endpoint will most likely be unreachable except through direct IP calling.

Table 134: Unregistered Endpoint Behaviour Values

Value Meaning

0 disablePort

1 enablePort

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SIP User Agent Header


The User-Agent header field contains information about the user agent client originating the request. For instance, the information of the User-Agent header could be something like the following:

User-Agent: Softphone Beta1.5

You can specify whether or not the Mediatrix Boss sends this information when establishing a communication.

To enable sending the SIP User Agent header:

1. Enable sending the SIP User Agent header by putting the following line in the configuration script:

sipEp.interopSendUaHeaderEnable="1"

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23 Endpoints State Configuration

This chapter describes how to set the administrative state of the Mediatrix Boss’s endpoints.

Endpoints Configuration

The endpoints configuration allows you to define the administrative state of the Mediatrix Boss’s endpoints.

To set the endpoints parameters:

1. In the web interface, click the SIP link, then the Endpoints sub-link.

Figure 139: SIP Configuration – Endpoints Web Page

2. In the Configuration section, select the permanent administrative state each endpoint will have when the Mediatrix Boss restarts in the Initial Administrative State column.

Table 135: Permanent Administrative State Settings

Setting Description

Unlocked Registers the endpoint to the SIP server.

Locked The endpoint is unavailable for normal operation. It cannot be used to make and/or receive calls.

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3. Select a temporary state for each endpoint in the corresponding Action column.

This command locks/unlocks an endpoint of the Mediatrix Boss. This state is kept until you modify it or the unit restarts. It offers the following settings:




The current endpoints configuration is displayed in the Status section.

Administration

The Administration section allows you to define endpoint operational state.

To set administration parameters:

1. In the Administration section of the Endpoints page, set the Disable Unit (All Endpoints) When No Gateways Are In State Ready drop-down menu with the proper behaviour.

Figure 140: Administration Section

2. Set the Shutdown Endpoint When Operational State is Disable And Its Usage State Is 'idle-unusable' drop-down menu with the proper behaviour.

Table 136: Action Settings

Setting Description

Force Lock Cancels the endpoint registration to the SIP server. All active calls in progress are terminated immediately. No new calls may be initiated.

Lock Cancels the endpoint registration to the SIP server. Active calls in progress remain established until normal call termination. No new calls may be initiated.

Unlock Registers the endpoint to the SIP server.

Table 137: Unit Operational State Parameters


Disable Signaling gateways have no impact on the unit operational state

Enable When all signaling gateways are not ready, the unit operational state is set to disabled.

Table 138: Endpoint Shutdown Parameters


Enable When the usage state becomes “Idle-unusable” and the operational state becomes “Disable”, the endpoint is physically shutdown.

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Session Timers


The session timer extension allows detecting the premature end of a call caused by a network problem or a peer’s failure by resending an INVITE at every n seconds.

A successful response (200 OK) to this INVITE indicates that the peer is still alive and reachable. A timeout to this INVITE may mean that there are problems in the signalling path or that the peer is no longer available. In that case, the call is shut down by using normal SIP means.

Session timers apply to the whole unit.

To set Session Timer information:

1. Enable the session timers by putting the following line in the configuration script:sipEp.defaultSessionTimerEnable="1"

2. Set the session timer minimum expiration delay, in seconds, by putting the following line in the configuration script:

sipEp.defaultSessionTimerMinimumExpirationDelay="Value"

where Value is the minimum expiration delay.

This is the minimum value, in seconds, for the periodical session refreshes. It must be equal to or smaller than the maximum value. This value is reflected in the Min-SE header.

The Min-SE value is a threshold under which proxies and user agents on the signalling path are not allowed to go. Increasing the minimum helps to reduce network traffic, but also makes “dead” calls longer to detect.

3. Set the session timer maximum expiration delay, in seconds, by putting the following line in the configuration script:

sipEp.defaultSessionTimerMaximumExpirationDelay="Value"

where Value is the maximum expiration delay.

This is the suggested maximum time, in seconds, for the periodical session refreshes. It must be equal to or greater than the minimum value. This value is reflected in the Session-Expires header.

Increasing the maximum helps to reduce network traffic, but also makes “dead” calls longer to detect.

To disable the Session Timer service:

1. Disable the session timers by putting the following line in the configuration script:sipEp.defaultSessionTimerEnable="0"

Disable When an endpoint's usage state becomes “Idle-unusable” whatever the value of its operational state, the endpoint remains physically up but the calls are denied.

Table 138: Endpoint Shutdown Parameters (Continued)


Note: When the defaultSessionTimerMaximumExpirationDelay value is lower than the defaultSessionTimerMinimumExpirationDelay value, the minimum and maximum expiration delay values in INVITE packets are the same as the value set in the defaultSessionTimerMinimumExpirationDelay variable.

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Background Information

The following explains how the session timers are used.

SDP in Session Timer reINVITEs

The reINVITE is sent with the last SDP that was negotiated. Receiving a session timer reINVITE should not modify the connection characteristics.

Relation Between Minimum and Maximum Values

A user agent that receives a Session-Expires header whose value is smaller than the minimum it is willing to accept replies a “422 Timer too low” to the INVITE and terminates the call. The phone does not ring.

It is up to the caller to decide what to do when it receives a 422 to its INVITE. The Mediatrix Boss will automatically retry the INVITE, with a Session-Expires value equal to the minimum value that the user agent server was ready to accept (located in the Min-SE header). This means that the maximum value as set in the Mediatrix Boss might not be followed. This has the advantageous effect of establishing the call even if the two endpoints have conflicting values. The Mediatrix Boss will also keep retrying as long as it gets 422 answers with different Min-SE values.

Who Refreshes the Session?

Re-sending a session timer INVITE is referred to as refreshing the session. Normally, the user agent server that receives the INVITE has the last word on who refreshes. The Mediatrix Boss always lets the user agent client (caller) perform the refreshes if the caller supports session timers. In the case where the caller does not support session timers, the Mediatrix Boss assumes the role of the refresher.

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24 SIP Authentication

This chapter describes how to configure authentication parameters of the Mediatrix Boss.

Authentication Configuration

Authentication information allows you to add some level of security to the Mediatrix Boss endpoints by setting user names and passwords.

You can add four types of authentication information:

The Authentication table may have between 20 and 64 rows. Each of these rows can either be associated with the unit, a specific gateway, a specific endpoint, or a specific user name. If you have less than 20 rows, the Mediatrix Boss automatically adds new rows up to the minimum of 20.

When a challenge occurs (either 401 or 407), the first entry in the Authentication table that matches the user name/password request is used to reply to the challenge. You can configure the use name and password in the web interface. The order of the tried entries in the Authentication table is from the first row to the last row.

The challenge matches an authentication entry if the realm of the challenge matches the realm specified in the Realm field or if the Validate Realm field is set to disable. For each entry matching certain criteria (described below), the challenge is replied with the entry's user name and password. If no entry matches the criteria, the authentication fails. To match the authentication request, the entry must also meet one of the following criteria:

The challenge needs to be for a SIP request related to the endpoint specified in the Endpoint column if the corresponding Apply To column is set to Endpoint.

The challenge needs to be for a SIP request performed on the SIP gateway specified in the Gateway column if the corresponding Apply To column is set to Gateway.

The challenge needs to be for a context that uses the user name specified in the User Name field if the corresponding Apply To column is set to Usename. The user name associated with a context is:

• the user name of the FROM if the context sent the original SIP request, or

Standards Supported • Basic and Digest authentication as per RFC 3261

Table 139: Authentication Information

Authentication Description

endpoint-specific Applies only to challenges received for SIP requests related to a specific endpoint. For instance, the registration associated with the endpoint in the user agent table or the INVITE sent to initiate a call from the endpoint. You can define several user names and passwords for each endpoint of the Mediatrix Boss. An endpoint can thus register with several different realms.

gateway-specific Applies only to challenges received for SIP requests on a specific SIP gateway. You can define several user names and passwords for each endpoint of the Mediatrix Boss. An endpoint can thus register with several different realms.

unit Applies to all challenges received for SIP dialog. You can define several user names and passwords for the Mediatrix Boss. These user names and passwords apply to all endpoints of the unit.

user name-specific

Applies only to challenges for a context that uses a specific user name.

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• the user name of the request URI if the context received the original SIP request

The challenge applies to a unit if the corresponding Apply To column is set to Unit.

Creating/Editing an Authentication Entry

The web interface allows you to create authentication entries or modify the parameters of an existing one.

To create or edit SIP authentication parameters:

1. In the web interface, click the SIP link, then the Authentication sub-link.

Figure 141: SIP Configuration – Authentication Web Page

2. Do one of the following:

• If you want to add an authentication entry before an existing entry, locate the proper

row in the table and click the button of this row.

• If you want to add an authentication entry at the end of the existing rows, click the button at the bottom right of the Authentication section.

• If you want to add several authentication entries at the same time, enter the number of entries you want to add in the Number of rows to add at the bottom of the page.

• If you want to edit a single authentication entry, locate the proper row in the table and

click the button.

• If you want to edit a several authentication entries of the current page at the same time, click the Edit All Entries button at the bottom of the page.

This brings you to the proper Authentication panel.

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Table 140: Authentication Panel – Single Entry

Table 141: Authentication Panel – Page

3. Select which criterion to use for matching an authentication request with an authentication entry in the Apply to column.

Table 142: Authentication Entity


Unit The authentication entry is used on all challenges.

Endpoint The authentication entry used for all challenges related to a specific endpoint.

43 5 6 7 8 9

43 5 6 7 8 9

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4. Enter a string that identifies an endpoint in other tables in the Endpoint column.

This field is available only if you have selected the Endpoint entity in the previous step for the specific row.

5. Enter a string that identifies a SIP gateway in other tables in the Gateway column.

This field is available only if you have selected the Gateway entity in the Apply to column for the specific row.

6. Select whether or not the current credentials are valid for any realm in the corresponding Validate Realm drop-down menu.

7. Enter a realm for each authentication row in the Realm column.

When authentication information is required from users, the realm identifies who requested it.

8. Enter a string that uniquely identifies this endpoint in the realm in the User Name column.

9. Enter a user password in the Password column.




Moving an Authentication Entry

The order of the tried entries in the Authentication table is from the first row to the last row. The rows sequence is thus very important. If you want the unit to try to match one row before another one, you must put that row first.

To move an authentication entry up or down:

1. Either click the or arrow of the row you want to move until the entry is properly located.

Deleting an Authentication Entry

You can delete an authentication row from the table in the web interface.

To delete an authentication entry:

1. Click the button of the row you want to delete.

Gateway The authentication entry is used for all challenges related to a specific SIP gateway.

Username The authentication entry is used for all challenges related to a specific user name. Only the username part is used if the value has the format 'username@domain'.

Table 143: Realm Authentication Parameters


Disable The current credentials are valid for any realm. The corresponding Realm field is read-only and cannot be modified.

Enable The credentials are used only for a specific realm set in the corresponding Realm field.

Table 142: Authentication Entity (Continued)


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25 SIP Transport Parameters

This chapter describes the SIP transport parameters you can set.

SIP Transport Type

You can globally set the transport type for all the endpoints of the Mediatrix Boss to either UDP (User Datagram Protocol), TCP (Transmission Control Protocol), or TLS (Transport Layer Security).

The Mediatrix Boss will include its supported transports in its registrations.

Please note that RFC 3261 states the implementations must be able to handle messages up to the maximum datagram packet size. For UDP, this size is 65,535 bytes, including IP and UDP headers. However, the maximum datagram packet size the Mediatrix Boss supports for a SIP request or response is 5120 bytes excluding the IP and UDP headers. This should be enough, as a packet is rarely bigger than 2500 bytes.

To set the SIP transport type parameters:

1. In the web interface, click the SIP link, then the Transport sub-link.

Figure 142: SIP Configuration – Transport Web Page

2. In the General Configuration section, enable or disable the transport registration in the Add SIP Transport in Registration drop-down menu.

When enabled, the Mediatrix Boss includes its supported transports in its registrations. It registers with one contact for each transport that is currently enabled. Each of these contacts contains a “transport” parameter.

Standards Supported • RFC 2246: The TLS Protocol Version 1.0

• RFC 3261: SIP, Session Initiation Protocol

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This is especially useful for a system where there are no SRV records configured to use a predefined transport order for receiving requests. When sending a request, the unit either follows the SRV configuration, or, if not available, any transport parameter received from a redirection or from a configured SIP URL.

3. Indicate whether or not the unit must include its supported transport in the Contact header in the Add SIP Transport in Contact Header drop-down menu.

The supported transports are included in all SIP messages that have the Contact header, except for the REGISTER message.

Available values are Enable and Disable. If you set the menu to Enable, the Mediatrix Boss will send SIP messages with the “transport” parameter in the Contact header set to:

• transport=tcp when TCP is enabled and UDP is disabled

• transport=udp when UDP is enabled and TCP disabled

• no transport parameter when both TCP and UDP are enabled

• transport=tls when secure transport (TLS) is selected

4. Define the base port used to establish TLS persistent connections with SIP servers when the TLS transport is enabled in the Persistent TLS Base Port field.

5. Set the time interval, in seconds, before retrying the establishment of a TLS persistent connection in the Persistent TLS Retry Interval field.

This is the interval that the Mediatrix Boss waits before retrying periodically to establish a TLS persistent connection using a single IP address or a FQDN. This timer is started when a TLS persistent connection goes down or fails to connect to the destination. The TLS persistent connect timeout applies only to TLS persistent connections.

When the destination is a single IP address and the TLS persistent connection goes down or fails to establish, the timer is started. When the timer expires, the Mediatrix Boss attempts to re-establish the TLS persistent connection.

When the destination is a FQDN and the TLS persistent connection goes down or fails to establish with the higher priority target received from a DNS answer, the timer is started and the lower priority targets are attempted. When the timer expires, a new DNS request is sent and depending on the DNS answer, the Mediatrix Boss retries to establish the TLS persistent connection with the higher priority target. The timer is unique for all TLS persistent connections using the same FQDN. This means that the timer is not restarted when a connection using a lower priority target fails while a connection using a higher priority target has already failed.

6. Set the TCP Connect Timeout field with the maximum time, in seconds, the unit should try to establish a TCP connection to SIP hosts.

This timeout value is useful to have a faster detection of unreachable remote hosts. This timer can also affect the TLS connection establishment time.

7. In the Protocol Configuration section, enable or disable the UDP, TCP, and TLS transport type to use in their corresponding drop-down menu.

UDP and TCP are mutually exclusive with TLS. Activating TLS automatically disables these unsecure protocols.

Note: If the Mediatrix Boss has the following configuration:

• the Add SIP Transport in Registration drop-down menu is set to Disable

• the UDP transport type is disabled

• the TCP transport type is enabled

The unit will not work properly unless the SIP server uses the TCP transport type by default.

This is also true if the Mediatrix Boss has the TCP transport disabled and the UDP transport enabled. In this case, the unit will not work properly unless the SIP server uses the UDP transport protocol by default.

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The successful configuration of a secure transport requires a little more than the activation of the TLS protocol itself. You need to:

• install the security certificates used to authenticate the server to which you will connect (see “Chapter 39 - Certificates Management” on page 527 for more details).

• Use secure media (see “Security” on page 391 for more details).

• configure the unit so that a “transport=tls” parameter is added to the Contact header of your SIP requests (see Step 3).

8. Set the priority order of each transport type in the corresponding QValue field.

A qvalue parameter is added to each contact. The qvalue gives each transport a weight, indicating the degree of preference for that transport. A higher value means higher preference.

The format of the qvalue string must follow the RFC 3261 ABNF (a floating point value between 0.000 and 1.000). If you specify an empty string, no qvalue is set in the contacts.


Additional Transport Parameters


UDP Source Port Behaviour

You can configure whether or not the Mediatrix Boss always uses the same local port (the port on which it is listening for incoming packets) when sending SIP traffic over UDP. This is called symmetric UDP source port. Symmetric UDP ports are sometimes needed to traverse NAT/Firewall devices.

When changing this setting, all destinations are automatically sent out of the penalty box, when applicable.

The following parameters are available:

To set the UDP source port behaviour:

1. Set whether or not the unit uses the symmetric UDP source port feature by putting the following line in a configuration script:sipEp.interopSymmetricUdpSourcePortEnable="Value"

where Value may be as follows:

Caution: If you have enabled Secure RTP (SRTP) on at least one line, it is acceptable to have the secure SIP transport (TLS) disabled for testing purposes. However, you must never use this configuration in a production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly recommended and is usually mandatory for security interoperability with third-party equipment.

Table 144: UDP Source Port Parameters


disable The SIP signalling over UDP uses a randomly-generated originating port. ICMP errors are processed correctly.

enable The SIP signalling sent over UDP originates from the same port as the port on which the user agent is listening. ICMP messages are not processed, which means that unreachable targets will take longer to detect.

Table 145: UDP Source Port Values

Value Meaning

0 disable

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2. Restart the SipEp service by putting the following line in the configuration script:

scm.serviceCommands.Restart[Name=SipEp]="10"

TLS Client Authentication

When acting as a TLS server, it is customary not to request from the clients that they authenticate themselves via the TLS protocol. However, if mutual authentication is required between client and server, you can set the Mediatrix Boss so that it requests client authentication when acting as a TLS server.


To set TLS client authentication:

1. Set whether or not the Mediatrix Boss requests client authentication when acting as a TLS server by putting the following line in a configuration script:sipEp.interopTlsClientAuthenticationEnable="Value"


1 enable

Table 145: UDP Source Port Values (Continued)

Value Meaning

Table 146: TLS Client Authentication Parameters


disable The Mediatrix Boss does not require TLS clients to provide their host certificate for the connection to be allowed. This is the default value.

enable The TLS clients must provide their host certificate for the connection to be allowed. In this case, the level of security used to validate the host certificate is TrustedCertificate, whatever the value set in the Certificate Validation drop-down menu of the TLS Interop section ( SIP > Interop web page). See “TLS Interop” on page 342 for more details.

Table 147: TLS Client Authentication Values

Value Meaning

0 disable

1 enable

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26 Interop Parameters

This chapter describes the interop parameters that allow the Mediatrix Boss to properly work, communicate, or connect with specific IP devices.

Behavior on T.38 INVITE Not Accepted

This section describes the unit’s behaviour after receiving an error to a SIP INVITE for T.38 fax.

To set the T.38 interop parameters:

1. In the web interface, click the SIP link, then the Interop sub-link.

Figure 143: SIP – Interop Web Page

2. In the Behavior on T.38 INVITE Not Accepted section, for each of 406, 488, and 606 SIP code, set the behaviour afte receiving the code in the error response to an INVITE for T.38 fax in the corresponding Behavior drop-down menu.

Standards Supported • draft-ietf-sipping-realtimefax-00

• ITU-T Recommendation T.38, section D.2.3

• RFC 3264: An Offer/Answer Model with the Session Description Protocol (SDP)

• RFC 3515: The Session Initiation Protocol (SIP) Refer Method

Table 148: Behavior on T.38 INVITE Not Accepted Parameters

Behavior Description

Drop Call The call is dropped by sending a BYE.

ReInviteForClearChannelOnly A re-INVITE is sent with audio codecs that support clear channel faxes.

Re-Establish Audio A re-INVITE is sent to re-establish the audio path. Also, fax detection is disabled for the remainder of the call.

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SIP Interop

This section describes the SIP interop parameters of the Mediatrix Boss .

To set the SIP interop parameters:

1. In the SIP Interop section of the Interop page, set whether or not the “x-Siemens-Call-Type” header is added to the SIP packets sent by the unit in the Secure Header drop-down header.

You can set the Mediatrix Boss so that it triggers the addition of the “x-Siemens-Call-Type” header to the SIP packets sent by the unit when secure transport is in use.


Figure 144: SIP Interop Section

2. Select the username to use when the username is empty or undefined in the Default Username Value drop-down menu.

UsePreviousMediaNegotiation

No re-INVITE is sent and the audio codec from the last successful negotiation is used. For the remainder of the call, T.38 is disabled and fax detection may trigger a switch to a clear channel codec that was available in the last successful negotiation.

Table 148: Behavior on T.38 INVITE Not Accepted Parameters (Continued)

Behavior Description


Table 149: Secure Transport Header Parameters


disable The “x-Siemens-Call-Type” header is not added to the SIP packets sent by the unit.

enable The “x-Siemens-Call-Type” header is added to the SIP packets sent by the unit, and assigned the value “ST-secure”, as soon as secure transport and secure payload are being used. If secure transport or secure payload are not used, the header is not added.

Table 150: Default Username Value


Anonymous Sets the username to “anonymous”.

Host Sets the username to the same value as the host.

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3. Define the behaviour of the Mediatrix Boss when answering a SIP OPTIONS request in the OPTIONS Method Support drop-down menu.

4. Define whether or not the SIP OPTIONS requests should be ignored when all endpoints are unusable in the Ignore OPTONS on no usable endpoints drop-down menu.

Note that this feature may be influenced by whether or not you have enabled the Monitor Link State parameter. For more information:

• ISDN PRI interface: “PRI Configuration” on page 232

• ISDN BRI interface: “BRI Configuration” on page 239

• R2 PRI interface: “R2 Channel Associated Signaling” on page 261

5. Set the value of the user parameter in SIP URIs sent by the unit in the SIP URI User Parameter Value field.

If you leave the field empty, the parameter is not added.

E.g : sip:[email protected];user=InteropSipUriUserParameterValue

Note that when the Map Plus To TON International drop-down menu is set to Enable, the parameter's value might be overwritten (“Misc Interop” on page 343).

6. Set the Behavior On Machine Detection drop-down menu with the SIP device’s behavior when a machine (fax or modem) is detected during a call.

Table 151: OPTIONS Method Support Parameters


None The Mediatrix Boss responds with an error 405 Method not allowed.

AlwaysOK The Mediatrix Boss responds with a 200 OK regardless of the content of the OPTIONS request.

Table 152: Ignore SIP Options Parameters


Enable The unit ignores SIP OPTIONS requests when all endpoints are unusable. When at least one endpoint is usable, then the SIP OPTIONS requests are answered as configured in the OPTIONS Method Support drop-down menu (see Step 10).

Disable The SIP OPTIONS requests are answered as configured in the OPTIONS Method Support drop-down menu (see Step 10) regardless of the state of the endpoints.

Table 153: Behavior on Machine Detection Parameters


Re-INVITE On Fax T38 Only A SIP re-INVITE is sent only on a fax detection and T.38 is enabled.

Re-INVITE On No Negotiated Data Codec

A SIP re-INVITE is sent on a fax or modem detection if no data codec was previously negotiated in the original SDP negotiation. In the case where at least one data codec was previously negotiated in the SDP negotiation, the device switches silently to a data codec without sending a SIP re-INVITE. Note sthat if there is no data codec enabled on the device, no SIP re-INVITE is sent and the call is dropped by sending a BYE.

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See “Data Codec Selection Procedure” on page 409 for more details on the procedure the Mediatrix Boss follows when selecting data codec.

7. Set the Registration Contact Matching field with the matching behaviour for the contact header received in positive responses to REGISTER requests sent by the unit.

8. Set the Transmission Timeout field with the time to wait for a response or an ACK before considering a transaction timed out.

This corresponds to timers B, F and H for all transport protocols and timer J for UDP. These timers are defined in section A of RFC 3261.

This timeout affects the number of retransmissions. Retransmissions continue to follow the timing guidelines described in RFC 3261.

If a DNS SRV answer contains more than one entry, the Mediatrix Boss will try these entries if the entry initially selected does not work. You can configure the maximum time, in seconds, to spend waiting for answers to messages, from a single source. Retransmissions still follow the algorithm proposed in RFC 3261, but the total wait time can be overridden by using this feature.

For example, if you are using DNS SRV and more than one entry are present, this timeout is the time it takes before trying the second entry.

Available values are from 1 to 32 seconds.


Re-INVITE Unconditional A SIP re-INVITE is sent with data codecs upon detection of a fax or modem even if a data codec was negotiated in the initial offer-answer. The T.38 codec is offered if it is enabled and a fax is detected.

Table 154: Registration Contact Matching Parameters


Strict Matches the complete contact's SIP URI including any URI parameters, if any, as per RFC 3261 sections '10.2.4 Refreshing Bindings' and '19.1.4 URI Comparison'. The contact's SIP URI of a 2XX positive response MUST match the contact's SIP URI of the REGISTER request.

IgnoreUriParams

Matches the username and the host port part of the contact's SIP URI. All URI parameters are ignored.

Table 153: Behavior on Machine Detection Parameters (Continued)


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SDP Interop

This section describes the SDP interop parameters of the Mediatrix Boss .

To set the SDP interop parameters:

1. In the SDP Interop section of the Interop page, Offer Answer Model part, select the codec negotiation rule when generating a SDP answer in the Answer Codec Negotiation drop-down menu.

Figure 145: SDP Interop Section

Standards Supported • RFC 3264: An Offer/Answer Model with Session Description Protocol (SDP)

Table 155: Answer Codec Negotiation Parameters


All Common - Local Priority

When generating an answer to an offered session, all common codecs are listed in the local order of priority. The local priority is defined for each codec in the Telephony > CODECS page – by clicking the button of each codec and looking in the Voice Priority and Data Priority fields. See “Chapter 29 - Voice & Fax Codecs Configuration” on page 373 for more details.

First Common - Local Priority

When generating an answer to an offered session, only the first common codec with the higher local priority is listed. The local priority is defined for each codec in the Telephony > CODECS page – by clicking the button of each codec and looking in the Voice Priority and Data Priority fields. See “Chapter 29 - Voice & Fax Codecs Configuration” on page 373 for more details.

All Common - Peer Priority

When generating an answer to an offered session, all common codecs are listed. The codecs order is the same as in the peer offer.

First Common - Peer Priority

When generating an answer to an offered session, only the first common codec is listed. The codecs order is the same as in the peer offer.

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2. Select whether or not the Mediatrix Boss requires strict adherence to RFC 3264 when receiving an answer from the peer when negotiating capabilities for the establishment of a media session in the Enforce Offer Answer Model drop-down menu.

The following values are available:

3. Define the behaviour of the Mediatrix Boss when receiving less media announcements in the response than in the offer in the Allow Less Media In Response drop-down menu.


4. Define the behaviour of the Mediatrix Boss when receiving a SDP answer activating a media that had been previously deactivated in the offer in the Allow Media Reactivation in Answer drop-down menu.

Table 156: Offer/Answer Model Parameters


Disable The peer can freely:

• Send back a brand new list of codecs or add new ones to the offered list.

• Add new media lines.

As long as at least one codec sent back was present in the initial offer, the call is allowed to go on. Any media line added by the peer is simply ignored.

Enable The following guidelines from the Offer-Answer Model must be strictly followed. An answer must:

• Include at least one codec from the list that the Mediatrix Boss sent in the offer.

• Contain the same number of media lines that the unit put in its offer.

Otherwise, the answer is rejected and the unit ends the call. This is the default value.

Table 157: Less Media Announcements Parameters


Disable The Mediatrix Boss rejects the response with less media announcements than in the offer.

Enable The Mediatrix Boss tries to find matching media when the response contains less media announcement than in the offer. This is a deviation from the Offer/Answer model.

Table 158: Media Reactivation Parameters


Enable A media reactivated in an incoming answer is ignored. This behaviour goes against the SDP Offer/Answer model described by IETF RFC 3264.

Disable A media reactivated in an incoming answer ends the current media negotiation and the call. This behaviour follows the SDP Offer/Answer model described by IETF RFC 3264.

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5. In the Multiple Active Media part, define the behaviour of the Mediatrix Boss when offering media or answering to a media offer with audio and image negotiation in the Allow Audio and Image Negotiation drop-down menu.

6. Define the behaviour of the Mediatrix Boss when answering a request offering more than one active media in the Allow Multiple Active Media in Answer drop-down menu.

7. In the Other part, define how to set the direction attribute and the connection address in the SDP when answering a hold offer with the direction attribute “sendonly” in the On Hold SDP Stream Direction in Answer drop-down menu.

The following parameters are supported:

In both cases, no direction attribute is present in the SDP if the interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on page 345 for more details.

8. Set the Codec vs Bearer Capabilities Mapping Preferred Codec Choice drop-down menu with the behaviour of the Codec vs. Bearer Capabilities Mapping table.

This modifies the selection of the preferred codec in the incoming SDP. This parameter is available only on ISDN interfaces.

Table 159: Audio and Image Negotiation Parameters


Enable The unit offers audio and image media simultaneously in outgoing SDP offers and transits to T.38 mode upon reception of a T.38 packet. Also, when the unit answers positively to a SDP offer with audio and image, it transits to T.38 mode upon reception of a T.38 packet.

Disable Outgoing offers never include image and audio simultaneously. Incoming offers with audio and image media with a non-zero port are considered as offering only audio.

Figure 146: Allow Multiple Active Media in Answer


disable The answer contains only one active media. The media specified as active in the answer is the top-most matching one in the offer. Other media are set to inactive.

enable Each matching active media in the offer is specified as active in the answer. Other media are set to inactive

Table 160: “sendonly” Direction Attribute


inactive The stream is marked as inactive and the connection address is set to '0.0.0.0'.

revconly If the stream is currently active or receive only, it is marked as recvonly and the connection address is set to the IP address of the unit.

If the stream is currently send only or inactive, it is marked as inactive and the connection address is set to '0.0.0.0'.

This method is in conformance with RFC 3264.

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The Codec vs. Bearer Capabilities Mapping table parameters are located in the Telephony > CODECS > CODEC vs. Bearer Capabilities Mapping section. See “Codec vs. Bearer Capabilities Mapping” on page 379 for more details.


TLS Interop

This section describes the TLS interop parameters of the Mediatrix Boss .

To set the TLS interop parameters:

1. In the TLS Interop section of the Interop page, select the level of security used to validate the TLS server certificate when the unit is acting as a TLS client in the Certificate Validation drop-down menu.

Figure 147: TLS Interop Section


Table 161: Codec vs Beareer Capabilities Mapping Preferred Codec Choice Parameters


First Codec The first valid codec in the incoming SDP is considered the preferred one and is used when looking up the Codec vs. Bearer Capabilities Mapping table.

Prioritize Clear Channel When a clear channel codec is in the incoming SDP, it is always considered as the preferred one, no matter where it stands in the codec list, and is used when looking up the Codec vs. Bearer Capabilities Mapping table.

Note: If you are experiencing media negotiation problems (because the Mediatrix Boss sends a BYE after receiving a 200 OK), try to set the Enforce Offer Answer Model value to Disable and the Allow Less Media In Response value to Enable.

1

Note: This parameter has no effect on the TLS client authentication when the unit is acting as a TLS server (see the interopTlsClientAuthenticationEnable variable in “TLS Client Authentication” on page 334).

Table 162: TLS Certificate Validation Parameters


No Validation No validation of the peer certificate is performed. All TLS connections are accepted without any verification. Note that at least one certificate must be returned by the peer even if no validation is made. This option provides no security and should be restricted to a lab use only.

Trusted Certificate

Allows a TLS connection only if the peer certificate is trusted. A certificate is considered trusted when the certificate authority (CA) that signed the peer certificate is present in the Management > Certificates page (“Chapter 39 - Certificates Management” on page 527). This option provides a minimum level of security and should be restricted to a lab use only.

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Misc Interop

This section describes miscellaneous interop parameters of the Mediatrix Boss .

To set the Misc interop parameters:

1. In the Misc Interop section of the Interop page, select whether or not the Mediatrix Boss enables the mapping between the “+” prefix of the user name and the “type of number” property in the Map Plus To TON International drop-down menu.

When enabled, the service has the following behaviour:

• For a call to SIP, the Mediatrix Boss prefixes the user name with '+' if the call has the call property “type of number” set to international. The unit also adds the “user” parameter with the value “phone” to the SIP URI. For instance: sip:[email protected];user=phone.

• For a call from SIP, the Mediatrix Boss sets the call property “type of number” to international if the user name has the prefix '+'.

Figure 148: Misc Interop Section

2. Define the Ignore Plus in Username drop-down menu as to whether or not the plus (+) character is ignored when attempting to match a challenge username with usernames in the Authentication table.

Dns Srv Response

Allows a TLS connection if the peer certificate is trusted and contains a known host name. A known host name can be the FQDN or IP address configured as the SIP server, or can also be returned by a DNS SRV request. In this case, the match is performed against the DNS response name. If it matches either one of the Subject Alternate Name (SAN) or Common Name (CN) in the peer certificate, the connection is allowed. This option provides an acceptable level of security, but not as good as Host Name.

HostName Allows a TLS connection if the peer certificate is trusted and contains a known host name. A known host name can only be the FQDN or IP address configured as the SIP server. If it matches either one of the Subject Alternate Name (SAN) or Common Name (CN) in the peer certificate, the connection is allowed. This option provides the highest level of security.

Table 162: TLS Certificate Validation Parameters


Table 163: Ignore Plus (+) Character in Username Parameters


Enable The plus (+) character is ignored when attempting to match a username in the authentication table.

Disable The plus (+) character is not ignored when attempting to match a username in the authentication table.

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3. Select whether or not the pound character (#) must be escaped in the username part of a SIP URI in the Escape Pound (#) in SIP URI Username drop-down menu.


Additional Interop Parameters


The interop parameters allow the Mediatrix Boss to properly work, communicate, or connect with specific IP devices.

Call Waiting Private Number Criteria for SIP INFO

You can specify the call waiting criteria, in the form of a regular expression, that defines a private number received in a SIP INFO.

To set the Call Waiting Private Number Criteria:

1. Set the Call Waiting Private Number Criteria by putting the following line in the configuration script:sipEp.interopCallWaitingSipInfoPrivateNumberCriteria="Value"

For example, the value "(Anonymous|anonymous)" would define a calling number that is either "Anonymous" or "anonymous" as private. The regular expression symbols to match the beginning and end of the number are implicit and do not need to be specified. See “Regular Expressions” on page 454 for more details.

The variable is effective only if the Default Hook-Flash Processing parameter of the SIP > Misc page is set to TransmitUsingSignalingProtocol (see “General Configuration” on page 419 for more details).

1. Set the proper timeout value, in seconds, by putting the following line in the configuration script:

Max-Forwards Header

Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request reaches its destination, it is rejected with a “483 (Too Many Hops)” error response. The Max-Forwards SIP header is always present and the default value is 70.

Table 164: Escape Pound Parameters


Enable The Pound character (#) is escaped in the username part of a SIP URI.

Disable The Pound character (#) is not escaped in the username part of a SIP URI.

Note that RFC 3261 specifies that the pound character (#) needs to be escaped in the username part of a SIP URI.


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Direction Attributes in a Media Stream

The Mediatrix Boss allows you to define various direction attributes pertaining to the media stream.

When Putting a Call on Hold

The Mediatrix Boss can provide the direction attribute and the meaning of the connection address “0.0.0.0” sent in the SDP when an endpoint is put on hold.


To define the direction attribute when putting a call on hold:

1. Set the proper SDP stream direction by putting the following line in the configuration script:sipEp.interopOnHoldSdpStreamDirection="Value"


This configuration has no effect if the interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on page 345 for more details).

Direction Attribute

You can define:

if the SDP direction attribute is present in the initial INVITE sent by the Mediatrix Boss

if the direction attribute present in the SDP received from the peer is ignored or not



Table 165: Direction Attributes


inactive The stream is put on hold by marking it as inactive. This is the default value. This setting should be used for backward compatibility issues.

sendonly The stream is put on hold by marking it as sendonly. This method allows the Mediatrix Boss to be in conformance with RFC 3264.

Table 166: Direction Attributes Values

Value Meaning

100 inactive

200 sendonly


• RFC 3264: An Offer/Answer Model with Session Description Protocol (SDP)

Table 167: SDP Direction Attribute


disable No direction attribute is present in the SDP sent by the Mediatrix Boss.

The Mediatrix Boss ignores any direction attribute found in the SDP received from the peer.

To put an endpoint on hold, a SDP containing a connection address of “0.0.0.0” is sent.

The method to put a session on hold is in conformance with RFC 2543.

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To define if the direction attribute is present:

1. Set the SDP direction attribute behaviour by putting the following line in the configuration script:sipEp.interopSdpDirectionAttributeEnable="Value"


Answering a Hold Offer with the Direction Attribute “sendonly”

You can define how to set the direction attribute and the connection address in the SDP when answering a hold offer with the direction attribute “sendonly”.


To define the behaviour with the “sendonly” direction attribute:

1. Define the behaviour with the “sendonly” direction attribute by putting the following line in the configuration script:sipEp.interopOnHoldAnswerSdpStreamDirection="Value"

enable The Mediatrix Boss always sends the direction attribute in the SDP of the initial INVITE.

The initial handshake determines whether or not the peer supports the direction attribute or not.

• If the direction attribute is present in the SDP received from the peer, the Mediatrix Boss sends the direction attribute in the SDP for the remainder of the session.

• If the direction attribute is not present in the SDP received from the peer, the Mediatrix Boss does not send the direction attribute in the SDP for the remainder of the session.

If present in the SDP, the direction attribute is preferred over the connection address to transmit session modification information.


Table 167: SDP Direction Attribute (Continued)


Table 168: SDP Direction Attribute

Value Meaning

0 disable

1 enable




inactive The stream is marked as inactive and the connection address is set to '0.0.0.0'.

revconly If the stream is currently active or receive only, it is marked as recvonly and the connection address is set to the IP address of the unit.

If the stream is currently send only or inactive, it is marked as inactive and the connection address is set to '0.0.0.0'.


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In both cases, no direction attribute is present in the SDP if the interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on page 345 for more details.

Local Ring Behaviour on Provisional Response

You can set the Mediatrix Boss so that it starts or not the local ring upon receiving a “18x Provisional” response without SDP.

This setting does not affect the behaviour when the “18x Provisional” response contains SDP, which allows establishing an early media session before the call is answered.


To define the local ring behaviour on provisional response:

1. Set the proper local ring behaviour by putting the following line in the configuration script:sipEp.interopLocalRingOnProvisionalResponse="Value"


Session ID and Session Version Number in the Origin Field of the SDP

You can define the maximum length of the session ID and the session version number in the origin line (o=) of the SDP. This allows the Mediatrix unit to be compatible with 3rd party vendor equipment.



Value Meaning

100 inactive

200 Recvonly

Figure 149: Local Ring Behaviour


disable The local ring is not started on a “18x Provisional” response without SDP, except for a “180 Ringing” message. This is the default value.

The 180 Ringing message is a provisional or informational response used to indicate that the INVITE message has been received by the user agent and that alerting is taking place.

enable The local ring is started on any “18x Provisional” response without SDP.

Figure 150: Local Ring Values

Value Meaning

0 disable

1 enable

Table 171: Maximum Length Parameters

Length Description

max-32bits The session ID and the session version number are represented with a 32 bit integer. They have a maximum length of 10 digits.

max-64bits The session ID and the session version number are represented with a 64 bit integer. They have a maximum length of 20 digits. This is the default value.

347


To set the maximum length of the session ID and the session version number:

1. Set the maximum length of the session ID by putting the following line in the configuration script:sipEp.interopSdpOriginLineSessionIdAndVersionMaxLength="Value"


Register Home Domain Override

By default, the address-of-record in the “To” header uses the value set in the Proxy Host field of the SIP/Configuration page for the host/port part. See “SIP Servers Configuration” on page 309 for more details. You can override this value if required.

To override the register home domain value:

1. Set the override home domain value by putting the following line in the configuration script:sipEp.interopRegisterHomeDomainOverride="IP_Address"

The address of record in the register will use this string instead of the home domain proxy. If the variable is empty, the value of the Proxy Host field is used.

The host is also overridden in the From and Call-Id headers since they match the To header.

Listening for Early RTP

You can set the Mediatrix Boss so that it listens for RTP before the reception of a response with SDP. This feature only applies to calls initiated from analog endpoints (FXS/FXO) with non-secure RTP.


To enable the Early RTP feature:

1. Set the early RTP behaviour by putting the following line in the configuration script:sipEp.InteropListenForEarlyRtpEnable="Value"

Table 172: Maximum Length Values

Value Meaning

100 max-32bits

200 max-64bits

Note: This feature applies to the following models:

• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731 / 3732 / 3741 / 3742

Table 173: Early RTP Parameters

Length Description

enable The RTP port is opened after the initial INVITE has been sent, without waiting for a provisional or final response with SDP to be received. No local ring is generated. This conforms to section 5.1 of RFC 3264.

disable The RTP port is opened only after a response with SDP is received.

Warning: Do not enable this feature unless the server supports early RTP (or early media). Failing so prevents any ringing to be heard for outgoing calls.

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Figure 151: Early RTP Values

Value Meaning

0 disable

1 enable

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C H A P T E R

27 Miscellaneous SIP Parameters

This chapter describes miscellaneous SIP parameters you can set:

SIP penalty box parameters

How to override the default mapping of error causes defined in RFC 3398.

SIP Keep Alive

PRACK & UPDATE

SIP Gateway Configuration

Timer Configuration

Diversion Configuration

DNS Configuration

SIP Penalty Box

The penalty box feature is used to “quarantine” a given host which address times out. During that time, the address is considered as “non-responding” for all requests.

This feature is most useful when using multiple servers and some of them are down. It ensures that users wait a minimal period of time before trying a secondary host.

The penalty box feature is applied only when the “non-responding” destination is associated with a non-persistent connection. A similar penalty box feature for the TLS persistent connections is available via the TLS Persistent Retry Interval parameter. See “SIP Transport Type” on page 331 for more details.

Penalty Box vs Transport Types

Media5 recommends to use this feature with care when supporting multiple transports (see “Chapter 25 - SIP Transport Parameters” on page 331 for more details) or you may experience unwanted behaviours.

When the Mediatrix Boss must send a packet, it retrieves the destination from the packet. If the destination address does not specify a transport to use and does not have a DNS SRV entry that configures which transport to use, then the Mediatrix Boss tries all transports it supports, starting with UDP. If this fails, it tries with TCP. The unit begins with UDP because all SIP implementations must support this transport, while the mandatory support of TCP was only introduced in RFC 3261.

Let’s say for instance that the Mediatrix Boss supports both the UDP and TCP transports. It tries to reach endpoint “B” for which the destination address does not specify a transport and there is no DNS SRV entry to specify which transports to use in which order. It turns out that this endpoint “B” is also down. In this case, the Mediatrix Boss first tries to contact endpoint “B” via UDP. After a timeout period, UDP is placed in the penalty box and the unit then tries to contact endpoint “B” via TCP. This fails as well and TCP is also placed in the penalty box.

Now, let’s assume endpoint “B” comes back to life and the Mediatrix Boss tries again to contact it before UDP and TCP are released from the penalty box. First, the unit tries UDP, but it is currently in the penalty box and there is another transport left to try. The Mediatrix Boss skips over UDP and tries the next target, which is TCP. Again, TCP is still in the penalty box, but this time, it is the last target the Mediatrix Boss can try, so penalty box or not, TCP is used all the same to try to contact endpoint “B”.

Note: It is not the destination itself that is placed in the penalty box, but the combination of address, port and transport. When a host is in the penalty box, it is never used to try to connect to a remote host unless it is the last choice for the Mediatrix Boss and there are no more options to try after this host.

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Chapter 27 - Miscellaneous SIP Parameters SIP Penalty Box

There is a problem if endpoint “B” only supports UDP (RFC 2543-based implementation). Endpoint “B” is up, but the Mediatrix Boss still cannot contact it: with UDP and TCP in the penalty box, the unit only tries to contact endpoint “B” via its last choice, which is TCP.

The same scenario would not have any problem if the penalty box feature was disabled. Another option is to disable TCP in the Mediatrix Boss, which makes UDP the only possible choice for the unit and forces to use UDP even if it is in the penalty box.

You must fully understand the above problem before configuring this feature. Mixing endpoints that do not support the same set of transports with this feature enabled can lead to the above problems, so it is suggested to either properly configure SRV records for the hosts that can be reached or be sure that all hosts on the network support the same transport set before enabling this feature.

Penalty Box Configuration

The following steps describe how to configure the penalty box feature.

To set the SIP penalty box parameters:

1. In the web interface, click the SIP link, then the Misc sub-link.

Figure 152: SIP Configuration – Misc Web Page

2. In the Penalty Box section, enable the SIP penalty box feature by selecting Enable in the Penalty Box Activation drop-down menu.

The penalty box is always “active”. This means that even if the feature is disabled, IP addresses are marked as invalid, but they are still tried. This has the advantage that when the feature is enabled, IP addresses that were already marked as invalid are instantly put into the penalty box.

3. Set the amount of time, in seconds, that a host spends in the penalty box in the Penalty Box Time field.

Changing the value does not affect IP addresses that are already in the penalty box. It only affects new entries in the penalty box.


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Error Mapping Software Configuration Guide

Error Mapping

You can override the default mapping of error causes defined in RFC 3398.The web interface offers two sections:

The SIP To Cause Error Mapping section allows you to override the default mapping for SIP code to ISDN cause.

The Cause To SIP Error Mapping section allows you to override the default mapping for ISDN cause to SIP code.

The following standard SIP codes are available:

You can also map any other custom code between 400 and 699.

Standards Supported • RFC 3398: Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mappinga

a. Only the ISDN to SIP error mapping is supported.

400: Bad Request

401: Unauthorized

402: Payment required

403: Forbidden

404: Not found

405: Method not allowed

406: Not acceptable

407: Proxy authentication required

408: Request timeout

410: Gone

413: Request Entity too long

414: Request-URI too long

415: Unsupported media type

416: Unsupported URI Scheme

420: Bad extension

421: Extension Required

423: Interval Too Brief

480: Temporarily unavailable

481: Call/Transaction Does not Exist

482: Loop Detected

483: Too many hops

484: Address incomplete

485: Ambiguous

486: Busy here

500: Server internal error

501: Not implemented

502: Bad gateway

503: Service unavailable

504: Server time-out

504: Version Not Supported

513: Message Too Large

600: Busy everywhere

603: Decline

604: Does not exist anywhere

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The following standard ISDN cause numbers specified in Q.931 are available:

You can also map any other custom code between 1 and 127.

Normal event:

1: Unassigned (unallocated) number.

2: No route to specified transit network.

3: No route to destination.

6: Channel unacceptable.

7: Call awarded and being delivered in an established channel.

17: User busy.

18: No user responding.

19: User alerting, no answer.

20: Subscriber absent.

21: Call rejected.

22: Number changed.

23: Redirection to new destination.

26: Non-selected user clearing.

27: Destination out of order.

28: Invalid number format (incomplete number).

29: Facility rejected.

30: Response to STATUS ENQUIRY.

31: Normal, unspecified.

Resource unavailable:

34: No circuit/channel available.

38: Network out of order.

41: Temporary failure.

42: Switching equipment congestion.

43: Access information discarded.

44: Requested circuit/channel not available.

47: Resource unavailable, unspecified.

Service or option not available:

55: Incoming calls barred within CUG.

57: Bearer capability not authorized.

58: Bearer capability not presently available.

63: Service or option not available, unspecified.

Service or option not implemented:

65: Bearer capability not implemented.

66: Channel type not implemented.

69: Requested facility not implemented.

70: Only restricted digital information bearer.

79: Service or option not implemented, unspecified.

Invalid Message

81: Invalid call reference value.

82: Identified channel does not exist.

83: A suspended call exists, but this call identity does not.

84: Call identity in use.

85: No call suspended.

86: Call having the requested call identity has been cleared.

87: user not member of CUG.

88: Incompatible destination.

91: Invalid transit network selection.

95: Invalid message, unspecified.

Protocol error

96: Mandatory information element is missing.

97: Message type non-existent or not implemented.

98: Message not compatible with call state or message type non-existent or not implemented.

99: Information element non-existent or not implemented.

100: Invalid information element contents.

101: Message not compatible with call state.

102: Recovery on time expiry.

111: Protocol error, unspecified.

Interworking

127: Interworking, unspecified

354 Mediatrix Boss


SIP to Cause Error Mapping

This section describes how to override the default mapping of ISDN error causes.

To override the default mapping of ISDN error causes:

1. In the SIP To Cause Error Mapping section of the Misc page, click the button to add a new row.

Figure 153: SIP To Cause Error Mapping Section

This brings you to the Configure New SIP To Cause Error Mapping panel.

2. Enter the SIP code in the SIP Code field, then the corresponding ISDN cause number in the Cause column.

You can use the Suggestion column’s drop-down menu to select between available code values.

Figure 154: Configure New SIP To Cause Error Mapping Panel

3. Click Submit.

This brings you back to the main Misc web page.

You can delete an existing row by clicking the button.

You can modify the Cause value by typing a new code in the field. See “SIP To Cause Default Error Mapping” on page 355 for the default mappings as per RFC 3398.


SIP To Cause Default Error Mapping

Table 174 lists the default mappings as per RFC 3398.

Table 174: SIP To Cause Default Error Mapping

SIP Response Received Cause Value

400 Bad Request 41 Temporary Failure

401 Unauthorized 21 Call rejected

402 Payment required 21 Call rejected

403 Forbidden 21 Call rejected

404 Not found 1 Unallocated number

405 Method not allowed 63 Service or option unavailable

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Cause to SIP Error Mapping

This section describes how to override the default mapping of SIP codes.

To override the default mapping of SIP codes:

1. In the Cause To SIP Error Mapping section of the Misc page, click the button to add a new row.

Figure 155: Cause To SIP Error Mapping Section

This brings you to the Configure New Cause To SIP Error Mapping panel.

406 Not acceptable 79 Service/option not implemented

407 Proxy authentication required 21 Call rejected

408 Request timeout 102 Recovery on timer expiry

410 Gone 22 Number changed (w/o diagnostic)

413 Request Entity too long 127 Interworking

414 Request-URI too long 127 Interworking

415 Unsupported media type 79 Service/option not implemented

416 Unsupported URI Scheme 127 Interworking

420 Bad extension 127 Interworking

421 Extension Required 127 Interworking

423 Interval Too Brief 127 Interworking

480 Temporarily unavailable 18 No user responding

481 Call/Transaction Does not Exist 41 Temporary Failure

482 Loop Detected 25 Exchange - routing error

483 Too many hops 25 Exchange - routing error

484 Address incomplete 28 Invalid Number Format

485 Ambiguous 1 Unallocated number

486 Busy here 17 User busy

500 Server internal error 41 Temporary failure

501 Not implemented 79 Not implemented, unspecified

502 Bad gateway 38 Network out of order

503 Service unavailable 41 Temporary failure

504 Server time-out 102 Recovery on timer expiry

504 Version Not Supported 127 Interworking

513 Message Too Large 127 Interworking

600 Busy everywhere 17 User busy

603 Decline 21 Call rejected

604 Does not exist anywhere 1 Unallocated number

Table 174: SIP To Cause Default Error Mapping (Continued)

SIP Response Received Cause Value

1

356 Mediatrix Boss


2. Enter the ISDN cause number in the Cause column, then the corresponding SIP code in the SIP Code field.

You can use the Suggestion column’s drop-down menu to select between available code values.

Figure 156: Configure New Cause To SIP Error Mapping Panel

3. Click Submit.

This brings you back to the main Misc web page.

You can delete an existing row by clicking the button.

You can modify the SIP Code value by typing a new code in the field. See “Cause To SIP Default Error Mapping” on page 357 for the default mappings as per RFC 3398.


Cause To SIP Default Error Mapping

Table 175 lists the default mappings as per RFC 3398.

Table 175: Cause To SIP Default Error Mapping

ISUP Cause Value SIP Response

Normal Event

1 unallocated number 404 Not Found

2 no route to network 404 Not Found

3 no route to destination 404 Not Found

16 normal call clearing --- BYE or CANCEL

17 user busy 486 Busy Here

18 no user responding 408 Request Timeout

19 no answer from the user 480 Temporarily unavailable

20 subscriber absent 480 Temporarily unavailable

21 call rejected 403 Forbidden

22 number changed (w/o diagnostic) 410 Gone

22 number changed (w/ diagnostic) 301 Moved Permanently

23 redirection to new destination 410 Gone

26 non-selected user clearing 404 Not Found

27 destination out of order 502 Bad Gateway

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28 address incomplete 484 Address incomplete

29 facility rejected 501 Not implemented

31 normal unspecified 480 Temporarily unavailable

Resource Unavailable

34 no circuit available 503 Service unavailable

38 network out of order 503 Service unavailable

41 temporary failure 503 Service unavailable

42 switching equipment congestion 503 Service unavailable

47 resource unavailable 503 Service unavailable

Service or Option not Available

55 incoming calls barred within CUG 403 Forbidden

57 bearer capability not authorized 403 Forbidden

58 bearer capability not presently available 503 Service unavailable

Service or Option not Implemented

65 bearer capability not implemented 488 Not Acceptable Here

70 only restricted digital available 488 Not Acceptable Here

79 service or option not implemented 501 Not implemented

Invalid message

87 user not member of CUG 403 Forbidden

88 incompatible destination 503 Service unavailable

Protocol error

102 recovery of timer expiry 504 Gateway timeout

111 protocol error 500 Server internal error

Interworking

127 interworking unspecified 500 Server internal error

Table 175: Cause To SIP Default Error Mapping (Continued)

ISUP Cause Value SIP Response

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SIP Keep Alive Software Configuration Guide

SIP Keep Alive

You can select the method used to perform the SIP keep alive mechanism. With this mechanism, the Mediatrix Boss sends messages periodically to the server to ensure that it can still be reached.

To use the SIP keep alive mechanism:

1. In the SIP Keep Alive section of the Misc page, select the keep alive method to use in the Keep Alive Method drop-down menu.

Figure 157: SIP Keep Alive Section

Table 176: Reason Header Support Parameters


None Silently ignores any incoming reason headers and does not send the reason header.

SendQ850 Silently ignores incoming reason codes and sends the SIP reason code when the original Q.850 code is available. The reason code sent is not affected by the entries in the Error Mapping SIP To Cause table.

ReceiveQ850 Uses the incoming Q.850 reason cause header. When received, the reason code supersedes any entrie s in the Error Mapping SIP To Cause table.

SendReceiveQ850 Uses the incoming Q.850 reason cause header and sends the SIP reason code when the original Q.850 code is available. When received, the reason code supersedes any entries in the Error Mapping SIP To Cause table. The reason code sent is not affected by the entries in the Error Mapping SIP To Cause table.

1

Table 177: SIP Keep Alive Parameters


None No keep alive is performed.

SipOptions SIP OPTIONS are sent periodically for each gateway to the corresponding server. Any response received from the server means that it can be reached. No additional processing is performed on the response. If no response is received after the retransmission timer expires (configurable via the Transmission Timeout field in “SIP Interop” on page 336), the gateway considers the server as unreachable. In this case, any call attempt through the gateway is refused. SIP OPTIONS are still sent when the server cannot be reached and as soon as it can be reached again, new calls are allowed.

12

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Chapter 27 - Miscellaneous SIP Parameters PRACK & UPDATE

2. Set the interval, in seconds, at which SIP OPTIONS are sent to verify the server status in the Keep Alive Interval field.

3. Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent to verify the server status in the Keep Alive Interval field.


PRACK & UPDATE

The Mediatrix Boss supports reliable provisional responses (PRACK) as per RFC 3262. You can define this support when acting as a user agent client and when acting as a user agent server.

The Mediatrix Boss supports the UPDATE as per RFC 3311; however, its support is limited to reception.

To define the PRACK support:

1. In the PRACK & UPDATE section of the Misc page, define the support of RFC 3262 (PRACK) when acting as a user agent server in the UAS PRACK Support drop-down menu.

Figure 158: PRACK & UPDATE Section

Receiving an UPDATE request to negotiate “early media” is supported only if you have selected Supported.

Ping A Ping is sent periodically for each gateway to the corresponding server. The response received from the server means that it is reachable. If no response is received after the retransmission timer expires (configurable via the Transmission Timeout field in “SIP Interop” on page 336), the gateway considers the server as unreachable. In this case, any call attempt through the gateway is refused. The Pings are still sent when the server is unreachable and as soon as it becomes reachable again, new calls are allowed.

Table 177: SIP Keep Alive Parameters (Continued)


Standards Supported • RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

• RFC 3311: The Session Initiation Protocol (SIP) UPDATE Methoda

a. Only support receiving UPDATE. Sending an UPDATE is not supported.

Table 178: PRACK User Agent Server Parameters


Unsupported The option tag “100rel” is ignored if present in the Supported or Required header of received initial INVITEs and provisional responses are not sent reliably as per RFC 3261.

Supported If the option tag “100rel” is present in the Supported or Required header of initial received INVITEs, provisional responses are sent reliably as per RFC 3262 by adding the option tag “100rel” to the Require header.

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SIP Gateway Configuration Software Configuration Guide

2. Define the support of RFC 3262 (PRACK) when acting as user agent client in the UAC PRACK Support drop-down menu.


SIP Gateway Configuration

You can define whether or not to override the SIP domain used.

To set the SIP domain override:

1. In the SIP Gateway Configuration section of the Misc page, define whether or not to override the SIP domain used in the SIP Domain field.

If not empty, the address of record uses this string instead of the home domain proxy (Proxy Host field of the Servers sub-page – SIP Default Servers section (“SIP Servers Configuration” on page 309).

Figure 159: SIP Gateway Configuration Section


Diversion Configuration

You can define call diversion parameters.

To set the call diversion parameters:

1. In the Diversion section of the Misc page, set the Methcd drop-down menu with the SIP method used to receive/send call diversion information in an INVITE.

The gateways available are those defined in “SIP Gateways Configuration” on page 305.

Table 179: PRACK User Agent Client Parameters


Unsupported The option tag “100rel” is not added in the Supported or Required header of sent INVITEs as per RFC 3261. If the provisional response contains a Require header field with the option tag “100rel”, the indication is ignored and no PRACK are sent.

Supported The option tag “100rel” is added to the Supported header of sent initial INVITEs as per RFC 3262. If the received provisional response contains a Require header field with the option tag “100rel”, the response is sent reliably using the PRACK method.

Required The option tag “100rel” is added to the Require header of sent initial INVITEs as per RFC 3262. If the received provisional response contains a Require header field with the option tag “100rel”, the response is sent reliably using the PRACK method.

1

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Chapter 27 - Miscellaneous SIP Parameters DNS Configuration

Figure 160: Diversion Configuration Section


DNS Configuration

You can define DNS-related parameters.

To set the DNS-related parameters:

1. In the DNS section of the Misc page, set the Supported DNS Queries drop-down menu with the type of DNS queries that the SipEp service supports and uses.

Figure 161: DNS Configuration Section


Table 180: Diversion Parameters


None No diversion information is sent in SIP messages.

Diversion Header The SIP gateway supports the SIP header 'Diversion' (RFC 5806) in received and sent INVITEs, as well as in 302 messages.

1

Table 181: DNS Parameters


Address Sends only Address requests (type A).

SRV Sends a Service request (type SRV) first and then Address requests (type A) if needed.

NAPTR Sends a Naming Authority Pointer request (type NAPTR) first and then Service requests (type SRV) or Address requests (type A) as needed.

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Telephony Parameters

C H A P T E R

28 DTMF Maps Configuration

This chapter describes how to configure and use the DTMF maps of the Mediatrix Boss.

DTMF maps syntax.

General DTMF maps parameters.

Allowed DTMF maps parameters.

Refused DTMF maps parameters.

Introduction

A DTMF map (also called digit map or dial map) allows you to compare the number users just dialed to a string of arguments. If they match, users can make the call. If not, users cannot make the call and get an error signal. It is thus essential to define very precisely a DTMF map before actually implementing it, or your users may encounter calling problems.

Because the Mediatrix Boss cannot predict how many digits it needs to accumulate before transmission, you could use the DTMF map, for instance, to determine exactly when there are enough digits entered from the user to place a call.

Syntax

The permitted DTMF map syntax is taken from the core MGCP specification, RFC 2705, section 3.4:

DigitMap = DigitString / '(' DigitStringList ')'

DigitStringList = DigitString 0*( '|' DigitString )DigitString = 1*(DigitStringElement)

DigitStringElement = DigitPosition ['.']

DigitPosition = DigitMapLetter / DigitMapRangeDigitMapLetter = DIGIT / '#' / '*' / 'A' / 'B' / 'C' / 'D' / 'T'

DigitMapRange = 'x' / '[' 1*DigitLetter ']'

DigitLetter ::= *((DIGIT '-' DIGIT ) / DigitMapLetter)

Where “x” means “any digit” and “.” means “any number of”.

For instance, using the telephone on your desk, you can dial the following numbers:

Standards Supported • RFC 2705: Media Gateway Control Protocol (MGCP) Version 1.0, section 3.4 (Formal syntax description of the protocol).

Note: Using certain Output Gain values may have an impact on the DTMF detection, so Media5 suggests to experiment with several values. The Output Gains cannot be set via the web interface, but only via a SNMP browser. See “Input/Output User Gain” on page 448 for more details.

Table 182: Number Examples

Number Description

0 Local operator

00 Long distance operator

xxxx Local extension number

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Chapter 28 - DTMF Maps Configuration Syntax

The solution to this problem is to load the Mediatrix Boss with a DTMF map that corresponds to the dial plan.

A Mediatrix Boss that detects digits or timers applies the current dial string to the DTMF map, attempting a match to each regular expression in the DTMF map in lexical order.

If the result is under-qualified (partially matches at least one entry in the DTMF map), waits for more digits.

If the result matches, dials the number.

If the result is over-qualified (i.e., no further digits could possibly produce a match), sends a fast busy signal.

Special Characters

DTMF maps use specific characters and digits in a particular syntax.

How to Use a DTMF Map

Let’s say you are in an office and you want to call a co-worker’s 3-digits extension. You could build a DTMF map that says “after the user has entered 3 digits, make the call”. The DTMF map could look as follows:

xxx

You could refine this DTMF map by including a range of digits. For instance, you know that all extensions in your company either begin with 2, 3, or 4. The corresponding DTMF map could look as follows:

[2-4]xx

If the number you dial begins with anything other than 2, 3, or 4, the call is not placed and you get a busy signal.

Combining Several Expressions

You can combine two or more expressions in the same DTMF map by using the “|” operator, which is equal to OR.

Let’s say you want to specify a choice: the DTMF map is to check if the number is internal (extension), or external (a local call). Assuming that you must first dial “9” to make an external call, you could define a DTMF map as follows:

([2-4]xx|9[2-9]xxxxxx)

8xxxxxxx Local number

#xxxxxxx Shortcut to local number at other corporate sites

91xxxxxxxxxx Long distance numbers

9011 + up to 15 digits International number

Table 182: Number Examples (Continued)

Number Description

Table 183: DTMF Map Characters

Character Use

Digits (0, 1, 2... 9) Indicates specific digits in a telephone number expression.

T The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished dialing and the SIP Server can make the call.

x Matches any digit, excluding “#” and “*”.

| Indicates a choice of matching expressions (OR).

. Matches an arbitrary number of occurrences of the preceding digit, including 0.

[ Indicates the start of a range of characters.

] Indicates the end of a range of characters.

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Syntax Software Configuration Guide

The DTMF map checks if:

the number begins with 2, 3, or 4 and

the number has 3 digits

If not, it checks if:

the number begins with 9 and

the second digit is any digit between 2 and 9 and

the number has 7 digits

Using the # and * Characters

It may sometimes be required that users dial the “#” or “*” to make calls. This can be easily incorporated in a DTMF map:

xxxxxxx#

xxxxxxx*

The “#” or “*” character could indicate users must dial the “#” or “*” character at the end of their number to indicate it is complete. You can specify to remove the “#” or “*” found at the end of a dialed number. See “General DTMF Maps Parameters” on page 368.

Using the Timer

The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished dialing and the Mediatrix Boss can make the call. A DTMF map for this could be:

[2-9]xxxxxxT

See “General DTMF Maps Parameters” on page 368 for more details.

Calls Outside the Country

If your users are making calls outside their country, it may sometimes be hard to determine exactly the number of digits they must enter. You could devise a DTMF map that takes this problem into account:

001x.T

In this example, the DTMF map looks for a number that begins with 001, and then any number of digits after that (x.).

Example

Table 182 on page 365 outlined various call types one could make. All these possibilities could be covered in one DTMF map:

(0T|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|91xxxxxxxxxx|9011x.T)

Validating a DTMF Map

The Mediatrix Boss validates the DTMF map as you are entering it and it forbids any invalid value.

Note: Enclose the DTMF map in parenthesis when using the “|” option.

Note: When making the actual call and dialing the number, the Mediatrix Boss automatically removes the “T” found at the end of a dialed number, if there is one (after a match). This character is for indication purposes only.

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Chapter 28 - DTMF Maps Configuration General DTMF Maps Parameters

General DTMF Maps Parameters

The following are the general DTMF maps parameters you can set.

To set the general DTMF map parameters:

1. In the web interface, click the Telephony link, then the DTMF Maps sub-link.

Figure 162: Telephony – DTMF Maps Web Page

2. In the General Configuration section, define the time, in milliseconds (ms), between the start of the dial tone and the receiver off-hook tone, if no DTMF is detected, in the First DTMF Timeout field.

Values range from 1000 ms to 180000 ms. The default value is 20000 ms.

If you want to set a different First DTMF Timeout value for one or more endpoints, click the Edit Endpoints button (see “Configuring Timeouts per Endpoint” on page 369 for more details).

3. Define the value, in milliseconds (ms), of the “T” digit in the Inter Digit Timeout field.

The “T” digit expresses a time lapse between the detection of two DTMFs. Values range from 500 ms to 10000 ms. The default value is 4000 ms.

If you want to set a different Inter Digit Timeout value for one or more endpoints, click the Edit Endpoints button (see “Configuring Timeouts per Endpoint” on page 369 for more details).

4. Define the total time, in milliseconds (ms), the user has to dial the DTMF sequence in the Completion Timeout field.

The timer starts when the dial tone is played. When the timer expires, the receiver off-hook tone is played. Values range from 1000 ms to 180000 ms. The default value is 60000 ms.

If you want to set a different Completion Timeout value for one or more endpoints, click the Edit Endpoints button (see “Configuring Timeouts per Endpoint” on page 369 for more details).

5. In the DTMF Maps Digit Detection (FXO/FXS) drop-down menu, define when a digit is processed through the DTMF maps.

This parameters is available only when the unit has FXS or FXO ports.

Table 184: DTMF Maps Digit Detection Parameters


When Pressed

Digits are processed as soon as they are pressed. This can lead to a digit leak in the RTP at the beginning of a call if the voice stream is established before the last digit is released.

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Configuring Timeouts per Endpoint

You can set a different timeout value for one or more endpoints.

To set a different value per endpoint:

1. In the General Configuration section of the DTMF Maps page, click the Edit Endpoints button.

The following window is displayed:

Figure 163: DTMF Map Timeout Section

2. Set the Override drop-down menu for the endpoint you want to set to Enable.

3. Change the value of one or more timeouts as required.

4. Repeat for each endpoint that you want to modify.

5. Click Submit when finished.

When Released

Digits are processed only when released. This option increases the delay needed to match a dialed string to a DTMF map. There is also an impact on the First DTMF Timeout, Inter Digit Timeout and Completion Timeout parameters since the timers are stopped at the end of a digit instead of the beginning.

Table 184: DTMF Maps Digit Detection Parameters (Continued)


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Chapter 28 - DTMF Maps Configuration Allowed DTMF Maps

Allowed DTMF Maps

You can create/edit ten DTMF maps for the Mediatrix Boss. DTMF map rules are checked sequentially. If a telephone number potentially matches two of the rules, the first rule encountered is applied.

To set up DTMF maps:

1. In the DTMF Map drop-down menu at the top of the window, select Allowed.

The Allowed DTMF Map section displays.

2. In the Allowed DTMF Map section – Enable column, enable one or more DTMF maps by selecting the corresponding Enable choice.

Figure 164: Allowed DTMF Map Section

3. Select the entity to which apply the allowed DTMF map in the Apply to column.



5. Define the DTMF map string that is considered valid when dialed in the DTMF Map column.

The string must use the syntax described in “DTMF Maps Configuration” on page 365. A DTMF map string may have a maximum of 64 characters.

6. Enter the DTMF transformation to apply to the signalled DTMFs before using it as call destination in the Transformation column.

The following are the rules you must follow; “x” represents the signalled number.

• Add before “x” the DTMF to prefix or/and after “x” the suffix to add. Characters “0123456789*# ABCD” are allowed.

• Use a sequence of DTMFs between “{}” to remove a prefix/suffix from the dialed number if present. Use before “x” to remove a prefix and after “x” to remove a suffix. Characters “0123456789*#ABCD” are allowed.

• Use a number between “()” to remove a number of DTMFs. Use before “x” to remove DTMFs at the beginning of the number and after “x” to remove DTMFs at the end. Characters “0123456789” are allowed.

The transformations are applied in order from left to right.

Table 185: DTMF Map Entity


Unit The DTMF map entry applies to the unit.

Endpoint The DTMF map applies to a specific endpoint. The endpoint is specified in the Endpoint column of the same row.

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The following table gives an example with “18195551111#” as signalled number.

7. Define the target to use when the DTMF map matches in the Target column.

This allows associating a target (FQDN) with a DTMF map. This defines a destination address to use when the DTMF map matches. This address is used as destination for the INVITEs in place of the “home domain proxy”. This is useful for such features as the speed dial and emergency call.

The default target is used when the value is empty.

The dialed DTMFs are not used if the target contains a user name.

8. Enable/Disable the emergency process of the call in the Emergency column.

• Disable: The call is processed normally.

• Enable: The call is processed as emergency.

The Emergency Call service (also called urgent gateway) allows a “911”-style service. It allows a user to dial a special DTMF map resulting in a message being sent to a specified urgent gateway, bypassing any other intermediaries.

If enabled, whenever the user dials the specified DTMF map, a message is sent to the target address.


Table 186: DTMF Map Transformation Examples

Action Transformation Result

Add the prefix “0” to the dialed number 0x 018195551111#

Remove the suffix “#” from the dialed number

x{#} 18195551111

Remove the first four DTMFs from the dialed number

(4)x 5551111#

Remove the international code and termination and replace the area code by another one

(1){819}514x{#} 5145551111

Replace the signalled DTMFs by “3332222” 3332222 3332222

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Chapter 28 - DTMF Maps Configuration Refused DTMF Maps

Refused DTMF Maps

A refused DTMF map forbids to call specific numbers; for instance, you want to accept all 1-8xx numbers except 1-801. You can create/edit ten refused DTMF maps for the Mediatrix Boss.

A refused DTMF map applies before an allowed DTMF map.

To set up refused DTMF maps:

1. In the DTMF Map drop-down menu at the top of the window, select Refused.

The Refused DTMF Map section displays.

2. In the Refused DTMF Map section – Enable column, enable one or more DTMF maps by selecting the corresponding Enable choice.

Figure 165: Refused DTMF Map Section

3. Select the entity to which apply the refused DTMF map in the Apply to column.



5. Define the DTMF map string that is considered valid when dialed in the DTMF Map column.

The string must use the syntax described in “DTMF Maps Configuration” on page 365. A DTMF map string may have a maximum of 64 characters.


Table 187: DTMF Map Entity


Unit The DTMF map entry applies to the unit.

Endpoint The DTMF map applies to a specific endpoint. The endpoint is specified in the Endpoint column of the same row.

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C H A P T E R

29 Voice & Fax CodecsConfiguration

This chapter describes the voice and fax codec configuration parameters.

Codec descriptions.

How to enable and disable the codecs.

How to set secure communication parameters.

How to set the individual codecs’ parameters.

Codec Descriptions

The Mediatrix Boss supports several voice and fax codecs. It also supports unicast applications, but not multicast ones. All voice transport is done over UDP.

All the endpoints of the Mediatrix Boss can simultaneously use the same codec (for instance, G.711 PCMA), or a mix of any of the supported codecs. Set and enable these codecs for each endpoint.

G.711 A-Law and µ-Law

The audio data is encoded as 8 bits per sample, after logarithmic scaling.

Standards Supported • RFC 3550: RTP: A Transport Protocol for Real-Time Applications

• RFC 3551: RTP Profile for Audio and Video Conferences with Minimal Control

Table 188: Codecs Comparison

Compression Voice Quality

G.711 None Excellent

G.723.1 Highest Good

G.726 Medium Fair

G.729a/ab High Fair/Good

Standards Supported • ITU-T Recommendation G.711

Table 189: G.711 Features

Feature Description

Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “G.711 Codec Parameters” on page 394 for more details.

For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the stream is not encrypted (SRTP).

Voice Activity Detection (VAD) Two levels of detection are available: transparent or conservative. See “Miscellaneous Parameters” on page 380 for more details.

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G.723.1

Dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s. This Recommendation specifies a coded representation that can be used to compress the speech signal component of multi-media services at a very low bit rate. The audio is encoded in 30 ms frames.

A G.723.1 frame can be one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4 octets. These 4-octet frames are called SID frames (Silence Insertion Descriptor) and are used to specify comfort noise parameters.

G.726

Algorithm recommended for conversion of a single 64 kbit/s A-law or U-law PCM channel encoded at 8000 samples/s to and from a 40, 32, 24, or 16 kbit/s channel. The conversion is applied to the PCM stream using an Adaptive Differential Pulse Code Modulation (ADPCM) transcoding technique.

Comfort noise Uses custom comfort noise as defined in RFC 3389.

Available for voice Yes

Available for fax Yes

Available for modem Yes

Table 189: G.711 Features (Continued)

Feature Description

Standards Supported • ITU-T Recommendation G.723.1

Table 190: G.723.1 Features

Feature Description



Voice Activity Detection (VAD) Supports the annex A, which is the built-in support of VAD in G.723.1.

Payload type 4


Available for fax No

Available for modem No

Standards Supported • ITU-T Recommendation G.726: 40, 32, 24, 16 kbit/s adaptive differential pulse code modulation (ADPCM)


Feature Description

Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “G.726 Codecs Parameters” on page 397 for more details.


Voice Activity Detection (VAD) Two levels of detection are available: transparent or conservative. See “Miscellaneous Parameters” on page 380 for more details.

Comfort noise Uses custom comfort noise as defined in RFC 3389.

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G.729

Coding of speech at 8 kbit/s using conjugate structure-algebraic code excited linear prediction (CS-ACELP). For all data rates, the sampling frequency (and RTP timestamp clock rate) is 8000 Hz.

A voice activity detector (VAD) and comfort noise generator (CNG) algorithm in Annex B of G.729 is recommended for digital simultaneous voice and data applications; they can be used in conjunction with G.729 or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, while the G.729 Annex B comfort noise frame occupies 2 octets.

The Mediatrix Boss supports G.729A and G.729AB for encoding and G.729, G.729A and G.729AB for decoding.

Clear Mode

The Clear Mode codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload (no encoding or decoding). The Clear Mode codec thus does not have echo cancellation and a fix jitter buffer. Clear Mode is a method to carry 64 kbit/s channel data transparently in RTP packets. This codec always uses the RTP transport.

Payload type Configurable as per “G.726 Codecs Parameters” on page 397.


Available for fax Yes (32 kbps and 40 kbps)

Available for modem Yes (32 kbps and 40 kbps)

Table 191: G.726 Features (Continued)

Feature Description

Standards Supported • ITU-T Recommendation G.729


Feature Description


For reception, the range is extended from 10 ms to 100 ms with increments of 10 ms only if the stream is not encrypted (SRTP).

Voice Activity Detection (VAD) Supports the annex B, which is the built-in support of VAD in G.729. See “G.729 Codec Parameters” on page 399 for more details.

Payload type 18


Available for fax No


Standards Supported • RFC 4040: RTP Payload Format for a 64 kbit/s Transparent Call

Table 193: Clear Mode Features

Feature Description

Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “Clear Mode Codec Parameters” on page 400 for more details.


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Clear Channel

The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload (no encoding or decoding). The Clear Channel codec thus does not have echo cancellation and a fix jitter buffer. Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP packets. The Clear Channel codec follows the specification of RFC 4040 and uses the “X-CLEAR-CHANNEL” mime type instead of the “CLEARMODE” mime type.

This codec always uses the RTP transport.

Voice Activity Detection (VAD) N/A

Comfort noise N/A

Payload type Configurable as per “Clear Mode Codec Parameters” on page 400.




Table 193: Clear Mode Features (Continued)

Feature Description


Table 194: Clear Channel Features

Feature Description

Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “Clear Channel Codec Parameters” on page 402 for more details.



Comfort noise N/A

Payload type Configurable as per “Clear Channel Codec Parameters” on page 402.




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X-CCD Clear Channel

The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload (no encoding or decoding). The X-CCD Clear Channel codec thus does not have echo cancellation and a fix jitter buffer. The X-CCD Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP packets. The Clear Channel codec follows the specification of RFC 4040 and uses the “X-CCD” mime type instead of the “CLEARMODE” mime type.

This codec always uses the RTP transport.

T.38

T.38 fax relay is a real-time fax transmission; that is, two fax machines communicating with each other as if there were a direct phone line between the two. T.38 is called a fax relay, which means that instead of sending inband fax signals, which implies a loss of signal quality, it sends those fax signals out-of-band in a T.38 payload, so that the remote end can reproduce the signal locally.

T.38 is an unsecure protocol, thus will not be used along with secure RTP (SRTP), unless the Allow Unsecure T.38 with Secure RTP parameter has been set to Enable. See “Security” on page 391 for more details.


Table 195: X-CCD Clear Channel Features

Feature Description

Packetization time Range of 10 ms to 100 ms with increments of 1 ms. See “X-CCD Clear Channel Codec Parameters” on page 404 for more details.


Comfort noise N/A

Payload type Configurable as per “X-CCD Clear Channel Codec Parameters” on page 404.




Standards Supported • ITU-T Recommendation T.38 version 0

Table 196: T.38 Features

Feature Description

Packetization time N/A


Payload type N/A

Available for voice No



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Chapter 29 - Voice & Fax Codecs Configuration Codec Parameters

Codec Parameters

The Codec section allows you to enable or disable the codecs of the Mediatrix Boss, as well as access the codec-specific parameters.



Specific configurations that override the default configurations. You can define specific configurations for each endpoint in your Mediatrix Boss.

To enable or disable the codecs:

1. In the web interface, click the Telephony link, then the CODECS sub-link.

Figure 166: Telephony – Codecs Web Page

2. Select to which endpoint (interface) you want to apply the changes in the Select Endpoint drop-down menu at the top of the window.

You have the choice between Default and the interfaces of your Mediatrix Boss. The number of interfaces available vary depending on the Mediatrix Boss model you have.

You can also perform this operation in the codec-specific pages.

3. Enable one or more codecs for voice transmission by selecting Enable in the Voice column of the corresponding codec(s).

This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as supported for this specific endpoint. Otherwise, it is ignored.

Standards Supported • draft-choudhuri-sip-info-digit-00

• ITU-T Recommendation Q.24: Multifrequency push-button signal reception

• RFC 2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

• RFC 1890: RTP Profile for Audio and Video Conferences with Minimal Control

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4. Enable one or more codecs for data transmission by selecting Enable in the Data column of the corresponding codec(s).

This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as supported for this specific endpoint. Otherwise, it is ignored.


5. Click the button to access the corresponding codec-specific parameters.

These parameters are described in the following sections.


Codec vs. Bearer Capabilities Mapping

The Codec vs. Bearer Capabilities Mapping section allows you to select the codec to prioritize or select in the outgoing INVITE when the incoming SETUP’s ITC (Information Transfer Capability) matches the configured one. On the other hand, you can also select the ITC value to set in the outgoing SETUP’s bearer capabilities when the incoming INVITE's codec matches the configured one.

Depending on the mapping type, a codec is prioritized or selected. Prioritized means it is the first in the list of offered codecs when initiating a call on the IP side and Selected means it is the only codec offered.

This parameter is available only on ISDN interfaces.

You can define up to three mappings.

To set the codec vs. bearer capabilities mapping

1. Define if the outgoing codec’s priority should reflect the incoming ITC and vice versa in the Enable column drop-down menu.

Figure 167: Codec vs. Bearer Capabilities Mapping Section

Table 197: Outgoing Codec Priority


Disable The mapping is not applied:

• The codec’s order in the outgoing INVITE follows the codec priority.

• The ITC in the outgoing SETUP is the default one (3.1 kHz) unless a mapping in the Call Routing table modifies it.

Enable If the ITC value in the incoming SETUP matches the value of the corresponding ITC drop-down menu, the first codec in the outgoing INVITE is the one set in the CODEC drop-down menu.

If the first codec in the incoming INVITE matches the value set in the CODEC drop-down menu, the ITC value in the outgoing SETUP is the one set in the ITC drop-down menu.

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2. Select a codec in the CODEC column drop-down menu.

This is the codec to be prioritized or selected in an outgoing INVITE when the incoming SETUP's ITC matches the value set in the corresponding ITC drop-down menu. This codec is also checked against an incoming INVITE's priority codec. If it matches, then the outgoing SETUP's ITC is set to the value in the corresponding ITC drop-down menu.

See Step 3 for a description of prioritization versus selection of a codec.

3. Set the mapping type of the codec in the Mapping Type drop-down value.

4. Select an ITC value in the ITC column drop-down menu.

This is the ITC value to be set in the outgoing SETUP when the incoming INVITE's priority codec matches the value of the corresponding CODEC drop-down menu. This value is also checked against an incoming SETUP's bearer capabilities. If it matches, then the outgoing INVITE's prioritized or selected codec is set to the value of the corresponding CODEC drop-down menu.

For an incoming ISDN call using UDI (Unrestricted Digital), the Codec vs Bearer Capabilities Mapping entries with the ITC set to Unrestricted is used only if the codec is Clear Mode, Clear Channel, XCCD, G.711 a-law or G.711 u-law. G.711 a-law and G.711 u-law are also used only if they match the ISDN port Preferred Encoding Scheme value (see “PRI Configuration” on page 232 and “BRI Configuration” on page 239 for more details).

See Step 3 for a description of prioritization versus selection of a codec.


Miscellaneous Parameters

The Misc section allows you to configure parameters that apply to all codecs.

To set the miscellaneous codec parameters:



Table 198: Codec Mapping Type


Prioritize The codec is set on top of the list in an outgoing INVITE when the incoming SETUP's ITC matches the value set in the corresponding ITC drop-down menu.

Select The codec is the only one offered in an outgoing INVITE when the incoming SETUP's ITC matches the value set in the corresponding ITC drop-down menu.

Table 199: Information Transfer Capability Values

Value Description

speech Voice terminals (telephones).

unrestricted Unrestricted digital information (64 kbps).

3.1Khz Transparent 3.1 kHz audio channel.

Standards Supported • draft-choudhuri-sip-info-digit-00.txt

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2. If you have selected a specific endpoint, select whether or not you want to override the jitter buffer parameters set in the Default configuration in the Endpoint Specific drop-down menu (Jitter Buffer sub-section of the Misc section).

This menu is available only in the specific endpoints configuration.

Figure 168: Misc Section

3. Select the jitter buffer level in the Level drop-down menu.

Jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval between successive pulses or the frequency or phase of successive cycles. An adaptive jitter buffer usually consists of an elastic buffer in which the signal is temporarily stored and then retransmitted at a rate based on the average rate of the incoming signal.

Jitter Buffer Configuration

Table 200: Jitter Buffer Levels

Level Description

Optimize Latency The jitter buffer is set to the lowest effective value to minimize the latency. Voice cut can be heard if the network is not optimal. The predefined values are as follows:

• Minimum value: 10 ms

• Maximum value: 40 ms

Normal The jitter buffer tries to find a good compromise between the latency and the voice quality. This setting is recommended in private networks. The predefined values are as follows:



Optimize Quality The jitter buffer is set to a high value to minimize the voice cuts at the cost of high latency. This setting is recommended in public networks. The predefined values are as follows:



4

3

10

2

13

5

11

1415

1817

16

1920

87

6

9

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4. If you have selected the Custom level, define the target jitter buffer length in the Minimum field of the Voice Call part.

The adaptive jitter buffer attempts to hold packets to the minimal holding time. This is the minimal delay the jitter buffer adds to the system. The minimal jitter buffer is in ms and must be equal to or smaller than the maximal jitter buffer.

Values range from 0 ms to 135 ms. The default value is 30 ms. You can change values by increments of 1 ms, but Media5 recommends to use multiples of 5 ms. The minimal jitter buffer should be a multiple of ptime.

It is best not to set the minimal jitter value below the default value. Setting a minimal jitter buffer below 5 ms could cause an error. Jitter buffer adaptation behaviour varies from one codec to another. See “About Changing Jitter Buffer Values” on page 387 for more details.

5. If you have selected the Custom level, define the maximum jitter buffer length in the Maximum field of the Voice Call part.

This is the highest delay the jitter buffer is allowed to introduce. The jitter buffer length is in ms and must be equal to or greater than the minimum jitter buffer.

Values range from 0 ms to 135 ms. The default value is 125 ms. You can change values by increments of 1 ms, but Media5 recommends to use multiples of 5 ms. The maximal jitter buffer should be a multiple of ptime.

The maximum jitter buffer value should be equal to the minimum jitter buffer value + 4 times the ptime value. Let’s say for instance that:

• Minimum jitter buffer value is 30 ms

• Ptime value is 20 ms

The maximum jitter buffer value should be: 30 + 4x20 = 110 ms

6. If you have selected the Custom level, define the voiceband data custom jitter buffer type in the Playout Type drop-down menu of the Data Call part.

This is the algorithm to use for managing the jitter buffer during a call. The Nominal field value serves as the delay at the beginning of the call and might be adapted afterwards based on the selected algorithm.

Fax / Modem The jitter buffer is set to maximum. The Fax/Modem transmission is very sensitive to voice cuts but not to latency, so the fax has a better chance of success with a high buffer. The predefined values are as follows:



Custom The jitter buffer uses the configuration of the Minimum and Maximum variables (Steps 4 and 5).

Voiceband Data Jitter Buffer Configuration

Table 201: Voiceband Data Custom Jitter Buffer Type

Level Description

Adaptive Immediately

The nominal delay varies based on the estimated packet jitter. Playout adjustment is done immediately when the actual delay goes out of bounds of a small window around the moving nominal delay.

Table 200: Jitter Buffer Levels (Continued)

Level Description

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7. If you have selected the Custom level, define the voiceband data jitter buffer minimal length (in milliseconds) in the Minimum field of the Data Call part.

The voiceband data jitter buffer minimal length is the delay the jitter buffer tries to maintain. The minimal jitter buffer MUST be equal to or smaller than the voiceband data maximal jitter buffer.

The minimal jitter buffer should be a multiple of ptime.

This value is not available when the Playout Type drop-down menu is set to Fixed.

8. If you have selected the Custom level, define the voiceband data custom jitter buffer nominal length in the Nominal field of the Data Call part.

The jitter buffer nominal length (in milliseconds) is the delay the jitter buffer uses when a call begins. The delay then varies depending on the type of jitter buffer.

In adaptive mode, the nominal jitter buffer should be equal to (voice band data minimal jitter buffer + voice band data maximal jitter buffer) / 2.

9. If you have selected the Custom level, define the default voiceband data custom jitter buffer maximal length in the Maximum field of the Data Call part.

The jitter buffer maximal length (in milliseconds) is the highest delay the jitter buffer is allowed to introduce. The maximal jitter buffer MUST be equal to or greater than the minimal jitter buffer.

The maximal jitter buffer should be a multiple of ptime.

The maximal jitter buffer should be equal to or greater than voiceband data minimal jitter buffer + (4 * ptime) in adaptive mode.

See “About Changing Jitter Buffer Values” on page 387 for more details.

10. In the DTMF Transport sub-section of the Misc section, select whether or not you want to override the DTMF transport parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Adaptive Silence The nominal delay varies based on the estimated packet jitter. Playout adjustment is done based on the actual delay going out of bounds of a small window around the moving nominal delay. The adjustment is deferred until silence is detected (either from playout buffer underflow or by analysis of packet content). Playout adjustment is also done when overflow or underflow events occur.

Fixed The nominal delay is fixed to the value of the Nominal field value and does not change thereafter. Playout adjustment is done when overflow or underflow events occur.

DTMF Transport Configuration

Table 201: Voiceband Data Custom Jitter Buffer Type (Continued)

Level Description

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11. Select the DTMF transport type in the Transport Method drop-down menu.

The following choices are available:

12. If you have selected the Out-of-band using SIP transport method, select the method used to transport DTMFs out-of-band over the SIP protocol in the SIP Transport Method drop-down menu.

This menu is available only in the default endpoint configuration.

Table 202: DTMF Transport Type Parameters

Transport Parameter Description

In-band The DTMFs are transmitted like the voice in the RTP stream.

Out-of-band using RTP The DTMFs are transmitted as per RFC 2833. This parameter also works with SRTP.

Out-of-band using SIP The DTMFs are transmitted as per draft-choudhuri-sip-info-digit-00.

Signaling protocol Dependant The signalling protocol has the control to select the DTMF transport mode. The SDP body includes both RFC 2833 and draft-choudhuri-sip-info-digit-00 in that order of preference.

Table 203: DTMF Out-of-Band Transport Methods

Method Description

draftChoudhuriSipInfoDigit00 Transmits DTMFs by using the method defined in draft-choudhuri-sip-info-digit-00. Only the unsolicited-digit part is supported.

Info DTMF Relay Transmits DTMFs by using a custom method. This custom method requires no SDP negotiation and assumes that the other peer uses the same method.

It uses a SIP INFO message with a content of type application/dtmf-relay. The body of the message contains the DTMF transmitted and the duration of the DTMF:

Signal= 1

Duration= 160

When transmitting, the duration is the one set in the interopDtmfTransportDuration variable (see “DTMF Transport over the SIP Protocol” on page 388).

When receiving, the duration of the DTMF received is not used.

DTMFs are transmitted one at a time.

Available digits are “0123456789ABCD*#”. The Mediatrix Boss also supports the “,;p” characters when receiving DTMFs.

DTMF out-of-band

Certain compression codecs such as G.723.1 and G.729 effectively distort voice because they lose information from the incoming voice stream during the compression and decompression phases. For normal speech this is insignificant and becomes unimportant. In the case of pure tones (such as DTMF) this distortion means the receiver may no longer recognize the tones. The solution is to send this information as a separate packet to the other endpoint, which then plays the DTMF sequence back by re-generating the true tones. Such a mechanism is known as out-of-band DTMF. The Mediatrix Boss receives and sends out-of-band DTMFs as per ITU Q.24. DTMFs supported are 0-9, A-D, *, #.

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13. If you have selected the Out-of-band using RTP transport method, set the payload type in the Payload Type field.

You can determine the actual RTP dynamic payload type used for the “telephone-event” in an initial offer. The payload types available are as per RFC 1890. Available values range from 96 to 127.

14. In the Generic Voice Activity Detection (VAD) sub-section of the Misc section, select whether or not you want to override the VAD parameters set in the Default configuration in the Endpoint Specific drop-down menu.


15. Enable the G.711 and G.726 Voice Activity Detection (VAD) by selecting the proper setting in the Enable (G711 and G726) drop-down menu.

The difference between transparent and conservative is how “aggressive” the algorithm considers something as an inactive voice and how “fast” it stops the voice stream. A setting of conservative is a little bit more aggressive to react to silence compared to a setting of transparent.

VAD defines how the Mediatrix Boss sends information pertaining to silence. This allows the unit to detect when the user talks, thus avoiding to send silent RTP packets. This saves on network resources. However, VAD may affect packets that are not really silent (for instance, cut sounds that are too low). VAD can thus slightly affect the voice quality.

16. In the Machine Detection sub-section of the Misc section, select whether or not you want to override the machine detection parameters set in the Default configuration in the Endpoint Specific drop-down menu.


17. Select whether or not you want to enable fax calling tone (CNG tone) detection in the CNG Tone Detection drop-down menu.

Generic VAD Configuration

Table 204: G.711/G.726 VAD Settings

Setting Description

Disable VAD is not used.

Transparent VAD is enabled. It has low sensitivity to silence periods.

Conservative VAD is enabled. It has normal sensitivity to silence periods.

Machine Detection Configuration

Table 205: CNG Tone Detection Settings

Setting Description

Enable Upon recognition of the CNG tone, the unit switches the communication from voice mode to fax mode and the CNG is transferred by using the preferred fax codec.

Note: This option allows for quicker fax detection, but it also increases the risk of false detection.

Disable The CNG tone does not trigger a transition from voice to data and the CNG is transferred in the voice channel.

Note: With this option, faxes are detected later, but the risk of false detection is reduced.

Mediatrix Boss 385


18. Select whether or not you want to enable CED tone detection in the CED Tone Detection drop-down menu.

19. Select whether or not you want to enable fax V.21 modulation detection in the V.21 Modulation Detection drop-down menu.

20. Define the behaviour of the unit upon detection of a CED tone in the Behavior on CED Tone Detection drop-down menu.

21. In the Base Ports sub-section of the Misc section, set the UDP port number you want to use as RTP/RTCP base port in the RTP field.

The RTP/RTCP ports are allocated starting from this base port.

RTP ports number are even and RTCP ports number are odd.

The default RTP/RTCP base port is 5004. For instance, assuming that the base port is defined on 5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the RTP/RTCP ports 5004 and 5005.

22. Set the port number you want to use as T.38 base port in the T.38 field.

The T.38 ports are allocated starting from this base port.

The default T.38 base port is 6004. For instance, assuming that the base port is defined on 6004 if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the T.38 port 6005.

Table 206: CNG CED Detection Settings

Setting Description

Enable Upon recognition of the CED tone, the unit behaves as defined in the Behavior on CED Tone Detection parameter Step ???).

Disable The CED tone does not trigger a transition to fax or voiceband data mode. The CED is transferred in the voice channel.

Table 207: V.21 Modulation Detection Settings

Setting Description

Enable Upon recognition of the V.21 modulation tone, the unit switches the communication from voice mode to fax mode and the signal is transferred by using the preferred fax codec.

Disable The V.21 modulation does not trigger a transition from voice to data and the signal is transferred in the voice channel.

Table 208: CED Tone Detection Settings

Setting Description

Passthrough The CED tone triggers a transition from voice to voice band data and is transferred in the voice channel.. Use this setting when any kind of analog device (i.e.: telephone, fax or modem) can be connected to this port.

Fax Mode Upon detection of a CED tone, the unit switches the communication from voice mode to fax mode and the CED is transferred by using the preferred fax codec. Only a fax can then be connected to this port.

Note: This parameter has no effect if the CED Tone Detection parameter is set to Disabled.

Base Ports Configuration

386 Mediatrix Boss


23. Set the UDP port number you want to use as SRTP/SRTCP base port in the SRTP field.

The SRTP/SRTCP ports are allocated starting from this base port.

SRTP ports number are even and SRTCP ports number are odd.

The default SRTP/SRTCP base port is 5004. For instance, assuming that the base port is defined on 5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the SRTP/SRTCP ports 5004 and 5005.

Using the same base port for RTP/RTCP and SRTP/SRTCP does not conflict.

Note that if the media transport is set to “Secure with fallback” (“Security” on page 391), both RTP and SRTP base ports are used at the same time when initiating an outgoing call. If there is currently no call and the default base ports are used, the RTP port is 5004 and the SRTP port is the next available port starting from the base port, which is 5006.



About Changing Jitter Buffer Values

Media5 recommends to avoid changing the target and maximum jitter buffer values unless experiencing or strongly expecting one of the following symptoms:

If the voice is scattered, try to increase the maximum jitter buffer value.

If the delay in the voice path (end to end) is too long, you can lower the target jitter value, but ONLY if the end-to-end delay measured matches the target jitter value.

For instance, if the target jitter value is 50 ms, the maximum jitter is 300 ms and the delay measured is 260 ms, it would serve nothing to reduce the target jitter. However, if the target jitter value is 100 ms and the measured delay is between 100 ms and 110 ms, then you can lower the target jitter from 100 ms to 30 ms.

Starting a Call in Voiceband Data Mode


This section describes configuration that is available only in the MIB parameters of the Mediatrix Boss. You can configure these parameters as follows:

by using a MIB browser

by using the CLI

by creating a configuration script containing the configuration variables

You can define whether or not a call should be started in voiceband data mode.


To start a call in voiceband data mode:

1. Set the voiceband data mode by putting the following line in the configuration script:telIf.InteropStartCallInVbdEnable="Value"

Table 209: Voiceband Data Mode Parameters


Disable The call is started in voice mode. A fax/modem tone detection triggers a transition from voice to voiceband data according to the configuration in the Machine Detection Group (“Miscellaneous Parameters” on page 380).

Enable The call is started in voiceband data mode.

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DTMF Transport over the SIP Protocol


You can select the method used to transport DTMFs out-of-band over the SIP protocol.

This feature is effective only if the Transport Method drop-down menu is set to Out-of-band using Signaling protocol (see “Miscellaneous Parameters” on page 380 for more details).

The following methods are available:

To select the DTMF transport method over the SIP protocol:

1. Set the DTMF transport type by putting the following line in the configuration script:sipEp.interopDtmfTransportMethod="Value"


2. Set the DTMF duration sent in the INFO message when using the infoDtmfRelay method to transmit DTMFs by putting the following line in the configuration script:

sipEp.interopDtmfTransportDuration="Value"

Table 210: Voiceband Data Mode Values

Value Method

0 Disable

1 Enable

Standards Supported • draft-choudhuri-sip-info-digit-00.txt

Table 211: DTMF Out-of-Band Transport Methods

Method Description

draftChoudhuriSipInfoDigit00 Transmits DTMFs by using the method defined in draft-choudhuri-sip-info-digit-00. Only the unsolicited-digit part is supported.

infoDtmfRelay Transmits DTMFs by using a custom method. This custom method requires no SDP negotiation and assumes that the other peer uses the same method.

It uses a SIP INFO message with a content of type application/dtmf-relay. The body of the message contains the DTMF transmitted and the duration of the DTMF:

Signal= 1

Duration= 160

When transmitting, the duration is the one set in the interopDtmfTransportDuration variable (see Step 2 below).

When receiving, the duration of the DTMF received is not used.

DTMFs are transmitted one at a time.

Available digits are “0123456789ABCD*#”. The Mediatrix Boss also supports the “,;p” characters when receiving DTMFs.

Table 212: DTMF Out-of-Band Transport Values

Value Method

100 draftChoudhuriSipInfoDigit00

200 infoDtmfRelay

388 Mediatrix Boss


This value is expressed in milliseconds (ms). The default value is 100 ms.

DTMF Detection


The default DTMF detection parameters of the Mediatrix Boss may sometimes not be enough to properly detect the DTMFs. This section describes how to set additional DTMF detection parameters.

DTMF Frequencies

The DTMF keypad is laid out in a 4x4 matrix, with each row representing a low frequency, and each column representing a high frequency. For example, pressing a single key (such as '1') sends a sinusoidal tone of the two frequencies (697 Hz and 1209 Hz). When the unit is configured to send DTMFs out-of-band, its DSP detects these DTMFs, removes them from the RTP stream, and sends them out-of-band.

DTMF Detection Configuration

Below is a frequency spectrum analysis of a DTMF (9) with the Frequency in Hertz on the x axis and the Power in dBm on the y axis. The low and high frequencies of the DTMF are in red and you can clearly see that they are the most powerful frequencies in the signal.

Figure 169: DTMF Detection Example

Table 213: DTMF Keypad Frequencies

Low/High (Hz) 1209 1336 1477 1633

697 1 2 3 A

770 4 5 6 B

852 7 8 9 C

941 * 0 # D

Mediatrix Boss 389


To configure the DTMF detection:

1. Set the DTMF transport type by putting the following line in the configuration script:sipEp.interopDtmfDetectionRiseTimeCriteria="Value"


2. Set the interopDtmfDetectionPositiveTwist variable by putting the following line in the configuration script:

sipEp.interopDtmfDetectionPositiveTwist="Value"

When the high-group frequency of a DTMF is more powerful than the low-group frequency, the difference between the high-group frequency absolute power and the low-group frequency absolute power must be smaller than or equal to the value set in this variable. Otherwise, the DTMF is not detected.

Raising this value increases the sensitivity of DTMF detection. Raising this value too high may also cause false detections of DTMFs.

Using the Payload Type Found in the Answer


The default behaviour when sending an initial offer that contains an RFC 2833 payload type is to keep using that payload type even if the response comes back with a different one. You can set the Mediatrix Boss to rather use the payload type found in the answer.

This feature is effective only if the Transport Method drop-down menu is set to Out-of-band using RTP (see “Miscellaneous Parameters” on page 380 for more details).


To use the payload type found in the answer:

1. Set the proper payload type behaviour by putting the following line in the configuration script:sipEp.interopUseDtmfPayloadTypeFoundInAnswer="Value"

Table 214: DTMF Detection Values

Value Method

100 CheckSr Enables the Step Rise criteria and disables the Confirm DTMF SNR criteria.

The Step Rise criteria compares the current frame energy to the high frequency power of the previous frame. If the current frame energy is high enough, then it passes the test, further validating the DTMF.

Disabling the Step Rise criteria may result in deteriorated talk-off performance, but increases the detection of malformed DTMF.

200 ConfirmSnr Enable the Confirm DTMF SNR criteria and disable the Step Rise criteria.

The Confirm DTMF SNR criteria is an additional Signal-to-noise ratio test performed before a confirmed DTMF report is sent to finally validate the DTMF.

Table 215: Payload Type in Answer


disable Keep using the initial payload type. This is the default value.

enable Use the RFC 2833 payload type found in the received answer.

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Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP


You can specify the quantity of packets sent at the beginning of an Out-of-Band DTMF using RTP. This variable also specifies the quantity of terminating packets that are sent at the end of the DTMF transmission.

Note that this variable has an effect only if the Transport Method drop-down menu is set to Out-of-band using RTP (see “Miscellaneous Parameters” on page 380 for more details).

To set the initial quantity of RTP packets:

1. Set the proper quantity by putting the following line in the configuration script:mipt.interopDtmfRtpInitialPacketQty="Value"

where Value may be between 1 and 3.

Security

You can define security features on the Mediatrix Boss. This section applies to media security parameters. Applying security on the Mediatrix Boss involves several steps:

Transfer a valid CA certificate into the Mediatrix Boss (“Chapter 39 - Certificates Management” on page 527).

Use secure signalling by enabling the TLS transport protocol (“Chapter 25 - SIP Transport Parameters” on page 331).

Use secure media by:

• Defining the SRTP/ SRTCP base port (“Miscellaneous Parameters” on page 380).

• Setting the RTP secure mode to “Secure” or “Secure with fallback” (this section).

Figure 170: Payload Type Values

Value Meaning

0 disable

1 enable

Standards Supported • RFC 3711: The Secure Real-time Transport Protocol (SRTP) (Supports only the AES-CM encryption)

• RFC 3830: MIKEY: Multimedia Internet KEYing (Compliant for method Pre-Shared Key only)

• RFC 4567: Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)

• RFC 4568: SDES: Security Descriptions for Media Streams

Caution: If you enable Secure RTP (SRTP) on at least one line, it is acceptable to have the secure SIP transport (TLS) disabled for testing purposes. However, you must never use this configuration in a production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly recommended and is usually mandatory for security interoperability with third-party equipments.

Caution: When using a codec other than G.711, enabling Secure RTP (SRTP) has an impact on the Mediatrix Boss’s overall performance as SRTP requires CPU power. The more lines use SRTP, the more overall performance is affected. This could mean that a user picking up a telephone on these models may not have a dial tone due to lack of resources.

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Chapter 29 - Voice & Fax Codecs Configuration Security

The Security section allows you to secure the RTP stream (media) of the Mediatrix Boss.

Since the SRTP encryption and authentication needs more processing, the number of calls that the Mediatrix Boss can handle simultaneously may be reduced, depending of the codecs enabled. You could set the Mediatrix Boss not to impact the number of simultaneous calls by enabling only G.711 codecs and disabling every other voice or data codec, even T.38.

The Mediatrix Boss supports the MIKEY protocol using pre-shared keys (MIKEY-PS) or the SDES protocol for negotiating SRTP keys.

To set the RTP stream security parameters:



2. In the Security section of the Codecs page, select the RTP payload mode in the Mode drop-down menu.

Figure 171: Security Section

The unit relies on these modes when negotiating an audio stream.

The TLS SIP transport must usually be enabled for secure audio negotiation via SDP (refer to the Caution box above). See “Chapter 25 - SIP Transport Parameters” on page 331 for more details.

The RTP mode is reflected in the SIP/SDP payload, with a RTP/AVP for unsecure RTP, and a RTP/SAVP for secure RTP.

The following basic rules apply when sending units capabilities via SDP:

• When the RTP mode is set to Unsecure, the Mediatrix Boss offers/answers with only one active RTP/AVP audio stream. Any other audio stream present in the offer is disabled in the answer.

• When the RTP mode is set to Secure, the Mediatrix Boss offers/answers with only one active RTP/SAVP audio stream. Any other audio stream present in the offer is disabled in the answer.

• When the RTP mode is set to Secure with fallback, the Mediatrix Boss offers one RTP/AVP and one RTP/SAVP audio streams. The unit answers with only the most secure stream.

• If the remote unit answers to an offer with both RTP/AVP and RTP/SAVP streams enabled, a new offer is sent with only RTP/SAVP enabled.

Table 216: Default RTP Mode

Mode Description

Unsecure The Mediatrix Boss supports only unsecure RTP. It rejects secure RTP offers it receives.

Secure The Mediatrix Boss supports only secure RTP. It rejects unsecure RTP offers it receives.

Secure with fallback The Mediatrix Boss supports both secure and unsecure RTP. It prioritizes secure RTP but permits unsecure RTP fallback when the remote peer does not support security.

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3. Select the key management protocol for SRTP in the Key Management drop-down menu.

This parameter has no effect if the Mode parameter is set to Unsecure.

If the unit receives an offer with both MIKEY and SDES, only the configured key management protocol is kept.

4. Select the encryption type to be used with SRTP in the Encryption drop-down menu.

This parameter has no effect if the Mode parameter is set to Unsecure.

5. Select whether or not to enable T.38 even if the call has been established previously in SRTP in the Allow Unsecure T.38 with Secure RTP drop-down menu.

Note that this parameter has no effect if the Mode parameter is set to Unsecure.


Enforcing Symmetric RTP


For each bi-directional RTP streams, you can define whether or not to enforce that incoming RTP packets are from the same source as the destination of outgoing RTP packets.

Enforcing symmetric RTP may prevent legitimate RTP streams coming from a media server from being processed, for example: Music and conferencing servers.


Table 217: Key Management Protocol


Mikey Use MIKEY (Multimedia Internet KEYing).

Sdes Use SDES (Security DEScriptions).


Encryption Description

Null No encryption. It is ignored for the Sdes Key Management as defined in Step 3. Use only for debug.

AesCm128 AES (Advanced Encryption Standard) Counter Mode 128 bits.


Mode Description

Disable T.38 is disabled for SRTP calls.

Enable T.38 is enabled for SRTP calls.

Caution: Enabling this parameter opens a security hole, because T.38 is an unsecure protocol.

Table 220: Enforce Symmetric RTP Parameters


disable Accept packets from all sources. This is the default value.

enable Silently discard incoming RTP packets with source address and port differing from the destination address and port of outgoing packets.

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Chapter 29 - Voice & Fax Codecs Configuration G.711 Codec Parameters

To enforce symmetric RTP:

1. Set the proper symmetric RTP behaviour by putting the following line in the configuration script:mipt.enforceSymmetricRtpEnable="Value"


G.711 Codec Parameters

The following are the G.711 codec parameters you can set. There are two sections for G.711:

G.711 a-law

G.711 u-law

These sections use the same parameters, so only one of them is described below.

To set the G.711 codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the corresponding G.711 codec to access the codec-specific parameters.



Figure 173: G.711 a-law Section

3. Select whether or not you want to override the G.711 parameters set in the Default configuration in the Use Endpoint Specific drop-down menu.


You can also perform this operation in the main CODEC section.

Figure 172: Symmetric RTP Values

Value Meaning

0 disable

1 enable

3

7

45

6

2

8

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G.723 Codec Parameters Software Configuration Guide

4. Enable the G.711 codec for voice transmission by selecting Enable in the Voice Transmission drop-down menu.



5. Set the default priority for voice in the Voice Priority field.

This sets the priority between different codecs. Codecs with a higher priority are used first, a priority of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority 2. The maximum priority is 10.

The Mediatrix Boss uses an internal order for codecs with the same priority.

6. Enable the G.711 codec for data transmission by selecting Enable in the Data Transmission drop-down menu.



7. Set the default priority for data in the Data Priority field.



8. Select the minimum and maximum packetization time values for the codec in the Minimum Packetization Time and Maximum Packetization Time drop-down menus.

The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice packet. The range is from 10 ms to 30 ms with increments of 10 ms.



You can also access the specific parameters of another codec by selecting the codec in the Select CODEC drop-down menu at the top of the page.


The following are the G.723 codec parameters you can set.


1. In the CODEC section of the CODECS page, click the button at the right of the G.723 codec to access the codec-specific parameters.



Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation, which decides on the codec to use.

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Chapter 29 - Voice & Fax Codecs Configuration G.723 Codec Parameters

Figure 174: G.723 Section










6. Select the G.723 bit rate in the Bit Rate drop-down menu.

You have the following choices:

• 53 Kbs

• 63 Kbs






3

7

45

6

2


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G.726 Codecs Parameters Software Configuration Guide

G.726 Codecs Parameters

The following are the G.726 codecs parameters you can set. There are four sections for G.726:

G.726 16 Kbps

G.726 24 Kbps

G.726 32 Kbps

G.726 40 Kbps

These sections offer almost the same parameters, except that you cannot use the G.726 16 Kbps and G.726 24 Kbps codecs for fax transmission.

To set the G.726 codecs parameters:








4. Enable the corresponding G.726 codec for voice transmission by selecting Enable in the Voice Transmission drop-down menu.



3

7

45

6

9

8

2

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Chapter 29 - Voice & Fax Codecs Configuration G.726 Codecs Parameters




6. Enable the codec for data transmission by selecting Enable in the Data Transmission drop-down menu.


This menu is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.





This field is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.

8. Set the G.726 actual RTP dynamic payload type used in an initial offer in the Payload Type field.

The payload types available are as per RFC 3551. The values range from 96 to 127. The default values are as follows:

9. Select the minimum and maximum packetization time values for the G.726 codec in the Minimum Packetization Time and Maximum Packetization Time drop-down menus.






Table 221: G.726 Default Payload Type

Codec Default Value

G.726 (16 kbps) 97

G.726 (24 kbps) 98

G.726 (32 kbps) 99

G.726 (40 kbps) 100

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G.729 Codec Parameters Software Configuration Guide


The following are the G.729 codec parameters you can set.


1. In the CODEC section of the CODECS page, click the button at the right of the G.729 codec to access the codec-specific parameters.




3. In the G.729 section, select whether or not you want to override the G.729 parameters set in the Default configuration in the Use Endpoint Specific drop-down menu.









3

6

45

7

2


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Chapter 29 - Voice & Fax Codecs Configuration Clear Mode Codec Parameters



For reception, the range is extended from 10 ms to 100 ms with increments of 10 ms only if the stream is not encrypted (SRTP).

7. Select the G.729 Voice Activity Detection (VAD) in the Built-in Voice Activity Detection (VAD) drop-down menu.

VAD defines how the Mediatrix Boss sends information pertaining to silence. This allows the unit to detect when the user talks, thus avoiding to send silent RTP packets. This saves on network resources. However, VAD may affect packets that are not really silent (for instance, cut sounds that are too low). VAD can thus slightly affect the voice quality.

G.729 has a built-in VAD in its Annex B version. It is recommended for digital simultaneous voice and data applications and can be used in conjunction with G.729 or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, while the G.729 Annex B frame occupies 2 octets. The CN packets are sent in accordance with annex B of G.729.



Clear Mode Codec Parameters

The following are the Clear Mode codec parameters you can set.

To set the Clear Mode codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the Clear Mode codec to access the codec-specific parameters.



Table 222: G.729 VAD


Disable G.729 uses annex A only.

Enable G.729 annex A is used with annex B. Speech frames are only sent during talkspurts (periods of audio activity). During silence periods, no speech frames are sent, but Comfort Noise (CN) packets containing information about background noise may be sent in accordance with annex B of G.729.

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Clear Mode Codec Parameters Software Configuration Guide

Figure 177: Clear Mode Section

3. Select whether or not you want to override the Clear Mode parameters set in the Default configuration in the Use Endpoint Specific drop-down menu.



4. Enable the Clear Mode codec for voice transmission by selecting Enable in the Voice Transmission drop-down menu.






6. Enable the Clear Mode codec for data transmission by selecting Enable in the Data Transmission drop-down menu.






3

7

45

6

2

8

9


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Chapter 29 - Voice & Fax Codecs Configuration Clear Channel Codec Parameters

8. Set the Clear Mode RTP dynamic payload type used in an initial offer in the Payload Type field.

The payload types available are as per RFC 3551. The values range from 96 to 127. The default value is 125.






Clear Channel Codec Parameters

The following are the Clear Channel codec parameters you can set.

To set the Clear Channel codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the Clear Channel codec to access the codec-specific parameters.



Figure 178: Clear Channel Section

3. Select whether or not you want to override the Clear Channel parameters set in the Default configuration in the Use Endpoint Specific drop-down menu.



3

7

45

6

2

8

9

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4. Enable the Clear Channel codec for voice transmission by selecting Enable in the Voice Transmission drop-down menu.






6. Enable the Clear Channel codec for data transmission by selecting Enable in the Data Transmission drop-down menu.






8. Set the Clear Channel RTP dynamic payload type used in an initial offer in the Payload Type field.








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Chapter 29 - Voice & Fax Codecs Configuration X-CCD Clear Channel Codec Parameters

X-CCD Clear Channel Codec Parameters

The following are the X-CCD Clear Channel codec parameters you can set.

To set the Clear Channel codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the X CCD codec to access the codec-specific parameters.



Figure 179: X CCD Section

3. Select whether or not you want to override the X CCD parameters set in the Default configuration in the Endpoint Specific drop-down menu.



4. Enable the X CCD codec for voice transmission by selecting Enable in the Voice Transmission drop-down menu.






3

7

45

6

2

8

9


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Fax Parameters Software Configuration Guide

6. Enable the X CCD codec for data transmission by selecting Enable in the Data Transmission drop-down menu.






8. Set the X CCD RTP dynamic payload type used in an initial offer in the Payload Type field.






Fax Parameters

The Mediatrix Boss handles G3 fax transmissions at speeds up to 14.4 kbps. Automatic fax mode detection is standard on all endpoints. Real-Time Fax Over UDP with the T.38 protocol stack is also available.

A fax call works much like a regular voice call, with the following differences:

1. The fax codec may be re-negotiated by using a re-INVITE.

2. The goal of the re-INVITE is to allow both user agents to agree on a fax codec, which is either:

a. Clear channel (G.711 or G.726) without Echo Cancellation nor Silence Suppression (automatically disabled).

b. T.38.

3. Upon fax termination, if the call is not BYE, the previous voice codec is recovered with another re-INVITE.

All endpoints of the Mediatrix Boss can simultaneously use the same codec (for instance, T.38), or a mix of any of the supported codecs. Set and enable these codecs for each endpoint.


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Chapter 29 - Voice & Fax Codecs Configuration Fax Parameters

Clear Channel Fax

The Mediatrix Boss can send faxes in clear channel. The following is a clear channel fax call flow:

Figure 180: Clear Channel Fax Call Flow

DSP Limitation

The Mediatrix Boss currently suffers from a limitation of its DSP. Because of this limitation, the voice does not switch back to the original negotiated codec after a clear channel fax is performed.

The Mediatrix Boss cannot detect the end of a clear channel fax, which means that the unit cannot switch back to the original negotiated codec if this codec was not a clear channel codec, e.g., a session established in G.729.

When the unit detects a fax, it automatically switches to a negotiated clear channel codec such as PCMU (if there is no T.38 or if T.38 negotiation failed). Once the fax is terminated, the Mediatrix Boss is not notified by the DSP. The unit thus stays in the clear channel codec and does not switch back to G.729.

T.38 Fax

The Mediatrix Boss can send faxes in T.38 mode over UDP. T.38 is used for fax if both units are T.38 capable; otherwise, transmission in clear channel over G.711 as defined is used (if G.711 µ-law and/or G.711 A-law are enabled). If no clear channel codecs are enabled and the other endpoint is not T.38 capable, the fax transmission fails.

The quality of T.38 fax transmissions depends upon the system configuration, type of call control system used, type of Mediatrix units deployed, as well as the model of fax machines used. Should some of these conditions be unsatisfactory, performance of T.38 fax transmissions may vary and be reduced below expectations.

INVITE

Ringing/Trying

ACK

[…]m=audio 5006 RTP/AVP 18 0 13a=rtpmap:18 G729/8000a=rtpmap:0 PCMU/8000

[…]m=audio 5004 RTP/AVP 18 0 13a=rtpmap:18 G729/8000a=rtpmap:0 PCMU/8000

BYE

200 OK

200 OK

RTP=G.729 (Voice Call)

Fax Tone Detected

RTP=PCMU (Echo Cancellation + Silence Suppression = disabled)

No re-INVITE!! There is no need for a re-INVITE since the far end already supports the

data codec (PCMU). When your SDP capabilities are inserted in a SIP packet, it implies that

you can receive any of these capabilities at any given time without notice. In this case, both ends should switch to clear channel automatically upon

detection of the fax transmission.

Fax is terminated

UserAgent

#1

UserAgent

#2

Caution: The Mediatrix Boss opens the T.38 channel only after receiving the “200 OK” message from the peer. This means that the Mediatrix Boss cannot receive T.38 packets before receiving the “200 OK”. Based on RFC 3264, the T.38 channel should be opened as soon as the unit sends the “INVITE” message.

Note: Media5 recommends not to use a fax that does not send a CNG tone. If you use such a fax to send a fax communication to the public network, this might result in a communication failure.

406 Mediatrix Boss


The following is a T.38 fax call flow:

Figure 181: T.38 Fax Call Flow

T.38 Parameters Configuration

The following are the T.38 codec parameters you can set.

To set the T.38 codec parameters:




INVITE

Ringing/Trying

ACK

INVITE

Trying

ACK

INVITE

[…]m=audio 5006 RTP/AVP 0 18 4 8 13a=rtpmap:0 PCMU/8000a=rtpmap:18 G729/8000a=rtpmap:4 G723/8000a=rtpmap:8 PCMA/8000


UserAgent

#1

UserAgent

#2

Trying

ACK

BYE

200 OK

200 OK

Fax Tone Detected

[…]m=image 6006 udptl t38a=T38MaxBitRate:14400a=T38FaxRateManagement:transferredTCFa=T38FaxFillBitRemoval:0a=T38FaxTranscodingMMR:0a=T38FaxTranscodingJBIG:0a=T38FaxVersion:0a=T38FaxUdpEC:t38UDPRedundancy

[…]m=image 6006 udptl t38a=T38MaxBitRate:14400a=T38FaxRateManagement:transferredTCFa=T38FaxFillBitRemoval:0a=T38FaxTranscodingMMR:0a=T38FaxTranscodingJBIG:0a=T38FaxVersion:0a=T38FaxUdpEC:t38UDPRedundancy 200 OK

Fax is terminated



200 OK

Mediatrix Boss 407


Figure 182: T.38 Section

3. In the T.38 section, select whether or not you want to override the T.38 parameters set in the Default configuration in the Use Endpoint Specific drop-down menu.



4. Enable the T.38 codec by selecting Enable in the Enable drop-down menu.


5. Set the default priority for fax in the Priority field.



6. Set the number of redundancy packets sent with the current packet in the Redundancy Level field.

This is the standard redundancy offered by T.38. Available values range from 1 to 5. Please see step 7 for additional reliability options for T.38.

7. Set the T.38 input signal detection threshold in the Detection Threshold drop-down menu.

Lowering the threshold allows detecting lower amplitude fax signals. The following values are available:

• Default: (-26 dB)

• Low: (-31 dB)

• Lowest: (-43 dB)

8. For additional reliability, define the number of times T.38 packets are retransmitted in the Frame Redundancy Level field.

This field is available only in the default endpoint configuration.

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Note: Currently, the only T.38 priority accepted is 10. Priority between 1 and 9 is refused.

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This only applies to the T.38 packets where the PrimaryUDPTL contains the following T.38 data type:

• HDLC_SIG_END,

• HDLC_FCS_OK_SIG_END,

• HDLC_FCS_BAD_SIG_END and

• T4_NON_ECM_SIG_END

9. Define whether or not the Mediatrix Boss sends no-signal packets during a T.38 fax transmission in the No Signal drop-down menu.


When enabled, the unit ensures that, during a T.38 fax transmission, data is sent out at least every time the No Signal Timeout delay expires. The Mediatrix Boss sends no-signal packets if no meaningful data have been sent for a user-specified period of time.

10. Set the period, in seconds, at which no-signal packets are sent during a T.38 transmission in the No Signal Timeout field.

This field is available only in the default endpoint configuration.

No-signal packets are sent out if there are no valid data to send.



Data Codec Selection Procedure

The Mediatrix Boss follows a procedure when selecting data codec. This procedure is the default behaviour of the Mediatrix Boss. Some interop variables may modify this procedure. Tones are detected on the analog ports only.

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Figure 183: Data Codec Selection Procedure

CNG tone detected?

CED tone detected?

V.21 tone detected?

Current voice codec is data -

capable?

List of negotiated

codecs contains a

data-capable codec?

T.38 is enabled?

Send a re-INVITEfor T.38

T.38 accepted by peer?

Yes

No

No

Yes

No

Yes

Continue call

Continue T.38 fax

Yes

Terminate call

Continue call with current codec

Switch to highest negotiated priority

data codec and continue call

Stop sending voice codec. Start

buffering T.38 packets

Yes

Continue call

Yes

Voice / Voiceband data call

Complete fax/modem codec selection procedure .

Tones are detected on the telephony ports except for the CED, which can also be detected on the IP side

Start sending T.38 packets

Send SIP re-INVITE for data-capable codecs

At least one data codec is enabled on the device ?

Yes

New codec accepted by

peer ?No

Switch to selected data

codec and continue call

Yes

Evaluate BehaviorOnT38Invite

NotAccepted.Behavior according to error

code

No

ReInviteForClearChannelOnly (default)

Disable fax/modem detection and send SIP re-INVITE for

voice-capable codecs

ReEstablishAudio

Restore the previouly used voice codec and

continue call

UsePreviousMediaNegiciation

DropCall

No Evaluate interopBehavior

OnMachineDetection

ReInviteOnFaxT38Only

ReInviteOnNoNegotiated

DataCodec

No

Switch to highest configured priority G.711

data codec(PCMU if all disabled )

No

No

Evaluate BehaviorOnCedToneDetection

Yes

PassthroughFaxmode

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C H A P T E R

30 Call Forward Configuration

This chapter describes how to set three types of Call Forward:

On Busy

On No Answer

Unconditional




Call Forward On Busy

You can automatically forward the incoming calls of your users to a pre-determined target if they are already on the line. The user does not have any feedback that a call was forwarded.

You can enable the Call Forward On Busy feature in two ways:

By allowing the user to configure the call forward activation and its destination via the handset (Steps 4-6).

By manually enabling the service (Steps 7-8).


• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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Chapter 30 - Call Forward Configuration Call Forward On Busy

To set the Call Forward On Busy feature:

1. In the web interface, click the Telephony link, then the Call Forward sub-link.

Figure 184: Telephony – Call Forward Web Page

2. Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at the top of the window.

You have the choice between Default and all FXS endpoints your Mediatrix Boss has.

3. In the Call Forward On Busy section, define whether or not you want to override the Call Forward On Busy parameters set in the Default configuration in the Endpoint Specific drop-down menu.


4. Enable the Call forward configuration via handset service by setting the Allow Activation via Handset drop-down menu to Enable.

You also need to configure the activation and deactivation DTMF maps (steps 5 and 6).

If you select Disable, this does not disable the call forward, but prevents the user from activating or deactivating the call forward service. The user will not be able to use the digits used to activate and deactivate the call forward service.

5. Define the digits that users must dial to start the service in the DTMF Map Activation field.

This field is available only in the Default configuration.

For instance, you could decide to put “*72” as the sequence to activate the service. This sequence must be unique and follow the syntax for DTMF maps (see “Chapter 28 - DTMF Maps Configuration” on page 365). Dialing this DTMF map does not have any effect unless the service’s status is “enabled”.

The activating sequence is set for all the endpoints of the Mediatrix Boss. You cannot have a different sequence for each endpoint.

6. Define the digits that users must dial to stop the service in the DTMF Map Deactivation field.


For instance, you could decide to put “*73” as the sequence to deactivate the service. This sequence must be unique and follow the syntax for DTMF maps (see “Chapter 28 - DTMF Maps Configuration” on page 365). Dialing this DTMF map does not have any effect unless the service’s status is “enabled”.

The deactivating sequence is set for all the endpoints of the Mediatrix Boss. You cannot have a different sequence for each endpoint.

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7. Set the call forward service in the Activation field to Inactive or Active.

To let the user activate or deactivate this service with his or her handset, see steps 4, 5, and 6. In that case, the field is automatically updated to reflect the activation status.

8. Define the address to which forward incoming calls in the Forwarding Address field.

Accepted formats are:

• telephone numbers (5551111)

• SIP URLs such as ”scheme:user@host”. For instance, “sip:[email protected]”.

This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be present.


Configuring Call Forward on Busy via Handset

The following is the procedure to use this service on the user’s telephone.

To forward calls:

1. Take the receiver off-hook.

2. Wait for the dial tone.

3. Dial the sequence implemented to activate the call forward on busy service.

This sequence could be something like *72.

4. Wait for the stutter dial tone (three “beeps”) followed by the dial tone.

5. Dial the number to which you want to forward your calls. Dial any access code if required.

6. Wait for three “beeps” followed by a silent pause.

The call forward is established.

7. Hang up your telephone.

To cancel the call forward:



3. Dial the sequence implemented to deactivate the call forward on busy service.


4. Wait for the transfer tone (three “beeps”) followed by the dial tone.

The call forward is cancelled.


Table 223: Activation State

State Description

Inactive The call forward service is not available on the telephone connected to the specific endpoint. A call to this endpoint is not forwarded if the endpoint is busy.

Active The call forward service is available on the telephone connected to the specific endpoint. A call to the endpoint is forwarded to the specified destination if the endpoint is busy. You must define the call forward destination in the Forwarding Address field (Step 8). The call forward service behaves as if it is inactive if the Forwarding Address is empty.

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Chapter 30 - Call Forward Configuration Call Forward On No Answer

Call Forward On No Answer

You can forward the incoming calls of your users to a pre-determined target if they do not answer their telephone before a specific amount of time. The user does not have any feedback that a call was forwarded.




To set the Call Forward On No Answer feature:



2. In the Call Forward On No Answer section, define whether or not you want to override the Call Forward On No Answer parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 185: Telephony – Call Forward on No Answer section












6. Define the time, in milliseconds, the telephone keeps ringing before the call forwarding activates in the Timeout field.

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7. Set the status of the service in the Activation field to Inactive or Active.








Configuring Call Forward on Answer via Handset


To forward calls:



3. Dial the sequence implemented to activate the call forward on no answer service.










3. Dial the sequence implemented to deactivate the call forward on no answer service.






State Description



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Chapter 30 - Call Forward Configuration Call Forward Unconditional

Call Forward Unconditional

The Call Forward Unconditional feature allows users to forward all of their calls to another extension or line.




To set the Call Forward Unconditional feature:



2. In the Unconditional section, define if you want to override the Call Forward Unconditional parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 186: Telephony – Call Forward Unconditional Section












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6. Set the status of the service in the Activation field to Inactive or Active.








Configuring Call Forward on Unconditional via Handset

When forwarding calls outside the system, a brief ring is heard on the telephone to remind the user that the call forward service is active. The user can still make calls from the telephone.

To forward calls:



3. Dial the sequence implemented to activate the call forward unconditional service.







To check if the call forward has been properly established:



3. Dial your extension or telephone number.

The call is forwarded to the desired telephone number.



State Description



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Chapter 30 - Call Forward Configuration Call Forward Unconditional




3. Dial the sequence implemented to deactivate the call forward – unconditional service.





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C H A P T E R

31 Telephony ServicesConfiguration

This chapter describes how to set the following subscriber services:

Hook Flash Processing

Automatic call

Call completion

Delayed Hotline

Call Transfer

Call Waiting

Conference

Direct IP address call

Hold

Second call

Message Waiting Indicator

Some of the subscriber services are not supported on all Mediatrix Boss models, so your specific model may not have all subscriber services listed in this chapter.




General Configuration

The General Configuration sub-section of the Services Configuration section allows you to define the Hook Flash Processing feature.

Standards Supported • RFC 2976: The SIP INFO Method

Note: Performing a flash hook and pressing the flash button means the same thing. However, not all telephone models have a flash button.

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Chapter 31 - Telephony Services Configuration General Configuration

To set general services parameters:

1. In the web interface, click the Telephony link, then the Services sub-link.

Figure 187: Telephony – Services Web Page



3. In the General Configuration sub-section, define whether or not you want to override the general services parameters set in the Default configuration in the Endpoint Specific drop-down menu.


4. Select how to process hook-flash detection in the Hook Flash Processing drop-down menu.

Hook flash processing allows hook flash signals to be transported over the IP network allowing to use advanced telephony services. Users normally press the “flash” button of the telephone during a call in progress to put this call on hold, transfer it, or even initiate a conference call.

You can define whether these subscriber services are handled by the unit or delegated to a remote party. If services are to be handled by a remote party, a SIP INFO message is sent to transmit the user's intention.

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Note: The hook-flash processing attribute is not negotiated in SDP.

Table 226: Hook Flash Settings

Setting Definition

Process Locally The hook-flash is processed locally. The actual behaviour of the “flash” button depends on which endpoint services are enabled for this endpoint.

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Automatic Call

The automatic call feature allows you to define a telephone number that is automatically dialed when taking the handset off hook.

When this service is enabled, the second line service is disabled but the call waiting feature is still functional. The user can still accept incoming calls.

To set the automatic call feature:



2. In the Automatic Call sub-section, define whether or not you want to override the automatic call parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 188: Telephony – Automatic Call Section

3. Enable the service by setting the Automatic Call Activation drop-down menu to Enable.

4. Define the string to dial when the handset is taken off hook in the Automatic Call Target field.






Transmit Using Signaling Protocol

The hook-flash is processed by a remote party. The hook-flash event is carried by a signaling protocol message. The actual behaviour of the “flash” button depends on the remote party.

The hook-flash event is relayed as a SIP INFO message as described in RFC 2976.

Table 226: Hook Flash Settings (Continued)

Setting Definition

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Call Completion

The call completion service allows you to configure the Completion of Calls on No Reply (CCNR) and Completion of Calls to Busy Subscriber (CCBS) features.

CCBS allows a caller to establish a call with a “busy” callee as soon as this callee is available to take the call. It is implemented by monitoring the activity of a UA and look for the busy-to-idle state transition pattern.

CCNR allows a caller to establish a call with an “idle” callee right after this callee uses his phone. It is implemented by monitoring the activity of a UA and look for the idle-busy-idle state transition pattern.

The information about the call completion is not kept after a restart of the EpServ service. This includes the call completion activation in the Pots service and the call completion monitoring in the SipEp service.

To set the call completion feature:



2. In the Call Completion sub-section, define whether or not you want to override the call completion parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 189: Telephony – Call Completion Section

3. Enable or disable the (CCBS) service by selecting the proper value in the Allow CCBS Activation Via Handset drop-down menu.


4. If the CCBS service is enabled, define the digits that users must dial to start the service in the CCBS DTMF Map Activation field.


You can use the same code in the CCNR DTMF Map Activation field.

Standards Supported • RFC 4235: An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP)a

• draft draft-poetzl-bliss-call-completion-00b

a. Implemented in client mode only and used for the call completion.

b. Implement the solution 1 in 5.1.

Note: This section applies only to the following models:

• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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5. Enable or disable the (CCNR) service by selecting the proper value in the Allow CCNR Activation Via Handset drop-down menu.


6. If the CCNR service is enabled, define the digits that users must dial to start the service in the CCNR DTMF Map Activation field.


You can use the same code in the CCBS DTMF Map Activation field.



7. Define the digits that users must dial to stop the CCBS and CCNR services in the DTMF Map Deactivation field.


For instance, you could decide to put “*94” as the sequence to deactivate the services. This sequence must be unique and follow the syntax for DTMF maps (see “Chapter 28 - DTMF Maps Configuration” on page 365). Dialing this DTMF map does not have any effect unless the service’s status is “enabled”.


8. Define the delay, in minutes, after the call completion activation to automatically deactivate the call completion if the call is not completed in the Expiration Timeout field.


9. Select the call completion method to detect that the call completion destination is ready to complete the call in the Method drop-down menu.


The monitoring method consists of using the protocol signalling to detect the destination state without using the call. When the destination is ready to complete the call, the local user is notified that the call is ready to be completed and the call to the destination is initiated when the user is ready to initiate the call.

The polling method consists of using periodic calls to the call completion destination until the destination responds with a ringing or connect. Upon receiving these responses, the local user is notified that the call is ready to be completed.

The polling mechanism can only be used for call completion to busy subscriber (CCBS).

Table 227: Call Completion Method Parameters

Method Desciption

Monitoring Only The call completion only uses the monitoring method to detect that the destination is ready to complete the call.

Monitoring And Polling The call completion only uses the monitoring method to detect that the destination is ready to complete the call. The polling mechanism is used if the call completion destination cannot be monitored.

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The retransmission of the polling mechanism is configurable with DefaultCallCompletionPollingInterval.

10. Enable or disable the call completion auto reactivation in the Auto Reactivate drop-down menu.


When enabled, the call completion busy subscriber is automatically activated if the call initiated by a call completion busy subscriber or call completion no response fails because of a busy destination.

11. Define the minimal delay to wait, in seconds, before executing a call completion after its activation in the Auto Reactivate Delay field.


This delay only applies to call completion activated via the call completion auto reactivation feature (See Step 9).

Media5 recommends to set a delay when the method to monitor the target state is based on the target calls instead of its ability to answer a call.

If the timeout is set to 0 and the target is off hook, the FXS endpoint always rings to notify that the call completion is ready to be completed. However the call is always busy and thus reactivated without the possibility for the user to cancel the call completion. The call completion will continue until the ringing or call completion timeout or if the target became ready to receive call.

12. Define how the call completion service needs to interpret the reception of a progress message with early media in the Early Media Behaviour drop-down menu.


13. Define the delay, in seconds, between the calls to the call completion target used for the polling mechanism in the Polling Interval field.


This parameter is used only if the Default Call Completion Method drop-down menu is set to Monitoring And Polling.


Special SIP Configuration

If you are using an Asterisk® IP PBX, it returns the error code 503 instead of 486 for a busy destination when the call limit is reached. The following error mapping can be required:

1. Go to the page SIP > Misc.

2. Insert a new mapping (with the plus button) in the SIP To Cause Error Mapping section.

3. Set the SIP code to 503 “Service Unavailable” and the cause to 17 “User busy”.

4. Click Submit.

Table 228: Call Completion Early Media Behaviour Parameters


None The progress message with early media is not considered as a busy or a ringing response.

CCBS The progress message with early media is interpreted as a busy response and the CCBS can be activated on the call.

CCNR The progress message with early media is interpreted as a ringing response and the CCNR can be activated on the call.

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Using the Call Completion Services

The following are the various procedures to use these services on the user’s telephone.

To start the CCBS (procedure 1)

The call has reached a busy destination and the busy tone is played.

1. Dial the sequence implemented to enable the CCBS.


The confirmation tone is played.

2. Hang up the telephone.

Alternatively, you can use procedure 2.

To start the CCBS (procedure 2)

The call has reached a busy destination and the busy tone is played.



The dial tone is played

3. Dial the sequence implemented to enable the CCBS.




Alternatively, you can use procedure 1.

To start the CCNR

The call has reached a destination but the call is still not yet established. A ring back or welcome message is generally played at this moment.




3. Dial the sequence implemented to enable the CCNR.




To stop the CCBS or CCNR



2. Dial the sequence implemented to disable the CCBS and CCNR.




Note: The CCBS and CCNR cannot be started to complete a second call.

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When the call completion target is ready to receive a call:

1. The telephone rings with the distinctive ringing “Bellcore-dr2” (0.8 On – 0.4 Off, 0.8 On – 4.0 Off).


The call is initiated to the call completion destination.

Call Transfer

The Call Transfer service offers two ways to transfer calls:

Blind Transfer

Attended Transfer

To enable the Call Transfer services:



2. In the Call Transfer sub-section, define whether or not you want to override the call transfer parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 190: Telephony – Call Transfer Web Page

3. Enable the Blind Transfer service by setting the Blind Transfer Activation drop-down menu to Enable.

The blind call transfer service is sometimes called Transfer without Consultation or Unattended Transfer. It allows a user to transfer a call on hold to a still ringing (unanswered) call. The individual at the other extension or telephone number does not need to answer to complete the transfer.

The call hold and second call services must be enabled for this service to work. See “Call Hold” on page 433 and “Second Call” on page 434.

4. Enable the Attended Transfer service by setting the Attended Transfer Activation drop-down menu to Enable.

The attended call transfer service is sometimes called Transfer with Consultation. It allows a user to transfer a call on hold to an active call. The individual at the other extension or telephone number must answer to complete the transfer.

The call hold and second call services must be enabled for this service to work. See “Call Hold” on page 433 and “Second Call” on page 434.



• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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Using Blind Call Transfer


To transfer a current call blind:

1. Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.

This puts the call on hold.

2. Wait for the transfer tone (three “beeps”).

3. Dial the number to which you want to transfer the call.

4. Wait for the ringback tone, then hang up your telephone.

The call is transferred. You can also wait for the third party to answer if you want. In this case, the call transfer becomes attended.

If you want to get back to the first call (the call on hold), you must perform a Flash-Hook.

You are back with the first call and the third party is released.

Using Attended Call Transfer


To transfer a current call attended:


This puts the call on hold.

2. Wait for the transfer tone (three “beeps”).

3. Dial the number to which you want to transfer the call.

The third party answers.


The call is transferred.

5. If you want to get back to the first call (the call on hold), you must perform a Flash-Hook before the target answers.

You are back with the first call and the third party is released.

Call Waiting

The call waiting tone indicates to an already active call that a new call is waiting on the second line.

Your users can activate/deactivate the call waiting tone for their current call. This is especially useful when transmitting faxes. The user that is about to send a fax can thus deactivate the call waiting tone to ensure that the fax transmission will not be disrupted by an unwanted second call. When the fax transmission is completed and the line is on-hook, the call waiting tone is automatically reactivated.

Note: If the number to which you want to transfer the call is busy or does not answer, perform a Flash-Hook. The busy tone or ring tone is cancelled and you are back with the first call.


• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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To set the Call Waiting services:



2. In the Call Waiting sub-section, define whether or not you want to override the call waiting parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 191: Call Waiting Section

3. Enable the service by setting the Call Waiting Activation drop-down menu to Enable.

This permanently activates the call waiting tone. When receiving new calls during an already active call, a special tone is heard to indicate that a call is waiting on the second line. The user can then answer that call by using the “flash” button. The user can switch between the two active calls by using the “flash” button.

The call hold service must be enabled for this service to work. See “Call Hold” on page 433.

If the user is exclusively using faxes, select Disable to permanently disable the call waiting tone.

4. Define the digits that users must dial to disable the Call Waiting tone in the Cancel DTMF Map field.


This allows a user who has call waiting enabled to disable that service on the next call only. If, for any reason, the user wishes to undo the cancel, unhook and re-hook the telephone to reset the service.

For instance, you could decide to put “*76” as the sequence to disable the call waiting tone. This sequence must be unique and follow the syntax for DTMF maps (see “Chapter 28 - DTMF Maps Configuration” on page 365). Dialing this DTMF map does not have any effect unless the service’s status is “enabled”.



Using Call Waiting

The call waiting feature alerts the user if he or she is already on the telephone and a second call happens. A “beep” (the call waiting tone) is heard and repeated every ten seconds to indicate there is a second incoming call.

To put the current call on hold:


This puts the call on hold and the second line is automatically connected to your line.

2. Answer the call on the second line.

To switch from one line to the other:

1. Perform a Flash-Hook each time you want to switch between lines.

To terminate the first call before answering the second call:


2. Wait for the telephone to ring.

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3. Answer the telephone.

The second call is on the line.

Removing the Call Waiting Tone

You can temporarily deactivate the call waiting tone indicating a call is waiting. This is especially useful when transmitting faxes. If you are about to send a fax, you can thus deactivate the call waiting tone to ensure that the fax transmission is not disrupted by an unwanted second call. When the fax transmission is completed and the line is on-hook, the call waiting tone is automatically reactivated.

To deactivate the call waiting tone:



3. Dial the sequence implemented to deactivate the call waiting tone.



The call waiting tone is disabled.

Conference

The Conference Call service allows a user to link two or more calls together to form a single conversation, called a conference.

Only 3-way conferences are currently supported.

A participant of the conference can put the conference on hold and attempt other calls. This participant may then rejoin the conference at a later time by unholding it. The participant who initiated the conference cannot put it on hold.

You must enable the call hold, second call and attended call transfer services for this service to work. See “Call Hold” on page 433, “Second Call” on page 434, and “Call Transfer” on page 426.

The following is a conference call flow example:


• Mediatrix 3208 / 3216

• Mediatrix 3308 / 3316

• Mediatrix 3731

• Mediatrix 3732

• Mediatrix 3741

• Mediatrix 3742

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Figure 192: Conference Call Flow

Enabling the Conference Call Feature

You must enable this service before your users can use it.

To enable the Conference service:



2. In the Conference sub-section, define whether or not you want to override the conference parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 193: Conference Section

3. Enable the service by setting the Conference Activation drop-down menu to Enable.


Managing a Conference Call

If you are on the telephone with one person and want to conference with a third one, you can do so. In the following examples, let’s assume that:

“A” is the conference initiator.

“B” is the person called on the first line.

“C” is the person called on the second line.

To initiate a three-way conference (“A” and “B” already connected):

1. “A” performs a Flash-Hook.

This puts “B” on hold and the second line is automatically connected. “A” hears a dial tone.

2. “A” dials “C’s” number.

“A” and “C” are now connected.

3. “A” performs another Flash-Hook.

The call on hold (“B”) is reactivated. “A” is now conferencing with “B” and “C”.

INVITE (G.729)

Trying/Ringing/200 OK

User Agent

#2(B)

User Agent

#3(C)

User Agent

#1(A)

ACK

Flash Hook

INVITE (HOLD)

Trying /200 OK

ACK

INVITE (G.729 )

Trying/Ringing/200 OK

ACK

Flash Hook

Trying200 OK

ACK

INVITE (UNHOLD-G.729 )

Trying/200 OK

ACK

3-way Conference Call Established

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“B” (or “C”) hangs up during the conference:

1. “B” (or “C”) hangs up during the conference.

The conference is terminated, but the call between “A” and “C” (or “B”) is not affected and they are still connected.

“A” (conference initiator) hangs up during the conference:

1. “A” hangs up.

The conference is terminated, both call "C" and "B" are also terminated.

Delayed Hot Line

The delayed hot line feature (also called warm line) is used to make an automatic call to a specified address on the two following conditions:

When the user picks up the phone but does not dial any digit. The configured destination is automatically called upon picking up the phone and after waiting for the configurable number of seconds without dialling.

When the user starts dialing but does not complete a valid number before the timeout set in the Delayed Hotline Condition drop-down menu expires.

The condition on which the delayed hotline is activated is configurable. This feature thus places an automatic call whenever the Delayed Hotline Condition timeout expires. It could be used as an alternative to the emergency number (for instance, the 911 number in North America).

To configure the basic delayed hot line feature:



2. In the Delayed Hotline sub-section, define whether or not you want to override the delayed hotline parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 194: Delayed Hotline Section

3. Enable the service by setting the Delayed Hotline Activation drop-down menu to Enable.

When the feature is disabled, a user picking up the phone but not pressing any telephone keys hears the Receiver Off-Hook tone after the amount of time specified in the digitMapTimeoutFirstDigit variable.



• Mediatrix 3208 / 3216

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To configure the delayed hotline activation condition:

1. In the Delayed Hotline sub-section, select the condition(s) that activate the delayed hotline in the Delayed Hotline Condition drop-down menu.



To configure the delayed hotline target:

1. In the Delayed Hotline sub-section, set the destination (address or telephone number) that is automatically called in the Delayed Hotline field.







Direct IP Address Call

The IP address call service allows a user to dial an IP address without the help of a SIP server. Using this method bypasses any server configuration of your unit.

The user can dial an IP address and enter an optional telephone number. Note that the optional telephone number is matched by using the same digit maps as a normal call.

The IP address call method can be used when a SCN user wants to reach a LAN endpoint.

Table 229: Delayed Hotline Conditions


FirstDtmfTimeout The delayed hotline is activated when the timeout configured in the First DTMF Timeout field of the Telephony > DTMF Maps page elapses (“General DTMF Maps Parameters” on page 368).

InterDtmfOrCompletionTimeout The delayed hotline is activated when the timeout configured in the Completion Timeout field of the Telephony > DTMF Maps page elapses or when the DTMFs collection fails because the Inter DTMF Timeout parameter elapses (“General DTMF Maps Parameters” on page 368).

AnyTimeout The delayed hotline is activated when the timeout configured in the Completion Timeout field of the Telephony > DTMF Maps page elapses and when the DTMFs collection fails because the Inter DTMF Timeout parameter elapses (“General DTMF Maps Parameters” on page 368).

1

1

432 Mediatrix Boss


To set the direct IP call feature:


This menu is available only in the default endpoints configuration.

2. Enable the service by setting the Direct IP Address Call drop-down menu to Enable.

Figure 197: Telephony – Direct IP Address Call Section

Dialing an IP Address

To make an IP address call:

1. Dial “**” (IP address prefix).

2. Dial the numerical digits of the IP address and use the “*” for the “.” of the IP address.

3. Dial “*” to terminate the IP address if you do not need to specify a phone number.

For instance, let’s say you want to reach a one-line access device or another LAN endpoint such as an IP Phone with the IP address 192.168.0.23. You must then dial the following digits:

**192*168*0*23*

4. If you need to specify the phone number of a specific line, dial “#” to terminate the IP address.

5. Dial the telephone number of the specific line you want to reach.

For example, let’s say you want to reach the telephone connected to Line 2 of the Mediatrix Boss with the IP address 192.168.0.23. The phone number assigned to Line 2 of this Mediatrix Boss is 1234. You must then dial the following digits:

**192*168*0*23#1234

In this case, the Mediatrix Boss sends an INVITE [email protected].

Call Hold

The Call Hold service allows the user to temporarily put an existing call on hold, usually by using the “flash” button of the telephone. The user can resume the call in the same way.

You must enable this service for the following services to work properly:

Call Waiting

Second Call

Blind Transfer

Attended Transfer

Conference

2


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• Mediatrix 3308 / 3316

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To enable the Call Hold service:



2. In the Hold sub-section, define whether or not you want to override the call hold parameters set in the Default configuration in the Endpoint Specific drop-down menu.


Figure 198: Hold Section

3. Enable the service by setting the Hold Activation drop-down menu to Enable.


Using Call Hold


To put the current call on hold:


This puts the call on hold. You can resume the call in the same way.

Second Call

The Second Call service allows a user with an active call to put the call on hold, and then initiate a new call on a second line. This service is most useful with the transfer and conference services.

The call hold service must be enabled for this service to work. See “Call Hold” on page 433.

You must enable this service for the following services to work properly:

Blind Transfer

Attended Transfer

Conference

To enable the Second Call service:



2. In the Second Call sub-section, define whether or not you want to override the second call parameters set in the Default configuration in the Endpoint Specific drop-down menu.


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Figure 199: Second Call Section

3. Enable the service by setting the Second Call Activation drop-down menu to Enable.


Using Second Call


To use the second call service:


This puts the call on hold and the second line is automatically connected to your line.

2. Initiate the second call.

Message Waiting Indicator


The Message Waiting Indicator (MWI) service alerts the user when new messages have been recorded on a voice mailbox. It is enabled by default.

After the message is recorded, the server sends a message (SIP NOTIFY request) to the Mediatrix Boss listing how many new and old messages are available. The Mediatrix Boss alerts the user of the new message in two different ways:

The telephone’s LED blinks (if present). A FSK signal is sent on the FXS line.

A message waiting stutter dial tone replaces the normal dial tone when the user picks up the FXS line.

The Mediatrix Boss supports to receive SIP MWI notifications via SIP NOTIFY requests as defined in RFC 3842 but with the following limitations/diversions:

Only NOTIFY without subscription is supported. The Mediatrix Boss does not send SIP

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• Mediatrix 3731

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Standards Supported • RFC 3265: Session Initiation Protocol (SIP)-Specific Event Notificationa

• RFC 3842: The Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)b

a. Supports receiving blind NOTIFY without subscribing. Sending blind NOTIFY is not supported.

b. Supports receiving blind NOTIFY without subscribing. Sending blind NOTIFY is not supported.

Note: The message waiting state does not affect the Second Call feature. When in an active call, performing a flash-hook to get access to the second line plays the usual dial tone.

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SUBSCRIBE request for this event package.

In addition to the SIP event string "message-summary" (RFC 3842), the string "simple-message-summary" is accepted. The significations of those strings are identical.

In addition to the SIP content type string "simple-message-summary" (RFC 3842), the string "message-summary" is accepted. The significations of those strings are identical.

Support of message-summary is not advertised in the SIP REGISTER.

Note that received SIP NOTIFY with an event different than "message-summary" or "simple-message-summary" is not interpreted as a valid MWI notification.






To disable the Message Waiting Indicator service:

1. Select the disabled activation by putting the following line in the configuration script:

pots.fxsDefaultMessageWaitingIndicatorActivation="100"

If you want to reactivate the feature, use the following:

pots.fxsDefaultMessageWaitingIndicatoreActivation="Value"

where Value may be one of the following:

2. If you want to set a different activation for one or more endpoints, put the following lines in the configuration script:

pots.fxsSpecificMessageWaitingIndicator.EnableConfig[Id="Specific_Endpoint"]="1"

pots.fxsSpecificMessageWaitingIndicator.Activation[Id="Specific_Endpoint"]="Valu

e"

where:

• Specific_Endpoint is the number of the endpoint you want to configure.

• Value is the activation as defined in Step 1.

Table 230: Message Waiting Indicator Parameters

Value Parameter Description

100 Disabled The user is not alerted of messages awaiting attention.

200 Tone When messages are awaiting attention, the user is alerted by a message waiting tone when picking up the handset.

300 Visual When messages are awaiting attention, the user is alerted by a Visual Message Waiting Indicator such as a blinking LED on the phone.

400 ToneAndVisual When messages are awaiting attention, the user is alerted by a Visual Message Waiting Indicator such as a blinking LED on the phone, and a message waiting tone when picking up the handset.

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32 Tone Customization ParametersConfiguration

This chapter describes how to override the pattern for a specific tone defined for the selected country (see “Appendix A - Country-Specific Parameters” on page 533 for more details). It covers the following topics:

Current Tone Definition

Tone Override

Current Tone Definition

The Tone Customization page allows you to both see the current definition and override the pattern of the following tones:

This includes the number of frequencies used, the tone value in Hertz (Hz), its power in dBm, as well as the states configured.

Busy

Call Waiting

Confirmation

Congestion

Dial

Hold

Intercept

Message Waiting

Preemption

Reorder

Ringback

Receiver Off Hook (ROH)

Special Information Tone (SIT)

Stutter

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To see the current definition of a tone:

1. In the web interface, click the Telephony link, then the Tone Customization sub-link.

Figure 200: Telephony – Tone Customization Web Page

2. Select the proper tone to see in the Select Tone drop-down menu at the top of the window.

The Current Tone Definition and Current Tone States sections describe the current definition of the selected tone.

Tone Override

You can override the pattern for a specific tone. This is done in two sections:

2

Table 231: Tone Override Sections

State Description

Overridden Tone Definition Allows you to define up to four frequencies (F1 to F4). You must enter at least one frequency.

Overridden Tone States Description of the tone state. You can define up to eight states. You must enter at least one state.

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To override the pattern of a tone:

1. Select which tone you want to override in the Override Current Tone Values drop-down menu.

Figure 201: Tone Override Sections

• You can use the current values of the selected tone as a starting point for your customization by clicking the Copy Current Tone Definition to Overridden button.

• You can clear all override fields by clicking the Reset Overridden Values button.

2. In the Overridden Tone Definition section, define the value of the proper Frequency used in the corresponding Value field.

The value is in Hz. The range is from 10 Hz to 4000 Hz.

3. Define the power level of the proper Frequency in dBm in the corresponding Power field.

The range is from -99 dBm to 3 dBm.

4. If applicable, enter a value for the loop counter in the Loop Count field.

The range is from 2 to 128. This value will be used in Step 8.

5. In the Overridden Tone States section, set the corresponding On/Off drop-down menu with the proper value for each state.

• On means the corresponding state plays a tone.

• Off means the corresponding state does not play a tone.

• CID means the moment where the Caller-ID will be sent to the analog port. This options is available only for the Call Waiting tone.

You may also want to perform the following operations:

• To add a state, click the button at the bottom of the Overridden Tone States section.

• To remove a state, click the button at the bottom of the Overridden Tone States section. This removes the last state in the list.

6. For the On states, select the frequency to play in the corresponding Frequencies column.

The frequencies defined in the Overridden Tone Definition section are listed as clickable buttons. You can use from one to four frequencies. A blue button indicates that the frequency is selected.

4

1

5 6 7 8 9

2 3

Note: You can use only two frequencies for the Call Waiting tone.

Note: You can use only one loop count for the Call Waiting tone.

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7. Set the corresponding Duration field with the number of times, in ms, to perform the action of the state.

The range is from 10 ms to 56000 ms. The tone stays indefinitely in the state (continuous) if no time is specified.

8. In the corresponding Loop drop-down menu, select whether or not to stop looping between states after a number of loops defined in Step 4.

When the number of loops is reached, the next state is s(n+1) for the state s(n) instead of the state defined in the Next State drop-down menu.

9. In the corresponding Next State drop-down menu, select the next tone state to use when the time has elapsed.

This value is not available if the Duration field is empty.


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33 Music on Hold ParametersConfiguration

This chapter describes how to configure the Music on Hold (MoH) parameters.

Music on Hold Configuration

The Music on Hold sub-page of the Telephony page allows you manage a MP3 file to be played while one or more local phones are on hold. Once one or more MP3 files have been downloaded on the unit through the Comdasys web interface.

To set the Music on Hold parameters:

1. In the web interface, click the Telephony link, then the Music on Hold sub-link.

Figure 202: Telephony – Music on Hold Web Page

2. In the Configuration section, indicate whether or not the unit should play music when being put on hold in the Streaming drop-down menu.

When enabled, music is played toward the telephony side when being put on hold from the network side.

3. In the File Name field, enter the name of one of the previously downloaded files.

Once one or more MP3 files have been downloaded on the unit through the Comdasys web interface,


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34 RTP Statistics Configuration

The Mediatrix Boss collects meaningful statistics that can be read via the web interface. This chapter describes how to read and configure the RTP statistics.

Statistics Displayed

The Mediatrix Boss collects two types of statistics:

statistics for the last 10 connections

statistics for the last 10 collection periods

The Connection Statistics section displays the statistics for the last 10 connections. You can use the Display All button to display more information or the Display Overview button to display less information.

The Connection Period Statistics section displays the statistics for the last 10 periods. The period duration is defined in the Statistics Configuration section. You can use the Display All button to display more information or the Display Overview button to display less information.

Figure 203: Telephony – RTP Stats Web Page

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The following table describes the statistics available.

Table 232: Statistics Displayed

Statistic Connection Statistics Collection Period Statistics

Octets Tx Number of octets transmitted during the connection.

Number of octets transmitted during the collection period. This value is obtained by cumulating the octets transmitted in all connections that were active during the collection period.

Octets Rx Number of octets received during the connection.

Number of octets received during the collection period. This value is obtained by cumulating the octets received in all connections that were active during the collection period.

Packets Tx Number of packets transmitted during the connection.

Number of packets transmitted during the collection period. This value is obtained by cumulating the packets transmitted in all connections that were active during the collection period.

Packets Rx Number of packets received during the connection.

Number of packets received during the collection period. This value is obtained by cumulating the packets received in all connections that were active during the collection period.

Packets Lost Number of packets lost during the connection. This value is obtained by substracting the expected number of packets based on the sequence number from the number of packets received.

Number of packets lost during the collection period. This value is obtained by cumulating the packets lost in all connections that were active during the collection period.

Min. Jitter Minimum interarrival time, in ms, during the connection. All RTP packets belonging to the connection and received at the RTP level are considered in the calculation.

Minimum interarrival time, in ms, during the collection period. This value is the lowest interarrival jitter for all connections that were active during the collection period.

Max. Jitter Maximum interarrival time, in ms, during the connection. All RTP packets belonging to the connection and received at the RTP level are considered in the calculation.

Maximum interarrival time, in ms, during the collection period. This value is the highest interarrival jitter for all connections that were active during the collection period.

Avg. Jitter Average interarrival time, in ms, during the connection. All RTP packets belonging to the connection and received at the RTP level are considered in the calculation.

Average interarrival time, in ms, during the collection period. This value is the weighted average of the interarrival jitter for all connections that were active during the collection period. For each connection, the total jitter of packets received during the collection period and the total number of packets received during the collection period are used in the weighted average calculation.

Min. Latency Minimum latency, in ms, during the connection. The latency value is computed as one half of the round-trip time, as measured through RTCP.

Minimum latency, in ms, during the collection period. This value is the lowest latency for all connections that were active during the collection period.

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Statistics Configuration

You can define how to collect the statistics. The statistics are sent as syslog messages, so you must properly set the syslog information before setting the statistics. You must set the Media IP Transport (MIPT) service to the Info or Debug level. See “Syslog Daemon Configuration” on page 219 for more details on how to configure the Syslog.

To configure how to collect statistics:

1. In the web interface, click the Telephony link, then the RTP Stats sub-link.

Figure 204: Telephony – RTP stats Web Page

Max. Latency Maximum latency, in ms, during the connection. The latency value is computed as one half of the round-trip time, as measured through RTCP.

Maximum latency, in ms, during the collection period. This value is the highest latency for all connections that were active during the collection period.

Avg. Latency Average latency, in ms, during the connection. The latency value is computed as one half of the round-trip time, as measured through RTCP.

Average latency, in ms, during the collection period. This value is the weighted average of the latency for all connections that were active during the collection period. For each connection, the total latency of packets received during the collection period and the total number of packets received during the collection period are used in the weighted average calculation.

Table 232: Statistics Displayed (Continued)

Statistic Connection Statistics Collection Period Statistics

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2. Set the Collection Period field with the collection period duration in minutes.

Putting a value of 0 disables the collection period statistics feature.

3. Set the End-of-Connection Notification drop-down menu with the proper behaviour.

4. Set the End-of-Period Notification drop-down menu with the proper behaviour.


• To save your settings, click Submit.

• To save your settings and reset the statistics of the current period., click Submit & Reset Current Collection Period Statistics.

The previous periods are left unchanged.

Table 233: End-of-Connection Notification


Enable Notifications are generated.

Disable Notifications are not generated.

Table 234: End-of-Period Notification


Enable Notifications are generated.

Disable Notifications are not generated.

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35 Country ParametersConfiguration

This chapter describes how to configure the country information:

Select a specific country.

Additional country settings.

Country Configuration

The Misc sub-page of the Telephony page allows you to configure the country in which the unit is located.

To set the miscellaneous parameters:

1. In the web interface, click the Telephony link, then the Misc sub-link.

Figure 205: Telephony – Misc Web Page

2. In the Country section, select the country in which the Mediatrix Boss is located in the Country Selection drop-down menu.

It is very important to set the country in which the unit is used because a number of parameter values are set according to this choice, such as tones, rings, impedances, and line attenuations. See “Appendix A - Country-Specific Parameters” on page 533 for more information on these country-specific settings.


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Additional Country Settings


Default vs. Specific Configurations






Input/Output User Gain

The user gain allows you to modify the input and output sound level of the Mediatrix Boss.

You can compensate with the user gain if there is no available configuration for the country in which the Mediatrix Boss is located. Because the user gain is in dB, you can easily adjust the loss plan, e.g., if you need an additional 1 dB for analog to digital, put 1 for user gain output.

You can use two types of configuration as described in “Default vs. Specific Configurations” on page 448.

To set user gain variables:

1. Define the default user output gain offset in dB (from analog to digital) by putting the following line in the configuration script:telIf.defaultCountryCustomizationUserGainOutputOffset="Value"

Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion depending on the country selected.

2. If you want to set a different output gain offset for one or more interfaces, put the following lines in the configuration script:

telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]

="1"

telIf.specificCountryCustomizationUserGain.OutputOffset[InterfaceId="Interface"]="Value"

where:

• Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).

• Value is the output gain offset.

3. Define the default user input gain offset in dB (from digital to analog) by putting the following line in the configuration script:

telIf.defaultCountryCustomizationUserGainInputOffset="Value"

Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion depending on the country selected.

Caution: Use these settings with great care. Media5 recommends not to modify the user gain variables unless absolutely necessary because default calibrations may no longer be valid.

Modifying user gains may cause problems with DTMF detection and voice quality – using a high user gain may cause sound saturation (the sound is distorted). Furthermore, some fax or modem tones may no longer be recognized. The user gains directly affect the fax communication quality and may even prevent a fax to be sent.

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4. If you want to set a different input gain offset for one or more interfaces, put the following lines in the configuration script:

telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]="1"

telIf.specificCountryCustomizationUserGain.InputOffset[InterfaceId="Interface"]=

"Value"

where:

• Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).

• Value is the input gain offset.

5. Restart the TelIf service by putting the following line in the configuration script:

scm.serviceCommands.Restart[Name=TelIf]="10"

Dialing Settings

Dialing settings allow you to configure how the Mediatrix Boss dials numbers.

When selecting a country (see “Country Configuration” on page 447 for more details), each country has default dialing settings. However, you can override these values and define your own dialing settings.


To set the dialing settings:

1. Add the following line to the configuration script:telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]="1"

where Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).

This allows overriding the default country settings.

2. If you want to change the override status for one or more interfaces, put the following lines in the configuration script:

telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=

"1"

telIf.specificCountryCustomizationDialing.Override[InterfaceId="Interface"]="1"


3. Set an inter-digit dial delay by putting the following line in the configuration script:

telIf.defaultCountryCustomizationDialing.InterDtmfDialDelay="Value"

This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone number. Values range from 50 ms to 600 ms.

4. If you want to set a different inter-digit dial delay for one or more interfaces, put the following lines in the configuration script:

telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=

"1"

telIf.specificCountryCustomizationDialing.InterDtmfDialDelay[InterfaceId="Slot3/Bri3"]="Value"


5. Set the DTMF duration value by putting the following line in the configuration script:

telIf.defaultCountryCustomizationDialing.DtmfDuration="Value"

This is the duration, in milliseconds (ms), a DTMF is played when dialing the destination phone number. Values range from 50 ms to 600 ms.

6. If you want to set a different DTMF duration value for one or more interfaces, put the following lines in the configuration script:

telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]="1"

telIf.specificCountryCustomizationDialing.DtmfDuration[InterfaceId="Interface"]=

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"Value"

7. Restart the TelIf service by putting the following line in the configuration script:

scm.serviceCommands.Restart[Name=TelIf]="10"

Fax Calling Tone Detection

You can enable the fax calling tone (CNG tone) detection.


To enable fax calling tone detection:

1. Add the following line to the configuration script:telIf.defaultMachineDetection.CngToneDetection="1"

Upon recognition of the CNG tone, the Mediatrix Boss switches the communication from voice mode to fax mode and the CNG is transferred by using the preferred fax codec. This option allows for quicker fax detection, but it also increases the risk of false detection.

If you do not want the Mediatrix Boss to detect the fax calling tone, set the variable to disable(0). In this case, the CNG tone does not trigger a transition from voice to data and the CNG is transferred in the voice channel. With this option, faxes are detected later, but the risk of false detection is reduced.

2. If you want to set a different calling tone detection setting for one or more interfaces, add the following lines in the configuration script:

telIf.specificMachineDetection.EnableConfig[InterfaceId="Interface"]="1"

telIf.specificMachineDetection.CngToneDetection[InterfaceId="Interface"]="Value"

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K C H A P T E R

36 Call Router Configuration

This chapter describes the call router service.

Introduction to the call router’s parts and types supported.

Routes parameters.

Mappings parameters.

Call signalling parameters.

SIP headers translation parameters.

Call properties translation parameters.

Hunt table parameters.

Introduction

The Mediatrix Boss’s call router allows you to route calls between interfaces. Based on a set of routing criteria, the call router determines the destination (interface) for every incoming call. The forwarding decisions are based on the following tables:

Standards Supported • ITU-T Recommendation E.164: The international public telecommunication numbering plan.

• ITU-T Recommendation F.69: List of Telex Destination Codes.

• ITU-T Recommendation X.121: International numbering plan for public data networks.

Table 235: Call Router Table Types

Table Description

Routing The routing table contains one or more routes. Each route associates a destination to a call that matches a set of criteria. See “Routes” on page 470 for more details.

Mapping The mapping table contains one or more mapping types and expressions. A mapping modifies call properties such as the calling and called party numbers according to the network requirements. These mappings are specifically called within a route. See “Mappings” on page 476 for more details.

Call Signalling

Call signalling specifies how to set up a call to the destination Mediatrix Boss or 3rd party equipment. Call signalling properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol level. See “Signalling Properties” on page 486 for more details.

SIP Headers Translation

A SIP headers translation overrides the default value of SIP headers in an outgoing SIP message. See “SIP Headers Translations” on page 490 for more details.

Call Properties Translation

A call properties translation overrides the default value of call properties in an incoming SIP message. See “Call Properties Translations” on page 493 for more details.

Hunt The hunt table contains one or more hunt entries, each with a set of possible destinations. A hunt tries the destinations until one of the configured destinations accepts the call. See “Hunt Service” on page 496 for more details.

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When a new call comes from one of the Mediatrix Boss interfaces, it is redirected to the routing table. The following figure illustrates the Mediatrix Boss call router:

Figure 206: Call Routing

Limitations

The call routing service has the following limitations:

A call coming from a SIP interface cannot be routed to another SIP interface. When that occurs, the call automatically fails.

A call automatically fails if it is redirected to a route or hunt more than 10 times.

Regular Expressions

Some of the routing types described in “Routing Type” on page 456 require that you enter them following the regular expression syntax. A regular expression is a string used to find and replace strings in other large strings. The Mediatrix Boss uses regular expressions to enter a value in several routing types, often by using wildcard characters. These characters provide additional flexibility in designing call routing and decrease the need for multiple entries in configuring number ranges.

The expression cannot begin by “^”, it is implicit in the expression. The following table shows some of the wildcard characters that are supported:

Hunt 1

Mapping 1

Interfaces

Call Router

Route 2 Route 3Route 1

SignallingProperties 1

Translation 1(SIP Headers orcall properties)

Standards Supported • IEEE Std 1003.1-2001: IEEE Standard for Information Technology---Portable Operating System Interface (POSIX®)

Table 236: Regular Expressions Wildcards

Character Description

. Single-digit place holder. For instance, 555 .... matches any dialed number beginning with 555, plus at least four additional digits. Note that the number may be longer and still match.

* Repeats the previous digit 0, 1, or more times. For instance, in the pattern:

1888*1

the pattern matches:

1881, 18881, 188881, 1888881

Note: If you are trying to handle the asterisk (*) as part of a dialed number, you must use \*.

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Introduction Software Configuration Guide

The matching criterion implicitly matches from the beginning of the string, but not necessarily up to the end. For instance, 123 will match the criterion 1, but it will not match the criterion 2.

If you want to match the whole string, you must end the criterion with “$”. For instance, 123 will not match the criterion 1$ and will match the criterion 123$.

Groups

A group is placed within parenthesis. It is used when replacing a string in a mapping. You can use up to nine groups (defined by “\1” to “\9”) and matching is not case sensitive. “\0” represents the whole string. Lets say for instance you have the following string:

9(123(45)6)

The following describes how the groups are replaced in a properties manipulation:

Groups can only be used with the following routing types:

Calling/Called E.164

Calling/Called Name

Calling/Called Host

Calling/Called URI

[ ] Range of digits.

• A consecutive range is indicated with a hyphen (-), for instance, [5-7].

• A nonconsecutive range is indicated without a delimiter, for instance, [58].

• Both can be used in combination, for instance [5-79], which is the same as [5679].

You may place a (^) symbol right after the opening bracket to indicate that the specified range is an exclude list. For instance, [^01] specifies the same range as [2-9].

Note: The call router only supports single-digit ranges. You cannot specify the range of numbers between 99 and 102 by using [99-102].

( ) Indicates a pattern (also called group), for instance, 555(2525). It is used when replacing a number in a mapping. See “Groups” on page 455 for more details.

? Matches 0 or 1 occurrence of the previous item. For instance, 123?4 matches both 124 and 1234.

+ Repeats the previous digit one or more time. For instance 12+345 matches 12345, 122345, etc. (but not 1345). If you use the + at the end of a number, it repeats the last number one or more times. For instance: 12345+ matches, 12345, 123455, 1234555, etc.

Table 236: Regular Expressions Wildcards (Continued)

Character Description

Note: You can use the “<undefined>” string if you want to match a property that is not defined.

Table 237: Groups Replacement Example

Replacement Result

\0 9123456

\1 123456

\2 45

\3

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Routing Type

The following sub-sections list the available routing types of the call router and their supported values. The routing types that offer choices use the choices as defined in the Q.931 standard. Q.931 is ISDN’s connection control protocol, roughly comparable to TCP in the Internet protocol stack. The values may also be a special tag, as described in “Special Tags” on page 461.

Media5 recommends to carefully define the routing requirements and restrictions that apply to your installation before starting the routing configuration. This will help you determine the types of routing you need. When this is done, define the routes and mappings, as well as the hunts that you need to fulfil these requirements. You may need several entries of the same type to achieve your goals.

See also “Call Properties Parameters” on page 462 for a description of the parameters used by the various routing types and interfaces of the call router.

Called / Calling E164

This is the Called/Calling Party Number. You can enter a regular expression (called/calling party E.164 number in the call setup message) as per “Regular Expressions” on page 454. Note that:

A PBX may insert or modify the calling party number. Sometimes there is no calling party number at all. This all depends on the equipment you connect to the device.

The Mediatrix Boss cannot filter the redirecting number information element of the SETUP message because it does not support the “calling-Redir-E164” and “Calling-Redir-Reason” routing properties criteria.

Standards Supported • ITU-T Recommendation Q.931: ISDN user-network interface layer 3 specification for basic call control

Table 238: Routing Types Locations

Routing Type Location

E164 “Called / Calling E164” on page 456

Type of Number (TON) “Called / Calling TON” on page 457

Numbering Plan Indicator (NPI) “Called / Calling NPI” on page 457

Name “Called / Calling Name” on page 457

Host “Called / Calling Host” on page 457

URI “Called / Calling URI” on page 457

Presentation Indicator (PI) “Calling PI” on page 458

Screening Indicator (SI) “Calling SI” on page 458

Information Transfer Capability (ITC) “Calling ITC” on page 458

Date and Time “Date/Time” on page 459

Phone Context “Called / Calling Phone Context” on page 459

SIP Username “Called / Calling SIP Username” on page 459

Diverting Reason “Last / Original Diverting Reason” on page 460

Diverting E.164 “Last / Original Diverting E.164” on page 460

Diverting Party Number Type “Last / Original Diverting Party Number Type” on page 460

Diverting Public Type Of Number “Last / Original Diverting Public Type Of Number” on page 460

Diverting Pivate Type Of Number “Last / Original Diverting Private Type Of Number” on page 461

Diverting Number Presentation “Last / Original Diverting Number Presentation” on page 461

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Called / Calling TON

Called or calling party type of number field in the ISDN setup message. The following values are available:

Called / Calling NPI

Called or calling party numbering plan indicator field in the ISDN setup message. The following values are available:

Called / Calling Name

Calling and called party name (display name). This is the human-readable name of the calling or called party. See “Regular Expressions” on page 454 for more details on how to enter a proper expression.

The Mediatrix Boss does not support the sending of the calling name in the user-to-user information element.

Called / Calling Host

IP address or domain name of the called or calling host in the following format:

Fqdn[:port]

If [:port] is missing, the call router uses the well-known port of the signalling protocol. Note that:

Incoming SIP calls use the calling party IP address property to store the IP address of the remote SIP user agent. Other interfaces such as ISDN set the IP address to 0.0.0.0.

You can use a regular expression to enter an IP address or a range of IP addresses.

Called / Calling URI

Uniform Resource Identifier (URI) of:

the called party, e.g., the To-URI.

Table 239: Type of Number Values

Value Description

unknown Unknown number type.

international International number.

national National number.

network Network specific number used to indicate an administration or service number specific to the serving network.

subscriber Subscriber number.

abbreviated Abbreviated number.

Note: The called type of number is set to international if the To username is an E.164 with the prefix “+”. The calling type of number is set to international if the From username is an E.164 with the prefix “+”.

Table 240: Numbering Plan Indicator Values

Value Description

unknown Unknown numbering plan.

isdn (E.164) ISDN/Telephony numbering plan according to ITU-T Recommendation E.164.

data (X.121) Data numbering plan according to ITU-T Recommendation X.121.

telex (F.69) Telex numbering plan according to ITU-T Recommendation F.69.

national Numbering plan according to a national standard.

private A private numbering plan.

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the originating VoIP peer, e.g., the From-URI of an incoming SIP call.

The URI follows the format described in RFC 3261.

Calling PI

Presentation indicator of the calling party number. The following values are available:

You may want to remove the calling party number when the user sets the presentation indicator to restricted. To achieve this, route restricted calls to a mapping that sets the Calling E164 to an empty string.

Calling SI

Screening indicator of the calling party number. The following values are available:

You may want to remove the calling party number when it is not screened or screening failed. To do so, route these calls to a mapping that sets the Calling E164 to an empty string. If you want to drop calls when the calling party number is not screened or screening failed, use the Calling Si as criteria for the route.

Calling ITC

The information transfer capability field of the bearer capability information element in the ISDN setup message. The following values are available:

The Mediatrix Boss currently supports the following Information Transfer Capabilities when receiving calls to and from the ISDN (named as in Q.931, 05/98):

Speech

Table 241: Presentation Indicator Values

Value Description

allowed Presentation of the calling party number is allowed.

restricted Presentation of the calling party number is restricted.

interworking The calling party number is not available due to interworking.

Table 242: Screening Indicator Values

Value Description

not-screened

The user provides the calling party number but the number is not screened by the network. Thus the calling party possibly sends a number that it does not own.

passed The calling party number is provided by the user and it passes screening.

failed The calling party number is set by the user and verification of the number failed.

network The originating network provides the number in the calling party number parameter.

Table 243: Information Transfer Capability Values

Value Description

speech Voice terminals (telephones).

unrestricted Unrestricted digital information (64 kbps).

restricted Restricted digital information (64 kbps).

3.1Khz Transparent 3.1 kHz audio channel.

udi-ta Unrestricted digital information with tones/announcements.

Note: This was formerly transparent 7.1 kHz audio channel.

video Video conference terminals.

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Unrestricted Digital Information

3.1 kHz Audio

Those are respectively referenced as Speech, Unrestricted and 3.1 kHz in the call routing configuration.

When initiating calls towards the ISDN, the Mediatrix Boss uses the calling ITC value if it is one of the three listed above. If none is set, it uses 3.1 kHz Audio. If the calling ITC set by the call router is different from the three listed above, the call is rejected.

Date/Time

Day of week and time period and/or date and time period. The following are the accepted formats:

Many of the formats above can be concatenated to form one expression. They must be separated by |. For instance: 25.12.2006 | SUN.

Called / Calling Phone Context

This is a user parameter in a URI. For instance:

sip:1234;[email protected];user=phone

You can enter a regular expression (called/calling party phone context in the call setup message) as per “Regular Expressions” on page 454.

Called / Calling SIP Username

Calling and called party SIP username. See “Regular Expressions” on page 454 for more details on how to enter a proper expression.

Note: Terminals connected to analog extensions (e.g. of a PBX) do not supply information transfer capability values in their call setup. The configuration of the analog port on the Terminal Adapter, NT or PBX is thus responsible to insert this value. The configuration of this value is however often omitted or wrong. The ITC value may therefore not be a reliable indication to differentiate between analogue speech, audio or Fax Group 3 connections. Furthermore, calls from SIP interfaces do not differentiate between bearer capabilities. They always set the information transfer capability property to 3.1Khz.

Table 244: Date/Time Accepted Formats

Format Description

Date/Time Period format • 'DD.MM.YYYY/HH:MM:SS-DD.MM.YYYY/HH:MM:SS'

• 'DD.MM.YYYY/HH:MM:SS-HH:MM:SS'

• 'DD.MM.YYYY-DD.MM.YYYY'

• 'DD.MM.YYYY'

• 'HH:MM:SS-HH:MM:SS'

Week Day/Time Period format • 'DDD'

• 'DDD,DDD...'

• 'DDD/HH:MM:SS-HH:MM:SS'

• 'DDD,DDD.../HH:MM:SS-HH:MM:SS'

DDD must be one of: SUN, MON, TUE, WED, THU, FRI, SAT.

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Last / Original Diverting Reason

This is the last or original diverting reason in ISDN setup and SIP INVITE messages. The following values are available:

Refer to “Diversion Configuration” on page 361 to select the SIP method used to receive/send call diversion information in an INVITE.

Last / Original Diverting E.164

Last or original party number to which the call was being routed when the first diversion occurred. You can enter a regular expression (called/calling party E.164 number in the call setup message) as per “Regular Expressions” on page 454. Note that:

A PBX may insert or modify the calling party number. Sometimes there is no calling party number at all. This all depends on the equipment you connect to the device.

The Mediatrix Boss cannot filter the redirecting number information element of the SETUP message because it does not support the “calling-Redir-E164” and “Calling-Redir-Reason” routing properties criteria.

Last / Original Diverting Party Number Type


Last / Original Diverting Public Type Of Number

Diverting or original called number public type of number field in the ISDN Setup message. Used only when the diverting or original called number type of number is 'public'. The following values are available:

Standards Supported • RFC 5806: Diversion Indication in SIP

Table 245: Diverting Reason Values

Value Description

cfb Call Forward on Busy – Allowed.

cfu Call Forward on Unavailable – Restricted

cfnr Call Forward on No Answer – Interworking

unknown unknown

Table 246: Diverting Party Number Type Values

Value Description


public Public number.

private Private number.

Table 247: Diverting Public Type of Number Values

Value Description


international International number.

national National number.

network-specific

Network specific number used to indicate an administration or service number specific to the serving network.

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Last / Original Diverting Private Type Of Number

Diverting or original called number private type of number field in the ISDN Setup message. Used when the diverting or original called party number type is 'private'. The following values are available:

Last / Original Diverting Number Presentation

Diverting or original called number presentation. The following values are available:

Special Tags

You can use the following special tags as routing types values.



Table 248: Diverting Private Type of Number Values

Value Description

unknown Unknown.

leg2-reg Leg2 reg.

leg1-reg Leg1 reg.

pisn-specific

PISN Specific.



Table 249: Diverting Presentation Values

Value Description

allowed Presentation of the party number is allowed.

restricted Presentation of the party number is restricted.

interworking The party number is not available due to interworking.

restricted-address

Restricted address.

Table 250: Special Tags

Tag Description

undefined Matches if the property is not defined for the call.

default Always matches. Generally used to set a default route if the previous criteria do not match.

Table 247: Diverting Public Type of Number Values (Continued)

Value Description

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Call Properties Parameters

The following sections describe the parameters used by the various call properties (routing types) and interfaces of the call router.

Call Properties to SIP

This section describes the information the call router uses for the various SIP fields.

Table 251: Call Properties to SIP

SIP Field Description

To The Mediatrix Boss uses the calling URI to populate the To field if not undefined. Otherwise, the unit does the following:

• Uses the called Name for the friendly name if not undefined.

• Uses the called SipUsername for the user name if not empty or undefined; otherwise, uses the called E164 for the username. If it is empty or undefined, the Mediatrix Boss rather uses the value defined in the Default Username Value field of the SIP > Interop > SIP Interop parameters as username (see “SIP Interop” on page 336 for more details). The unit uses the called Phone Context for the user's 'phone-context' parameter if not empty. If a 'phone-context' parameter is added, the URI parameter 'user' is also automatically added. Its value is defined in the SIP URI User Parameter Value field of the SIP > Interop > SIP Interop parameters. If empty, then the value 'phone' is used

• Uses the called Host for the host if not undefined, otherwise uses the configured home domain proxy host.

• Prefixes the user name with “+” and adds the URI parameter “user” with the value “phone” if the called TON is “international”.

• If there is no URI parameter “user” yet and the SIP URI User Parameter Value field of the SIP > Interop > SIP Interop parameters is not empty, then the parameter is added with the value defined by the field.

From The Mediatrix Boss uses the called URI to populate the From field if not undefined. Otherwise, the unit does the following:

• Uses the calling Name for the friendly name if not undefined.

• Uses the calling SipUsername for the user name if not empty or undefined; otherwise, uses the calling E164 for the username. If it is empty or undefined, the Mediatrix Boss rather uses the value defined in the Default Username Value field of the SIP > Interop > SIP Interop parameters as username (see “SIP Interop” on page 336 for more details).The unit uses the calling Phone Context for the user's 'phone-context' parameter if not empty. If a 'phone-context' parameter is added, the URI parameter 'user' is also automatically added. Its value is defined in the SIP URI User Parameter Value field of the SIP > Interop > SIP Interop parameters. If empty, then the value 'phone' is used.

• Uses the calling Host for the host if not undefined, otherwise uses the configured home domain proxy host.

• Prefixes the user name with “+” and adds the URI parameter “user” with the value “phone” if the calling TON is “international”.

• If there is no URI parameter “user” yet and the SIP URI User Parameter Value field of the SIP > Interop > SIP Interop parameters is not empty, then the parameter is added with the value defined by the field.

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SIP to Call Properties

This section describes the SIP information the call router uses for the various call properties.

Request URI The Mediatrix Boss uses the same information as the To field.

Contact The Mediatrix Boss uses the same information as the From field, but with the current IP address/port for the host.

Diversion A Diversion header is added if the Last Diverting E.164 property is present and not empty. This Diversion header is constructed as follows:

• The username of the URI is set to the value of the Last Diverting E.164 property.

• The host of the URI is set to the configured home domain proxy host.

• The reason field is set according to value of the Last Diverting Reason property:

• cfu: "unconditional"

• cfb: "user-busy"

• cfnr: "no-answer"

• All other values or when undefined: "unknown'.

• The field counter is set to the value of DivertingCounter if the Original Diverting E.164 property is set to empty or undefined, otherwise it is set to DivertingCounter -1.

A second Diversion header is added if the Last Diverting E.164 and Original Diverting E.164 properties are present and not empty. This Diversion header is constructed as follows:

• The username of the URI is set to the value of the Original Diverting E.164 property.

• The host of the URI is set to the configured home domain proxy host.

• The reason field is set according to the value of the Original Diverting Reason property:

• cfu: "unconditional"

• cfb: "user-busy"

• cfnr: "no-answer"

• All other values or when undefined: "unknown'.

The field counter is set to 1.

Table 252: SIP to Call Properties

Property SIP Information

Called URI The URL of the To field.

Calling URI The URL of the From field.

Called Name The friendly name in the To field. The property is undefined if there is no friendly name.

Calling Name The friendly name in the From field. The property is undefined if there is no friendly name.

Called E164 The user name of the Request-Uri field if the user name is a compatible E.164. The prefix “+” and separator “-” are removed. The property is undefined if there is no user name or if it is not compatible.

Table 251: Call Properties to SIP (Continued)

SIP Field Description

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Calling E164 The user name of the From field if the user name is a compatible E.164. The prefix “+” and separator “-” are removed. The property is undefined if there is no user name or if it is not compatible.

Called Host The host of the To field.

Calling Host The host of the Contact field.

Called TON Set to “international” if the To user name is an E.164 with the prefix “+”; otherwise, the property is undefined.

Calling TON Set to “international” if the From user name is an E.164 with the prefix “+”; otherwise the property is undefined.

Called Phone Context Set to the parameter “phone-context” of the user name of the To if the user name is an E.164, otherwise the property is undefined.

Calling Phone Context Set to the parameter “phone-context” of the user name of the From if the user name is an E.164, otherwise the property is undefined.

Called SIP Username Set to the username of the Request-Uri. Note that this does not include the username parameter like the “phone-context”.

Calling SIP Username Set to the username of the From. Note that this does not include the username parameter like the “phone-context”.

Last Diverting Reason If the INVITE contains at least one Diversion header, this value is set according to the reason field value of the first Diversion header:

• "user-busy": cfb

• "unconditional":cfu

• "no-answer": cfna

• All other values: unknown

Otherwise, the property is undefined.

The reason field comparison is not case sensitive.

Original Diverting Reason

If the INVITE contains more than one Diversion header, this value is set according to the reason field value of the last Diversion header:

• "user-busy": cfb

• "unconditional":cfu

• "no-answer": cfna

• All other values: unknown


The reason field comparison is not case sensitive.

Last Diverting E.164 If the INVITE contains at least one Diversion header, this value is set to the username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the first Diversion header converted into an E.164. It can be set to empty if there is no username or if the username is not an E.164.


Original Diverting E.164 If the INVITE contains more than one Diversion header, this value is set to the username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the last Diversion header converted into an E.164. It can be set to empty if there is no username or if the username is not an E.164.


Table 252: SIP to Call Properties (Continued)


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Call Properties to ISDN

This section describes the information the call router uses for the various ISDN information elements.

ISDN to Call Properties

This section describes the ISDN information the call router uses for the various call properties.

Diverting Counter If the INVITE contains at least one Diversion header, this value is set to the sum of the counter field of all Diversion headers. If a diversion header does not contain the counter field, the value 1 is assumed for the header.

All others The property is undefined.

Table 253: Call Properties to ISDN

Information Element Description

Bearer Capabilities If valid, the calling ITC is used to fill the “information transfer capability” (octet 3 [5:1]). Otherwise, the ITC is set to “3.1 kHz audio”. If more than one bearer capability information elements is provided in a prioritized list, they all receive the same ITC. This information element is included in the SETUP message only for outgoing calls.

Calling Party Number Uses the calling E164 to fill the field “number digits” (octet 4).

Uses the calling TON to fill the field “type of number” (octet 3 [7:5]).

Uses the calling PI to fill the field “presentation indicator” (octet 3a [7:6).

Uses the calling SI to fill the field “screening indicator” (octet 3a [2:1]).

Uses the calling NPI to fill the field “numbering plan identification” (octet 3 [4:1]).

Called Party Number Uses the called E164 to fill the field “number digits” (octet 4).

Uses the called TON to fill the field “type of number” (octet 3 [7:5]).

Uses the called NPI to fill the field “numbering plan identification” (octet 3 [4:1]).

Display Uses the calling E164 to fill the field “display information” (octet 3).

Table 254: ISDN to Call Properties

Property ISDN Information

Calling Name Field “display information” (octet 3) of the Display information element, if included in the SETUP Q.931 message.

Called E164 Field “number digits” (octet 4) of the called party information element included in the SETUP Q.931 message.

Calling E164 Field “number digits” (octet 4) of the calling party information element included in the SETUP Q.931 message.

Called TON Field “type of number” (octet 3 [7:5]) of the called party information element included in the SETUP Q.931 message.

Calling TON Field “type of number” (octet 3 [7:5]) of the calling party information element included in the SETUP Q.931 message.

Calling PI Field “presentation indicator” (octet 3a [7:6) of the calling party information element included in the SETUP Q.931 message.

Table 252: SIP to Call Properties (Continued)


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Call Properties to FXS

This section describes the information the call router uses for the various call properties to FXS.

FXS to Call Properties

This section describes the information the call router uses for the various FXS to call properties.

Calling SI Field “screening indicator” (octet 3a [2:1]) of the calling party information element included in the SETUP Q.931 message.

Calling ITC Field “information transfer capability” (octet 3 [5:1]) of the bearer capability information element included in the SETUP Q.931 message.

Called NPI Field “numbering plan identification” (octet 3 [4:1]) of the called party information element included in the SETUP Q.931 message.

Calling NPI Field “numbering plan identification” (octet 3 [4:1]) of the calling party information element included in the SETUP Q.931 message.

All others The property is undefined.

Table 255: Call Properties to FXS

Caller ID Description

Number If the PI property is present and not set to "allowed", the number is "P". Otherwise, the number is set to the value of the E164 property (truncated to the first 20 characters). See “Auto-Routing” on page 507 for details.

Name If the PI property is present and not set to "allowed", the name is "Anonymous". Otherwise, the name is set to the value of the Name property (truncated to the first 50 characters). See “Auto-Routing” on page 507 for details.

Table 256: FXS to Call Properties


Calling E164 If the auto routing is enabled and the E164 field of the Call Router > Auto-routing page is not empty (see “Auto-Routing” on page 507 for details), the value of the E164 field. Otherwise, the property is not present.

Calling Name If the auto routing is enabled and the Name field of the Call Router > Auto-routing page is not empty (see “Auto-Routing” on page 507 for details), the value of the Name field. Otherwise, the property is not present.

Calling SIP Username If the auto routing is enabled and the SIP Username field of the Call Router > Auto-routing page is not empty (see “Auto-Routing” on page 507 for details), the value of the SIP Username field. Otherwise, the property is not present.

Called E164 For automatic calls, the E.164 defined in the Automatic Call Target field of the Telephony > Services page (see “Automatic Call” on page 421 for more details).

For other calls, the dialed digit after the transformation defined in the Transformation field of the Allowed DTMF Map section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on page 370 for more details).

Table 254: ISDN to Call Properties (Continued)

Property ISDN Information

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Call Properties to FXO

This section describes the information the call router uses for the various call properties to FXO.

FXO to Call Properties

This section describes the information the call router uses for the various FXO to call properties.

Called Name For automatic calls, the name specified in the Automatic Call Target field of the Telephony > Services page (see “Automatic Call” on page 421 for more details). The property is not present if the target address does not contain a name.

For other calls, the property is not present.

Called Host For automatic calls, the host specified in the the Automatic Call Target field of the Telephony > Services page (see “Automatic Call” on page 421 for more details). The property is not present if the target address does not contain a host.

For other calls, the host defined in the Target field of the Allowed DTMF Map section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on page 370 for more details). The property is not present if the target host is not configured for the matching DTMF map.

Table 257: Call Properties to FXO


Dialled number The Called E164 property.

Table 258: FXO to Call Properties


Calling E164 If the caller ID is detected, the numbers provided by the caller ID.

If the auto routing is enabled and the E164 field of the Call Router > Auto-routing page is not empty (see “Auto-Routing” on page 507 for details), the value of the E164 field. Otherwise, the property is not present.

Calling Name If the caller ID is detected, the name provided by the caller ID.

If the auto routing is enabled and the Name field of the Call Router > Auto-routing page is not empty (see “Auto-Routing” on page 507 for details), the value of the Name field. Otherwise, the property is not present.

Calling SIP Username If the caller ID is detected, the property is not present.

If the auto routing is enabled and the SIP Username field of the Call Router > Auto-routing page is not empty (see “Auto-Routing” on page 507 for details), the value of the SIP Username field. Otherwise, the property is not present.

Called E164 For automatic calls, the E.164 defined in the Automatic Call Target field of the Telephony > Services page (see “Automatic Call” on page 421 for more details).

For other calls, the dialed digit after the transformation defined in the Transformation field of the Allowed DTMF Map section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on page 370 for more details).

Table 256: FXS to Call Properties (Continued)


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SIP/ISDN Call Default Values

When performing a call from SIP to ISDN or ISDN to SIP, some ISDN informations are missing from the SIP packet. The Mediatrix Boss sets the following default values when the information is missing. You cannot filter on these default values, but you can filter with the “<undefined>” or “<default>” values.

Called Name For automatic calls, the name specified in the Automatic Call Target field of the Telephony > Services page (see “Automatic Call” on page 421 for more details). The property is not present if the target address does not contain a name.

For other calls, the property is not present.

Called Host For automatic calls, the host specified in the the Automatic Call Target field of the Telephony > Services page (see “Automatic Call” on page 421 for more details). The property is not present if the target address does not contain a host.

For other calls, the host defined in the Target field of the Allowed DTMF Map section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on page 370 for more details). The property is not present if the target host is not configured for the matching DTMF map.

Table 258: FXO to Call Properties (Continued)


Table 259: SIP/ISDN Calls Default Values

Parameter Default Value

SIP to ISDN Calls

TON (calling) unknown

TON (called) unknown

NPI (calling and called) unknown

SI (calling) User-side: not-screened

Network-side: network

ITC (calling) 3.1 kHz audio

PI (calling) 1. When the Calling Party Number E.164 is missing: interworking. In this case, this value overrides any value set by the call router.

2. When CLIR is enabled (user-side only): restricted. In this case, this value overrides any value set by the call router.

3. All other cases: allowed. This is the default value if the two cases above do not apply and no value has been set by the call router.

ISDN to SIP Calls

SI (calling) Network-side: The SI in the incoming Calling Party information element is ignored and replaced by one of the following:

1. No calling IA5 digits received: network.2. NPI is not “unknown” nor “ISDN telephony”: network.3. TON is not “international” nor “national”: network, called IA5 digits are

discarded.4. PI is set to “interworking”: network.5. Otherwise: passed.

User-side: not-screened.

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Note that the calling PI, SI, TON and NPI are present in Calling Party information elements in SETUP messages sent by the network-side only when CLIP is enabled. They should always be present in messages sent by the user-side. See “Chapter 17 - ISDN Configuration” on page 229 for more details on CLIP.

PI (calling) Network-side:

1. CLIR enabled: restricted. The PI is set to restricted no matter if a PI is present in the incoming Calling Party IE.

2. CLIR disabled, no IA5 digits provided: interworking.3. CLIR disabled, IA5 digits provided: allowed.

User-side:

1. CLIR disabled, no IA5 digits provided: interworking.2. CLIR disabled, IA5 digits provided: allowed.

ITC (calling) Must be provided in the incoming Bearer Capabilities information element provided by the ISDN peer that initiated the call. There is no default value, the call should be rejected if missing.

TON (called) The Called TON must be provided by the ISDN peer that initiated the call.

TON (calling) unknown

NPI (called The Called NPI must be provided by the ISDN peer that initiated the call.

NPI (calling) unknown

Table 259: SIP/ISDN Calls Default Values (Continued)

Parameter Default Value

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Call Routing Status

The routes, mappings, and hunts currently in use, as well as the available interfaces, are displayed in the Call Router > Status page.

Figure 207: Call Router – Status Web Page

Routes

The routing table contains one or more routes. These routes forward an incoming or outgoing call to another route, interface, or hunt based on a specific call property such as the called party number. It may also use a mapping to modify the call setup message of a call and a signalling property to modify the behaviour of the call at the SIP protocol level.

Once the call router finds a route that matches, it does not check the other routes, even if some of them may still match. The routes sequence is thus very important. The call router follows the routing table rows (routes) as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

When a call arrives, the call router proceeds as follows:

1. It examines the call property as specified with the routes.

To select a route, the call must match all three of the Source, Properties Criteria, and Expression Criteria parameters.

2. It selects the first matching route in the list of routes.

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3. It routes the call to the specified destination interface, hunt, or route.

Creating/Editing a Route

The web interface allows you to create a route or modify the parameters of an existing one.

To create or edit a route:

1. In the web interface, click the Call Router link, then the Route Configuration sub-link.

Figure 208: Call Router – Route Configuration Web Page

2. Locate the Route section.


• If you want to add a route before an existing entry, locate the proper row in the table

and click the button of this row.

• If you want to add a route at the end of the existing rows, click the button at the bottom right of the Route section.

• If you want to edit an existing route, locate the proper row in the table and click the button.

This brings you to the Configure Route panel.

Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route Configuration page will be lost.

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Figure 209: Configure Route Panel

4. Enter one or more sources to compare with the call and match in order to select the route in the Source field.

You can use the Suggestion column’s drop-down menu to select between suggested values, if any. A source may be:

• route-name: The call uses the route name.

• sip-name: The call comes from the SIP interface name.

• isdn-name: The call comes from the ISDN interface name.

• fxs-name: The call destination is set to the FXS interface name.

If you want to use multiple sources, you must separate them by commas.

For instance, if you want to route calls that come from the SIP interface “default”, enter the following value:

sip-default

If you want to route calls that come from the SIP interfaces “default” and “other”, enter the following value:

sip-default,sip-other

Keep in mind that to select a route, the call must match all three of the Source, Properties Criteria, and Expression Criteria parameters.

5. Select a call property to compare with the call and match in order to select the route in the Properties Criteria drop-down menu.

The call router offers several different routing types. Each type specifies which call property the call router examines.

Table 260: Routing Types

Type Description

Called E164 Routes calls based on the called party E.164 number.

Calling E164 Routes calls based on the calling party E.164 number.

Called TON Routes calls based on the called party type of number.

Calling TON Routes calls based on the calling party type of number.

Called NPI Routes calls based on the called party numbering plan indicator.

Calling NPI Routes calls based on the calling party numbering plan indicator.

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6. Enter the expression (related to the call properties selected in the previous step) to compare with the call and match in order to select the route in the Expression Criteria field.

You can use the Suggestion column’s drop-down menu to select between suggested values, if any. See “Routing Type” on page 456 for a list of available values for each call property.

For instance, if the property is Calling TON, you could instruct the call router to look for the following expression:

international

If you have selected the Date/Time property in the above step, you can click the Time criteria editor link and use the editor to easily configure the Date/Time parameters.

Called Name Routes calls based on the display name of the called party.

Calling Name Routes calls based on the display name of the calling party.

Called Host Routes calls based on the signalling IP address or domain name.

Calling Host Routes calls based on the signalling IP address or domain name.

Called URI Routes calls based on the To-URI.

Calling URI Routes calls based on the From-URI.

Calling PI Routes calls based on the presentation indicator.

Calling SI Routes calls based on the screening indicator.

Calling ITC Routes calls based on the information transfer capability.

Date/Time Routes calls based on the date and/or time the call arrived at the call router. A link called Time criteria editor appears on the right of the Expression criteria field. Use it to easily configure the Date/Time type.

Called Phone Context

Routes calls based on the called party phone context.

Calling Phone Context

Routes calls based on the calling party phone context.

Called SIP Username

Routes calls based on the called party SIP username.

Calling SIP Username

Routes calls based on the calling SIP username.

Table 260: Routing Types (Continued)

Type Description

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Figure 210: Date/Time Criteria Editor (Day Time)

• Select between the Day-Time or Time-Period settings in the Select Criteria Type drop-down menu. If you select Time-Period, the editor changes as follows:

Figure 211: Date/Time Criteria Editor (Time Period)

• Select or enter the parameters you want, then click the Add to List button. If a parameter is invalid (for instance, the end date is inferior to the start date), it is displayed in red in the Time Criteria List field.

• To remove an existing parameter, select it in the Time Criteria List field, then click the Remove Selected button.

• To update an existing parameter, select it in the Time Criteria List field, then click the Update Selected button.

• To remove all parameters, click the Clear Parameters button.

• When done, click the Submit button.

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7. If applicable, enter the name of mappings to apply to the call in the Mappings field.

You can enter more than one mapping by separating them with commas. These mappings are executed in sequential order.

You can use the Suggestion column’s drop-down menu to select an existing mapping, if any.

The manipulations are executed before sending the call to the new destination. See “Mappings” on page 476 for more details.

If you leave this field empty, no mapping is required.

8. Select the call signalling property of the route used to modify the behaviour of the call at the SIP protocol level in the Call Signaling drop-down menu.

You must set call signaling properties as defined in “Signalling Properties” on page 486. You can use the Suggestion column’s drop-down menu to select between existing properties, if any.

9. Select the destination of the call when it matches in the Destination field.

You can use the Suggestion column’s drop-down menu to select between suggested values, if any. The destination can be:

• route-name: The call destination is set to the route name.

• hunt-name: The call destination is set to the hunt name.

• sip-name: The call destination is set to the SIP interface name.

• isdn-name: The call destination is set to the ISDN interface name.

• fxs-name: The call destination is set to the FXS interface name.

For instance, if you want to route calls to the hunt “CallCenter”, enter the following:

hunt-CallCenter

10. Click the Submit button.

This brings you back to the main Call Router > Route Configuration web page.

You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that the configuration has been modified but not applied (i.e., the Call Router > Status differs from the Call Router > Route Configuration). The Route Configuration sub-menu is a working area where you build up a Call Router configuration. While you work in this area, the configured parameters are saved but not applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the configuration has been modified but is not applied.

11. Click the Apply button to enable the route.

The current routes applied are displayed in the Call Router > Status web page. You can also see that the yellow Config Modified Yes flag is cleared.

Examples

The following are some examples of routes:

Figure 212: Routes Examples

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Moving a Route

Once the call router finds a routing entry that matches, it does not check the other entries, even if some of them may still match. The routes sequence is thus very important. The call router follows the routing table rows as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

To move a routing entry up or down:


2. Click the Apply button to update the Call Router > Status web page.

Deleting a Route

You can delete a routing row from the table in the web interface.

To delete a routing entry:



Mappings

Mapping entries modify the call setup message of a call. They thus influence the routing decision and/or the setup message leaving the call router. They are specifically called within a route.

Like the routing table, the mapping table finds the first matching entry. It then executes it by manipulating a call property. A mapping always examines one call property and changes another property.

The call router executes all mapping entries that match by following the mapping table rows as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

The mapping may work with three types of call properties:

calling party properties

called party properties

generic properties

Generic properties are used for call properties that apply to both calling and called parties.

The web interface mapping configuration is separated in two parts: Mapping Type and Mapping Expression. You must properly configure both parts for the mapping to work as required.

When a call arrives at the mapping table, the call router proceeds as follows:

1. It examines the call property as specified in the Criteria (input) value of the Mapping Type part.

2. It selects the first matching entry.

3. It replaces the property specified in the Transformation (output) value of the Mapping Expression part with the value of the selected entry.


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Creating/Editing a Mapping Type

The Mapping Type part allows you to define the input call property to match and to define which call property to change. The mapping type then uses one or more corresponding mapping expressions that you can define in “Creating/Editing a Mapping Expression” on page 478.

To create or edit a mapping type:



2. Locate the Mapping Type section.


• If you want to add a mapping type entry before an existing entry, locate the proper row

in the table and click the button of this row.

• If you want to add a mapping type entry at the end of the existing rows, click the button at the bottom right of the Mapping Type section.

• If you want to edit an existing entry, locate the proper row in the table and click the button.

This brings you to the Configure Mapping Type panel.

Figure 214: Configure Mapping Type Panel

4. Enter the name of the mapping in the Name field.

This is the name used in a route when calling a mapping. It must be unique. Media5 suggests to use the type as part of the name for ease of identification.

There must be at least one corresponding mapping expression in the Mapping Expression table with the exact same name. See “Creating/Editing a Mapping Expression” on page 478 for more details.

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5. Select the input call property to compare with the call and match in order to select the mapping in the Criteria drop-down menu.

6. Select the call property to transform in the Transformation drop-down menu.


• Click the Submit button to go back to the main Call Router > Route Configuration web page. You can now define a corresponding mapping expression.

• Click the Submit and Insert Expression button to directly access the proper mapping expression dialog.

Creating/Editing a Mapping Expression

The Mapping Expression part defines the actual transformation to apply to the corresponding mapping type. Each mapping expression must match a mapping type as defined in “Creating/Editing a Mapping Type” on page 477.

To create or edit a mapping expression:



2. Locate the Mapping Expression section.


• If you want to add a mapping expression entry before an existing entry, locate the

proper row in the table and click the button of this row.

• If you want to add a mapping expression entry at the end of the existing rows, click the

button at the bottom right of the Mapping Expression section.


This brings you to the Configure Mapping Expression panel.

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Figure 216: Configure Mapping Expression Panel

4. Enter the name of the mapping expression in the Name field.

This name must match a mapping type as defined in “Creating/Editing a Mapping Type” on page 477. You can use the Suggestion column’s drop-down menu to select an existing mapping type. When a name matches a mapping type, its type is displayed in the Type row as follows:

input type to output type

You can define several mapping expressions with the same name. In that case, the first row matching the call is used. The rows are used in ascending order.

5. Enter the expression (related to this specific input type) to compare with the call and match in order to select the mapping in the Criteria field.

This string differs depending on the input type selected in the Mapping Type part (Criteria drop-down menu). For instance, if your input type is Calling TON, you could instruct the call router to look for the following expression:

international

You can use the Suggestion column’s drop-down menu to select between suggested values, if any. See “Routing Type” on page 456 for a list of available transformation values.

Table 261: Input Type Criteria

Input Type Criteria

None No criteria, always matches.

E164 If the Transformation value of the Mapping Type part is also a generic property, this is applied to both this Calling E164 and Called E164 property.

Called E164 Selects an entry based on the called party E.164 number. You can use wildcards to summarize entries as per “Called / Calling E164” on page 456.

Calling E164 Selects an entry based on the calling party E.164 number. You can use wildcards to summarize entries as per “Called / Calling E164” on page 456.

Name If the Transformation value of the Mapping Type part is also a generic property, this is applied to both this Calling Name and Called Name property.

Called Name Selects an entry based on the display name of the called party. You can use wildcards to summarize entries as per “Called / Calling Name” on page 457.

Calling Name Selects an entry based on the display name of the calling party. You can use wildcards to summarize entries as per “Called / Calling Name” on page 457.

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TON If the Transformation value of the Mapping Type part is also a generic property, this is applied to both this Calling TON and Called TON property.

Called TON Selects an entry based on the called party type of number as per “Called / Calling TON” on page 457.

Calling TON Selects an entry based on the calling party type of number as per “Called / Calling TON” on page 457.

NPI If the Transformation value of the Mapping Type part is also a generic property, this is applied to both this Calling NPI and Called NPI property.

Called NPI Selects an entry based on the called party numbering plan indicator as per “Called / Calling NPI” on page 457.

Calling NPI Selects an entry based on the calling party numbering plan indicator as per “Called / Calling NPI” on page 457.

Host If the Transformation value of the Mapping Type part is also a generic property, this is applied to both this Calling Host and Called Host property.

Called Host Selects an entry based on the remote signalling IP address or domain name of the destination VoIP peer. You can use wildcards to summarize entries as per “Called / Calling Host” on page 457.

Calling Host Selects an entry based on the remote signalling IP address or domain name of the originating VoIP peer. You can use wildcards to summarize entries as per “Called / Calling Host” on page 457.

Calling PI Selects an entry based on the presentation indicator as per “Calling PI” on page 458.

Calling SI Selects an entry based on the screening indicator as per “Calling SI” on page 458.

Calling ITC Selects an entry based on the information transfer capability as per “Calling ITC” on page 458.

URI If the Transformation value of the Mapping Type part is also a generic property, this is applied to both this Calling URI and Called URI property.

Called URI Selects an entry based on the called SIP URI properties. You can use wildcards to summarize entries as per “Called / Calling URI” on page 457.

Calling URI Selects an entry based on the calling SIP URI properties. You can use wildcards to summarize entries as per “Called / Calling URI” on page 457.

Date/Time Selects an entry based on the date and/or time the call arrived at the call router as per “Date/Time” on page 459.

Phone Context

Selects an entry based on the called or calling phone context properties as per “Called / Calling Phone Context” on page 459.


Selects an entry based on the called phone context properties as per “Called / Calling Phone Context” on page 459.


Selects an entry based on the calling phone context properties as per “Called / Calling Phone Context” on page 459.

SIP Username

Selects an entry based on the called or calling SIP username properties as per “Called / Calling SIP Username” on page 459.

Called SIP Username

Selects an entry based on the called SIP username properties as per “Called / Calling SIP Username” on page 459.

Table 261: Input Type Criteria (Continued)

Input Type Criteria

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If you are editing a Date/Time property, you can click the Time criteria editor link and use the editor to easily configure the Date/Time parameters.


Selects an entry based on the calling SIP username properties as per “Called / Calling SIP Username” on page 459.

Last Diverting Reason

Selects an entry based on the last diverting reason properties as per “Last / Original Diverting Reason” on page 460.

Last Diverting E164

Selects an entry based on the last diverting E.164 properties as per “Last / Original Diverting E.164” on page 460.

Last Diverting Party Number Type

Selects an entry based on the party number type of the last diverting number properties as per “Last / Original Diverting Party Number Type” on page 460.

Last Diverting Public Type Of Number

Selects an entry based on the public type of number of the last diverting number properties as per “Last / Original Diverting Public Type Of Number” on page 460.

Last Diverting Private Type Of Number

Selects an entry based on the private type of numbekr of the last diverting number properties as per “Last / Original Diverting Private Type Of Number” on page 461.

Last Diverting Number Presentation

Selects an entry based on the presentation of the last diverting number properties as per “Last / Original Diverting Number Presentation” on page 461.

OriginalDivertingReason

Selects an entry based on the original diverting reason properties as per “Last / Original Diverting Reason” on page 460.

OriginalDivertingE164

Selects an entry based on the original diverting E.164 properties as per “Last / Original Diverting E.164” on page 460.

Original Diverting Party Number Type

Selects an entry based on the party number type of the original diverting number properties as per “Last / Original Diverting Party Number Type” on page 460.

Original Diverting Public Type Of Number

Selects an entry based on the public type of number of the original diverting number properties as per “Last / Original Diverting Public Type Of Number” on page 460.

Table 261: Input Type Criteria (Continued)

Input Type Criteria

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Figure 217: Date/Time Criteria Editor (Day Time)

• Select between the Day-Time or Time-Period settings in the Select Criteria Type drop-down menu. If you select Time-Period, the editor changes as follows:

Figure 218: Date/Time Criteria Editor (Time Period)

• Select or enter the parameters you want, then click the Add to List button. If a parameter is invalid (for instance, the end date is inferior to the start date), it is displayed in red in the Time Criteria List field.

• To remove an existing parameter, select it in the Time Criteria List field, then click the Remove Selected button.

• To update an existing parameter, select it in the Time Criteria List field, then click the Update Selected button.

• To remove all parameters, click the Clear Parameters button.

• When done, click the Submit button.

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6. Enter the transformation (related to this specific output type) to apply in the Transformation field.

You can use the Suggestion column’s drop-down menu to select between suggested values, if any.

If the transformation is to replace part of an expression, it can use the matched group of the criteria. “\0” will be replaced by the whole criteria capability and “\1” to “\9” by the matched group. See “Groups” on page 455 for more details.

See “Routing Type” on page 456 for a list of available transformation values.

Table 262: Output Type Transformation

Output Type Transformation

None No transformation is applied.

E164 If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling E164 and Called E164 properties.

Called E164 Modifies the called party E.164 number as per “Called / Calling E164” on page 456.

Calling E164 Modifies the calling party E.164 number as per “Called / Calling E164” on page 456.

Name If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling Name and Called Name properties.

Called Name Sets the display name of the called party as per “Called / Calling Name” on page 457.

Calling Name Sets the display name of the calling party as per “Called / Calling Name” on page 457.

TON If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling TON and Called TON properties.

Called TON Sets the called party type of number as per “Called / Calling TON” on page 457.

Calling TON Sets the calling party type of number as per “Called / Calling TON” on page 457.

NPI If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling NPI and Called NPI properties.

Called NPI Sets the called party numbering plan indicator as per “Called / Calling NPI” on page 457.

Calling NPI Sets the calling party numbering plan indicator as per “Called / Calling NPI” on page 457.

Host If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling Host and Called Host properties.

Called Host Sets the remote IP address or domain name of the destination VoIP peer as per “Called / Calling Host” on page 457.

Calling Host Sets the remote IP address or domain name of the originating VoIP peer as per “Called / Calling Host” on page 457.

Calling PI Sets the presentation indicator as per “Calling PI” on page 458.

Calling SI Sets the screening indicator as per “Calling SI” on page 458.

Calling ITC Sets the information transfer capability as per “Calling ITC” on page 458.

URI If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling URI and Called URI properties.

Called URI Sets the called URI as per “Called / Calling URI” on page 457.

Calling URI Sets the calling URI as per “Called / Calling URI” on page 457.

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Phone Context

If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling Phone Context and Called Phone Context properties.


Sets the called Phone Context as per “Called / Calling Phone Context” on page 459.


Sets the calling Phone Context as per “Called / Calling Phone Context” on page 459.

SIP Username

If the Criteria value of the Mapping Type part is also a generic property, this is applied to both the Calling SIP Username and Called SIP Username properties.

Called SIP Username

Sets the called SIP Username as per “Called / Calling SIP Username” on page 459.


Sets the calling SIP Username as per “Called / Calling SIP Username” on page 459.

Last Diverting Reason

Sets the last diverting reason properties as per “Last / Original Diverting Reason” on page 460.

Last Diverting E164

Sets the last diverting E.164 properties as per “Last / Original Diverting E.164” on page 460.

Last Diverting Party Number Type

Sets the party number type of the last diverting number properties as per “Last / Original Diverting Party Number Type” on page 460.

Last Diverting Public Type Of Number

Sets the public type of number of the last diverting number properties as per “Last / Original Diverting Public Type Of Number” on page 460.

Last Diverting Private Type Of Number

Sets the private type of number of the last diverting number properties as per “Last / Original Diverting Private Type Of Number” on page 461.

Last Diverting Number Presentation

Sets the presentation of the last diverting number properties as per “Last / Original Diverting Number Presentation” on page 461.

Original Diverting Reason

Sets the original diverting reason properties as per “Last / Original Diverting Reason” on page 460.

Original Diverting E164

Sets the original diverting E.164 properties as per “Last / Original Diverting E.164” on page 460.

Original Diverting Party Number Type

Sets the party number type of the original diverting number properties as per “Last / Original Diverting Party Number Type” on page 460.

Table 262: Output Type Transformation (Continued)


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You cannot use Date/Time as an output type transformation.

7. If applicable, enter the name of one or more subsequent mappings to execute in the Sub Mappings field.

You can enter more than one mapping by separating them with commas. The mappings are executed in sequential order.

You can use the Suggestion column’s drop-down menu to select between existing values, if any.

You may want to send the result of the first mapping to another one. Once the subsequent mapping is finished, the call router continues to check the mapping entries for matching entries. For instance, if the call router is checking the fourth mapping entry and that entry uses subsequent mapping, the call router executes the subsequent mapping, then resumes checking the fifth mapping entry, and so on.

The maximal number of subsequent interleaved mapping is 3.


• Click the Submit button to go back to the main Call Router > Route Configuration web page.


• Click the Submit and Insert Expression button to create another expression for the same type.

9. Click the Apply button to enable the mapping entry.

The current mappings applied are displayed in the Call Router > Status web page. You can also see that the yellow Config Modified Yes flag is cleared.

Original Diverting Public Type Of Number

Sets the public type of number of the original diverting number properties as per “Last / Original Diverting Public Type Of Number” on page 460.

Original Diverting Private Type Of Number

Sets the private type of number of the original diverting number properties as per “Last / Original Diverting Private Type Of Number” on page 461.

Original Diverting Number Presentation

Sets the Presentation of the original diverting number properties as per “Last / Original Diverting Number Presentation” on page 461.

Table 262: Output Type Transformation (Continued)


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Examples

The following are some examples of mappings:

Figure 219: Mappings Examples

Moving a Mapping Type or Expression Row

The mapping entries sequence is very important. The call router follows the mapping table rows as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

To move a mapping entry up or down:

1. In the Mapping Type or Mapping Expression table, either click the or arrow of the row you want to move until the entry is properly located.


Deleting a Mapping Type or Expression Row

You can delete a mapping row from the Mapping Type or Mapping Expression table in the web interface.

To delete a mapping entry:



Signalling Properties

Call signalling specifies how to set up a call to the destination Mediatrix Boss or 3rd party equipment. Call signalling properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol level.

Signaling Properties are applied after mappings rules.

Like the routing table, the signalling properties table finds the first matching entry. It then executes it by modifying the behaviour of the call.

Standards Supported • RFC 3323: A Privacy Mechanism for the Session Initiation Protocol (SIP) (only supports 'none' as Privacy level)

• RFC 3325: Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks (supports 'id' as Privacy level. Accept/send P-Asserted-Identity and P-Preferred-Identity.)


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Creating/Editing a Signalling Property

The web interface allows you to create a signalling property or modify the parameters of an existing one. The signalling properties are called from a route as described in “Routes” on page 470.

To create or edit a signalling property:



2. Locate the Signaling Properties section.


• If you want to add a signalling property entry before an existing entry, locate the proper


• If you want to add a signalling property entry at the end of existing rows, click the button at the bottom right of the Signaling Properties section.


This brings you to the Configure Signaling Properties panel.

Figure 221: Configure Signaling Properties Panel

4. Enter the name of the signalling property in the Name field.

The name must be unique. It will be used in routes to call a specific signalling property as described in “Routes” on page 470.

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5. Select whether or not the early connect feature is enabled in the Early Connect drop-down menu.

When early connect is enabled, the SIP call is connected by sending a 200 OK message instead of a 183 Session Progress message with early media, if the called party answers the call. It allows interoperability with units that do not support the 183 Session Progress with SDP message.

When early connect is disabled, call progress tones or announcements are transmitted in the early SIP dialog.

6. Select whether or not the early disconnect feature is enabled in the Early Disconnect drop-down menu.

This feature is useful to avoid hearing the end of call tone when the far end party terminates the call during a conference.

When early disconnect is:

• enabled, the SIP BYE message is sent upon receiving the ISDN “Disconnect” signal.

• disabled, the SIP BYE message is sent upon receiving the ISDN “Call release” signal.

If early disconnect is enabled but no ISDN “Disconnect” message is received, the SIP BYE message is sent upon receiving an ISDN “Call release” signal as if the early disconnect was disabled.

7. Define the SIP messages destination (where an INVITE is sent) in the Destination Host field.

It can override the Called Host property set by a mapping rule because signalling properties are applied after mappings.

8. Define whether or not to enable the 180 with SDP allowed feature in the Allow 180 SDP drop-down menu.

9. Define whether or not to enable the 183 without SDP allowed feature in the Allow 183 No SDP drop-down menu.

10. Set the privacy level of the call in the Privacy drop-down menu.

Table 263: 180 with SDP Parameters


Enable The unit can send a SDP in the provisional response 180. Thus when the ISDN peer sends an alerting with indication to open the voice (or if the voice is already opened), the unit sends a 180 with SDP. This is the default value.

Disable A SIP 183 with SDP is sent instead of a 180 with SDP. This does not affect the 180 without SDP. This is useful if your proxy has issues receiving 180 with SDP messages.

The SIP 183 with SDP replacing the SIP 180 with SDP is not sent if a 183 with SDP has already been sent.

Table 264: 183 without SDP Parameters


Enable When enabled, the unit sends a 183 without SDP upon receiving an ISDN progress indicator without any indication to open a voice stream. This is the default value.

Disable When disabled, nothing is sent instead of a 183 without SDP. This does not affect the 183 with SDP. This is useful if your proxy has issues receiving 183 without SDP messages.

Table 265: Privacy Levels

Level Effects on incoming SIP call Effects on outgoing SIP call

Disable None None

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11. Enter the name of one or more SIP headers translation to apply to the call in the SIP Headers Translations field.

You must define SIP headers translations as defined in “SIP Headers Translations” on page 490. You can use the Suggestion column’s drop-down menu to select between existing translations, if any.

You can enter more than one translation. In that case, the translations are separated with “,” and are executed in sequential order.

12. Enter the name of one or more call properties translation to apply to the call in the Call Properties Translations field.

You must set call properties translations as defined in “Call Properties Translations” on page 493. You can use the Suggestion column’s drop-down menu to select between existing translations, if any.

You can enter more than one translation. In that case, the translations are separated with “,” and are executed in sequential order.




14. Click the Apply button to enable the signalling property entry.

The current properties applied are displayed in the Call Router > Status web page. You can also see that the yellow Config Modified Yes flag is cleared.

None None Adds two headers:

• Privacy: none

• P-Asserted-Identity: p_asserted_identity_value

p_asserted_identity_value is the call's From URI unless a SIP header translation has been added to the Signaling Properties for the Identity-header.

Id The calling-name is empty and the PI is set to restricted.

Always adds one header:

• P-Preferred-Identity: p_preferred_identity_value

p_ preferred _identity_value is the call's From URI unless a SIP header translation has been added to the Signaling Properties for the Identity-header.

If the incoming call's PI property is restricted, another header is added:

• Privacy: id

Table 265: Privacy Levels (Continued)

Level Effects on incoming SIP call Effects on outgoing SIP call

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Examples

The following are some examples of signalling properties:

Figure 222: Signalling Properties Examples

Moving a Signalling Property Row

The signalling properties entries sequence is very important. The call router follows the signalling properties table rows as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

To move a signalling property entry up or down:



Deleting a Signalling Property Row

You can delete a signalling property row from the table in the web interface.

To delete a signalling property entry:



SIP Headers Translations

A SIP Headers Translation overrides the default value of SIP headers in an outgoing SIP message. It modifies the SIP headers before the call is sent to its destination.

Like the routing table, the SIP headers translation table finds the first matching entry. It then executes it by modifying the behaviour of the call.

Creating/Editing a SIP Headers Translation

The web interface allows you to create a SIP header translation or modify the parameters of an existing one. The SIP headers translations are called from a signalling property as described in “Signalling Properties” on page 486.

To create or edit a SIP headers translation:




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2. Locate the SIP Headers Translations section.


• If you want to add a SIP headers translation before an existing entry, locate the proper


• If you want to add a SIP headers translation at the end of existing rows, click the button at the bottom right of the SIP Headers Translations section.


This brings you to the Configure SIP Headers Translation panel.

Figure 224: Configure SIP Headers Translation Panel

4. Enter the name of the SIP headers translation in the Name field.

5. Set which SIP header is modified by this translation in the SIP Header drop-down menu.

6. Set what information is used to build the selected SIP header in the Built From drop-down menu.

Table 266: SIP Headers

SIP Header Description

From Header (Host Part) Host part of the From header's URI.

From Header (User Part) User part of the From header's URI.

Identity Header (Host Part) Host part of the Identity header's URI.

Identity Header (User Part) User part of the Identity header's URI.

Identity Header (Phone Number) Phone number in the Identity header's tel URL.

Request Line (Host Part) Host part of the Request line's URI.

Request Line (User Part) User part of the Request line's URI.

To Header (Host Part) Host part of the To header's URI.

To Header (User Part) User part of the To header's URI.

Table 267: Built From Information

Built From Description

Called E164 Use the called party E.164 property.

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7. If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted in the SIP header in the Fix Value field.

For instance, you could hide the caller’s name in a SIP message by using the From Header (User Part) SIP header and entering “anonymous” in the Fix Value field.




9. Click the Apply button to enable the SIP headers translation entry.


Example

The following is an example of SIP headers translations:

Figure 225: SIP Headers Translations Example

Moving a SIP Headers Translation Row

The SIP headers translation entries sequence is very important. The signalling properties table follows the SIP headers translation table rows as they are entered in the web interface. If you want the signalling properties table to try to match one row before another one, you must put that row first.

To move a SIP headers translation entry up or down:



Destination Host Use the destination host configured in the signalling properties of which this translation is part.

Domain Use the domain name configured in the unit.

Fix Value Use a fix value as defined in the Fix Value field (see Step 7).

Host Name Use the host name configured in the unit.

Local Ip Use the local IP address.

Table 267: Built From Information (Continued)


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Deleting a SIP Headers Translation Row

You can delete a SIP headers translation row from the table in the web interface.

To delete a SIP headers translation entry:



Call Properties Translations

A Call Properties Translation overrides the default value of call properties in an incoming SIP message. It modifies the call properties before the call is sent to its destination.

Like the routing table, the call properties translation table finds the first matching entry. It then executes it by modifying the behaviour of the call.

Creating/Editing a Call Properties Translation

The web interface allows you to create a call properties translation or modify the parameters of an existing one. The call properties translations are called from a signalling property as described in “Signalling Properties” on page 486.

To create or edit a call properties translation:



2. Locate the Call Properties Translations section.


• If you want to add a call properties translation before an existing entry, locate the

proper row in the table and click the button of this row.

• If you want to add a call properties translation at the end of existing rows, click the button at the bottom right of the Call Properties Translations section.


This brings you to the Configure Call Properties Translation panel.


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Figure 227: Configure Call Properties Translation Panel

4. Enter the name of the call properties translation in the Name field.

5. Set which call property is modified by this translation in the Call Property drop-down menu.

6. Set what information is used to build the selected call property in the Built From drop-down menu.

Table 268: Call Properties

Call Property Description

Called E164 Called party E.164 property.

Calling E164 Calling party E.164 property.

Called Name Called party name property.

Calling Name Calling party name property.

Called Uri Called URI name property.

Calling Uri Calling URI name property.

Table 269: Built From Information


Domain Use the domain name configured in the unit.

Fix Value Use a fix value as defined in the Fix Value field (see Step 7).

From Header (Uri) Use the From header's URI.

From Header (Friendly Name) Use the friendly name part of the From header.

From Header (User Part) Use the user part of the From header's URI.

Identity Header (Uri) Use the Identity header's URI.

Identity Header (User Part) Use the user part of the Identity header's URI.

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7. If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted in the call property in the Fix Value field.

For instance, you could hide the callee’s name in a SIP message by using the From Header (User Part) SIP header and entering “anonymous” in the Fix Value field.




9. Click the Apply button to enable the call properties translation entry.


Example

The following is an example of call properties translations:

Figure 228: Call Properties Translations Example

Identity Header (Phone Number) Use the phone number in the Identity header's tel URL. The phone number is not retrieved if the received tel URL is invalid. Only the phone number part is retrieved. Examples:

• Received header: P-Preferred-Identity: <tel:8298749;phone-context=819>

Retrieved phone number: 8298749

• Received header: P-Preferred-Identity: <tel:+8298749>

Retrieved phone number: 8298749

• Received header: P-Preferred-Identity: <tel:8298749>

Retrieved phone number: None, the received header is invalid.

Local Ip Use the local IP address.

Request Line (Uri) Use the Request line's URI.

Request Line (User Part) Use the user part of the Request line's URI.

To Header (Uri) Use the To header's URI.

To Header (Friendly Name) Use the friendly name part of the To header.

To Header (User Part) Use the user part of the To header's URI.

Table 269: Built From Information (Continued)


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Moving a Call Properties Translation Row

The call properties translation entries sequence is very important. The signalling properties table follows the call properties translation table rows as they are entered in the web interface. If you want the signalling properties table to try to match one row before another one, you must put that row first.

To move a call properties translation entry up or down:



Deleting a Call Properties Translation Row

You can delete a call properties translation row from the table in the web interface.

To delete a SIP headers translation entry:



Hunt Service

Routes and mappings only manipulate address properties of a call. The hunt service hunts an incoming call to multiple interfaces. It accepts a call routed to it by a route or directly from an interface and creates another call that is offered to one of the configured destination interfaces. If this destination cannot be reached, the hunt tries another destination until one of the configured destinations accepts the call. When an interface accepts a call, the interface hunting is complete and the hunt service merges the original call with the new call to the interface that accepted the call.

The hunt sequence is very important. The call router follows the hunt rows as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

Creating/Editing a Hunt

The web interface allows you to create a hunt or modify the parameters of an existing one.

To create or edit a hunt:



2. Locate the Hunt section.


• If you want to add a hunt entry before an existing entry, locate the proper row in the

table and click the button of this row.


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• If you want to add a hunt entry at the end of existing rows, click the button at the bottom right of the Hunt section.


This brings you to the Configure Hunt panel.

Figure 230: Configure Hunt Panel

4. Enter the name of the hunt in the Name field.

The name must be unique. If more than one hunt have the same name, only the first hunt is used.

5. Define a list of hunt destinations separated by commas in the Destinations field.

This is the interface, route, or hunt that is tried during the hunt’s interface hunting. The destination can either be:

• route-name: The call destination is the route name.

• hunt-name: The call destination is the hunt name.

• sip-name: The call destination is the SIP interface name.

• isdn-name: The call destination is the ISDN interface name.

• fxs-name: The call destination is the FXS interface name.

For instance:

isdn-Slot2/Pri1, route-test


6. Select the algorithm used to select the order of the destination in the Selection Algorithm drop-down menu.

The algorithm can be:

• Sequential: The hunt tries the destination in the same order as listed. The first destination hunted is the first listed.

• Cyclic: The Mediatrix Boss starts from the destination that follows the destination used for the last hunt. Subsequent calls try another first destination in a round-robin method. For instance, if the destination is set to 'x, y, z', the destination the hunt tries is in the following order:

1. x,y,z

2. y,z,x

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3. z,x,y

4. x,y,z

7. Set the maximal time, in seconds (s), allowed to an interface to handle the call in the Timeout field.

After this timeout has elapsed, the next destination is tried when the current destination does not answer. This feature is useful to ensure a minimal time of response and fallback to other destinations. Some interfaces (e.g. SIP, which has a default timeout of 32 seconds) may wait an arbitrary long time until an answer is returned.

Setting the field to 0 disables the timeout, which means that the call router waits indefinitely for the interface to respond. This does not affect the internal interface timeouts (the ISDN timeout as defined in ITU norms or the SIP transmission timeout) that will eventually stop the call and the call router will try another destination.

Example:

You want a call from ISDN to SIP to fallback to another ISDN interface when the SIP destination cannot be contacted within 5 seconds.

You thus create a hunt with the following destinations in order:

sip-[gateway name], isdn-[fallback interface]

and set the timeout to 5. The Selection Algorithm drop-down menu must be set to Sequential to always try the SIP destination first.

Figure 231: Hunt Timeout Example

The Mediatrix Boss has the following behaviour if the SIP transmission timeout has the default value (32 seconds):

a. A new call comes from an ISDN interface and the call router sets the destination of the call to the isdn-to-sip hunt.

b. The call router starts the hunt timeout (5 s) and tries the first destination sip-default.

c. The SIP interface performs a DNS query to resolve the server name. The DNS result returns server A and server B.

d. The SIP interface sends an INVITE to the server A.

e. The hunt timeout elapses, so the call router cancels the call to the SIP interface and tries the second destination isdn-Slot3/Bri2. The hunt timeout is restarted.

f. The SIP interface continues to send the INVITE retransmission until the SIP transmission timeout elapses. RFC 3261 states that an INVITE request cannot be cancelled until the destination sends a response. If the destination responds before the SIP transmission timeout elapses, a CANCEL or BYE request is sent. The SIP interface will not try to use the server B location.

The Mediatrix Boss has the following behaviour if the SIP transmission timeout is set to 3 seconds:

a. A new call comes from an ISDN interface and the call router sets the destination of the call to the isdn-to-sip hunt.

b. The call router starts the hunt timeout (5 s) and tries the first destination sip-default.

c. The SIP interface performs a DNS query to resolve the server name. The DNS result returns server A and server B.

d. The SIP interface sends an INVITE to the server A.

e. A SIP transmission timeout occurs after 4 seconds and the SIP interface sends an INVITE to the server B.

f. The hunt timeout elapses, so the call router cancels the call to the SIP interface and tries the second destination isdn-Slot3/Bri2. The hunt timeout is restarted.

g. The SIP interface continues to send the INVITE retransmission until the SIP transmission timeout elapses. RFC 3261 states that an INVITE request cannot be cancelled until the

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destination sends a response. If the destination responds before the SIP transmission timeout elapses, a CANCEL or BYE request is sent.

8. Select call rejection causes to continue the hunt in the Causes field.

When an interface has a problem placing a call to the final destination, it drops the call by specifying a drop cause based on Q.850 ISUP drop causes. Separate the causes with commas.

See “Call Rejection (Drop) Causes” on page 499 for a list of drop causes.





10. In the main Call Routing Config web page, click the Apply button to enable the hunt.

The current hunts applied are displayed in the Call Router > Status web page. You can also see that the yellow Config Modified Yes flag is cleared.

Examples

The following are some examples of hunts:

Figure 232: Hunt Example

Call Rejection (Drop) Causes

When a destination interface drops the call, the hunt service must supply a call rejection cause based on Q.850 ISUP drop causes. The Mediatrix Boss offers the following drop causes categories:

Normal Event


Service or Option Not Available

Service or Option Not Implemented

Invalid Message

Protocol Error

Interworking

Note: The maximal response time of a SIP interface is the transmission timeout total of all SIP destination locations + the DNS query time.

Note: You can use any custom code between 1 and 127.

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Normal Event

The following table lists all normal events drop causes. These causes are used to drop the original call.

Table 270: Normal Event Drop Causes

# Cause Description

1 Unassigned (unallocated) number

The calling user requested a destination that cannot be reached because the number is unassigned.

2 No route to specified transit network

The destination is asked to route the call through an unrecognized network. This may mean that:

• The wrong transit network code was dialed.

• The transit network does not serve this equipment.

• The transit network does not exist.

3 No route to destination The called party cannot be reached because the network through which the call has been routed does not serve the destination address.

6 Channel unacceptable The sending entity cannot accept the channel most recently identified for use in this call.

7 Call awarded and being delivered in an established channel

The user has been awarded the incoming call, which is being connected to a channel already established to that user for similar calls.

16 Normal call clearing The call is being cleared because one of the users involved with the call has requested that the call be cleared (usually, a call participant hung up).

17 User busy The called party is unable to accept another call because all channels are in use. It is noted that the user equipment is compatible with the call.

18 No user responding The called party does not respond to a call establishment message with either an alerting or connect indication within the time allotted. The number that is being dialed has an active D-channel, but the far end chooses not to answer.

19 User alerting, no answer

The called party has been alerted but does not respond with a connect indication within the time allotted.

21 Call rejected The remote equipment can accept the call but rejects it for an unknown reason, although it could have accepted it because the equipment sending this cause is neither busy nor incompatible.

22 Number changed The called number indicated by the calling party is no longer assigned.

26 Non-selected user clearing

The user has not been awarded the incoming call.

27 Destination out of order

The destination indicated by the user cannot be reached because the destination's interface is not functioning correctly. This can be a temporary condition, but it could last for an extended period.

28 Invalid number format (incomplete number)

The called party cannot be reached because the called party number is not in a valid format or is not complete.

29 Facility rejected The network cannot provide the facility requested by the user.

30 Response to STATUS ENQUIRY

The STATUS message is generated in direct response to receiving a STATUS ENQUIRY message.

31 Normal, unspecified Reports a normal event only when no other cause in the normal class applies.

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The following table lists all resource unavailable drop causes. These causes are used to hunt the next destination.

Service or Option Not Available

The following table lists all service or option not available drop causes. These causes are used to drop the original call.

Service or Option Not Implemented

The following table lists all service or option not implemented drop causes. These causes are used to drop the original call.

Table 271: Resource Unavailable Drop Causes

# Cause Description

34 No circuit/channel available

There is no appropriate circuit or channel presently available to handle the call (usually, no B-channels are available to make the selected call).

38 Network out of order The network is not functioning properly and the condition is likely to last for an extended period.

41 Temporary failure The network is not functioning properly and the condition should be resolved quickly.

42 Switching equipment congestion

Cannot reach the destination because the network switching equipment is temporary experiencing high traffic.

43 Access information discarded

The network could not deliver access information to the remote user as requested.

44 Requested circuit/channel not available

The other side of the interface cannot provide the circuit or channel indicated by the requested entity.

47 Resource unavailable, unspecified

The requested channel or service is unavailable for an unknown reason.

Table 272: Service or Option Not Available Drop Causes

# Cause Description

57 Bearer capability not authorized

The user has requested a bearer capability that is implemented on the equipment but the user is not authorized to use it.

58 Bearer capability not presently available

The user has requested a bearer capability that is implemented by the equipment and is currently unavailable.

63 Service or option not available, unspecified

The network or remote equipment cannot provide the requested service option for an unspecified reason.

Table 273: Service or Option Not Implemented Drop Causes

# Cause Description

65 Bearer capability not implemented

The remote equipment does not support the requested bearer capability.

66 Channel type not implemented

The remote equipment does not support the requested channel type.

69 Requested facility not implemented

The remote equipment does not support the requested supplementary service.

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Invalid Message

The following table lists all invalid message drop causes. These causes are used to drop the original call.

Protocol Error

The following table lists all protocol error drop causes. These causes are used to drop the original call.

70 Only restricted digital information bearer capability is available

The calling party has requested an unrestricted bearer service but the remote equipment only supports the restricted version of the requested bearer capacity.

79 Service or option not implemented, unspecified

The network or remote equipment cannot provide the requested service option for an unspecified reason. This can be a subscription problem.

Table 274: Invalid Message Drop Causes

# Cause Description

81 Invalid call reference value

The remote equipment has received a message with a call reference that is not currently in use on the user-network interface.

82 Identified channel does not exist

Indicates a call attempt on a channel that is not configured.

83 A suspended call exists, but this call identity does not

Attempted to resume a call with a call identity that differs from the one in use for any presently suspended calls.

84 Call identity in use The network has received a call suspended request containing a call identity that is already in use for a suspended call.

85 No call suspended The network has received a call resume request containing a call identity information element that does not indicate any suspended call.

86 Call having the requested call identity has been cleared

The network has received a call identity information element indicating a suspended call that has in the meantime been cleared while suspended.

88 Incompatible destination

The remote equipment has received a request to establish a call with compatibility attributes that cannot be accommodated.

91 Invalid transit network selection

Received a transit network identification of an incorrect format was received.

95 Invalid message, unspecified

Received an invalid message event.

Table 275: Protocol Error Drop Causes

# Cause Description

96 Mandatory information element is missing

The remote equipment has received a message that is missing an information element (IE). This IE must be present in the message before the message can be processed.

97 Message type non-existent or not implemented

The remote equipment has received a message with a missing information element that must be present in the message before the message can be processed.

Table 273: Service or Option Not Implemented Drop Causes (Continued)

# Cause Description

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Interworking

The following table lists all interworking drop causes. These causes are used to drop the original call.

Moving a Hunt

The hunt sequence is very important. The call router follows the hunt rows as they are entered in the web interface. If you want the call router to try to match one row before another one, you must put that row first.

To move a hunt entry up or down:



Deleting a Hunt

You can delete a hunt row from the table in the web interface.

To delete a hunt entry:

1. Click the button of the row you want to move.


98 Message not compatible with call state or message type non-existent or not implemented

The remote equipment has received a message that is not allowed while in the current call state.

99 Information element non-existent or not implemented

The remote equipment has received a message that includes information elements or parameters that are not recognized.

100 Invalid information element contents

The remote equipment has received a message that includes invalid information in the information element or call property.

101 Message not compatible with call state

Received an unexpected message that is incompatible with the call state.

102 Recovery on time expiry

A procedure has been initiated by the expiration of a timer in association with error handling procedures.

111 Protocol error, unspecified

An unspecified protocol error with no other standard cause occurred.

Table 276: Interworking Drop Causes

# Cause Description

127 Interworking, unspecified

An event occurs, but the network does not provide causes for the action it takes. The precise problem is unknown.

Table 275: Protocol Error Drop Causes (Continued)

# Cause Description

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Hairpinning

Hairpinning is defined as a call between two telephony endpoints without using SIP.

Hairpinning is only supported between ISDN and R2 endpoints.

The Call Router does not produce an error when configuring a route between telephony interfaces that do not support the hairpinning, but the call will fail if it uses the configured route. This is not limited to direct routes in the route table configuration, but also for calls that use multiple routes and hunts where the source and the final destination are telephony interfaces that do not support hairpinning.

Calls between the following telephony interfaces are allowed:

ISDN -> ISDN

ISDN -> R2

ISDN -> SIP

R2 -> ISDN

R2 -> R2

R2 -> SIP

FXS -> SIP

SIP -> ISDN

SIP -> R2

SIP -> FXS

Hairpinning is thus possible with the Mediatrix 34xx and Mediatrix 36xx models.

Calls between the following telephony interfaces are not supported:

ISDN -> FXS

R2 -> FXS

FXS -> FXS

FXS -> ISDN

FXS -> R2

SIP -> SIP

Hairpinning is not possible with the Mediatrix 33xx models.

The Mediatrix 37xx models partially support hairpinning with their ISDN card.

You can still make a loopback call on the same unit between two interfaces that do not support hairpinning by performing a SIP loopback. To do this, you need to:

make a route from the source telephony interface to a SIP interface (“Routes” on page 470)

associate a “Signalling Properties” to override the SIP destination (“Signalling Properties” on page 486)

make a route from the SIP interface to the destination telephony interface (“Routes” on page 470)

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Configuration Examples

The following are examples of configuration you could do with the call router.

Figure 233: Configuration Examples

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C H A P T E R

37 Auto-Routing Configuration

This chapter describes the auto-routing feature.

Auto-Routing

The auto-routing feature is an aid to call routing configuration. When this feature is enabled, routing rules are automatically generated for all endpoints marked as "Auto-routable". For each auto-routable endpoint, two rules are generated and added to the Call Router: one directing incoming calls from the associated auto-routing SIP gateway to the endpoint, and one sending outgoing calls from the endpoint to the associated auto-routing SIP gateway.

The auto-routing routes are not displayed in the Route Configuration page because you cannot edit them. They are however listed in the Status page and are attributed a type:

User: the route has been manually entered by the user.

Auto: this is an auto-routing route.

Note: Auto-routing can only be used if the username of the endpoint is an E.164 string and the username part of the request-URI of the received INVITE can be converted into an E.164. See “Manual Routing” on page 510 for more details.

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Chapter 37 - Auto-Routing Configuration Auto-Routing

To activate auto-routing:

1. In the web interface, click the Call Router link, then the Auto-routing sub-link.

Figure 234: Call Router – Auto-Routing Web Page

2. In the top section, set the Auto-routing drop-down menu with the proper behaviour.

If you select Enable, routes are automatically added to the Route Table in order to connect the endpoints marked as eligible for auto-routing (see Step 3) and the designated SIP gateway (see Step 4). These automatic routes are displayed in the Call Router > Status page, but do not show up in the Call Router > Route Configuration page.

3. Select the type of criteria to use to create automatic rules from SIP to the telephony endpoints in the Criteria Type drop-down menu.

4. Set the Incoming Mappings field with the name of the properties manipulations associated with the route from the SIP gateway to the endpoint.

You can specify more than one mapping by separating them with ','. They are executed in sequential order. See “Mappings” on page 476 for more details.


Table 277: Criteria Types


E164 The E.164 associated with the endpoint is used as criterion.

Sip Username The SIP username associated with the endpoint is used as criterion.

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5. Set the Outgoing Mappings field with the name of the properties manipulations associated with the route from the endpoint to the SIP gateway.

You can specify more than one mapping by separating them with ','. They are executed in sequential order. See “Mappings” on page 476 for more details.


6. Set the Incoming Signaling Properties field with the name of the signaling properties associated with the route from the SIP gateway to the endpoint.

See “Signalling Properties” on page 486 for more details.


7. Set the Outgoing Signaling Properties field with the name of the signaling properties associated with the route from the endpoint to the SIP gateway.

See “Signalling Properties” on page 486 for more details.


8. Select whether or not automatic routes are generated for the endpoint when auto-routing is enabled in the Auto-routable drop-down menu.

9. Select the SIP gateway to use as the destination of outgoing calls and the source of incoming calls when generating auto-routing rules in the Auto-routing Gateway drop-down menu.

If you leave the field blank, it is the same as disabling the auto-routing feature.

If available, two additional parameters are displayed:

• If an endpoint has a telephone number that is associated with it, it is displayed in the corresponding E164 column. This is the User Name field as configured in the SIP > Registration page as long as the name follows the E.164 syntax.

• If an endpoint has a friendly name that is associated with it, it is displayed in the corresponding Name column. This is the Friendly Name field as configured in the SIP > Registration page.

See “Endpoints Registration” on page 315 for more details.

10. Click the Submit button to enable auto-routing.

The current routes applied are displayed in the Call Router > Status web page. They are added at the end of the routes that are already present, if any. This ensures that the user-defined routes always have precedence over the automatic routes when both types of routes apply to the same endpoint.

Table 278: Auto-routable Parameters


Enable Automatic routes allowing incoming and outgoing calls to and from the endpoint are added to the Route Table when auto-routing is enabled.

Disable Automatic route generation is turned off for this endpoint.

HardwareDependent Automatic routes are generated if the endpoint belongs to an FXS interface.

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Manual Routing

Auto-routing can only be used if the username of the endpoint is an E.164 string and the username part of the request-URI of the received INVITE can be converted into an E.164.

The conversion of a username into an E.164 follows these rules:

The prefix “+” is removed. Note that if the Map Plus To TON International drop-down menu is set to Enable, the call property ‘type of number’ is set to ‘international’. See “Misc Interop” on page 343 for more details.

The visual separator “-” is removed.

The username parameter is removed. The username parameter is a suffix beginning with “;”.

All remaining characters need to be “0123456789*#abcdABCD”.

Examples of conversion:

5551234 --> 5551234

#20 --> #20555-1234 --> 5551234

+1-819-555-1234 --> 18195551234

5551234;parameter --> 55512345551234_parameter --> cannot convert

To use a username not compatible with E.164, you must disable the auto-routing and use manual routes.

Figure 235 gives an example of manual routes for an endpoint using “5550001_paramter” as user.

Figure 235: Manual Routes Example

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C H A P T E R

38 Configuration Script

This chapter describes the configuration script download feature, which allows updating the Mediatrix Boss configuration by transferring a configuration script from a remote server or from the local file system. The Mediatrix Boss is the session initiator, which allows NAT traversal. You can also configure the Mediatrix Boss to automatically update its configuration.

You can also generate a configuration script from the running configuration of the Mediatrix Boss.

Configuration scripts are files containing textual commands that are sent over the network to a Mediatrix Boss. Upon receiving the file, the unit executes each command line in sequence. Script commands can assign values to configuration variables, or execute configuration commands. See “Creating a Configuration Script” on page 526 for more details on how to create a configuration script.

Scripts are written by the system administrator and can be used to accomplish various tasks, such as automating recurrent configuration tasks or batch-applying configuration settings to multiple devices. Scripts can be executed once or periodically at a specified interval. They can also be scheduled to execute when the Mediatrix Boss restarts.

This chapter describes the following:

Configuration script server setup.

Configuration script server parameters.

Configuration download procedure.

Generating a configuration script from the running configuration.

Automatic configuration update parameters.

How to create a configuration script from scratch.

Configuration Script Server

To download a configuration script, you may need to setup the following applications on your computer:

TFTP server with proper root path

SNTP server properly configured

HTTP server with proper root path

HTTPS server with proper root path

Configuring the TFTP Server

When you perform a configuration script download by using the TFTP (Trivial File Transfer Protocol) protocol, you must install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you know how to set the TFTP root path. If not, refer to your TFTP server’s documentation.

Standards Supported • RFC 959: File Transfer Protocol (client-side only)

• RFC 1350: The TFTP Protocol (Revision 2) (client-side only)

• RFC 2616: Hypertext Transfer Protocol - HTTP/1.1 (client-side only)

• RFC 2617: HTTP Authentication: Basic and Digest Access Authentication

• RFC 3617: Uniform Resource Identifier (URI) Scheme for the Trivial File Transfer Protocol

• draft-ietf-http-authentication-03

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Configuring the SNTP Server

When you use the automatic configuration script update feature (see “Automatic Configuration Update” on page 522 for more details) or the HTTPS protocol, you need to have a time server SNTP that is accessible and properly configured. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP server’s documentation.

Configuring the HTTP Server

When you to perform a configuration script download by using the HTTP protocol, you must install a HTTP server running on the PC designated as the server host. It is assumed that you know how to set the root path. If not, refer to your HTTP server’s documentation.

Configuring the HTTPS Server

When you perform a configuration script download that requires authentication or privacy by using the HTTP over the Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the PC designated as the server host. It is assumed that you know how to set the root path and SSL/TLS security configuration. If not, refer to your HTTPS server’s documentation.

When two peers establish a HTTPS connection, they negotiate and decide on a cipher suite to use for data encryption. The client suggests a list of cipher suites and the server selects one that it supports. Some cipher suites are more secured than others. The Mediatrix Boss acts as a client.

The Mediatrix Boss suggests a wide range of cypher suites, which includes cipher suites that are not very secure. The final choice rests with the server and it is thus possible that the transfer uses a SSL/TLS link that is not very secure.

Media5 recommends to use cipher suites based on the RSA key exchange mechanism, because the Diffie-Hellman key exchange mechanism introduces a noticeable delay in the HTTPS session establishment. Furthermore, Media5 recommends using cipher suites based on the following SSL/TLS algorithms:

Note: The Mediatrix Boss hardware does not include a real time clock. The unit uses the SNTP client to get and set its clock. As certain services need correct time to work properly (such as HTTPS), you should configure your SNTP client with an available SNTP server in order to update and synchronise the local clock at boot time.

Standards Supported • RFC 2246: The TLS Protocol Version 1.0

• RFC 2459: X.509 Digital Certificates

• RFC 2818: HTTP Over TLS (client side only)

• RFC 3268: Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS)

• RFC 3280: Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile

Caution: You must have a time server SNTP that is accessible and properly configured, or the automatic configuration update feature may not work properly. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP server’s documentation.

Table 279: Suggested Secure Parameters

Suggested Parameter Description

Key Exchange Mechanism • RSA

• Diffie-Hellman

Ciphers • AES (128 and 256 bits)

• 3DES (168 bits)

Message Digests • SHA-1

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The following six recommended cipher suites are based on the algorithms of Table 279:

Certificates

The Mediatrix Boss contains embedded security certificates formatted as per ITU x.509 and RFC 3280. The certificates are factory-installed. You can also add new certificates as described in “Chapter 39 - Certificates Management” on page 527.

When contacting a HTTPS server, the Mediatrix Boss establishes a TLS connection by (among others):

negotiating cipher suites

checking the server certificates validity (dates)

The Mediatrix Boss then checks the server’s identity by validating the host name used to contact it against the information found in the server’s certificate, as described in RFC 2818, section 3.1.

If any of the above does not succeed, the Mediatrix Boss refuses the secure connection. To help detect such errors, you can increase the syslog messages level.

Generating a Configuration Script from the Running Configuration

The Export Script section allows you to generate a configuration script from the running configuration of the Mediatrix Boss and export it.

You can export the configuration in two ways:

To a URL you specify with one of the supported transfer protocols.

By directly downloading the exported script via your web browser. This option uses the protection provided by your web browser (it is protected if you log on to the unit via HTTPS).

To use this option, you must use one of the following supported browsers:

• Internet Explorer 61, 72 and 82

• Mozilla Firefox

• Google Chrome

Table 280: Recommended Cipher Suites

ID Name

0x0035 TLS_RSA_WITH_AES_256_CBC_SHA

0x0039 TLS_DHE_RSA_WITH_AES_256_CBC_SHA

0x000a TLS_RSA_WITH_3DES_EDE_CBC_SHA

0x0016 TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA

0x002f TLS_RSA_WITH_AES_128_CBC_SHA

0x0033 TLS_DHE_RSA_WITH_AES_128_CBC_SHA

1. Supported from SP2 of Internet Explorer 6.

2. Internet Explorer 7 and 8 must have the “Automatic Prompting” feature enabled. It is located in Tools / Internet Options / Security / Custom Level.

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To export a configuration to a URL:

1. In the web interface, click the Management link, then the Configuration Scripts sub-link.

Figure 236: Management – Configuration Scripts Web Page

2. Select the content to export in the generated configuration script in the Content drop-down menu.

3. Set the Send To URL field with the URL where to send the exported configuration script.

The URL should follow this format:

protocol://[user[:password]@]hostname[:port]/[path/]filename

The brackets [ ] denote an optional parameter.

The filename may contain a %mac% macro that is substituted by the MAC address of the unit at the moment of sending the configuration script. For instance, the “%mac%.cfg” value for a Mediatrix Boss with MAC address “0090f12345ab” will be “0090f12345ab.xml”.

The transfer protocols supported are:

• TFTP

• FTP

• FILE

Examples of valid URLs:

• tftp://tftpserver.com:69/folder/script.cfg

• ftp://[email protected]/script.cfg

• ftp://username:[email protected]/script.cfg

• file://script.cfg

The protocol’s default port is used if none is specified.

4. Set the Privacy Key field with the key used to encrypt the configuration script to export.

The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F, and a-f. All other characters are not supported.

Table 281: Exported Configuration Script Content


All Config Exports everything.

Modified Config Export only the configuration that has been modified (differs from the default values).

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The maximum key length is 64 characters, which gives a binary key of 32 bytes (256 bits). It is the maximum key size accepted by the MxCryptFile application.

For instance, a 32-bit key could look like the following: A36CB299.

If the field is empty, the configuration script is not encrypted.

To decrypt the exported configuration script, you must use the MxCryptFile application. MxCryptFile is a command line tool that encrypts or decrypts files to be exchanged with the Mediatrix Boss. Contact your sales representative for more details.

5. Initiate the configuration scripts exportation by clicking the Submit & Export Now button at the bottom of the page.

The Mediatrix Boss immediately generates and transfers a configuration script based on the export settings set in the previous steps.

To export a configuration and download it with your web browser:

1. In the web interface, click the Management link, then the Configuration Scripts sub-link.

Figure 237: Management – Configuration Scripts Web Page

2. Select the content to export in the generated configuration script in the Content drop-down menu.

3. Initiate the configuration scripts exportation by clicking the download link at the left of the Content field.

The Mediatrix Boss immediately generates a configuration script based on the export settings set in the previous steps and sends it to the web browser, using the MAC address of the unit as the default filename.

The download link allows you to directly download the exported script via your web browser instead of using the URL defined in the Send To URL field. This option does not require a Privacy Key because the protection is provided via the web (it is protected if you log on to the unit via HTTPS).

Table 282: Exported Configuration Script Content


All Config Exports everything.

Modified Config Export only the configuration that has been modified (differs from the default values).

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Configuration Scripts Server Settings

This section describes how to configure the IP address and port number of the configuration script server. This server contains the configuration scripts the Mediatrix Boss will download.

When performing a configuration script download, you can download two different scripts:

A generic configuration script that should be used to update a large number of units with the same configuration.

A specific configuration script that contains the configuration for a single unit, for instance the telephone numbers of its endpoints.

You can use a specific configuration script but no generic configuration script or vice-versa. You can also use both generic and specific configuration scripts. When both the generic and specific configuration scripts are downloaded, settings from the specific configuration script always override the settings from the generic configuration script. These scripts must be located in the same directory.

Each script is executed independently from the other one. A script that is empty, cannot be found or has an invalid syntax does not prevent the execution of the other script. If one or both scripts fail, error messages are sent.

To set the configuration scripts server parameters:

1. In the Configuration Scripts section of the Configuration Scripts page, set the name of the generic configuration script to download in the Generic File Name field.

This script should be used to update a large number of units with the same configuration. The script name is case sensitive hence it must be entered properly.

This field may contain some macros that are substituted by the actual value at the moment of fetching the configuration script. The supported macros are:

• %mac% - the MAC address of the unit

• %version% - the MFP version of the unit

• %product% - the Product name of the unit.

• %productseries% - the Product series name of the unit.

For instance, the “%mac%.xml” value for a Mediatrix Boss with MAC address “0090f12345ab” will be “0090f12345ab.xml”.

If you leave the field empty, the Mediatrix Boss does not download the generic configuration script.

Figure 238: Automatically Update Scripts Section

2. Set the name of the specific configuration script to download in the Specific File Name field.

This script should be used to update the configuration of a single unit. The script name is case sensitive hence it must be entered properly.

This field may contain a macro that is substituted by the actual value when downloading the configuration script. The Mediatrix Boss supports the %mac% macro, which will be substituted by the MAC address of the unit. For instance, the “%mac%.xml” value for a Mediatrix Boss with MAC address “0090f12345ab” will be “0090f12345ab.xml”.

This field may contain some macros that are substituted by the actual value when downloading the configuration script. The supported macros are:



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If the variable is empty (after macro substitution), the Mediatrix Boss does not download the specific configuration script.

3. Set the path of the directory where the configuration scripts are located in the Location field.

The path is case sensitive hence it must be entered properly. It is relative to the root of the configuration scripts server. Use the “/” character when defining the path to indicate sub-directories.

This field may contain some macros that are substituted by the actual value when downloading the configuration script. The supported macros are:






The path differs depending on the transfer protocol selected (see Step 5).

Let’s consider the following example for all protocols except File:

• The directory that contains the configuration script is called: Config_Script.

• This directory is under C:/Root/Download.

The following are some tips to help your download process:

• Use the “/” character when defining the path to indicate sub-directories. For instance, root/download.

• If you are using the TFTP protocol to download the software, note that some TFTP servers on Windows do not recognize the “/” character and produce an error. In this case, use the “\” character.

• Use basic directory names, without spaces or special characters such as “~”, “@”, etc., which may cause problems.

• Cut and paste the path and/or name of the directory that contains the extracted scripts into the configuration download path of the Mediatrix Boss (you may have to convert “\” into “/”) to eliminate typographical errors.

Note that you can define the C:/Root/Download part as you want. The script names may also differ from the example shown above.

When the Transfer Protocol is set to File, you may prefix the path by one of the following to indicate storage media:

• Persistent: for onboard persistent storage. The configuration script is saved into the persistent file system of the Mediatrix Boss (in flash memory). This is the default value.

Example: All Transfer Protocols Except File

Table 283: Path Configurations Example

Root Path Corresponding Path Name

c:/root/download Config_Script

c:/ root/download/Config_Script

c:/root download/Config_Script

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• Volatile: for onboard non-persistent storage. The configuration script is saved into the non-persistent RAM memory of the Mediatrix Boss. All information is lost the next time the unit restarts.

4. Set the transfer protocol to transfer the configuration scripts in the Transfer Protocol field.

You can select from five different transfer protocols:

• HTTP: HyperText Transfer Protocol.

• HTTPS: HyperText Transfer Protocol over Transport Layer Security.

• TFTP: Trivial File Transfer Protocol.

• FTP: File Transfer Protocol. Note that the Mediatrix Boss FTP client does not support the EPSV command.

• File: Complete path to a configuration image in a storage device.

Note that if the firewall is enabled on the Comdasys device, you cannot use the TFTP protocol to submit and run configuration scripts. If the firewall is disabled, you can use the TFTP protocol.

HTTP and HTTPS support basic or digest authentication mode as described in RFC 2617. HTTPS requires a valid certificate.

If you have selected HTTP or HTTPS, please note that your server may activate some caching mechanism for the script download. This mechanism caches the initial script download for later processing, thus preventing changes or update of the original script. This can cause strange problems if you want to edit a configuration script to modify values and upload it immediately. The result will still return the original script and not the new one.

5. If your server requires authentication when downloading the configuration script, set the following:

• The user name in the User Name field.

• The password in the Password field.

6. Set the static configuration scripts server IP address or domain name and port number in the Host Name field.

This is the current address of the PC that hosts the configuration scripts.

Use the special port value zero to indicate the protocol default. For instance, the TFTP default port is 69, the HTTP default port is 80, and the HTTPS default port is 443.

The default value is 0.0.0.0:0.

7. Set the key used to decrypt configuration scripts when they are encrypted in the Privacy key field.

You can secure the exchange of configuration scripts between the server and the Mediatrix Boss. A privacy key allows the unit to decrypt a previously encrypted configuration script.

To encrypt a configuration script, you must use the MxCryptFile application. MxCryptFile is a command line tool that encrypts files before sending them to the Mediatrix Boss. Contact your sales representative for more details.

The password is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F, and a-f. All other characters are not supported.

Each character encodes 4 bits and the maximum key length is 112 characters, which gives a binary key of 56 bytes. It is the maximum required by the MxCryptFile application.

Table 284: Path Configurations Example (File)

Location Corresponding Path Name

Onboard persistent storage of the Mediatrix Boss under the directory “Script-1”

Persistent:Script-1

or

Script-1

Note: The configuration script download via TFTP can only traverse NATs of types “Full Cone” or “Restricted Cone”. If the NAT you are using is of type “Port Restricted Cone” or “Symmetric”, the script transfer will not work.

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For instance, a 32-bit key could look like the following: A36CB299.

This key must match the key used for the encryption of the relevant configuration script.

The Mediatrix Boss will be able to decrypt the next encrypted configuration script.

If the field is empty, the configuration script is not decrypted by the unit and the configuration update fails.

Encryption is auto-detected.


Certificate Validation


When downloading a script from an HTTPS server, you can define the level of security to use when validating the server's certificate.

To set the certificate validation parameter:

1. Set the Call Waiting Tone configuration by putting the following line in the configuration script:conf.ScriptsTransferCertificateValidation="Value"


Table 285: Certificate Validation Parameters


NoValidation Allow a connection to the server without validating its certificate. The only condition is to receive a certificate from the server. This option provides partial security and should be selected with care.

HostName Allow a connection to the server by validating its certificate is trusted and valid. The validations performed on the certificate include the expiration date and that the Subject Alternate Name (SAN) or Common Name (CN) matches the FQDN or IP address of the server.

Table 286: Certificate Validation Values

Value Meaning

100 NoValidation

200 HostName

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Configuration Download Procedure

The following steps explain how to download configuration scripts from the web interface.

To download configuration scripts:

1. Place the configuration scripts to download on the computer hosting the configuration scripts server.

These scripts must be in a directory under the server’s root path.

2. Initiate the configuration scripts download by clicking the Submit & Run Now button at the bottom of the page.

The Mediatrix Boss immediately downloads the configuration scripts.

Automatic Configuration Update

This section describes how to configure the Mediatrix Boss to automatically update its configuration. This update can be done:

Every time the Mediatrix Boss restarts.

At a specific time interval you can define.

NAT Variations

NAT treatment of UDP varies among implementations. The four treatments are:

• Full Cone: All requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host by sending a packet to the mapped external address.

• Restricted Cone: All requests from the same internal IP address and port are mapped to the same external IP address and port. Unlike a full cone NAT, an external host (with IP address X) can send a packet to the internal host only if the internal host had previously sent a packet to IP address X.

• Port Restricted Cone: Similar to a restricted cone NAT, but the restriction includes port numbers. Specifically, an external host can send a packet, with source IP address X and source port P, to the internal host only if the internal host had previously sent a packet to IP address X and port P.

• Symmetric: All requests from the same internal IP address and port, to a specific destination IP address and port, are mapped to the same external IP address and port. If the same host sends a packet with the same source address and port, but to a different destination, a different mapping is used. Furthermore, only the external host that receives a packet can send a UDP packet back to the internal host.

For more details on NAT treatments, refer to RFC 3489.

Note: The configuration download via TFTP can only traverse NATs of types “Full Cone” or “Restricted Cone”. If the NAT you are using is of type “Port Restricted Cone” or “Symmetric”, the file transfer will not work.

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Automatic Update on Restart

The Mediatrix Boss may download new configuration scripts each time it restarts.

To set the automatic update every time the Mediatrix Boss restarts:

1. Set the configuration scripts parameters as defined in “Configuration Scripts Server Settings” on page 518.


These scripts must be in a directory under the root path.

3. In the Automatically Update Scripts section of the Configuration Scripts page, select Enable in the Update On Restart drop-down menu.


The automatic configuration update will be performed each time the Mediatrix Boss restarts.

The unit configuration is only updated if at least one parameter value defined in the downloaded configuration scripts is different from the actual unit configuration.


Automatic Update at a Specific Time Interval

You can configure the Mediatrix Boss to download new configuration scripts at a specific day and/or time.

To set the automatic update at a specific time interval:

1. Set the configuration scripts parameters as defined in “Configuration Scripts Server Settings” on page 518.


These scripts must be in a directory under the root path.

3. In the Automatically Update Scripts section of the Configuration Scripts page, select Enable in the Update Periodically drop-down menu.


4. Select the time base for configuration updates in the Time Unit drop-down menu.

You can specify the x value in the Period field (see Step 5).

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Table 287: Time Unit Parameters


Minutes Updates the unit’s configuration every x minutes.

Hours Updates the unit’s configuration every x hours.

Days Updates the unit’s configuration every x days. You can define the time of day when to perform the update in the Time of Day field (see Step 6).

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5. Set the waiting period between each configuration update in the Period field.

Available values are from 1 to 60. The time unit for the period is specified in the Time Unit field (see Step 4).

6. If you have selected Days in Step 4, set the time of the day when to initiate a configuration update in the Time of Day field.

You must have a time server SNTP that is accessible and properly configured or the automatic configuration update feature may not work properly. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP server’s documentation.

The configuration scripts are downloaded at the first occurrence of this value and thereafter at the period defined by the Period field. Let’s say for instance the automatic unit configuration update is set with the time of day at 14h00 and the update period at every 2 days.

• If the automatic update is enabled before 14h00, the first update will take place the same day at 14h00, the second update two days later at the same hour, and so on.

• If the automatic update is enabled after 14h00, the first update will take place the day after at 14h00, the second download two days later at the same hour, and so on.

Available values are -1, and from 0 to 23.

When setting the variable to -1, the time of the day at which the Mediatrix Boss first downloads the configuration scripts is randomly selected.


DHCP Auto-Provisioning

You can configure the Mediatrix Boss to automatically download new configuration scripts upon receiving options 66 (tftp-server) or 67 (bootfile) in a DHCP answer. A DHCP answer includes both Bound and Renew.

The contents of the option 66 or 67 defines which script to download. The unit’s configuration is not used to download the script. This allows the unit, for instance, to download a script from a server after a factory reset and to reconfigure itself without a specific profile.

The syntax of options 66 and 67 is as follows:

[FileType] = [protocol]://[username] :[password]@[fqdn server]/[path]

For instance:

Script=https://admin :[email protected]/Mx3000config/%mac%.cfg

The Mediatrix Boss supports only the Script file type for now.

The following is an example of a valid option 67 (Bootfile):

Option: (t=67, l=53) Bootfile name = "Script=http://192.168.50.1/digest/%mac%__2.0.6.84.cfg"

Option: (67) Bootfile name

Length: 53Value: 5363726970743D687474703A2F2F3139322E31136382E3530...


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To set DHCP auto-provisioning:

1. In the Automatically Update Scripts section of the Configuration Scripts page, set the DHCP Download Enable drop-down menu with the proper behaviour.


When enabled, the DHCP options tftp-server (option 66) and bootfile (option 67) are used to download a configuration script. If this configuration script is identical to the last executed script, it will not be run again. The script retry mechanism is not enabled for the DHCP triggered scripts (see “Configuration Scripts Server Settings” on page 518 for more details).

If the two options are received, both scripts are executed independently. The script defined by the tftp-server (option 66) option is executed first.

If you are using HTTPS to transfer scripts, you must have a time server SNTP that is accessible and properly configured. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP server’s documentation.

When a DHCP download script is configured in HTTPS, the script execution is deferred and a 30 seconds timer is started to let enough time for the NTP synchronization.

This timer is independent for each HTTPS script launched. If, for instance, a DHCP answer has both option 66 and 67 configured in HTTPS and if the Update on Restart feature is used, up to 1 min 30 seconds can pass before any other operation such as backup, restore, script execution can be processed.

Once the NTP synchronization is established, the deferred scripts are started immediately one after the other, ending the timer.

When synchronization is already established, there is no timer, even in HTTPS.


Number of Retries


When using the automatic configuration update (on restart or at a specific interval), the Mediatrix Boss may encounter a problem upon restarting the unit (such as a DHCP server problem) that prevents the update to succeed. You can define a maximum number of attempts to retry a script transfer until it succeeds when it fails upon an automatic transfer on restart or automatic periodic transfer. The retries are only attempted if the server is unreachable. Unreachable port or file not found errors don't trigger the retry mechanism.The time interval between each retry is 30 seconds.

To set the number of retries:

1. Set the number of retries by putting the following line in the configuration script:conf.scriptsTransferRetriesNumber="Value"

where Value may be as follows

• -1 means a retry to infinity.

• 0 means no retry.

The maximum number of retries is 100.

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Chapter 38 - Configuration Script Creating a Configuration Script

Creating a Configuration Script

Configuration scripts are text files that contain command lines interpreted by the Mediatrix Boss. Most commands contained in a script assign values to configuration variables. Script commands can also execute configuration commands. This configuration script can then be downloaded into the Mediatrix Boss as described in the current chapter.

Writing configuration scripts requires a bit of knowledge about the Mediatrix Boss’s configuration variables tree structure.

Configuration scripts use the Media5 proprietary scripting language, as described in “Appendix B - Scripting Language” on page 547.

Refer to “Appendix B - Scripting Language” on page 547 for samples of configurations you can use in a configuration script. The samples include the configuration required to perform a basic call between an ISDN telephone and an analog telephone.

526 Mediatrix Boss

C H A P T E R

39 Certificates Management

This chapter describes how to transfer and manage certificates into the Mediatrix Boss.

Introduction

The Mediatrix Boss uses digital certificates, which are a collection of data used to verify the identity of the holder or sender of the certificate.

The certificates contain the following information:

certificate name

issuer and issued to names

Validity period (the certificate is not valid before or after this period)

Usage of the certificate (Identifies in which role or context a certificate can be used by the host it authenticates).

• TlsClient: The certificate identifies a TLS client. A host authenticated by this kind of certificate can act as a client in a SIP over TLS connection when mutual authentication is required by the server.

• TlsServer: The certificate identifies a TLS server. A host authenticated by this kind of certificate can serve files or web pages using the HTTPS protocol or can act as a server in a SIP over TLS connection.

whether or not the certificate is owned by a CA (Certification Authority)

The Mediatrix Boss uses two types of certificates:

The transferred certificate must be in Privacy Enhanced Mail (PEM) (host or others) or Distinguished Encoding Rules (DER) (others) format. When transferring a host certificate, the certificate must be appended to the private key to form one PEM file. The private key must not be encrypted.

You can transfer a certificate by using the HTTP or HTTPS protocol, but Media5 recommends to use HTTPS.

To access the unit via HTTPS, your browser must support RFC 2246 (TLS 1.0). Commonly used browsers such as Mozilla Firefox (1.5 and 2.x), Microsoft Internet Explorer 7.x, or Opera 9.x all support HTTPS browsing.

Standards Supported • RFC 3280: Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile

Table 288: Certificates Types

Type Description

Host Certificates used to certify the unit (e.g.: a web server with HTTPS requires a host certificate).

Others Any other certificate including trusted CA certificates used to certify peers (e.g.: a SIP server with TLS).

Mediatrix Boss 527

Chapter 39 - Certificates Management Managing Certificates

Managing Certificates

You can view certificates information and you can delete certificates.

To view and manage certificates:

1. In the web interface, click the Management link, then the Certificates sub-link.

Figure 242: Management – Certificates Information Web Page

The Host Certificates section contains the certificates used to certify the unit. The Others Certificates section contains any other certificate used to certify peers.

2. If applicable, delete a certificate in the Host Certificates or Others Certificates sections by clicking the button of the certificate you want to delete.

3. If applicable, delete a certificate in the Other Certificates section by clicking the button of the certificate you want to delete.


Transferring a Certificate

The following steps explain how to transfer (add) a certificate from the web interface.

To transfer a certificate:

1. If you are currently using an unsecure HTTP access, the Certificate Transfer section is disabled. This is to avoid transferring the certificate in clear text. To enable the section, you can:

• Access the secure site (recommended) by clicking the corresponding link at the top of the window. This is the recommended way to proceed.

• Activate unsecure certificate transfer by clicking the corresponding link at the top of the window. This is not recommended.

2. In the Certificate Transfer section of the Certificates page, select the type of the certificate in the Type drop-down menu.

Before transferring the certificate, you must indicate whether this is a Host or Others certificate.

3

2

528 Mediatrix Boss

Transferring a Certificate Software Configuration Guide

Figure 243: Certificates Transfer Section

3. Use the Browse button to select the certificate to transfer.

The maximum certificate name is 50 characters.

4. Initiate the certificate transfer by clicking the Submit button at the bottom of the page.

The Mediatrix Boss immediately transfers the certificate. Once the certificate is transferred, you must restart the SipEp and Web services in the System > Services page (“Chapter 13 - Services” on page 211) before using the newly transferred certificate. Click the link in the message that is displayed to access the Services web page.

Transferring a Certificate via Configuration Script


You can use a special command to transfer a certificate by configuration script or CLI. This command has the following parameters: URL of the certificate to download, its Type (Host/ Others), the username and password. See “Appendix B - Scripting Language” on page 547 for more details on the Media5 proprietary scripting language

To transfer a certificate via configuration script:

1. Set the Call Waiting Tone configuration by putting the following line in the configuration script:Cert.DownloadCertificate FileUrl=Value UserName=Value Password=Value Type=Value

where the different values may be as follows:

2 3

Table 289: Certificate Transfer Values

Value Description

FileUrl URL to a Certificate file that is loaded upon executing the execution of Download command. The transfer protocols supported are:

• HTTP

• HTTPS

• TFTP

• FTP

Examples of valid URLS:

• http://www.myserver.com/Cert_MxDefault001.der

• tftp://myserver.com:69/myfolder/Cert_MxDefault001.der

When the port is not included in the URL, the default port for the chosen protocol is used.

This field may contain some macros that are substituted by the actual value at the moment of fetching the configuration script. The supported macros are:

• %mac% - the MAC address of the unit.


UserName When authentication is required by the remote file server, this variable is used as the username.

Password When authentication is required by the remote file server, this variable is used as the password.

Mediatrix Boss 529

Chapter 39 - Certificates Management Host Certificate Associations

For instance, a valid command would be:

Cert.DownloadCertificate FileUrl=http://www.myserver.com/Cert_MxDefault001.der UserName=MyName Password=MyPassword Type=Host

Host Certificate Associations

The Host Certificate Associations section allows you to define which services can use the host certificates.

To set host certificate associations:

1. In the Host Certificate Associations section of the Certificates page, check the services that can use a given host certificate.

Figure 244: Host Certificate Associations Section


Type Type of certificate to transfer.

• Host: Certificate used to certify the host system.

• Other: Remote systems certificates and issuers certificates.

Table 289: Certificate Transfer Values (Continued)

Value Description

Table 290: Host Certificate Associations Parameters


SIP Specifies if this certificate can be used for SIP security.

Web Specifies if this certificate can be used for Web security.

EAP Specifies if this certificate can be used for EAP security.

530 Mediatrix Boss

Appendices

A P P E N D I X

A Country-Specific Parameters

The following parameters differ depending on the country in which you are.

Definitions

The following are some useful definitions.

Conventions

The following conventions apply to this Appendix.

Frequencies

Symbol “*” means modulated. For instance: 425 Hz * 25 means 425 Hz modulated at 25 Hz.

Symbol “+” means added. For instance: 425 Hz + 330 Hz means that both 425 Hz and 330 Hz

Table 291: Definitions

Term Description

Dial Tone Indicates the endpoint is ready to receive dialing.

Busy Tone Indicates the endpoint or equipment is in use, engaged or occupied.

Ringback Tone Indicates the called line is ringing out.

Special Information Tone Identifies network-provided announcements.

Stutter Dial Tone Notifies the user that they have a voice mail message when the phone does not or cannot have a message-waiting light.

Confirmation Tone Confirms a command performed by the user (such as activate a service).

Receiver Off Hook (ROH) Tone Indicates that the telephone is not hung up correctly.

Message Waiting Indicator Tone Indicates there is a message waiting somewhere for the owner of the phone

Network Congestion Tone Indicates that all switching paths are busy, all toll trunks are busy, or there are equipment blockages.

Intercept Tone Indicates that you have dialed incorrectly or that the feature you've requested is not available on your terminal.

Preemption Tone In military telephone systems, a distinctive tone that is used to indicate to connected users, i.e., subscribers, that their call has been preempted by a call of higher precedence.

Reorder Tone Indicates that all switching paths are busy, all toll trunks are busy, there are equipment blockages, the caller dialled an unassigned code, or the digits dialled got messed up along the way.

FED Tone Indicates the far end tone detection.

Mediatrix Boss 533

Appendix A - Country-Specific Parameters Definitions

sines are played at the same time.

When a tone is composed of more than one frequency, if not otherwise specified, the given electrical level applies to each frequency taken separately.

Impedance

Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.

When representing an impedance, the following applies:

Symbol “//” means parallel.

Symbol “+” means serial.

Furthermore, there are two types of impedances:

Input Impedance

Terminal Balance Return Loss (TBRL) Impedance

Input Impedance

Impedance of the Mediatrix Boss at the Tip and Ring wires.

Terminal Balance Return Loss (TBRL) Impedance

Balance return loss attributable to transmission loss between two points. It is used to characterize an impedance balancing property of the 2-wire analog equipment port.

Each country has its own definition of the TBRL value. For instance, in North America, TIA/EIA 464 (and TIA/EIA 912) define two TBRL values:

600 Ω for “on-premise” or short loop ports.

350 Ω + (1000 Ω || 21 nF) for “off-premise” or long loop ports.

A wire length above 2.5 km is considered long loop according to TIA/EIA 912 section 6.4 (7)(b)).

In Europe, ETSI 300 439 also mentions a TBRL value. However, most European countries have different requirements regarding the TBRL Impedance. This is also true for other countries around the world. Each one of them has different requirements.

Line Attenuation

Values are given in dBr (deciBel relative):

A “+” for input means that the digital side is attenuated by x decibels relative to the analog side.

A “+” for output means that the analog side is amplified by x decibels relative to the digital side.

A “-” for input means that the digital side is amplified by x decibels relative to the analog side.

A “-” for output means that the analog side is attenuated by x decibels relative to the digital side.

On-Off Sequences

Values in bold are “on” cycles, where tones are audible. Values in normal style are “off” cycles, where tones are not audible. When not otherwise specified, sequences repeat forever. A “x” symbol means that the sequences between parenthesis is repeated x times. The next cycle(s) repeat forever, unless otherwise specified. Values are in seconds.

For instance:

3*(0.1 – 0.1) then 0.6 – 1.0 - 0.2 – 0.2

means that the 0.1s on and 0.1s off sequence is repeated 3 times, afterwards the 0.6s on, 1.0s off, 0.2s on and 0.2s off sequence repeats forever.

534 Mediatrix Boss

Definitions Software Configuration Guide

Distinctive Ring

The distinctive ring service allows you to have three different numbers with each their own ring. The numbers ring through a single line coming into the business or residence and each number can be distinguished by the pattern of the ring. These ring patterns are made up of various combinations of ring bursts.

This feature uses the “Alert-Info” header from the initial INVITE of a call to know if the call requires a distinctive ringing.

The supported value of the “Alert-Info” are:

The Mediatrix Boss plays the default ring of the country selected if the Alert-Info value is not present or the value is not supported.

Table 292: Distinctive RIng Patterns

Alert-Info value Ring Name On – Off Sequence (s)

<http://127.0.0.1/Bellcore-dr2> Bellcore-dr2 0.8 – 0.4, 0.8 – 4.0

<http://127.0.0.1/Bellcore-dr3> Bellcore-dr3 0.4 – 0.2, 0.4 – 0.2, 0.8 – 4.0

<http://127.0.0.1/Bellcore-dr4> Bellcore-dr4 0.3 – 0.2, 1.0 – 0.2, 0.3 – 4.0

Note: Since the first pause of the distinctive ring is lower that 1 second, a splash ring followed by an Off of 1 second precedes the distinctive ring pattern.

Mediatrix Boss 535

Appendix A - Country-Specific Parameters Austria

Austria

The following parameters apply if you have selected Austria1 as location.

Table 293: Austria1 Parameters

Parameter Value On – Off Sequence (s) Elect. Levels

Busy Tone 450 Hz 0.3 – 0.3 -20 dBm

Confirmation Tone 450 Hz (0.1 – 0.1) x 3 End -20 dBm

Dial Tone 450 Hz CONTINUOUS -20 dBm

Message Waiting Indicator Tone 450 Hz (0.1 – 0.1) x 10, CONTINUOUS -20 dBm

Network Congestion Tone 450 Hz 0.3 – 0.3 -20 dBm

Receiver Off Hook (ROH) Tone 1400+2060+2450+2600 Hz 0.1 – 0.1 -19 dBm

Reorder Tone 950 Hz1400 Hz1800 Hz

0.330.330.33 – 1.0

-20 dBm-20 dBm-20 dBm

Ringback Tone 450 Hz 1.0 – 5.0 -20 dBm

Special Information Tone 950 Hz1400 Hz1800 Hz

0.330.330.33 – 1.0

-20 dBm

Stutter Dial Tone 450 Hz (0.1 – 0.1) x 3, CONTINUOUS -20 dBm

Call Waiting Tone 440 Hz 2.0 – 0.3, 10.0 – 0.3, 10.0 -17 dBm

Ring (FXS) AC: 45 VRMS, 50 HzDC: 15 Vdc

1.0 – 5.0

Loop Current 30 ma

Flash Hook Detection Range Min: 170 msMax: 900 ms

Input Impedance (FXS) 270 Ω + 750 Ω // 150 nF

Input Impedance (FXO) 270 Ω + 750 Ω // 150 nF

FED Tone 450 Hz 8.0

Default Caller ID (FXS) BELLCORE

FXS Line Attenuation (Input) -3 dBr

FXS Line Attenuation (Output) -10 dBr

FXO Line Attenuation (Input) +6 dBr

FXO Line Attenuation (Output) -1 dBr

Delay Before Answering 0 seconds

Delay Before Dialing (No Dial Tone Detection)

4 seconds

536 Mediatrix Boss

Czech Republic Software Configuration Guide

Czech Republic

The following parameters apply if you have selected Czech Republic1 as location.

Table 294: Czech Republic1 Parameters


Busy Tone 425 Hz 0.33 – 0.33 -12 dBm

Confirmation Tone 425 Hz (0.1 – 0.1) x 3, End -12 dBm

Dial Tone 425 Hz 0.33 – 0.33, 0.66 – 0.66 -12 dBm

Message Waiting Indicator Tone 425 Hz (0.1 – 0.1) x 10, 0.33 – 0.33, 0.66 – 0.66 -12 dBm


Receiver Off Hook (ROH) Tone 425 Hz 0.165 – 0.165 -12 dBm



0.330.330.33 – 1.0

-12 dBm

Stutter Dial Tone 425 Hz (0.165 – 0.165) x 3, 0.66 – 0.66 -12 dBm



1.0 – 4.0

Loop Current 30 ma


Input Impedance 220 Ω + 820 Ω // 115 nF

FED Tone 425 Hz 0.165 – 0.165

Default Caller ID (FXS) ETSI FSK

FXS Line Attenuation (Input) 0 dBr






4 seconds

Mediatrix Boss 537

Appendix A - Country-Specific Parameters France

France

The following parameters apply if you have selected France1 as location.

Table 295: France1 Parameters


Busy Tone 440 Hz 0.5 – 0.5 -19.9 dBm

Confirmation Tone 440 Hz (0.1 – 0.1) x 3, End -16.9 dBm

Dial Tone 440 Hz CONTINUOUS -16.9 dBm

Message Waiting Indicator Tone 440 Hz (0.1 – 0.1) x 10, CONTINUOUS -16.9 dBm

Network Congestion Tone 440 Hz 0.25 – 0.25 -19.9 dBm


Ringback Tone 440 Hz 1.5 – 3.5 -19.9 dBm


(3 x 0.3 – 2 x 0.03) – 1.0 -19.9 dBm

Stutter Dial Tone 440 Hz (0.1 – 0.1) x 3, CONTINUOUS -16.9 dBm



1.5 – 3.5

Loop Current 30 ma




Tbrl-Impedance 600 Ω

FED Tone 440 Hz 8.0

Default Caller ID (FXS) FRANCE: BELLCOREFRANCE_ETSI_FSK: ETSI_FSKFRANCE_ETSI_DTMF:ETSI_DTMF

FXS Line Attenuation (Input) +1.9 dBr

FXS Line Attenuation (Output) -8.9 dBr





4 seconds

538 Mediatrix Boss

Germany Software Configuration Guide

Germany

The following parameters apply if you have selected Germany as location.

Germany 1

The following parameters apply if you have selected Germany 1 as location.

Table 296: Germany 1 Parametersa

a. The Germany 2 choice in the MIB is exactly the same as Germany 1.


Busy Tone 425 Hz 0.48 – 0.48 -16 dBm








0.330.330.33 – 1.0

-16 dBm


Call Waiting Tone 440 Hz 0.3 – End -17 dBm


1.0 – 4.0

Loop Current 30 ma




Tbrl-Impedance 220 Ω + 820 Ω // 115 nF

FED Tone 425 Hz 8.0








4 seconds

Mediatrix Boss 539

Appendix A - Country-Specific Parameters Germany

Germany2

The following parameters apply if you have selected Germany 2 as location.

Table 297: Germany 2 Parameters


Busy Tone 425 Hz 0.48 – 0.48 -13 dBm








0.330.330.33 – 1.0

-13 dBm




1.0 – 4.0

Loop Current 30 ma




Tbrl-Impedance

FED Tone 425 Hz 8.0




FXO Line Attenuation (Input) -99 dBr




4 seconds

540 Mediatrix Boss

Italy Software Configuration Guide

Italy

The following parameters apply if you have selected Italy1 as location.

Table 298: Italy1 Parameters


Busy Tone 425 Hz 0.5 – 0.5 -13 dBm


Dial Tone 425 Hz 0.2 – 0.2, 0.6 – 1.0 -13 dBm

Message Waiting Indicator Tone 425 Hz (0.1 – 0.1) x 10, 0.2 – 0.2, 0.6 – 1.0 -13 dBm





0.330.330.33 – 1.0

-20 dBm

Stutter Dial Tone 425 Hz (0.1 – 0.1) x 3, 0.2 – 0.2, 0.6 – 1.0 -13 dBm



1.0 – 4.0

Loop Current 30 ma




Tbrl-impedance 750 Ω // 18 nF

FED Tone 425 Hz 0.2 – 0.2, 0.6 – 1.0








4 seconds

Mediatrix Boss 541

Appendix A - Country-Specific Parameters North America

North America

The following parameters apply if you have selected North America as location.

North America 1

The following parameters apply if you have selected North America 1 as location.

Table 299: North America 1 Parameters


Busy Tone 480+620 Hz 0.5 – 0.5 -21 dBm

Confirmation Tone 350+440 Hz (0.1 – 0.1) x 3, End -17 dBm

Dial Tone 350+440 Hz CONTINUOUS -17 dBm

Intercept Tone 440+620 Hz 0.5 – 0.5 -14 dBm

Message Waiting Indicator Tone 350+440 Hz (0.1 – 0.1) x 10, CONTINUOUS -17 dBm

Network Congestion Tone 480+620 Hz 0.25 – 0.25 -21 dBm


Reorder Tone 480+620 Hz 0.3 – 0.2 -21 dBm

Ringback Tone 440+480 Hz 2.0 – 4.0 -19 dBm


0.330.330.33 – 1.0

-14 dBm

Stutter Dial Tone 350+440 Hz (0.1 – 0.1) x 3, CONTINUOUS -17 dBm



2.0 – 4.0

Loop Current 30 ma


Input Impedance (FXS) 600 Ω

Input Impedance (FXO) 600 Ω

Tbrl-Impedancea (FXS)

a. TBRL-Impedance for “on-premise” or short loop ports.

600 Ω

FED Tone 440 Hz 8.0




FXO Line Attenuation (Input) 0 dBr

FXO Line Attenuation (Output) 0 dBr



0 seconds

542 Mediatrix Boss

Spain Software Configuration Guide

Spain

The following parameters apply if you have selected Spain1 as location.

Table 300: Spain1 Parameters


Busy Tone 425 Hz 0.2 – 0.2 -13 dBm




Network Congestion Tone 425 Hz 0.2 – 0.2, 0.2 – 0.2, 0.2 – 0.6 -13 dBm


Reorder Tone 425 Hz 0.2 – 0.2, 0.2 – 0.6 -13 dBm



0.330.330.33 – 1.0

-20 dBm




1.5 – 3.0

Loop Current 30 ma




FED Tone 425 Hz 8.0








4 seconds

Mediatrix Boss 543

Appendix A - Country-Specific Parameters Switzerland

Switzerland

The following parameters apply if you have selected Switzerland as location.

Table 301: Switzerland Parameters


Busy Tone 425 Hz 0.5 – 0.5 -13 dBm








0.3330.3330.333 – 1.0

-13 dBm




1.0 – 4.0

Loop Current 30 ma




FED Tone 425 Hz 8.0



FXS Line Attenuation (Output) -6.5 dBr





4 seconds

544 Mediatrix Boss

United Arab Emirates Software Configuration Guide

United Arab Emirates

The following parameters apply if you have selected the United Arab Emirates 2 as location.

Table 302: United Arab Emirates 2 Parameters


Dial Tone 350+440 Hz CONTINUOUS -13 dBm

Busy Tone 400 Hz 0.375 – 0.375 -13 dBm

Ringback Tone 425 Hz 0.4 – 0.2, 0.4 – 2.0 -13 dBm


0.33 – 0.33, 0.33 – 1.00.33 – 0.33, 0.33 – 1.00.33 – 0.33, 0.33 – 1.0

-13 dBm-13 dBm-13 dBm

Stutter Dial Tone 350+440 Hz (0.4 – 0.04-) x 5 CONTINUOUS -13 dBm

Confirmation Tone 400 Hz (0.1 – 0.1) x 3 End -13 dBm

Receiver Off Hook (ROH) Tone 1400+2060+2450+2600 Hz (0.1 – 0.1) -19 dBm

Message Waiting Indicator Tone 350+440 Hz (0.1 – 0.1) x 10 CONTINUOUS -13 dBm

Network Congestion Tone 400 Hz 0.4 – 0.35, 0.225 – 0.525 -13 dBm

Reorder Tone 400 Hz CONTINUOUS -13 dBm

Ring AC: 45 VRMS, 25 HzDC: 15 Vdc

0.4 – 0.2, 0.4 – 2.0

Call Waiting Tone 425 Hz ( 0.2 – 12.0, 0.2 –12.0)x2 End -13 dBm

Loop Current 30 ma


Input Impedance (FXS) 600 Ω

Input Impedance (FXO) 600 Ω

FED Tone 440 Hz 8.0

Default Caller ID BELLCORE



FXO Line Attenuation (Input) Not calibrated

FXO Line Attenuation (Output) Not calibrated



4 seconds

Mediatrix Boss 545

Appendix A - Country-Specific Parameters United Arab Emirates

546 Mediatrix Boss

A P P E N D I X

B Scripting Language

This appendix describes the Media5 proprietary scripting language. It also lists a few configuration samples that can be downloaded into the Mediatrix Boss via the Configuration Script feature (see “Chapter 38 - Configuration Script” on page 513 for more details).

This appendix covers the following topics:

General Scripting Language Syntax

Assigning scalar values

Assigning table cell values

Executing commands

Call Router Specific Information

Examples

General Scripting Language Syntax

The Media5 proprietary scripting language can be used to assign values to configuration variables and execute configuration commands. The scripting language may be used when creating configuration scripts (“Creating a Configuration Script” on page 526).

Using the scripting language requires a bit of knowledge about the Mediatrix Boss’s configuration variables tree structure.

The scripting language uses the following general syntax:

[keyword] [Context_Name [separator expression [operator constant]]] [#comment]

All specific syntaxes in this Appendix are derived from this general syntax.

Note that the brackets ([ and ]) are used to mark optional arguments. They are not part of the syntax.

Table 303: Scripting Language Syntax

Token Types Description

keyword A token that defines the type of operation to execute on expression or the type of data to retrieve from expression. Currently, only the set keyword is supported, which assigns value of constant to expression.

Context_Name Defines to which service the following expression belongs. For instance, the Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and the Host Configuration context is Hoc.

separator Delimiter defined as “.” (dot).

expression String that describes a configuration object. It can resolve to either:

• a scalar variable

• a cell

• a column

• a row

• a table

When a service, a scalar, a command or a table variable has the same name as a keyword, you should use the "get" and "set " keywords to access this variable.

operator Only the assignment operator (=) is defined.

Mediatrix Boss 547

Appendix B - Scripting Language General Scripting Language Syntax

Supported Characters

When using the scripting language, the following ASCII codes are supported:

All other ASCII codes are invalid.

Note that you must escape the XML reserved characters when inserting them in a configuration script:

< : <

> : >

% : %

& : &

constant A textual string or a number to assign to expression.

comment Anything following the comment marker (#) up to the end of the line is ignored.

Table 303: Scripting Language Syntax (Continued)

Token Types Description

10 LF, line feed13 CR, carriage return

32 space

33 !, exclamation mark34 ", double quote

35 #, hash

36 $, dollar37 %, percent

38 &, ampersand

39 ', quote40 (, open parenthesis

41 ), close parenthesis

42 *, asterisk43 +, plus

44 ,, comma

45 -, minus46 ., full stop

47 /, oblique stroke

48 0, zero49 1

50 2

51 352 4

53 5

54 655 7

56 8

57 958 :, colon

59 ;, semicolon

60 <, less than61 =, equals

62 >, greater than63 ?, question mark

64 @, commercial at

65 A66 B

67 C

68 D69 E

70 F

71 G72 H

73 I

74 J75 K

76 L

77 M78 N

79 O

80 P81 Q

82 R

83 S84 T

85 U

86 V87 W

88 X

89 Y90 Z

91 [, open square bracket

92 \, backslash93 ], close square bracket

94 ^, caret95 _, underscore

96 `, back quote

97 a98 b

99 c

100 d101 e

102 f

103 g104 h

105 i

106 j107 k

108 l

109 m110 n

111 o

112 p113 q

114 r

115 s116 t

117 u

118 v119 w

120 x

121 y122 z

123 {, open curly bracket

124 |, vertical bar125 }, close curly bracket

126 ~, tilde

548 Mediatrix Boss

Assigning Scalar Values Software Configuration Guide

Assigning Scalar Values

The following is a sample script command assigning a value to a scalar configuration variable:

Service_Name.Scalar_Name=value

A valid script line would be:

Cli.InactivityTimeOut=25

Assigning Table Cell Values

When you want to get the value of a specific table cell or set the value of a specific table cell, you must follow a particular syntax:

[get]Service_Name.Table_Name[Index=key].Column_Name[set]Service_Name.Table_Name[Index=key].Column_Name=<value>

Let’s take for instance the NetworkInterfacesStatus table:

If you want to get the IP address value of Interface 3, you would have to enter the following command:

Table 304: Scalar Syntax

Command Description

Service_Name Defines which service should process the expression. For instance, the Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and the Host Configuration context is Hoc.

Scalar_Name Name of the specific variable to which assign a value.

value A textual string or a number to assign to the argument.

Table 305: Table Cell Syntax

Command Description

Service_Name Defines which service should process the expression. For instance, the Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and the Host Configuration context is Hoc.

Table_Name Name of the table that contains the cell.

index=key List of index values identifying the row on which the cell is located. The list of indexes is in the name=value form, separated by spaces and enclosed within brackets. The index is always the first column of a table.

Column_Name Name of the column that contains the cell.

value A textual string or a number to assign to the argument.

InterfaceName InterfaceStatus LinkName IpAddr

Interface1 100a

a. This enum means “disabled”

eth1-4 10.1.1.1

Interface2 400b

b. This enum means “ok”

eth1-4 10.1.1.2

Interface3 400 eth5 10.1.1.3

Mediatrix Boss 549

Appendix B - Scripting Language Executing Commands

get Bni.NetworkInterfacesStatus[InterfaceName=Interface3].IpAddr

Executing Commands

Configuration commands are used to make the Mediatrix Boss perform actions such as restarting the unit, restarting a service, refreshing its SIP registration, etc.

There are two types of commands you can execute:

Normal Commands

Row Commands

Normal Commands

The normal command feature has the following syntax:

Service_Name.Command_Name arg1=value1 –b arg2=[value2 value3 value4]

For instance, a valid command would be:

Conf.BackupImage FileName=backup_test_1 Location=/testfiles/Conf/v1/ManualTests/

TransferProtocol=400 TransferUsername=testuser TransferPassword=test TransferSrvHostname=”test1.mediatrix.com”

Another valid command (without arguments) would be:

SipEp.RegistrationRefresh

Double Quotes

You must use double quotes when the text parameter contains a dot. For instance, entering the following command results in a bad command:

Conf.BackupImage FileName=test.cfg Location=config TransferProtocol=400 TransferUsername=Usr1 TransferPassword=Pwd1 TransferSrvHostname=192.168.6.3

You must enclose each text parameter that contains a dot with double quotes. In the above example, you must enclose FileName=test.cfg and TransferSrvHostname=192.168.6.3 in double quotes:

Table 306: Normal Command Syntax

Command Description

Service_Name Defines which service should process the command. For instance, the Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and the Host Configuration context is Hoc.

Command_Name The command to execute.

argn Name of the argument for which you want to assign a value. Three types of arguments are allowed:

• flags (beginning with ‘-‘, without anything else, for instance, “-b” in the command syntax above). Flags are optional.

• scalar arguments (with mandatory ‘=’ and following value). They are mandatory unless they have a default value, in which case they are optional.

• vector arguments (with mandatory ‘=’ and following a list of values enclosed within brackets and separated by spaces). They are mandatory unless they have a default value, in which case they are optional.

The number and types of arguments depend on the specific command you are using.

valuen A textual string or a number to assign to the argument.

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Call Router Specific Information Software Configuration Guide

Conf.BackupImage FileName="test.cfg" Location=config TransferProtocol=400

TransferUsername=Usr1 TransferPassword=Pwd1 TransferSrvHostname="192.168.6.3"

Row Commands

Row commands appear as table cells and allow you to perform an action on a specific row of the relevant table.

Row commands are available in several services of the Mediatrix Boss. For instance, the Call Router service uses the Up, Down, Insert, and Delete commands in its various tables.

The row command feature has the following syntax:

Context_Name.Table_Name[index1=value1 index2=value2].Row_Command=execute_value

For instance, the following executes the service Dhcp’s StaticLeases Delete row command on one of the table’s rows. The StaticLeases table only has one index column: the MacAddress column. The Delete row command is an enum that has two possible values: noOp (0) and delete (10). The command is executed by assigning the execute value (10) to the Delete cell.

Dhcp.StaticLeases[MacAddress="0090F8001234"].Delete=10

DeleteAllRows Command

The DeleteAllRows command is a table command that you can use to delete all rows of a specific table to start anew. You can use it as follows:

Service_Name.Table_Name.DeleteAllRows

A valid command would be:

CRout.MappingExpression.DeleteAllRows

Call Router Specific Information

When working with call router parameters, you must be aware of the following:

You must prefix the name of a route with “route-”, for instance: route-isdn_sip.

You must prefix the name of a SIP interface with “sip-”, for instance: sip-default.

You must prefix the name of an ISDN interface with “isdn-”, for instance: isdn-default.

You must prefix the name of a hunt with “hunt-”, for instance: hunt-hunt1.

Table 307: Row Command Syntax

Command Description

Context_Name Defines which service should process the command. For instance, the Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and the Host Configuration context is Hoc.

Table_Name Table where the row command is located.

indexn=valuen List of index values identifying the row on which to execute the command. The list of indexes is in the name=value form, separated by spaces and enclosed within brackets. The index is always the first column of a table. See “Assigning Table Cell Values” on page 549 for more details.

Row_Command The row command to execute.

execute_value Numerical value of the enum.

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Appendix B - Scripting Language Examples

Examples

This section gives a few configuration samples that can be used both in the CLI or as part of a configuration script.

Basic Call

The following sections describe the minimal configuration you must apply to perform a basic call between an ISDN telephone and an analog telephone.

Performing a Firmware Upgrade

The first step to do upon receiving a new Mediatrix Boss is to upgrade its firmware to the latest version available.

Fpu.MfpVersion="1.1.8.73"

Fpu.MfpLocation=""

Fpu.MfpTransferProtocol=300Fpu.MfpTransferUsername=""

Fpu.MfpTransferPassword=""

Fpu.MfpTransferSrvHostname="192.168.3.4"Fpu.AutomaticRestartEnable=1

Fpu.Install

Isdn.BasicRateInterface[Name=Slot2/E1T1].EndpointType=200

Isdn.BasicRateInterface[Name=Slot2/E1T1].ConnectionType=200

Configuration of the SIP Endpoint

Configuring the SIP endpoint allows you to register your ISDN telephone to a SIP server.

SipEp.DefaultStaticRegistrarServerHost=”192.168.3.3”SipEp.DefaultStaticProxyHomeDomainHost=”192.168.3.3”

sipEp.userAgent[EpId=Slot2/E1T1].Username=4401sipEp.userAgent[EpId=Slot2/E1T1].FriendlyName=4401

sipEp.userAgent[EpId=Slot2/E1T1].GatewayName="default"

sipEp.userAgent[EpId=Slot2/E1T1].Publish=0sipEp.userAgent[EpId=Slot2/E1T1].Register=1

SipEp.registrationrefresh

Configuration of the Call Router: Routes

You must create routes that will route calls from ISDN to SIP and from SIP to ISDN.

Crout.InsertRouteCrout.Route[Index=1].SourceCriteria="Slot2/E1T1"

Crout.Route[Index=1].PropertiesCriteria=200

Crout.Route[Index=1].ExpressionCriteria=""

Crout.Route[Index=1].Destination="sip-default"

Crout.InsertRoute

Crout.Route[Index=2].SourceCriteria="sip-default"Crout.Route[Index=2].PropertiesCriteria=200

Crout.Route[Index=2].ExpressionCriteria=""

Crout.Route[Index=2].Destination="Slot2/E1T1"

Crout.ApplyConfig

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Examples Software Configuration Guide

Configuration of the Call Router: Mapping

You must create mappings that will allow you to properly communicate between an ISDN telephone and an analog telephone. In this example, it is assumed that the ISDN telephone number is 800 and the analog telephone number is 101.

Crout.InsertMappingType

Crout.mappingtype[Index=1].Name="ISDN_To_SIP"

Crout.mappingtype[Index=1].Criteria=300Crout.mappingtype[Index=1].Transformation=300

Crout.InsertMappingExpressionCrout.mappingexpression[Index=1].Name="ISDN_To_SIP"

Crout.mappingexpression[Index=1].Criteria=4400

Crout.mappingexpression[Index=1].Transformation=101Crout.mappingexpression[Index=1].SubMappings=""

Crout.InsertMappingTypeCrout.mappingtype[Index=2].Name="SIP_to_ISDN"

Crout.mappingtype[Index=2].Criteria=300

Crout.mappingtype[Index=2].Transformation=300

Crout.InsertMappingExpression

Crout.mappingexpression[Index=2].Name="SIP_to_ISDN"Crout.mappingexpression[Index=2].Criteria=4400

Crout.mappingexpression[Index=2].Transformation=800

Crout.mappingexpression[Index=2].SubMappings=""

Crout.Route[Index=1].Mappings="ISDN_To_SIP“

Crout.Route[Index=2].Mappings="SIP_to_ISDN“

Crout.ApplyConfig

Debugging

The following sections allow you to enable two useful debugging tools of the Mediatrix Boss: syslog messages and PCM traces.

Enabling Syslog

This example assumes that you run a syslog server at address 192.168.3.4.

Nlm.SyslogRemoteHost=”192.168.3.4”Cli.MinSeverity=300

Bni.MinSeverity=400

Hoc.MinSeverity=500

Configuring PCM Capture

The PCM traces are two different RTP streams made specifically to record all analog signals that are either sent or received on the analog side of the Mediatrix Boss. Only the configured port, port #1 and/or #2 are sending the PCM traces for a maximum of four simultaneous RTP streams.

The RTP streams are sent to a configurable IP address, normally an IP address on your network where it can be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP streams of the VoIP call.

All streams are sent instantly at startup with an average ptime of 15 ms. This means that until the PCM traces are disabled, even an idle unit will continuously send up to 66.6 packets/s X 4 streams = 267 packets/s using approximately 174 bytes each, for a total of 46 Kbytes of upstream bandwidth.

Mipt.PcmCaptureEnable=1Mipt.PcmCaptureEndpoint=”Bri1-1”

Mipt.PcmCaptureIpAddr=”192.168.3.3”

Mipt.restart

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Advanced Configuration

The following sections describe more advanced configuration such as creating and enabling network interfaces and configuring the ISDN interface of the Mediatrix Boss.

Network Interfaces

Bni.pppServiceName="test"

Bni.pppRetryInterval=20

Bni.pppAuthenticationProtocol=100Bni.pppIdentity="user"

Bni.pppSecret="pass"

Bni.networkInterfaces[InterfaceName=Lan1].LinkName=eth1-4Bni.networkInterfaces[InterfaceName=Lan1].ConnectionType=100

Bni.networkInterfaces[InterfaceName=Lan1].StaticIpAddr=”192.168.0.100/24”

Bni.networkInterfaces[InterfaceName=Lan1].Activation=1Bni.networkInterfaces[InterfaceName=Lan1].StatusInterfaceName=Loop

Bni.networkInterfaces[InterfaceName=Rescue].LinkName=eth1-4

Bni.networkInterfaces[InterfaceName=Rescue].ConnectionType=100Bni.networkInterfaces[InterfaceName=Rescue].StaticIpAddr=”192.168.0.110/24”

Bni.networkInterfaces[InterfaceName=Rescue].Activation=0

Bni.networkInterfaces[InterfaceName=Rescue].StatusInterfaceName=LoopBni.networkInterfaces[InterfaceName=Uplink].LinkName=eth5

Bni.networkInterfaces[InterfaceName=Uplink].ConnectionType=100

Bni.networkInterfaces[InterfaceName=Uplink].StaticIpAddr=”192.168.0.120/24”Bni.networkInterfaces[InterfaceName=Uplink].Activation=1

Bni.networkInterfaces[InterfaceName=Uplink].StatusInterfaceName=Loop

Bni.MinSeverity=100

ISDN Service – Mediatrix 3600

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].EndpointType=200

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].LineType=100

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].LineCoding=300Isdn.PrimaryRateInterface[Name=Slot2/E1T1].LineFraming=300

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].NetworkLocation=500

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].PreferredEncodingScheme=200Isdn.PrimaryRateInterface[Name=Slot2/E1T1].FallbackEncodingScheme=100

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ChannelRange=1-15

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ChannelAllocationStrategy=400Isdn.PrimaryRateInterface[Name=Slot2/E1T1].MaxActiveCalls=30

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SignalInformationElementEnable=1

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].InbandToneGenerationEnable=0Isdn.PrimaryRateInterface[Name=Slot2/E1T1].InbandDtmfDialingEnable=0

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].OverlapDialingEnable=0

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].CallingNameMaxLength=82Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ExclusiveBChannelSelectionEnable=1

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SendingCompleteEnable=0

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ClipEnable=1

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ClirEnable=1Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ClirOverrideEnable=1

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SendRestartOnStartupEnable=0

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SupplementaryServicesEnable=1Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInSetupEnable=0

Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInSetupAckEnable=0

Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInCallProgressForSetupEnable=0

Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInAlertingEnable=0

Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInConnectEnable=0Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].MaximumFacilityWaitingDelay=1500

Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].Protocol=400

Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].ColpEnable=1Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].ColrEnable=1

Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].ColrOverrideEnable=1

Isdn.PhysicalLink[InterfaceName=Slot2/E1T1].L1TimerT3=30000

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Examples Software Configuration Guide

Isdn.PhysicalLink[InterfaceName=Slot2/E1T1].ClockMode=300

Isdn.MinSeverity=500Isdn.Restart

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A P P E N D I X

C Glossary

10 BaseT

An Ethernet local area network that works on twisted pair wiring.

100 BaseT

A newer version of Ethernet that operates at 10 times the speed of a 10 BaseT Ethernet.

Access Device

Device capable of sending or receiving data over a data communications channel.

Accounting

Accounting measures the resources a user consumes during access. This can include the amount of system time or the amount of data a user has sent and/or received during a session. Accounting is carried out by logging of session statistics and usage information and is used for authorization control, billing, trend analysis, resource utilization, and capacity planning activities.

A-Law

The ITU-T companding standard used in the conversion between analog and digital signals in PCM (Pulse Code Modulation) systems. A-law is used primarily in European telephone networks and contrasts with the North American mu (µ)-law standard. See also mu (µ)-law.

ANI

In CAS signalling, the sending of the calling numbers is known as Automatic Number Identification.

Area Code

The preliminary digits that a user must dial to be connected to a particular outgoing trunk group or line. In North America, an area code has three digits and is used with a NXX (office code) number. For instance, in the North American telephone number 561-955-1212, the numbers are defined as follows:

Outside North America, the area code may have any number of digits, depending on the national telecommunication regulation of the country. In France, for instance, the numbering terminology is xZABPQ 12 34, where:

Table 308: North American Numbering Plan

No. Description

561 Area Code, corresponding to a geographical zone in a non-LNP (Local Number Portability) network.

955 NXX (office code), which corresponds to a specific area such as a city region.

1212 Unique number to reach a specific destination.

Table 309: France Numbering Plan

No. Description

x Operator forwarding the call. This prefix can be made of 4 digits.

Z Geographical (regional) zone of the number (in France, there are five zones). It has two digits.

ABPQ First four digits corresponding to a local zone defined by central offices.

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Appendix C - Glossary Authentication

In this context, the area code corresponds to the Z portion of the numbering plan. Because virtually every country has a different dialing plan nomenclature, it is recommended to identify the equivalent of an area code for the location of your communication unit.

Authentication

Authentication provides a way of identifying a user, typically by having the user enter a valid user name and valid password before access is granted. The process of authentication is based on each user having a unique set of criteria for gaining access. The AAA server compares a user's authentication credentials with other user credentials stored in a database. If the credentials match, the user is granted access to the network. If the credentials are at variance, authentication fails and network access is denied.

Basic Rate Interface (BRI)

An Integrated Services Digital Network configuration defined in the physical layer standard I.430 produced by the ITU. This configuration consists of two 64 kbit/s “bearer” channels (B channels) and one 16 kbit/s “data” channel (D channel). The B channels are used for voice or user data, and the D channel is used for any combination of: data, control/signalling and X.25 packet networking. The two B channels can be bonded together giving a total data rate of 128 kbit/s. BRI is the kind of ISDN interface most likely to be found in residential service.

Call Routing

Calls through the Mediatrix Boss can be routed based on a set of routing criteria.

Channel Associated Signalling (CAS)

With this method of signalling, each traffic channel has a dedicated signalling channel. In other words the signalling for a particular traffic circuit is permanently associated with that circuit. Channel-associated call-control is still widely used today mostly in South America, Africa, Australia and in Europe.

Country Code (CC)

In international direct telephone dialing, a code that consists of 1-, 2-, or 3-digit numbers in which the first digit designates the region and succeeding digits, if any, designate the country.

Dialed Number Identification Service (DNIS)

DNIS is a telephone service that identifies for the receiver of a call the number that the caller dialed. It's a common feature of 800 and 900 lines. If you have multiple 800 or 900 numbers to the same destination, DNIS tells which number was called. DNIS works by passing the touch tone digits (dual tone multi frequency or MF digits) to the destination where a special facility can read and display them or make them available for call center programming.

Digital Subscriber Lines (DSL)

A technology for bringing high-bandwidth information to homes and small businesses over ordinary copper telephone lines. xDSL refers to different variations of DSL, such as ADSL, HDSL, and RADSL.

Domain Name Server (DNS)

Internet service that translates domain names into IP addresses. For instance, the domain name www.example.com might translate to 198.105.232.4.

12 34 Unique number to reach a specific destination.

Table 309: France Numbering Plan (Continued)

No. Description

558 Mediatrix Boss

Dual-Tone Multi-Frequency (DTMF) Software Configuration Guide

Dual-Tone Multi-Frequency (DTMF)

In telephone systems, multi-frequency signalling in which a standard set combinations of two specific voice band frequencies, one from a group of four low frequencies and the other from a group of four higher frequencies, are used. Although some military telephones have 16 keys, telephones using DTMF usually have 12 keys. Each key corresponds to a different pair of frequencies. Each pair of frequencies corresponds to one of the ten decimal digits, or to the symbol “#” or “*”, the “*” being reserved for special purposes.

Dynamic Host Configuration Protocol (DHCP)

TCP/IP protocol that enables PCs and workstations to get temporary or permanent IP addresses (out of a pool) from centrally-administered servers.

Echo Cancellation

Technique that allows for the isolation and filtering of unwanted signals caused by echoes from the main transmitted signal.

Far End Disconnect

Refers to methods for detecting that a remote party has hung up. This is also known as Hangup Supervision. There are several methods that may be used by a PBX/ACD/CO to signal that the remote party has hung up, including cleardown tone, or a wink.

Federal Communications Commission (FCC)

U.S. government regulatory body for radio, television, interstate telecommunications services, and international services originating in the United States.

Firewall

A firewall in a networked environment prevents some communications forbidden by the security policy. It has the basic task of controlling traffic between different zones of trust. Typical zones of trust include the Internet (a zone with no trust) and an internal network (a zone with high trust).

Foreign Exchange Office (FXO)

A network-provided service in which a telephone in a given local exchange area is connected, via a private line, to a central office in another, i.e., “foreign”, exchange, rather than the local exchange area’s central office. This is the office end of an FX circuit (frequently a PBX).

Foreign Exchange Service/Station (FXS)

A network-provided service in which a telephone in a given local exchange area is connected, via a private line, to a central office in another, i.e., “foreign”, exchange, rather than the local exchange area’s central office. This is the station (telephone) end of an FX circuit. An FXS port will provide dial tone and ring voltage.

Full-Duplex Connection

Refers to a transmission using two separate channels for transmission and reception and that can transmit in both ways at the same time. See also Half-Duplex Connection.

G.703

ITU-T recommendation for the physical and electrical characteristics of hierarchical digital interfaces at rates up to 140Mbit/s.

G.704

ITU-T recommendation for synchronous frame structures on G.703 interfaces up to 45Mbit/s. The conventional use of G.704 on a 2Mbit/s primary rate circuit provides 30 discrete 64kbit/s channels, with a further 64kbit/s channel available for common channel signalling.

G.711

Algorithm designed to transmit and receive A-law PCM (Pulse Code Modulation) voice at digital bit rates of 48 kbps, 56 kbps, and 64 kbps. It is used for digital telephone sets on digital PBX and ISDN channels.

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Appendix C - Glossary G.723.1

G.723.1

A codec that provides the greatest compression, 5.3 kbps or 6.3 kbps; typically specified for multimedia applications such as H.323 videoconferencing.

G.726

An implementation of ITU-T G.726 standard for conversion linear or A-law or µ-law PCM to and from a 40, 32, 24 or 16 kbit/s channel.

G.729

A codec that provides near toll quality at a low delay which uses compression to 8 kbps (8:1 compression rate).

Gateway

A device linking two different types of networks that use different protocols (for example, between the packet network and the Public Switched Telephone Network).

Half-Duplex Connection

Refers to a transmission using the same channel for both transmission and reception therefore it can't transmit and receive at the same time. See also Full-Duplex Connection.

Hunt Group

The hunt group hunts an incoming call to multiple interfaces. It accepts a call routed to it by a routing table or directly from an interface and creates another call that is offered to one of the configured destination interfaces. If this destination cannot be reached, the hunt group tries another destination until one of the configured destinations accepts the call. When an interface accepts a call, the interface hunting is complete and the hunt group service merges the original call with the new call to the interface that accepted the call.

Impedance

Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.

Information Transfer Capability (ITC)

A request to the network exchange equipment to ask if a particular type of encoding is allowed. It is also called ISDN bearer capability or ISDN service.

Integrated Services Digital Network (ISDN)

A set of digital transmission protocols defined by the international standards body for telecommunications, the ITU-T (formerly called the CCITT). These protocols are accepted as standards by virtually every telecommunications carrier all over the world.

ISDN complements the traditional telephone system so that a single pair of telephone wires is capable of carrying voice and data simultaneously. It is a fully digital network where all devices and applications present themselves in a digital form.

International Telecommunication Union (ITU)

Organization based in Geneva, Switzerland, that is the most important telecom standards-setting body in the world.

Internet-Drafts

Internet-Drafts are working documents of the IETF, its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts.

Internet Protocol (IP)

A standard describing software that keeps track of the Internet’s addresses for different nodes, routes outgoing messages, and recognizes incoming messages.

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IP Forwarding Software Configuration Guide

IP Forwarding

Allows the packet to be forwarded to a specific network based on the packet’s criteria (source IP address and source Ethernet link).

Jitter

A distortion caused by the variation of a signal from its references which can cause data transmission errors, particularly at high speeds.

Light Emitting Diode (LED)

A semiconductor diode that emits light when a current is passed through it.

Local Area Network (LAN)

Data-only communications network confined to a limited geographic area, with moderate to high data rates. See also WAN.

Local Firewall

Allows you to dynamically create and configure rules to filter incoming packets with the unit as destination. The traffic is analyzed and filtered by all the rules configured.

Management Information Base (MIB)

Specifications containing definitions of management information so that networked systems can be remotely monitored, configured and controlled.

Media Access Control (MAC) Address

A layer 2 address, 6 bytes long, associated with a particular network device; used to identify devices in a network; also called hardware or physical address.

Mu (µ)-Law

The PCM (Pulse Code Modulation) voice coding and companding standard used in Japan and North America. See also A-Law.

Music on Hold (MoH)

Refers to the practice of playing pre-recorded music to fill the silence that would be heard by telephone callers who have been placed on hold. It is especially common in situations involving customer service.

Network

A group of computers, terminals, and other devices and the hardware and software that enable them to exchange data and share resources over short or long distances. A network can consist of any combination of local area networks (LAN) or wide area networks (WAN).

Network Address Translation (NAT)

NAT, also known as network masquerading or IP masquerading, rewrites the source and/or destination addresses/ports of IP packets as they pass through a router or firewall. It is most commonly used to connect multiple computers to the Internet (or any other IP network) by using one IP address. This allows home users and small businesses to cheaply and efficiently connect their network to the Internet. The basic purpose of NAT is to multiplex traffic from the internal network and present it to the Internet as if it was coming from a single computer having only one IP address.

There are two types of NAT rules:

Source rules: They are applied on the source address of outgoing packets.

Destination rules: They are applied on the destination address of incoming packets.

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Appendix C - Glossary Network Firewall

Network Firewall

Allows dynamically creating and configuring rules to filter packets forwarded by the unit. Since this is a network firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and filtered by all the rules configured.

Off-hook

A line condition caused when a telephone handset is removed from its cradle.

On-hook

A line condition caused when a telephone handset is resting in its cradle.

Packet

Includes three principal elements: control information (such as destination, origin, length of packet), data to be transmitted, and error detection. The structure of a packet depends on the protocol.

Plain Old Telephone System (POTS)

Standard telephone service used by most residential locations; basic service supplying standard single line telephones, telephone lines, and access to the public switched network.

Point to Point Protocol over Ethernet (PPPoE)

A proposal specifying how a host personal computer interacts with a broadband modem (i.e., DSL, cable, wireless, etc.) to access the growing number of Highspeed data networks. Relying on two widely accepted standards, Ethernet and the point-to-point protocol (PPP), the PPPoE implementation requires virtually no more knowledge on the part of the end user other than that required for standard Dialup Internet access. In addition, PPPoE requires no major changes in the operational model for Internet Service Providers (ISPs) and carriers. The base protocol is defined in RFC 2516.

Port

Network access point, the identifier used to distinguish among multiple simultaneous connections to a host.

Portable Operating System Interface (POSIX)

POSIX is a set of standard operating system interfaces based on the UNIX operating system. The need for standardization arose because enterprises using computers wanted to be able to develop programs that could be moved among different manufacturer's computer systems without having to be recoded.

Primary Rate Interface (PRI)

A telecommunications standard for carrying multiple DS0 voice and data transmissions between two physical locations. All data and voice channels are (ISDN) and operate at 64 kbit/s.

North America and Japan use a T1 of 23 B channels and one D channel which corresponds to a T1 line. Europe, Australia and most of the rest of the world use the slightly higher capacity E1, which is composed of 31 B channels and one D channel.

Fewer active B channels (also called user channels) can be used for a fractional T1. More channels can be used with more T1's, or with a fractional or full T3 or E3.

Presentation Indicator (PI)

An information element (IE) field that determines whether a caller’s CLI can be displayed on a Caller ID device or otherwise presented to the called party.

Private Branch Exchange (PBX)

A small to medium sized telephone system and switch that provides communications between onsite telephones and exterior communications networks.

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Protocol Software Configuration Guide

Protocol

A formal set of rules developed by international standards bodies, LAN equipment vendors, or groups governing the format, control, and timing of network communications. A set of conventions dealing with transmissions between two systems. Typically defines how to implement a group of services in one or two layers of the OSI reference model. Protocols can describe low-level details of machine-to-machine interfaces or high-level exchanges between allocation programs.

Proxy Server

An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.

Public Switched Telephone Network (PSTN)

The local telephone company network that carries voice data over analog telephone lines.

Quality of Service (QoS)

Measure of the telephone service quality provided to a subscriber. This could be, for example, the longest time someone should wait after picking up the handset before they receive dial tone (three seconds in most U.S. states).

Real Time Control Protocol (RTCP)

RTCP is the control protocol designed to work in conjunction with RTP. It is standardized in RFC 1889 and 1890. In an RTP session, participants periodically send RTCP packets to convey feedback on quality of data delivery and information of membership.

Realtime Transport Protocol (RTP)

An IETF standard for streaming realtime multimedia over IP in packets. Supports transport of real-time data like interactive voice and video over packet switched networks.

Registrar Server

A server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and MAY offer location services.

Request for Comment (RFC)

A formal document from the IIETF that is the result of committee drafting and subsequent review by interested parties. Some RFCs are informational in nature. Of those that are intended to become Internet standards, the final version of the RFC becomes the standard and no further comments or changes are permitted. Change can occur, however, through subsequent RFCs that supersede or elaborate on all or parts of previous RFCs.

Screening Indicator (SI)

A service provided by ISDN that can be used to test the trustworthiness of the calling party’s number. This signalling-related information element is found in octet 3a of the ISDN SETUP message.

Server

A computer or device on a network that works in conjunction with a client to perform some operation.

Session Description Protocol (SDP)

Describes multimedia sessions for the purpose of session announcement, session invitation and other forms of multimedia session initiation. SDP communicates the existence of a session and conveys sufficient information to enable participation in the session. SDP is described in RFC 2327.

Session Initiation Protocol (SIP)

A protocol for transporting call setup, routing, authentication, and other feature messages to endpoints within the IP domain, whether those messages originate from outside the IP cloud over SCN resources or within the cloud.

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Appendix C - Glossary Simple Network Management Protocol (SNMP)

Simple Network Management Protocol (SNMP)

A standard of network management that uses a common software agent to manage local and wide area network equipment from different vendors; part of the Transmission Control Protocol / Internet Protocol (TCP/IP) suite and defined in RFC 1157.

Simple Network Time Protocol (SNTP)

SNTP, which is an adaptation of the Network Time Protocol (NTP), is widely used to synchronize computer clocks in the global Internet. It provides comprehensive mechanisms to access national time and frequency dissemination services, organize the time-synchronization subnet and adjust the local clock in each participating subnet peer. In most places of the Internet of today, NTP provides accuracies of 1-50 ms, depending on the characteristics of the synchronization source and network paths.

Subnet

An efficient means of splitting packets into two fields to separate packets for local destinations from packets for remote destinations in TCP/IP networks.

Switched Circuit Network (SCN)

A communication network, such as the public switched telephone network (PSTN), in which any user may be connected to any other user through the use of message, circuit, or packet switching and control devices.

T.38

An ITU-T Recommendation for Real-time fax over IP. T.38 addresses IP fax transmissions for IP-enabled fax devices and fax gateways, defining the translation of T.30 fax signals and Internet Fax Protocols (IFP) packets.

Telephony

The science of translating sound into electrical signals, transmitting them, and then converting them back into sound.

Transmission Control Protocol/Internet Protocol (TCP/IP)

The basic communication language or protocol of the Internet. It can also be used as a communications protocol in a private network (either an intranet or an extranet).

Trivial File Transfer Protocol (TFTP)

A simplified version of FTP that transfers files but does not provide password protection, directory capability, or allow transmission of multiple files with one command.

User Datagram Protocol (UDP)

An efficient but unreliable, connectionless protocol that is layered over IP, as is TCP. Application programs are needed to supplement the protocol to provide error processing and retransmission of data. UDP is an OSI layer 4 protocol.

Virtual LAN (VLAN)

A network of computers that behave as if they are connected to the same wire even though they may actually be physically located on different segments of a LAN. One of the biggest advantages of VLANs is that when a computer is physically moved to another location, it can stay on the same VLAN without any hardware reconfiguration.

Virtual Private Network (VPN)

A private communications network usually used within a company, or by several different companies or organizations, to communicate over a public network. VPN message traffic is carried on public networking infrastructure (e.g. the Internet) using standard (often insecure) protocols, or over a service provider's network providing VPN service guarded by well defined Service Level Agreement (SLA) between the VPN customer and the VPN service provider.

564 Mediatrix Boss

Voice Over IP (VoIP) Software Configuration Guide

Voice Over IP (VoIP)

The technology used to transmit voice conversations over a data network using the Internet Protocol. Such data network may be the Internet or a corporate Intranet.

Wide Area Network (WAN)

A large (geographically dispersed) network, usually constructed with serial lines, that covers a large geographic area. A WAN connects LANs using transmission lines provided by a common carrier.

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Appendix C - Glossary Wide Area Network (WAN)

566 Mediatrix Boss

A P P E N D I X

D List of Acronyms

AES Advanced Encryption StandardANI Automatic Number IdentificationAOC-E Advice of Charge End-of-Call

BRI Basic Rate Interface

CA Certification AuthorityCAS Channel Associated SignallingCCBS Completion of Calls to Busy SubscriberCCNR Completion of Calls on No ReplyCLIP Calling Line Information PresentationCLIR Calling Line Information RestrictionCNG Comfort Noise GeneratorCNIP Calling Name Identity PresentationCOLP Connected Line Identification PresentationCOLR Connected Line Identification RestrictionCS-ACELP Conjugate Structure-Algebraic Code Excited Linear Prediction

DER Distinguished Encoding RulesDNIS Dialed Number Identification ServiceDSCP Differentiated Services Code PointDSS1 Digital Subscriber Signaling System No.1DTMF Dual Tone Multi-Frequency

FQDN Fully Qualified Domain NameFSK Frequency Shift Keying

HTML Hyper Text Markup LanguageHTTP HyperText Transfer ProtocolHTTPS HTTP over the Transport Layer Security

ISDN Integrated Services Digital NetworkITC Information Transfer CapabilityITU International Telecommunication Union

kbps KiloBits Per Second

MFC Multi-Frequency CodeMIKEY Multimedia Internet KEYingMSN Multiple Subscriber NumberMWI Message Waiting Indicator

NAPTR Naming Authority Pointer requestNAT Network Address TranslationNPI Numbering Plan IndicatorNT Network Termination

PEM Privacy Enhanced MailPI Presentation IndicatorPRACk Provisional Response AcknowledgementPRI Primary Rate Interface

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Appendix D - List of Acronyms

QoS Quality of Service

RFC Request For CommentRTCP Real Time Control Protocol

SDES Secure DescriptionSDP Session Description ProtocolSHA Secure Hash AlgorithmSI Screening IndicatorSIP Session Initiation ProtocolSRTCP Secure Real-Time Transport Control ProtocolSRTP Secure Real-Time Transport ProtocolSSL Secure Sockets Layer

TBRL Terminal Balance Return LossTE Terminal EquipmentTEI Terminal Endpoint IdentifierTFTP Trivial File Transfer ProtocolTLS Transport Layer SecurityTON Type of Number

UDP User Datagram ProtocolUNI User-Network InterfaceURI Uniform Resource Identifier

VAD Voice Activity Detector

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A P P E N D I X

E List of Standards Supported

Ddraft draft-poetzl-bliss-call-completion-00 ......................................................................................................................... 422draft-choudhuri-sip-info-digit-00 ........................................................................................................................ 378, 380, 388draft-ietf-http-authentication-03 ........................................................................................................................................ 513draft-ietf-sipping-realtimefax-00 ........................................................................................................................................ 335

EETSI 300 659-1 January 2001 (Annex B) ......................................................................................................................... 290ETSI EN 300659-3 ........................................................................................................................................................... 294

IIEEE Std 1003.1-2001 - IEEE Standard for Information Technology---Portable Operating System Interface (POSIX®) 454ITU-T Recommendation E.164 - The international public telecommunication numbering plan ........................................ 453ITU-T Recommendation F.69 - List of Telex Destination Codes ...................................................................................... 453ITU-T Recommendation G.703 - Physical/Electrical Characteristics of Hierarchical Digital Interfaces ............ 232, 239, 259ITU-T Recommendation G.704 - Synchronous Frame Structures Used at Primary and Secondary Hierarchy Levels ... 232,

239, ............................................................................................................................................................. 259ITU-T Recommendation G.711 ........................................................................................................................................ 373ITU-T Recommendation G.723.1 ..................................................................................................................................... 374ITU-T Recommendation G.726 ........................................................................................................................................ 374ITU-T Recommendation G.729 ........................................................................................................................................ 375ITU-T Recommendation I.430 - Basic user-network interface - Layer 1 specification ...................................................... 239ITU-T Recommendation I.431 - Primary rate user-network interface - Layer 1 specification ........................................... 232ITU-T Recommendation Q.24 - Multifrequency push-button signal reception .................................................................. 378ITU-T Recommendation Q.421 - Digital line signalling code ............................................................................................ 259ITU-T Recommendation Q.441 - Signalling code ............................................................................................................. 259ITU-T Recommendation Q.921 - ISDN user-network interface - Data link layer specification ................................. 232, 239ITU-T Recommendation Q.931 - ISDN user-network interface layer 3 specification for basic call control ....... 232, 239, 456ITU-T Recommendation T.38 ................................................................................................................................... 335, 377ITU-T Recommendation X.121 - International numbering plan for public data networks ................................................. 453

RRFC 1350 - The TFTP Protocol (Revision 2) ................................................................................................................... 513RFC 1890 - RTP Profile for Audio and Video Conferences with Minimal Control ............................................................ 378RFC 2246 - The TLS Protocol Version 1.0 ............................................................................................................... 331, 514RFC 2459 - X.509 Digital Certificates ............................................................................................................................... 514RFC 2543 - SIP, Session Initiation Protocol ............................................................................................. 307, 312, 315, 345RFC 2616 - Hypertext Transfer Protocol - HTTP/1.1 ............................................................................................... 203, 513RFC 2617 - HTTP Authentication - Basic and Digest Access Authentication .................................................................. 513RFC 2705 - Media Gateway Control Protocol (MGCP) Version 1.0 ................................................................................. 365RFC 2818 - HTTP Over TLS ............................................................................................................................................ 514RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals ................................................... 378RFC 2976 - The SIP INFO Method .................................................................................................................................. 419RFC 3164 - The BSD Syslog Protocol ............................................................................................................................. 219RFC 3261 - SIP, Session Initiation Protocol ..................................................................... 307, 312, 315, 327, 331, 336, 344RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP) ................................................ 360RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP) .................................. 335, 339, 345, 346RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification ......................................................................... 435RFC 3268 - Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS) ............................. 514RFC 3280 - Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile ...... 514, 527RFC 3311 - The Session Initiation Protocol (SIP) UPDATE Method ................................................................................ 360RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP) ..................................................................... 486RFC 3325 - Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks .. 486RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping 353RFC 3515 - The Session Initiation Protocol (SIP) Refer Method ..................................................................................... 335

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Appendix E - List of Standards Supported

RFC 3550 - RTP - A Transport Protocol for Real-Time Applications ............................................................................... 373RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control ............................................................ 373RFC 3617 - Uniform Resource Identifier (URI) Scheme for the Trivial File Transfer Protocol ......................................... 513RFC 3711 - The Secure Real-time Transport Protocol (SRTP) ....................................................................................... 391RFC 3830 - MIKEY - Multimedia Internet KEYing ............................................................................................................ 391RFC 3842 - The Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

..................................................................................................................................................................... 435RFC 3863 - Presence Information Data Format (PIDF) ................................................................................................... 315RFC 3903 - Session Initiation Protocol (SIP) Extension for Event State Publication ............................................... 307, 315RFC 4040 - RTP Payload Format for a 64 kbit/s Transparent Call .................................................................. 375, 376, 377RFC 4235 - An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP) .................................... 422RFC 4567 - Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol

(RTSP) ........................................................................................................................................................ 391RFC 4568 - SDES - Security Descriptions for Media Streams ......................................................................................... 391RFC 5806 - Diversion Indication in SIP ............................................................................................................................ 460RFC 959 - File Transfer Protocol ...................................................................................................................................... 513

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Index
Numerics10 BaseT, defined 557100 BaseT, defined 557
AAccess via CLI 130acronyms 567A-Law 373

defined 557Announcement 106Announcement Menu 108Announcement Time Circuit 110Announcement Upload 119answering on caller ID, on FXO port 298AOC-E support, ISDN interface 254ApplyConfig via CLI 133authentication information 327

creating 328deleting 330editing 328moving 330

auto cancel timeout, ISDN interface 250Autoattendant 9

integration 8automatic

call 421configuration update

number of retries 525on restart 523specific time interval 523

auto-routingauto-routable parameter 509enabling 508introduction 507SIP gateway 509

BB2BUA 100Backup

Configuration Backup 17Failover PBX 62Remote Backup 20VRRP (Routers) 42

Bandwidth 36FIFO 38

Bandwidth and CLI 142Bandwidth Daemon and CLI 143Bandwidth Management Overview 37base ports 386Basic Settings (SBC) 85Basic Settings System 13bearer capabilities mapping, vs. codec 379

preferred codec choice 341Billing 179Bottom Menu Bar 7Branch SBC (Template) 78BRI interface configuration

bypass connection 246calling name max length 244channel allocation strategy 243clock mode 241

BRI interface configuration (continued)clock reference 215connection type 242copy config to all interfaces 240encoding scheme, fallback 243encoding scheme, preferred 242endpoint type 241exclusive B-channel selection 244inband DTMF dialing 244inband tone generation 243maximum active calls 243network location 242overlap dialing 244restart on startup 244select interface 240sending complete information 244signal information element 243signalling protocol 242

Browser 5Burst (Bandwidth) 38bypass

activation 295activation DTMF map 296deactivation timeout 296

bypass connection, in BRI 246

CCA Certificates 47call

automatic 421completion

CCBS 422CCBS, using 425CCNR 422CCNR, using 425

emergency, enabling 371forward

on busy 411, 413on no answer 414, 415unconditional 416, 417

hold 433direction attributes 345

putting on hold 428, 434second 434, 435transfer

attended 426, 427blind 426, 427

waiting 427disabling 429enabling 429using 428

Call Admission Control Template 60Call Detail Records 179call failure timeout, on FXO port 300Call Forwarding, Hunt Groups 116call properties translation, in call routing

built from 494creating 493defined 493deleting 496editing 493modified property 494

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Index

call properties translation, in call routing (continued)moving 496name 494

call rejection (drop) causes 499call router

call properties translationbuilt from 494creating 493defined 493deleting 496editing 493modified property 494moving 496name 494

hairpinning 504hunts

call rejection (drop) causes 499creating 496defined 496deleting 503destinations 497editing 496moving 503selection algorithm 497timeout 498

introduction 453limitations 454mappings, defined 476mappings, expression

accessing 478criteria 479deleting 486moving 486sub mappings 485transformation 483

mappings, typeaccessing 477input call property 478moving 486transformation 478

regular expression 454routes

call property to compare 472creating 471defined 470deleting 476destination, of call 475editing 471expression to compare 473mapping, apply 475moving 476signalling, of call 475source to compare 472

routing typeCalled E164 456Called Host 457Called Name 457Called NPI 457Called Phone Context 459Called SIP username 459Called TON 457Called URI 457Calling E164 456Calling Host 457Calling ITC 458Calling Name 457

call router (continued)routing type

Calling NPI 457Calling Phone Context 459Calling PI 458Calling SI 458Calling SIP username 459Calling TON 457Calling Uri 457Date/Time 459Last Diverting E.164 460Last Diverting Number Presentation 461Last Diverting Party Number Type 460Last Diverting Private Type of Number 461Last Diverting Public Type of Number 460Last Diverting Reason 460Original Diverting E.164 460Original Diverting Number Presentation 461Original Diverting Party Number Type 460Original Diverting Private Type of Number 461Original Diverting Public Type of Number 460Original Diverting Reason 460special tags 461

signalling properties180 with SDP 488183 without SDP 488call properties translation 489creating 487defined 486deleting 490destination host 488early connect 488early disconnect 488editing 487moving 490name 487privacy level 488SIP headers translation 489

SIP headers translationbuilt from 491creating 490defined 490deleting 493editing 490modified header 491moving 492name 491

status information 470call waiting, private number for SIP INFO 344Called E164, routing type 456Called Host, routing type 457Called Name, routing type 457Called NPI, routing type 457Called Phone Context, routing type 459Called SIP username, routing type 459Called TON, routing type 457Called URI, routing type 457caller ID

customization, in POTS 288DTMF signalling 290FSK generation 290generation 290

Calling E164, routing type 456Calling Host, routing type 457Calling ITC, routing type 458

572 Mediatrix Boss


calling name max length, ISDN interfaceBRI configuration 244PRI configuration 237

Calling Name, routing type 457Calling NPI, routing type 457Calling Phone Context, routing type 459Calling PI, routing type 458Calling SI, routing type 458Calling SIP username, routing type 459Calling TON, routing type 457Calling Uri, routing type 457CAS. see R2 configurationCCBS

setting up 422using 425

CCNRsetting up 422using 425

CED tone detection 386behaviour 386

Ceiling (Bandwidth) 38Ceiling Burst (Bandwidth) 38certificate, validation, in TLS 342certificates

deleting 528transferring 528, 530validation, in configuration script download 521

Chaeck Configuration via CLI 134Change Password 15Change Source IP 178channel allocation strategy

ISDN interfaceBRI configuration 243PRI configuration 236

R2 interface, configuration 263channel range

ISDN interface, PRI configuration 236R2 interface, configuration 262

Class Description (Bandwidth) 38Clear Channel

defined 376enabling

for data transmission 402for voice transmission 402

packetization time 402payload type 402priority

for data codecs 402for voice codecs 402

clear channel faxcall flow 406limitation 406

Clear Modedefined 375enabling




CLIBranch SBC 154Custom 153ENUM 154

CLI (continued)IP Tables, Firewall Rules 134QoS 176Session Border Controller 154SIP Proxy Scenarios 153SIP Trunking 154Standalone 153Survivability 153Traffic Control 176

CLIP, ISDN interface 252CLIR, ISDN interface 252

override 253clock mode


R2 interface, configuration 261clock reference, PRI configuration 217clock reference, in ISDN, BRI configuration 215CNG tone detection 385, 450codec

bearer capabilities mapping 379preferred codec choice 341

Clear Channeldefined 376enabling




Clear Modedefined 375enabling




DTMF payload type 383DTMF transport type 383G.711

defined 373enabling


packetization time 394priority


voice activity detection 385G.723

bit rate 395defined 374enabling for voice transmission 395packetization time 396priority for voice codecs 395

G.726defined 374enabling


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Index

codec (continued)G.726




defined 375enabling for voice transmission 399packetization time 400priority for voice codecs 399voice activity detection 400

T.38defined 377enabling 407number of redundancy packets 407priority 407

X-CCD Clear Channeldefined 377enabling




COLP, ISDN interface 253COLR, ISDN interface 253

override 254Conference Room with Moderator 113Conferencing 10configuration, exporting to script 515Configuration Backup 17configuration script, exporting configuration 515configuration script download

automatic updatenumber of retries 525on restart 523specific time interval 523

certificate validation 521decryption 518DHCP auto-provisioning

DHCP, auto-provisioning (scripts) 524downloading 522location path 518privacy key 518remote server

HTTP server, configuring 514HTTPS server, configuring 514SNTP server, configuring 514TFTP server, configuring 513

script example 526server, IP address 518transfer protocols 518

connection type, ISDN interface, BRI configuration 242Country 294country selection, R2 interface 264country-specific parameters

CNG tone detection 450country 447custom tone configuration 437dialing settings 449list of countries 533

country-specific parameters (continued)user gain

input 448output 448

Creating a Conference Room 112Custom Firewall Rules 57Custom Port Numbers 136Custom Template (Voice) 98custom tones, configuring 437

DDate/Time, routing type 459Debugging 123decryption, configuration script download 518Default Password 5delayed hot line 431Demilitarized Zone 30detection custom repetition, on FXO port 302DHCP Server 28diagnostic traces, enabling 219

PCM capture 221dial tone detection mode, on FXO port 297dial tone detection timeout, on FXO port 298dialing

settingsDTMF duration value 449inter-digit dial delay 449

direct IP address, call 432direction attribute "sendonly" 341, 346direction attributes, in SIP 345disabling endpoints 323distinctive ring 535DMZ 30DNS 33

supported queries 362DNS (System) 14Domain Name Server 14Dropped Packages 126DSCP 44DTMF

base ports 386defined 559detection 389duration value 449machine detection 385

CED tone detection 386CED tone detection, behaviour 386CNG tone detection 385V.21 modulation detection 386

out-of-band 384signalling, caller ID 290transport type 383

over the SIP protocol 388payload type 383

DTMF mapallowed 370defined

# and * characters 367combining two expressions 366special characters 366timer 367using 366validating 367

definition 365general parameters 368

574 Mediatrix Boss


DTMF map (continued)refused 372

Dynamic DNS 32Dynamic IP Address (LAN Interface 1) 26

Eemergency call, enabling 371enabling endpoints 323encoding scheme, R2 interface, configuration 263encoding scheme, ISDN interface

fallbackBRI configuration 243PRI configuration 236

preferredBRI configuration 242PRI configuration 236

encoding scheme, preferred, R2 interface, configuration 263endpoint

administration 324locking/unlocking 323unregistered, behaviour when 321

endpoint servicesautomatic call 421call completion

CCBS 422CCNR 422

call forwardon busy 411on no answer 414unconditional 416

call transferattended transfer 426blind transfer 426

call waiting 427conference call 429direct IP address call 432hold 433

direction attributes 345hook-flash processing 420second call 434

endpoint type, ISDN interfaceBRI configuration 241PRI configuration 233

ENUM 95ENUM Template 95error causes mapping, setting 353

cause to SIP 356SIP to cause 355

Error Diagnostics 123Error Logs 122Ethereal 126exclusive B-channel selection, ISDN interface

BRI configuration 244PRI configuration 237

Extended Options 129

Ffacility ISDN services, enabling 251Factory Reset 175factory reset

reverting to 201see also partial reset

Failover PBX 62

faxcall waiting tone, disabling 429calling tone detection, enabling 450clear channel

call flow 406limitation 406

T.38defined 377enabling 407number of redundancy packets 407priority 407

user gain vs communication quality 448FIFO 38Filter (Bandwidth) 39Firewall 56Firewall CLI 150Firewall Report 126Firewall Rules 56Firewall Support 126flash hook detection range 294force end of call on call failure, on FXO port 300force_send_notifies 176forced end of call on silence detection mode, on FXO port 300forced end of call on tone detection mode, on FXO port 301Foreign Exchange Office (FXO)

defined 559see also endpoint

Foreign Exchange Service/Station (FXS)defined 559see also endpoint

FSK generation, caller ID 290Function Keys 130FXO configuration

answering on caller ID 298call failure timeout 300detection custom repetition 302dial tone detection mode 297dial tone detection timeout 298force end of call on call failure 300forced end of call on silence detection mode 300forced end of call on tone detection mode 301link state verification 299link state verification timeout 299not allowed behavior 299pre dial delay 297silence detection timeout 301tone detection custom cadence 302tone detection custom frequency 301wait before answering delay 298wait for callee to answer 299

FXSauto cancel timeout 292bypass

activation 295activation DTMF map 296deactivation timeout 296

country overrideflash hook detection range 294loop current 294

disconnect delay 291inband ringback 292line supervision mode 291override country customization 293power drop on disconnect duration 292service activation 292shutdown behavior 292

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Index

GG.711

defined 373enabling


packetization time 394priority



bit rate 395defined 374enabling for voice transmission 395packetization time 396priority for voice codecs 395

G.726defined 374enabling





defined 375enabling for voice transmission 399packetization time 400priority for voice codecs 399voice activity detection 400

Gateway, Internal Architecture 8gateways, multiple SIP

adding 305messaging servers 310proxy servers 311registrar servers 310removing 305

group port management 323GUI Concepts 7

Hhairpinning 504hardware configuration

BRI interface, clock reference 215PRI interface

clock reference 217line type 217, 260signaling 217, 261

header, SIP user agent, sending 322Hierarchical Token Bucket (Bandwidth) 36hold, putting a call on 428, 433, 434

direction attributes 345home domain proxy override 361hook-flash processing 420HTB 36HTTP, remote server, configuring 514HTTPS, remote server, configuring 514HTTPS Port 23Hunt Group 116

hunts, in call routingcall rejection (drop) causes 499creating 496defined 496deleting 503destinations 497editing 496moving 503selection algorithm 497timeout 498

Iinband DTMF dialing, ISDN interface


inband tone generation, ISDN interfaceBRI configuration 243PRI configuration 237

Initial Setup via CLI 131inter-digit dial delay 449IP (LAN Interface 1) 27IP address

configuration script download server 518SIP outbound proxy 309SIP proxy 309SIP registrar 309syslog daemon 219vocal identification of 289

IP address call 432ISDN configuration

auto calcel timeout 250BRI interface

bypass connection 246calling name max length 244channel allocation strategy 243clock mode 241connection type 242copy config to all interfaces 240encoding scheme, fallback 243encoding scheme, preferred 242endpoint type 241exclusive B-channel selection 244inband DTMF dialing 244inband tone generation 243maximum active calls 243network location 242overlap dialing 244restart on startup 244select interface 240sending complete information 244signal information element 243signalling protocol 242

interop parameterscopy config to all interfaces 247, 250, 251maximum facility waiting delay 248Progress Indicator in Alerting 248Progress Indicator in Call Proceeding 248Progress Indicator in Connect 248Progress Indicator in Progress 248Progress Indicator in Setup 247Progress Indicator in Setup ACK 247select interface 247

introduction 229Layer 1 Timer 3 250

576 Mediatrix Boss


ISDN configuration (continued)PRI interface

calling name max length 237channel allocation strategy 236channel range 236clock mode 234copy config to all interfaces 233encoding scheme, fallback 236encoding scheme, preferred 236endpoint type 233exclusive B-channel selection 237inband DTMF dialing 237inband tone generation 237line coding 235line framing 235line type 231link establishment 237, 244maximum active calls 236network location 235overlap dialing 237restart on startup 237select interface 233sending complete information 237signal information element 236signalling protocol 235

servicesAOC-E support 254CLIP 252CLIR 252CLIR override 253COLP 253COLR 253COLR override 254facility services, enabling 251Maintenance Service Call Termination 254MSN 255select interface 251supplementary services, enabling 251

Jjitter buffer protection 380

Kkeep alive, in SIP 359

LL2TP 52L2TP Server via CLI (setup) 146L2TP Users (CLI) 147LAN Interface 1 26LAN Interface 2 29Last Diverting E.164, routing type 460Last Diverting Number Presentation, routing type 461Last Diverting Party Number Type, routing type 460Last Diverting Private Type of Number, routing type 461Last Diverting Public Type of Number, routing type 460Last Diverting Reason, routing type 460Layer 1 Timer 3, ISDN interface 250Leaf Queuing Discipline (Bandwidth) 38License Information 24limitations, DSP 195line coding

ISDN interface, PRI configuration 235

line coding (continued)R2 interface, configuration 262

line framingISDN interface, PRI configuration 235R2 interface, configuration 262

line signalling protocol, R2 interface, configuration 263line type, PRI configuration 217, 231, 260Link 123link establishment, ISDN interface, PRI configuration 237, 244link state verification timeout, on FXO port 299link state verification, on FXO port 299local ring behaviour, in SIP 347locking endpoints 323Log Level (Bandwidth) 40Logging (CLI) 152Logging Messages 122loop current 294

MMAC address

defined 561vocal identification of 289

machine detection 385CED tone detection 386CED tone detection, behaviour 386CNG tone detection 385V.21 modulation detection 386

Main Frame 7Main Navigation 7Maintenance Service Call Termination, ISDN interface 254Management (Bandwidth) 36mappings, in call routing

defined 476expression

accessing 478criteria 479deleting 486moving 486sub mappings 485transformation 483

typeaccessing 477input call property 478moving 486transformation 478

Max-Forwards header, in SIP 344maximum active calls


R2 interface, configuration 263maximum facility waiting delay, ISDN interface 248Maximum Transmission Unit 140Media Stream Trace 126message waiting indicator 435Meta Rules (CLI) 148MSN, ISDN interface 255MTU 140MTU (Bandwidth) 39Mu (µ)-Law 373

defined 561music on hold, configuring 441Music on Hold (upload) 118

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Index

NName Server 33NAT, in configuration script download 522Netmask (LAN Interface 1) 27network location, ISDN interface


Network Trace 126New IPsec Connection 49New OpenVPN Connection 49not allowed behavior, on FXO port 299NTP (Server) 41Number of Samples (Bandwidth) 40

OOffer/Answer model 336, 339, 340, 343Original Diverting E.164, routing type 460Original Diverting Number Presentation, routing type 461Original Diverting Party Number Type, routing type 460Original Diverting Private Type of Number, routing type 461Original Diverting Public Type of Number, routing type 460Original Diverting Reason, routing type 460outbound proxy server, configuring 309out-of-band DTMF 384overlap dialing, ISDN interface


Ppacketization time

Clear Channel 402Clear Mode 400G.711 394G.723 396G.726 398G.729 400X-CCD Clear Channel 404

Page Header 6Parent Class (Bandwidth) 38partial reset 200

disabling 200see also factory reset

Password 15RADIUS 180

Password Reset 175Passwords, System, Separate Passwords. See Change Passwordpayload type

Clear Channel 402Clear Mode 400G.726 398using the one found in answer 390X-CCD Clear Channel 404

PBX Extensions 115PCM capture 221penalty box, in SIP 351performance and usage recommendation 3Perturbation (Bandwidth) 38Ping 123port number

configuration script download server 518SIP outbound proxy 309SIP proxy 309SIP registrar 309

Port Numbers 23POTS

auto cancal timeout 292caller ID customization 288FXO configuration

answering on caller ID 298call failure timeout 300detection custom repetition 302dial tone detection mode 297dial tone detection timeout 298force end of call on call failure 300forced end of call on silence detection mode 300forced end of call on tone detection mode 301link state verification 299link state verification timeout 299not allowed behavior 299pre dial delay 297silence detection timeout 301tone detection custom cadence 302tone detection custom frequency 301wait before answering delay 298wait foe callee to answer 299

FXS bypassactivation 295activation DTMF map 296deactivation timeout 296

FXS Country Override Flash Hook Detection Range 294FXS Country Override Loop Current 294FXS Disconnect Delay 291FXS Inband Ringback 292FXS Line Supervision Mode 291FXS Override Country Customization 293FXS power drop on disconnect duration 292FXS service activation 292FXS shutdown behavior 292status information 287

PRACK, support 360pre dial delay, on FXO port 297PRI interface configuration

calling name max length 237channel allocation strategy 236channel range 236clock mode 234clock reference 217copy config to all interfaces 233encoding scheme, fallback 236encoding scheme, preferred 236endpoint type 233exclusive B-channel selection 237inband DTMF dialing 237inband tone generation 237line coding 235line framing 235line type 217, 231, 260link establishment 237, 244maximum active calls 236network location 235overlap dialing 237restart on startup 237select interface 233sending complete information 237signal information element 236signaling 217, 261signalling protocol 235

Priority (Bandwidth) 38privacy key, configuration script download 518

578 Mediatrix Boss


privacy level of calls 488Progress Indicator in Alerting, ISDN interface 248Progress Indicator in Call Proceeding, ISDN interface 248Progress Indicator in Connect, ISDN interface 248Progress Indicator in Progress, ISDN interface 248Progress Indicator in Setup ACK, ISDN interface 247Progress Indicator in Setup, ISDN interface 247Provisioning 17proxy server

SNMP, configuring via, override 348proxy server, configuring 309

QQoS 36, 44

defined 563VLAN 225

QSIG, signalling protocol 235, 242

RR2 configuration

channel allocation strategy 263channel range 262clock mode 261country selection 264encoding scheme 263encoding scheme, preferred 263introduction 259line coding 262line framing 262line signalling protocol 263maximum active calls 263R2 digit timers 272R2 link timers 274R2 signaling 264

C bit setting 266D bit setting 266override default country settings 265

R2 timers 268R2 tones backward groups 279R2 tones forward groups 278register signalling protocol 263, 264selecting R2 signaling protocol 260

RADIUS 180Rate (Bandwidth) 38Redundancy

Failover PBX 62VRRP 42

register home domain override, in SIP 348register signalling protocol, R2 interface 263, 264registrar server, configuring 309registration

expiration, in SIP 318, 320refresh, in SIP 317, 320

regular expression, defined 454Remote Backup 20Remote Update 16reset

factory reset procedure 201partial reset procedure 200

disabling 200Restart 22restart, unit 207, 211restart on startup, ISDN interface

BRI configuration 244

restart on startup, ISDN interface (continued)PRI configuration 237

Restart Services via CLI 133Restore Configuration 17Restore Values 24Revert Configuration 18ring, distinctive 535Rollback 18routes, in call routing

call property to compare 472creating 471defined 470deleting 476destination, of call 475editing 471expression to compare 473mapping, apply 475moving 476signalling, of call 475source to compare 472

RTCP 225RTP, enforcing symmetric 393RTP statistics

collection period 446end-of-connection notification 446end-of-period notification 446statistics displayed 443

RTP, early, listening to 348

SSample Rate (Bandwidth) 40Save Configuration via CLI 135second call, service 435secure communication

introduction 208secure media 391secure signalling 332

select interface 250sending complete information, ISDN interface


Serial Console 15Serial Line Access 130services, ISDN

AOC-E support 254CLIP 252CLIR 252

override 253COLP 253COLR 253

override 254enabling 251Maintenance Service Call Termination 254MSN 255select interface 251

services, managing 211Session Border Controller 83session ID/number in Origin field, in SIP 347session timer, enabling 325Shared Secrets (OpenVPN) 48signal information element, ISDN interface


signaling, PRI configuration 217, 261

Mediatrix Boss 579

Index

signalling properties, in call routing180 with SDP 488183 without SDP 488call properties translation 489creating 487defined 486deleting 490destination host 488early connect 488early disconnect 488editing 487moving 490name 487privacy level 488SIP headers translation 489

signalling protocolISDN interface


silence detection timeout, on FXO port 301SIP headers translation, in call routing

built from 491creating 490defined 490deleting 493editing 490modified header 491moving 492name 491

SIP INFO, call waiting private number for 344SIP Proxy 59SIP Server (standalone) 93SIP Server and CLI 153SIP Traces 126SIP Trunk (Template) 89SIP, setting

allow audio and image negotiation 341allow less media in response 340allow media reactivation in answer 340authentication information

creating 328deleting 330editing 328moving 330

call waiting private number for SIP INFO 344calling party name of the caller ID 294default username value 336direction attribute "sendonly" 341, 346direction attributes 345direction attributes present 345early RTP, listening to 348error causes mapping 353

cause to SIP 356SIP to cause 355

escape pound (#) in SIP URI Username 344gateways

adding 305auto-routing, setting for 509messaging servers 310proxy servers 311registrar servers 310removing 305

home domain proxy override 361Ignore OPTONS on no usable endpoints 337ignore plus (+) character in username 343keep alive 359

SIP, setting (continued)local ring behaviour 347map plus to TON International 343Max-Forwards header 344Offer/Answer model 336, 339, 340, 343OPTIONS Method Support 337outbound proxy server 309

loose router status 312payload type in answer, using 390penalty box 351persistent TLS connection port 332persistent TLS retry interval 332PRACK 360proxy server 309quantity of RTP packets 391register home domain override 348registrar server 309registration configuration 317registration contact matching 338registration expiration 318, 320registration refresh 317, 320secure header 336session ID/number in Origin field 347session timer, enabling 325supported DNS queries 362symmetric RTP, enforcing 393TCP/TLS connect timeout 332, 361TLS certificate validation 342TLS client authentication 334TLS persistent connections status 308transmission timeout 338transport type 331

TCP 331UDP 331

UDP source port behaviour 333unit registration 317UPDATE 360user agents

authentication information 327friendly user name 315header, enabling to send 322main user name 315

voiceband data mode, call in 387SipEp, service

endpoint behaviour when unregistered 321registration parameters 320

SNMPcommand

syntax, normal 550syntax, row 551

SNMP (Daemon) 124SNMP Daemon (CLI) 152SNTP, remote server, configuring 514special tags, in routing types values 461special vocal features

IP address 289MAC address 289

SSH 15SSH Access 130SSH Port 23Static Routes 142supplementary ISDN services, enabling 251Survivability 60Survivability Basic Settings 60Syslog 122

580 Mediatrix Boss


syslog daemonconfiguring the application 222diagnostic traces 219

PCM capture 221IP address 219messages level 219

Syslog File 121System 136System Log Information 122System Name 14

TT.38

defined 377enabling 407number of redundancy packets 407priority 407

Tabs of the WebGUI 11TCP

connect timeout 332, 361transport type 331

tcpdump 126Templates (Voice) 59Terminal Emulator Program 130TFTP, remote server, configuring 513Time and Date 13Time Zone 14TLS

certificate validation 342client authentication 334connect timeout 332, 361persistent connection port 332persistent connections status 308persistent retry interval 332

tone detection custom cadence, on FXO port 302tone detection custom frequency, on FXO port 301Top Menu Bar 6TOS 44Trace Parameters 126Traceroute 123Traces 123Traffic Shaping (Bandwidth) 36transfer protocols, configuration script download 518transferring a call

attended transfer 427blind 427

transmission timeout, setting 338

UUDP

source port behaviour 333transport type 331

unitlocking/unlocking endpoints 323restarting 207, 211

unlocking endpoints 323Update 16UPDATE, support 360user gain

input 448output 448

VV.21 modulation detection 386Virtual Interface 31Virtual Interfaces via CLI 139VLAN, in QoS 225vocal features, special

IP address 289MAC address 289

vocal unit information 289voice activity detection

G.711 385G.726 385G.729 400

voiceband data mode, starting a call in 387VPN 45VPN and CLI 143VPN Connections 49VRRP 42, 178

Wwait before answering delay, on FXO port 298wait for calle to answer, on FXO port 299WAN and CLI 139WAN Interface 25web interface

automatic call 421call completion

CCBS 422CCNR 422

call forwardon busy 411on no answer 414unconditional 416

call hold 433call transfer

attended 426blind 426

call waiting 427conference call 429introduction 203second call 434using 204

Web, service, TCP port 203WebGUI Access 5Wireshark 126

XX.509 Certificates 47X-CCD Clear Channel

defined 377enabling




Mediatrix Boss 581

Index

582 Mediatrix Boss

administrator and user manual - media5...

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