book4 integrate partner technologies

Integrate Partner Technologies

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4Book

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H T T P : / / W W W . S P H E R E C O M . C O MP A R T N U M B E R 5 4 0 - 4 0 4 R 7

V E R S I O N 6 . 2

Integrate Partner Technologies

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Copyright © 2008NEC Sphere Communications Inc., a wholly owned subsidiary of NEC Corporation, Japan. All Rights Reserved. Printed in USA.NEC Sphere Communications Inc. (or “Sphere”) is continually upgrading and developing the products described in this publication. The information contained herein is specifically designed for the Sphericall v6.2 release and is subject to change without notice. Written permission is required prior to reproduction of any of the work covered here by copyright.For warranty information, see the License Agreement on the Sphericall software DVD media.Sphere, Sphericall, Sphericall Voice Mail, Sphericall Manager, Sphericall Desktop, PhoneHub, COHub, BranchHub, MeetingHub, VG3, and the Sphere logo are trademarks of NEC Sphere Communications Inc. a wholly owned subsidiary of NEC Corporation.NEC is a trademark of NEC Corporation, Japan. Windows, Microsoft, Outlook, and Exchange are trademarks or registered trademarks of Microsoft Corporation. Other products mentioned in this document are the property of their respective owners and are also subject to copyright, trademark, and intellectual property protection as applicable by law.

U.S. Patent Numbers 5,892,764 and 6,735,208 and related Foreign Patents. Other U.S. and Foreign Patents Pending.

D O C U M E N T R E V I S I O N H I S T O R Y

540-404r4 26 February 2006 Version 5.0 product, updates, edits & corrections

540-404r5 25 July 2006 Version 5.1 product, updates, edits & corrections

540-404r6 01 August 2007 Version 6.0 product, updates, edits & corrections

540-404r7 17 January 2008 Version 6.2 product, updates, edits, corrections and branding changes.

iv 540-404r7 Integrate Partner Technologies

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. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C O N T A C T N E C S P H E R E C O M M U N I C A T I O N S

M A I L I N G A D D R E S SNEC Sphere Communications Inc.300 Tri-State International Suite 150Lincolnshire, IL 60069 USAPhone 1-847-793-9600; Toll Free Phone 1-888-774-3732; Fax 1-847-793-9690NEC Sphere Communications Inc. is a wholly owned subsidiary of NEC Corporation.

C O R P O R A T E C O N T A C T S

N E C H E A D Q U A R T E R S & A S I A O P E R A T I O N SNEC Corporation7-1, Shiba 5-chome Minato-ku, Tokyo 108-8001 JapanTelephone: +81-3-3454-1111Fax: +81-3-3798-1510Website: http://www.nec.co.jp/

E U R O P E , M I D D L E E A S T A N D A F R I C ANEC Philips Unified SolutionsThe corporate headquarters of NEC Philips is located at: NEC Philips Unified Solutions Nederland B.V.Anton Philipsweg 11223 KZ Hilversum, NetherlandsTelephone: +31 35 689 9111Fax: +31 35 689 1450Website: http://www.nec-philips.comFor a complete list of countries served and local contact information, please visit NEC Philips at: http://content.nec-philips.com/hq/office-locations/

N O R T H A M E R I C ANEC Unified Solutions, Inc.6535 N. State Highway 161Irving, TX 75039-2402Telephone: 1-800-240-0632 (800-2400-NEC)Fax: 1-888-318-7932

Integrate Partner Technologies 540-404r7 v

Authorized Associates and Resellers:Phone: 1-800-752-6275Fax: 214-262-5566Website: http://www.necunified.com

vi 540-404r7 Integrate Partner Technologies

Contents. . . .

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Notices ................................................................................................................ ivDocument Revision History ............................................................................. iv

Contact NEC Sphere Communications................................................................ vCorporate Contacts ......................................................................................... -v

1 Integrate Partner Technologies ................................................1-1Document Index ................................................................................................1-1

2 MGCP IP Phones........................................................................2-5I. Polycom SoundPoint IP Phone ......................................................................2-5Planning ............................................................................................................2-6

Final Planning ............................................................................................2-6Preparing...........................................................................................................2-6

FTP Server, DHCP Server, SNTP Services ..............................................2-6FTP server—to create login and password for IP phones on FTP server .2-8FTP—to change the IP Phone User Account ............................................2-8

Installing ..........................................................................................................2-11System Properties .......................................................................................2-11

Sphericall Manager—to configure System properties..............................2-11MGC-To-MGCP Phone Connection Control................................................2-11

Sphericall Manager—to configure MGC-to-MGCP phone connections on a Sphere system .........................................................................................2-12SoundPoint MGCP Phone—to assemble and power the phone .............2-12To install the MGCP phone with dynamic IP addressing you will need to: ..2-12To install the SoundPoint MGCP phone with a static IP address you will need to: .............................................................................................................2-13Sphericall Manager—to complete configuration ......................................2-14

MGCP Phone Installation Test........................................................................2-14To test for successful extension and station configuration ......................2-14

IP Phone Troubleshooting...............................................................................2-14Normal Boot Screens ..................................................................................2-15Installation Issues ........................................................................................2-15Restarting the Ip Phone...............................................................................2-16

SoundPoint phone—to restart the IP phone ............................................2-16Sphericall Manager—to restart the IP phone remotely ............................2-16Sphericall Manager—to sync IP Phone Files...........................................2-16Sphericall Manager—to Revert to Default Configuration .........................2-16Sphericall Manager—to view Default Parameters ...................................2-17

IP Phone Configuration File Upgrades ........................................................2-17Convert Polycom MGCP to SIP Command .................................................2-18

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IP Phone Failover ........................................................................................2-19IP Phone Upgrades.........................................................................................2-19Using ...............................................................................................................2-19II. Aastra 480i IP Phone ..................................................................................2-20Planning ..........................................................................................................2-20

Final Planning ..........................................................................................2-21Preparing.........................................................................................................2-21

Verify permissions required .....................................................................2-21FTP server—to create login and password for IP phones on the FTP Server2-23FTP—to change the IP Phone User Account ..........................................2-23FTP Server—to add required Sayson files to FTP server........................2-23TFTP Server—to copy files to the TFTP server.......................................2-23Sphericall Manager—to add FTP server location for use by Aastra phones ..2-24Sphericall Manager—to add Aastra FTP login and password .................2-24

MGC-To-MGCP Phone Connection Control................................................2-24Sphericall Manager—to configure MGC-to-MGCP phone connections on a Sphere system .........................................................................................2-24

Installing ..........................................................................................................2-25480i IP Phone—to assemble and power the phone.................................2-25480i IP Phone—to configure the 480i phone ...........................................2-25To install the Aastra IP phone with a static IP address you will need to: .2-26On the phone: ..........................................................................................2-26Open Internet Explorer:............................................................................2-26

Optional Configurations ...............................................................................2-27Web Browser—to continue using the web browser for optional configurations2-27

Busy-Lamp-Field Configuration ...................................................................2-28STEP ONE: Setup Address Group(s) on the Sphericall Manager ...........2-28STEP TWO: Setup the CTIP Multicast IP Address on the Web Browser 2-28STEP THREE: Web Browser—to configure Busy Lamp Field Keys........2-29Sphericall Manager—to complete configuration ......................................2-30

IP Phone Installation Test ...............................................................................2-30To test for successful extension and station configuration ......................2-30

Troubleshooting ..............................................................................................2-30For overview of phone key options: .........................................................2-30Web Browser—to Reset the 480i phone..................................................2-33Sphericall Manager—to Reset the 480i phone from the Sphericall Administrator............................................................................................2-33480i phoneset—to Reset the 480i phone back to factory defaults...........2-33

Changing the Discovery IP Address ...............................................................2-38STEP ONE: Sphericall Manager—to change the Discovery IP Address .2-38STEP TWO: From the Phone—to change the Discovery IP Address......2-38

IP Phone Upgrades.........................................................................................2-39Using ...............................................................................................................2-39

3 SIP Phones ...............................................................................3-41SIP Configuration Notes..................................................................................3-42

General SIP Configuration Information........................................................3-42Section I - SIP User Agents ............................................................................3-45

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To configure SIP User Agents .................................................................3-45User Agent Maintenance.................................................................................3-48

To remove a User Agent Parameter ........................................................3-49SIP Failover.....................................................................................................3-49

create a DNS Record ..................................................................................3-49On DNS Server—To create A DNS Record.............................................3-50

Section III - Aastra SIP Phone Configuration ..................................................3-52Overview .........................................................................................................3-52

To integrate the Aastra phone with the Sphere system ...........................3-52To enable MWI on an Aastra SIP phone .................................................3-53

Softkeys/Programmable Keys .....................................................................3-54Hardkeys .....................................................................................................3-54

To save programmable keys from the phone ..........................................3-55To program keys from the Aastra web interface ......................................3-55

Configuration Files.......................................................................................3-56Upgrades .....................................................................................................3-56

To upgrade the Aastra Phone..................................................................3-57Sphere System Upgrades ...........................................................................3-57Section IV - Grandstream GXP-2000 SIP Phone Configuration..................3-58

Overview .........................................................................................................3-58Planning ..........................................................................................................3-58Installing ..........................................................................................................3-59

To integrate the Grandstream GXP-2000 with the Sphere system..........3-59Web Configuration ..........................................................................................3-59

Account Page ..............................................................................................3-60Basic Settings Page ....................................................................................3-62

To configure the GXP-2000 basic settings ..............................................3-62To configure speed dial............................................................................3-63To configure the time zone ......................................................................3-63

Advanced Settings.......................................................................................3-63To upgrade the Grandstream GXP-2000.................................................3-64

Grandstream GXV-3000 Phone ......................................................................3-66Overview .........................................................................................................3-66Planning ..........................................................................................................3-66Installing ..........................................................................................................3-67

To integrate the Grandstream GXV-3000 with the Sphere system..........3-67Web Configuration ..........................................................................................3-67

Account Page ..............................................................................................3-68Basic Settings Page ....................................................................................3-70

To configure the GXP-3000 basic settings ..............................................3-70To configure speed dial............................................................................3-71To configure the time zone ......................................................................3-71

Advanced Settings.......................................................................................3-71To upgrade the Grandstream GXV-3000.................................................3-72

Section V - Polycom SIP Phone Configuration ...............................................3-74Planning ..........................................................................................................3-74Overview .........................................................................................................3-74

Structure of Polycom Configuration Files ....................................................3-75System Overrides ........................................................................................3-78Localization..................................................................................................3-79Convert Polycom MGCP to SIP Command .................................................3-81Adding a Polycom SIP Phone to A Station..................................................3-81


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To add a Polycom SIP phone to the Sphere system ...............................3-81To add directory information to the Polycom phone ................................3-84

Export Phone Distribution Map....................................................................3-86To export phone distribution map ............................................................3-86

Upgrades .....................................................................................................3-87Sphere System Upgrades ...........................................................................3-88

Section VI - UTStarcom F1000G/F3000 SIP Phone Configuration.................3-89Overview .........................................................................................................3-89

Things To Consider .....................................................................................3-89A. WiFi and Network Settings......................................................................3-90

On the F1000/3000—To configure WiFi and Network Settings ...............3-90B. F1000/3000 Web Configuration ..............................................................3-91

Web Browser—To configure the F1000/3000 from the web....................3-91Web Browser—To configure F1000/3000 user settings ..........................3-93Web Browser—To configure wireless access point settings ...................3-93

C. F1000/3000 and Sphericall Voice Mail ...................................................3-94Sphericall Manager—To configure Sphericall Voice Mail settings...........3-94

D. Firmware UpDates ..................................................................................3-94To update the F1000/3000 firmware version ...........................................3-94

Sphere System Upgrades ...........................................................................3-95Section VII - SIP Phone Compatibility and Capability with Spherciall Desktop...3-96Section VIII - SIP Phone Administrative Star Codes.......................................3-97Troubleshooting SIP Connections...................................................................3-98

4 ATI RG6XX or iMG6XX...........................................................4-101Planning ........................................................................................................4-101Preparing.......................................................................................................4-101Installing ........................................................................................................4-102

To verify firmware version of RG or iMG................................................4-102To update flash of RG............................................................................4-102To configure RG.....................................................................................4-103To set the flash-hook and on-hook timing..............................................4-104To restart device and apply changes .....................................................4-104To update firmware after upgrading to Sphericall v6.0.0.6 ....................4-104

Using .............................................................................................................4-105

5 USB Devices...........................................................................5-107Eutectics IPP200...........................................................................................5-107

Planning.....................................................................................................5-107Preparing ...................................................................................................5-107Installing ....................................................................................................5-108Using .........................................................................................................5-108

Eutectics IPP520...........................................................................................5-108Planning.....................................................................................................5-108Overview of Operation...............................................................................5-109Installation .................................................................................................5-109Test The Installation ..................................................................................5-109

To test the installation and make sure the IPP520 is properly recognized by


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your PC ..................................................................................................5-109Configuration .............................................................................................5-109Functionality ..............................................................................................5-110

Plantronics Wireless set CS50-USB .............................................................5-112Planning.....................................................................................................5-112Overview of Operation...............................................................................5-112Installation .................................................................................................5-112

6 Microsoft Windows Messenger Client .................................6-115Planning ........................................................................................................6-115Preparing.......................................................................................................6-116Installing ........................................................................................................6-116

Sphericall Manager—to prepare for Windows Messenger endpoints....6-117Client PC—to install Windows Messenger to run in the Sphericall environement .........................................................................................6-117Client PC—to allow any clients to text message Messenger clients......6-118Sphericall Manager—to configure a primary extension for the messenger client.......................................................................................................6-118Client PC—to allow Messenger clients to monitor presence state ........6-119Client PC—to add phone equipment to the Windows Messenger client6-119

Testing ..........................................................................................................6-120Using Windows Messenger...........................................................................6-120

7 Music On Hold........................................................................7-123Music-on-Hold ...............................................................................................7-123

Section I - Hardware-based Music-on-Hold...............................................7-123To Configure Music-on-Hold Sources....................................................7-125To assign an extension to a Music-on-Hold station ...............................7-128

Section II - Music-on-Hold Installation Test ...............................................7-129To test functionality of the Music-on-Hold feature..................................7-129

Section III - Music-on-Hold and Zones ......................................................7-129

8 Paging.....................................................................................8-133Paging Lines .................................................................................................8-133

Overview....................................................................................................8-133Configuring a Paging Line .........................................................................8-134

To add a paging line to a Sphere system ..............................................8-134To configure a paging line for a Sphere system ....................................8-134

Installing and Integrating a Paging Device ................................................8-136To integrate a Sphere system with a paging device ..............................8-136

Installation Test .........................................................................................8-138To verify successful paging system installation and configuration.........8-138

9 SMDI Planning & Preparation ...............................................9-139About SMDI integration with Sphericall Voice Mail .......................................9-139

SMDI and Sphericall Voice Mail ................................................................9-139


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SMDI Overview .............................................................................................9-139Manuals .....................................................................................................9-140

SMDI Integration Requirements....................................................................9-140SMDI Platform Hardware Considerations..................................................9-142

SMDI Operation ............................................................................................9-143Overview....................................................................................................9-143Call Processing..........................................................................................9-144

SMDI and System Planning ..........................................................................9-146Planning for Voice Ports ............................................................................9-146

Call Traffic Calculation: ..........................................................................9-147Integration Notes .......................................................................................9-149

10 SMDI Installation..................................................................10-153Before You Begin ........................................................................................10-153

This chapter at a glance.......................................................................10-153Preparing the Sphere system......................................................................10-154

Preparing the Sphere system ..................................................................10-154To set flow control settings ..................................................................10-154To enable the SMDI process ...............................................................10-156

Configuring a Sphericall Numbering Plan for Voice Mail ............................10-158To configure a voice mail extension.....................................................10-158To configure a voice mail station .........................................................10-159To assign an extension to a station .....................................................10-161To configure the voice mail line setting................................................10-161To re-configure the VM and AA hunt order ..........................................10-162To verify Monitor privileges for the SMDI instance ..............................10-163

Configuring a Media Server ........................................................................10-164To create a media server within a Sphere system...............................10-164

Export User Info from Sphericall .................................................................10-166To export to CSV .................................................................................10-166

Testing the Voice Messaging Platform........................................................10-166To test direct subscriber access of mailboxes .....................................10-167To test direct calls into voice mail by the operator ...............................10-167To test the setting and cancelling of MWI ............................................10-167To test forwarding to voice mail and to test call disconnect.................10-168To test voice messaging integration ....................................................10-168To test auto attendant integration ........................................................10-169

Restarts & Refreshes ..............................................................................10-169SMDI Call Records ..................................................................................10-169

11 SMDI Voice Mail Troubleshooting......................................11-173Troubleshooting SMDI Voice Messaging ................................................11-173

To troubleshoot a voice messaging platform .......................................11-173To troubleshoot COM port settings ......................................................11-173To verify the sending of information from the Sphericall Manager.......11-175To verify the sending of information from the voice mail server...........11-176

Troubleshooting SMDI Voice Messaging ................................................11-176To troubleshoot a voice messaging platform .......................................11-177To troubleshoot COM port settings ......................................................11-177


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To verify the sending of information from the Sphericall Manager.......11-179To verify the sending of information from the voice mail server...........11-179

Restarts and refreshes ............................................................................11-180Voice Mail Troubleshooting Steps ...........................................................11-180Common Voice Mail Issues .....................................................................11-181

Summary.....................................................................................................11-183

12 SNMP Integration.................................................................12-185Before you begin .........................................................................................12-185

In this chapter, you will learn how to ....................................................12-185Simple Network Management Protocol.......................................................12-185

Overview..................................................................................................12-185Requirements ..........................................................................................12-186SNMP Community Name ........................................................................12-186

To change the SNMP access write community name .........................12-186SNMP Manager and Agent Configuration ...............................................12-187

The Manager-to-Agent-to-Manager Monitoring Process .....................12-187The Agent-to-Manager Trap Function..................................................12-187

Sphere SNMP Traps ...................................................................................12-188Critical Trap Information ..........................................................................12-188

Sphere SNMP MIBs ....................................................................................12-189Overview..................................................................................................12-189installation................................................................................................12-189

To install Sphere SNMP MIBs on an SNMP manager.........................12-189Contents ..................................................................................................12-190

Sphere SNMP Object IDs ...........................................................................12-191

13 Microsoft Windows Installer ...............................................13-197Windows Installer And Sphericall Desktop Installation ...............................13-197

I.Group Policy ..........................................................................................13-197II. Installation components.......................................................................13-200III. Differences Between MSI and the Desktop Manager.........................13-200

To install the Sphericall Desktop Installation .......................................13-200Instructions for upgrading a user’s Sphericall Desktop........................13-201

14 AudioCodes..........................................................................14-203AudioCodes MP11X Access & Setup .........................................................14-203

Overview of Operation.............................................................................14-203Choose this method if the installer doesn’t have access to the DHCP lease list.........................................................................................................14-203Choose this method if the installer has access to the DHCP lease list......14-204To update and configure: independent of the method of access .........14-204Using the AudioCodes FXO MP-11X Quick Setup Screen..................14-205Using the Automatic Dialing Table.......................................................14-206

Advanced Configuration..............................................................................14-207CHANNELSELECTMODE* .....................................................................14-207


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ISPROXYUSED.......................................................................................14-207ProxyName..............................................................................................14-207PROXYIP.................................................................................................14-208SIPGATEWAYNAME ..............................................................................14-208ISREGISTERNEEDED............................................................................14-208REGISTRARNAME .................................................................................14-208REGISTRATIONTIME .............................................................................14-208uSERname ..............................................................................................14-209AuthenticationMODE ...............................................................................14-209CODERNAME .........................................................................................14-209ENABLECURRENTDISCONNECT .........................................................14-209Disconnectonbrokenconnection ..............................................................14-209Currentdisconnectduration ......................................................................14-210TimeToSampleAnalogLineVoltage ..........................................................14-210enablecallerID..........................................................................................14-210pSTNPREFIX ..........................................................................................14-210TRUNKGROUP_1 ...................................................................................14-210enablecallerid_<Port>..............................................................................14-210targetofchannel<Port> .............................................................................14-211ISTWOSTSTAGEDIAL ............................................................................14-211ISWAITFORDIALTONe ...........................................................................14-211DTMFTransportType ...............................................................................14-211MFTransportType ....................................................................................14-211

AudioCodes MP104 Access & Setup..........................................................14-212Planning ......................................................................................................14-212

Overview of Operation.............................................................................14-212Choose this method if the installer doesn’t have access to the DHCP lease list.........................................................................................................14-212Choose this method if the installer has access to the DHCP lease list......14-213To update and configure: independent of the method of access .........14-213Using the AudioCodes FXO MP104 Quick Setup Screen ...................14-214Fax Signaling Method Using T.38........................................................14-215Using the Automatic Dialing Table.......................................................14-215

Advanced Configuration..............................................................................14-216CHANNELSELECTMODE* .....................................................................14-216ISPROXYUSED.......................................................................................14-216ProxyName..............................................................................................14-216PROXYIP.................................................................................................14-217SIPGATEWAYNAME ..............................................................................14-217ISREGISTERNEEDED............................................................................14-217REGISTRARNAME .................................................................................14-217REGISTRATIONTIME .............................................................................14-218uSERname ..............................................................................................14-218AuthenticationMODE ...............................................................................14-218CODERNAME .........................................................................................14-218ENABLECURRENTDISCONNECT .........................................................14-218Currentdisconnectduration ......................................................................14-218TimeToSampleAnalogLineVoltage ..........................................................14-219enablecallerID..........................................................................................14-219pSTNPREFIX ..........................................................................................14-219TRUNKGROUP_1 ...................................................................................14-219


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enablecallerid_<Port>..............................................................................14-219targetofchannel<Port> .............................................................................14-220ISTWOSTSTAGEDIAL ............................................................................14-220ISWAITFORDIALTONe ...........................................................................14-220DTMFTransportType ...............................................................................14-220MFTransportType ....................................................................................14-220

15 SpectraLink Wireless Telephones......................................15-223Planning ......................................................................................................15-223Required Materials ......................................................................................15-223

Spectralink Overview...............................................................................15-223

16 Visual Basic Type Libraries ................................................16-227Introduction..............................................................................................16-227Constants ................................................................................................16-227Sample Applications for Sphericall Type Libraries ..................................16-228

Getting Started ............................................................................................16-228To install and verify the Sphericall COM API .......................................16-228

17 SIP Trunking.........................................................................17-231Overview of SIP ..........................................................................................17-231SIP Terminal Location in Sphericall ............................................................17-232NEC Sphere SIP Trunking ..........................................................................17-233

Before You Begin Any SIP Integration ....................................................17-234To configure SIP User Agents .............................................................17-234

User Agent Maintenance.............................................................................17-237To remove a User Agent Parameter ....................................................17-238

SIP Trunking to SIP Service Provider .........................................................17-239To plan to install SIP trunking ..............................................................17-240Before you begin the trunk configuration .............................................17-241To configure a SIP trunk ......................................................................17-241To configure Service Provider information for the softtrunk.................17-245To configure SIP trunk properties ........................................................17-246Vendor Specific SIP Trunking Configuration........................................17-252

SIP Trunking Tie Line..................................................................................17-253To install SIP trunking or tie lines.........................................................17-253To configure SIP User Agents .............................................................17-254To configure SIP for tie line..................................................................17-255To configure SIP trunk properties ........................................................17-260To verify the softtrunk is registered on the far end of tie line ...............17-263

SIP tie line to Third-Party App.....................................................................17-266To install SIP trunking or tie lines.........................................................17-266To configure SIP ..................................................................................17-267To configure SIP trunk general properties ...........................................17-271

Troubleshooting SIP Connections...............................................................17-276


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18 Quintum VoIP Gateways .....................................................18-279Overview .....................................................................................................18-279Operation ....................................................................................................18-281Planning ......................................................................................................18-282Preparing.....................................................................................................18-282

To connect analog phone devices .......................................................18-283To connect SIP phone devices ............................................................18-283

Installing ......................................................................................................18-283To install with Sphere system ..............................................................18-283IP Address Configuration .....................................................................18-283Dial Plan Configuration ........................................................................18-283Phone Port Configuration.....................................................................18-284Multi Path Configuration.......................................................................18-284Line Port Configuration ........................................................................18-284Survivability Configuration (only available for configuration on survivable units) ....................................................................................................18-284VoIP Routing Configuration .................................................................18-285Channel Configuration .........................................................................18-285Configuration Summary .......................................................................18-285The following items must be configured for SIP phone Reg: ...............18-285To configure SIP Signaling Groups......................................................18-286To create a second SIP signaling group and move the FXS ports: .....18-286Under: Circuit Configuration/Trunk Routing Configuration/Hopoff Number Directory...............................................................................................18-286To set CAS Signaling Group-line .........................................................18-287To disable Silence Suppression on the Quintum: ................................18-287To enable Caller ID name display for calls inbound from the PSTN:...18-288To adjust the volume of PSTN calls:....................................................18-288To enable Hook Flash and Caller ID on the station ports: ...................18-288To set the Quintum to pass Caller ID name to the Quintum FXS ports .....18-288To adjust the system for gateways that do not support T.38 Fax Relay ....18-289The following items may optionally be configured: ..............................18-289The following setting must be verified in the Sphericall Administrator application............................................................................................18-289The Trunk Capacity must be set in the Sphericall Administrator application ..18-289To verify the latest firmware version of Quintum Gateway ..................18-290

Configuration at the Sphericall Manager.....................................................18-290Quintum feature Codes for Analog Phones ................................................18-291

To configure codes ..............................................................................18-291Configuring Inbound routing........................................................................18-292

To configure inbound routing of trunk or PSTN calls ...........................18-292To create a local registration through the DN Channel Map FXS port:18-292To create a local registration through the DN Channel Map FXO port: .....18-293To create a static route, add the following two lines to your existing var_config.cfg file (if any): ....................................................................18-293To create a var_config.cfg file..............................................................18-293To load the file onto the Quintum gateway ..........................................18-293To enable authentication for the FXO ports in the Quintum gateway: .18-294


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Configuring Survivable Outbound Routing (Survivable models) .................18-294To enable survivable outbound routing................................................18-294To load the file onto the Quintum gateway ..........................................18-294

Understanding the Environment..................................................................18-295Testing ........................................................................................................18-296

A FTP, SNTP & DHCP Notes.................................................... A-297FTP Server Configuration ............................................................................ A-297

Configure FTP Server............................................................................... A-297FTP server—to create login and password for IP phones on FTP server .. A-297To configure the FTP service on a Microsoft Windows 2003 Server .... A-298To verify FTProot directory security ...................................................... A-300To verify FTP service functionality ........................................................ A-300

SNTP Server Configuration ...................................................................... A-300DHCP Configuration..................................................................................... A-301

Configure DHCP....................................................................................... A-301To install DHCP service on a Microsoft 2003 Server............................ A-301To create a scope with a range of IP addresses in Windows 2003: ..... A-301To configure the Microsoft Windows 2003 Server DHCP scope for IP Phones only ....................................................................................................... A-304

Time Offset Values ................................................................................... A-307

Index.............................................................................................I-1Document Feedback ...........................................................................................-1

Document Information .....................................................................................-1


C O N T E N TS


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .INTEGRATE PARTNER TECHNOLOGIES 1

The Sphericall telecommunications software suite delivers integrated communications to the enterprise desktop. Sphericall provides a foundation of traditional PBX capabilities combined with desktop video, text messaging, on-line phone book, status presence controls, on-demand conference bridging, IP-phone synchronization, and integration with enterprise tools such as Exchange & Outlook.

• MGCP, SIP Standards-based VOIP Protocols• SIMPLE messaging protocol - Windows Messenger• Supports range of industry leading IP Telephones• Multiple enterprise and SOHO gateway options• General SIP Configuration Notes

I N T H I S M A N U A LIn this manual, you will find information on how to integrate several third-party products into the Sphere system. Our suite of Alliance products prove that the Sphere system is more than adequate for meeting your telecommunications needs over the network.All end points on the network can take advantage of the Sphericall Desktop for multiple forms of communication including voice, video, and text messaging, in addition to Exchange and Outlook.Alliance products also include SNMP and SMDI integrations, Music-on-Hold and Paging integrations, and the Windows Messenger phone client.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D O C U M E N T I N D E X

For complete instructions on installing the Sphere system, refer to our documentation. The documentation is listed here in the order in which a system is installed.

Integrate Partner Technologies 1-1

I N T E G R A T E P A R T N E R TE C H N O L O G I E SDocument Index

DoOrde

Reres

des

t

;

tion s;

s;

p &

ce tion

e

ll

port

ces, s

S

S

R

cument r of Use: Sphere Document Name: Document Description

Adminferences

Release Notes & Upgrade Procedures

Sphere System Requirements

Sphere Star Codes

Release Notes for current product & upgrade procedu

System Requirements, minimums, limits, requisites

Telephone Set, Administrative and Diagnostic Star Co

Book 1 Plan & Prepare the Sphere System Project Planning & System Requirements; Site Walk Workbook; Customer Implementation; Import & ExporUtility; Stage & Cut Information; Logins & Passwords Required for Installation; Planning for SIP InstallationsNetwork Planning Information; Suppliers

Book 2 Install & Configure the Sphere System Full information and process for a new installation—everything from software installation to detailed integraincluding: Install Sphericall Software; System PropertieLANs; Number Plan; Telephony Area Settings; StationTemplate Settings; Station Port Settings; PSTN SettingMapping Lists; Secondary Managers; User Rights & Permissions for Sphericall Desktop; Sphericall DesktoSoftphone; ARS; Media Server Options; Optional Connections; Call Recording; Call Admission Control; Optional Desktop Connections

Book 3 Install Sphericall Voice Mail Planning, installing and troubleshooting Sphericall VoiMail (Exchange 2000/2003); Auto Attendant ConfiguraTroubleshooting; Theory of Operation; Sphericall VoicMail User Quick Reference

Book 4 Integrate Partner Technologies Content may vary depending on third-party partners. Athird-party product integrations are documented for reference

Book 5 Manage, Monitor & Support Sphericall Moves, Adds & Changes to the System; Daily Management; Sphere Services & Utilities; Supporting Sphericall Desktop; Reports, Statistics and Tools; SupResources; User, System & Hardware Changes; Emergency Backup Plans; Additional Support ResourQuick Reference Charts, Settings, Microsoft Resource

Book 6 Emergency Service Installations Emergency Service Planning, Installation and setup; MLPP or CallNOW implementation

phericallDesktop

Sphericall Desktop Users Manual Using the telephoneSphericall Desktop SoftwareSphericall Desktop Options

phericallQuick

eferenceGuides

• Sphericall Desktop Quick Reference Guide• Sphericall Voice Mail Quick Reference Guide English• Sphericall Voice Mail Quick Reference Guide Spanish (Mex-

ico)

Guides to aid in use of Sphericall

1-2 Integrate Partner Technologies

. . .

. .I N T E G R A T E P A R T N E R TE C H N O L O G I E SDocument Index

Th

s;

s;

s;

s;

d

D

Seers, r

DoOrde

ird PartyResourceManuals

Analog Phones• Aastra 390 & 480 Quick Reference GuideMGCP Phones• Aastra 480i phone for Sphericall User Guide• SoundPoint IP30x Quick Reference Guide• SoundPoint IP50x Quick Reference Guide• SoundPoint IP60x Quick Reference GuideSIP Phones• Aastra 480i phone for Sphericall User Guide• Aastra 9112i phone for Sphericall Quick Reference• Aastra 9133i phone for Sphericall Quick Reference• Grandstream GXP2000 Quick Reference• Grandstream GXV3000 Quick Reference• Polycom SIP Phone Quick Reference—generic all SIP phones• SoundStation IP4000 Quick Reference Guide• UT StarCom F1000/3000G Quick Reference

Reference Guides related to Sphericall integration withpartner products

Hardware COHub Manual Install & Configure; Troubleshooting; Display MessageSpecifications

Hardware PhoneHub Manual Install & Configure; Troubleshooting; Display MessageSpecifications; Emergency Failover

Hardware BranchHub Manual Install & Configure; Troubleshooting; Display MessageSpecifications; Emergency Failover

Hardware MeetingHub Manual Install & Configure; Troubleshooting; Display MessageSpecifications

Hardware MG CLI Reference Manual A complete reference to the Media Gateway CommanLine Interface; SNMP traps

Help • Admin Help• Phone Help• Visual Basic ComAPI Help

Searchable help file with Admin Help onlySearchable help file of Desktop Help onlyHelp file for generating VB com objects

Featureocument

• NEC Sphere document(s) with all the features listed TBD 2008

archableIndex

• Document_INDEX.pdfNote: must be named this file name exactly

• located in CD:\\...Documents\Document_INDEX.pdf• Must run the index on a local PC to all document fold

then place it on the Installs\Documents\version folde• Source files are required in that folder

cument r of Use: Sphere Document Name: Document Description


I N T E G R A T E P A R T N E R TE C H N O L O G I E SDocument Index


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .MGCP IP PHONES 2

This chapter supports the following MGCP IP phones:I. Polycom SoundPoint IP phoneII. Aastra 480i IP phone

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I . P O L Y C O M S O U N D P O I N T I P P H O N E

The SoundPoint IP phone is a standards-based internet protocol phone that delivers rich applications to the desktop as part of the Sphere system. The phone is from Polycom, a leader in advanced telephony. Platform-independent design enables a seamless interface to all leading protocols and platforms, making the SoundPoint the perfect complement to the Sphere system. The SoundPoint is a full-featured, intelligent endpoint on the converged network. IP phones connect to your Ethernet network. A single Cat 5 connection to the desk serves both the PC and phone.The SoundPoint provides excellent voice quality with the latest microprocessor technology and support for quality-of-service processing. Programmable feature keys and context-sensitive keys let you access Sphericall advanced functions with a simple touch. It also features speakerphone functionality and an extra large graphic display for time, date, caller ID data and future data streams.

FailoverThe SoundPoint IP phone supports failover. If a Primary Sphericall Manager becomes unavailable, all IP phones automatically fail over to the Secondary Sphericall Manager (if one exists within that Sphere system). When the Primary Sphericall Manager becomes available, the phones will automatically move back.


M G C P I P P H O N E SPlanning

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

The following information needs to be considered before installing an IP phone:Each IP phone needs three addresses in order to begin its configuration:

• IP address• FTP server address• SNTP server address

Note: Microsoft Windows Server has built-in DHCP and FTP servers which ease the installation process. Sphere recommends that the FTP server is not housed on the Primary Sphericall Manager.

Make sure an FTP server is set up on your network. For notes related to configuration of FTP, SNTP or DHCP see Appendix A of this manual, or refer to Microsoft documentation.

Note: Every system requires adherence to the Sphere System Requirements.

Final P lanning1 Review Sphere System Requirements for domain password issues.2 Verify that the rest of the system is operational prior to starting voice mail services.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P R E P A R I N G

FTP Server , DHCP Server , SNTP Services1 Install Static or Dynamic IP Address components to the Sphere system. For more

information, review Appendix A of this manual.2 Setup SNTP services for the Sphere system.3 Verify FTP server on the network.4 Verify permissions required.

Table 2.1 Third-party permissions required

Account Logins/Group

Needed:Minimum Permissions Required: Account Creation:

Group Policy Support for Microsoft Windows Installer and Group Policy Snap-in for installing, updating or uninstalling Sphericall Desktop or Sphericall Desktop Softphone clients.See Microsoft product documentation for full permission information.

Domain Administrator permissions required for Group Policy Administration


. . .

. .M G C P I P P H O N E SPreparing

FTP ServerFTProot

Directory

Organizations using a separate FTP server must do the following:• Copy the FTProot directory from the Sphericall Manager to the FTP Server.• Grant FULL CONTROL Security access to the FTProot directory, based on the

type of phone(s) you are using (i.e. if using SoundPoint IP phones, you must cre-ate that account; if using Aastra 480i phones, you must create that account; if using both, you must create both accounts).

Required Folder(s):If you are installing Windows FTP server, the ftproot folder will be located by default at: c:\inetpub\ftproot. This default setting needs to be changed as follows:• For systems with the FTP Server on the Primary Sphericall Manager, the following

folder is required for the location of IP phone resource files:<drive>:\\Program Files\Sphere\ftproot

• For systems with the FTP Server on any other server (third-party or Secondary Sphericall Manager), the following folder is required for the location of IP phone resource files:<drive>:\\ftproot\

See below based on manufacturer

Aastra 480i:Used for IP

phones

Aastra480i

See ActiveDirectory User

Accounts NotesBelow

NEW AastraSIP 9112i &

9133i IP Phone

Aastra 480i FTP login: Login: Sayson; Password: Aastra480iEither of the following two:1. Must be a Domain User if FTP server is also the Domain/Active Directory server.2. Local User if on the FTP server and FTP server is not the Domain/Active Direc-

tory server. Create a user account with username Sayson; create password: Aastra480i

Default Administrative Passcode on 480i phone: 22222Configurable Administrative Passcode on 480i phone only (not web interface):1) Options2) Option #9 - MGCP Settings3) Option #9 - Admin PasswordAdministrative Passcode & web interface login on 480i phones: Login: admin Password: 22222

Default Administrative Passcode on 9112i and 9133i phone: 22222

Administrative Passcode & web interface login on 9112i and 9133i phones: Login: admin Password: 22222

Manually on a Domain or onthe FTP server

Entered manually on the phone

Entered manually on the web interface

Entered manually on the phone Entered manually on the web interface

GrandstreamBT series&

GXP2000

Default administrative password: admin Entered manually on the web interface

Polycom: Usedfor IP phonesPlcmSpIp

(case sensitive)


Accounts NotesBelow

Login & Password: PlcmSpIp (case sensitive)Either of the following two:1. PlcmSpIp Must be a Domain User if FTP server is also the Domain/Active

Directory server.2. PlcmSpIp Local User if on the FTP server and FTP server is not the

Domain/Active Directory. Create a user account with username PlcmSpIp; create password PlcmSpIp

Administrative Passcode on the Polycom SoundPoint IP phone and SoundStation 4000 IP Conference Phone: 456More detailed instructions for this account are contained in Book 4.






M G C P I P P H O N E SPreparing

Ac••

)

•

-

3 Complete the following steps to:

C R E A T E A L O C A L U S E R A C C O U N T ( W I T H P A S S W O R D ) F O R T H E I P P H O N E S O N T H E F T P S E R V E R

FTP server—to create login and password for IP phones on FTP server1 Create a local user account on the FTP server with username PlcmSpIp.2 Create password PlcmSpIp.3 Deselect “User must change password at next logon.”4 Select “User cannot change password.”5 Select “Password never expires.”

FTP—to change the IP Phone User AccountIf you wish to change the login name and password used by the IP phone(s), you must change:• The user account defined on the local machine • You must also change the login and password in the Sphericall Administrator

application so that it can use this new account to communicate with the FTP service.

• Finally, you must also change the default login and password on the Polycom SoundPoint phoneset.

For systems using Windows 2003 AND Password Complexity, you may need to standardize on a password that meets the Windows 2003 password complexity criteria as stated by Microsoft.

Quintum SSG Username: adminPassword: admin

UTStarcomF1000/G

DNS Domain name requiredMore detailed instructions for this account are contained in Book 4.User Name: userPassword: 888888

Manually on the web interface

tive Directory User Accounts: Sphere-MS & Sphere-DB Accounts MUST be set for: “User cannot change password” and “Password never expires”.Microsoft Password Complexity (affecting IP phones, Sphere-DB and Sphere-MS passwords):New Windows 2003 Domain/Active Directory servers (versus Windows 2000 Domain/Active Directory that has been upgraded to W2003by default enables password complexity. Default passwords for IP phones do not satisfy the complexity requirement. It is recommendedthat password complexity be disabled at the domain level on new Windows 2003 Domain/Active Directory.Options: An alternative solution to disabling password complexity on the Windows 2003 Domain/Active Directory would be to redefine the pass-word on all IP phones. Redefining the password on all IP phones also requires that you redefine the password on the Sphericall Administrator-> System - View Properties -> IP Phones tab.




. . .


Refer to Appendix B for more information on FTP, SNTP and DHCP.

F T P P R E P A R A T I O N N O T E SIf the IP phone has been used previously in the network (i.e. has communicated with an FTP server), FLASH-stored XML files exist on the IP phone. XML files contain information about the last FTP server located by this IP phone. If the IP phone cannot locate the “new” FTP server, the IP phone will use the information stored locally in the XML files.When the IP phone initializes, it always attempts to locate the most current information. If it can not find the most current information in a new XML file, it will use the information in the locally-stored XML file.

• If the new FTP server address was entered at the IP phone, the IP phone will re-initialize and download a generic XML file containing the information necessary for normal runtime operations.

• If the IP phone is using locally stored information, the phone will be active within the Sphere system and attempt to communicate with the designated primary Sphericall Manager. At this point, the installer will not know whether or not the IP phone is using new or old FTP server information.

• The IP phone appears in the Stations tab of the Sphericall Administration application, ready for extension and station configuration. If changes are made, new information in the XML file may never reach the IP phone. The IP phone is constantly looking to the old primary MGC as opposed to the MGC responsible, now, for its configuration and runtime.

V L A N U S E F O R I P P H O N EVLAN setup on an IP phone must be performed at each phone by the setup menu.



P O L Y C O M P H O N E K E Y C A P S

Figure 2.1 IP301 - Two Line Phone

Figure 2.2 IP501- Three Line Phone

Figure 2.3 IP600 - Six Line Phone

Redial

Directories

Menu

Directory

Services

Call Lists

Transfer

Redial

Conference

Call 2

Call 1

Call 3

Menu

Messages

DND

Transfer

Redial

Conference

Directories

Do Not Disturb

Messages

Menu

Hold

Services


. . .

. .M G C P I P P H O N E SInstalling

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I N S T A L L I N G

S Y S T E M P R O P E R T I E SBefore configuring your organization’s IP phones, you need to define certain characteristics of the Sphere system. These settings will dictate functionality of the connected MGs and IP phones.

Spher ical l Manager—to conf igure System proper t iesFrom the Sphericall Administrator application window:

1 Click the General tab.

In the System details (highest level on the tree):2 Right-click System to View Properties.

From the System Properties window:3 Click the General tab.

In the FTP Server area:4 Type the name or address (or Browse to select) of the FTP server used to store the XML

configuration files for your organization’s IP phones in the Server Name field.

IP phones on a Sphere system must download XML configuration files from an FTP server on the network. These files are installed into the FTP root directory upon Sphere system installation and are responsible for setting the functional parameters of the individual IP phones.

5 Click the IP Phones tab.6 Verify the login name to the FTP server in the Polycom Login Name field.7 Verify the password for access to the FTP server in the Polycom Password field.8 Click Apply.9 Click OK.

M G C - T O - M G C P P H O N E C O N N E C T I O N C O N T R O LBefore defining Media Gateway properties, it is necessary to configure the connections between MGCP phones and the Sphericall Manager.With the Polycom MGCP phones, this step is required. With the Sayson MGCP phones, this step is optional since the MGCP phone automatically discovers the MGC and is assigned its MGC by load.

Unique Load Balancing for Aastra MGCP PhoneThe following procedure is optional for Sayson 480i phone users. Administrators may take the extra time to assign unique or specific MGCs as Primary, Secondary or Tertiary to the 480i phone. The 480i phone is designed to automatically find the MGC that has the appropriate load for that phone. Administrators may want to override the systems auto-balancing of the 480i phone additions based on the location of the phone in relation to its MGC (i.e. via a WAN link), or perhaps the location of a Secondary Sphericall Manager for manager failover.


M G C P I P P H O N E SInstalling

Spher ical l Manager—to conf igure MGC-to-MGCP phone connect ions on a Sphere systemFrom the Sphericall Administrator:

1 Click the General tab.2 Expand Media Gateway Controllers from the tree list.3 Highlight the Media Gateway Controller you wish to configure.4 Right-click to View Properties.5 Click the MGCP Phones tab.

From the Properties for Media Gateway Controllers window:In the Set As Default area:

6 Select the Primary MGC check box if this Sphericall Manager is to be considered the default Primary Sphericall Manager that THIS MGCP phone will use to obtain its configuration information upon initialization.

Note: Settings configured for individual MGCP phones will override this default setting.

7 Select the Secondary MGC check box if this Sphericall Manager is to be considered the default Secondary Sphericall Manager THIS MGCP phone will use to obtain its configuration information upon initialization if the Primary Sphericall Manager is unavailable.


8 Click ADD under each category to add this MGCP phone to its Primary, Secondary or Tertiary MGC.

9 Click Apply.

You will receive a message asking if you want to restart the MGCP phones.Select the appropriate option based on your system. The MGCP phone will need to be restarted to apply all these settings.

10 Click OK.

You may also choose the MGC to which each MGCP phone will be assigned via their Station Properites.

SoundPoint MGCP Phone—to assemble and power the phone1 Please use the quick start guide provided with your SoundPoint phone for phone

assembly and cable connections.2 See the above section for the keycap layout of the phones.

M G C P P H O N E I N S T A L L A T I O N W I T H D Y N A M I C I P A D D R E S S I N GFor systems requiring dynamic IP addresses for their MGCP phones, this procedure is a general idea of how to accomplish dynamic IP address assignment.

To insta l l the MGCP phone wi th dynamic IP addressing you wi l l need to:1 Turn on the phone by connecting power to the phone.


. . .


As the phone initializes, a “Welcome” screen is displayed with a count-down timer and three softkeys: “START,” SETUP,” and “ABOUT.”

2 Press the SETUP softkey.3 A menu is displayed on the phone screen. Use the arrow keys on the upper right of the

phone to scroll through the menu options.

You MUST press the EDIT softkey to make configuration changes and the SAVE softkey to accept changes to that option.

4 Configure the DHCP server with the appropriate scope and information to pass to the various endpoints on the network:

• IP address range• Default gateway and subnet mask addresses• FTP server address• SNTP server address• SNTP time offset (GMT offset)

When the MGCP phone initializes, it broadcasts to locate a DHCP server. The DHCP server will pass the IP address, FTP server address, and SNTP server address to the phone based upon the configuration of the DHCP scope.The MGCP phone re-initializes and downloads a generic XML file. This file contains the information necessary for normal runtime operations including the location of the MGC(s) to which this phone will look for address identification. After the MGCP phone re-initializes, the MGCP phone appears in the Sphericall Administration application.

M G C P P H O N E I N S T A L L A T I O N W I T H S T A T I C I P A D D R E S S I N GFor systems requiring static IP addresses for their MGCP phones, this procedure is a general idea of how to accomplish static IP address assignment.

To insta l l the SoundPoint MGCP phone wi th a stat ic IP address you wi l l need to:1 Turn on the phone by connecting power to the phone.

As the phone initializes, a “Welcome” screen is displayed with a count-down timer and three softkeys: “START,” SETUP,” and “ABOUT.”

2 Press the SETUP softkey.3 A menu is displayed on the phone screen. Use the arrow keys on the upper right of the

phone to scroll through the menu options.

Note: You MUST press the EDIT softkey to make configuration changes and the SAVE softkey to accept changes to that option.

4 Manually enter the setup information at the MGCP phone.

• IP address of the MGCP phone• Default gateway and subnet mask addresses• FTP server address• SNTP server address• SNTP time offset (GMT offset)


M G C P I P P H O N E SMGCP Phone Installation Test

The MGCP phone re-initializes and downloads a generic XML file from the FTP server. This file contains the information necessary for normal runtime operations including the location of the MGC(s) to which this phone will look for address identification.After the MGCP phone re-initializes, the MGCP phone appears in the Sphericall Administration application.

Spher ical l Manager—to complete conf igurat ion1 Return to the Properties for Station and Properties for Extension windows to configure

settings including

• Forwarding conditions• Emergency group• Telephony area association (for stations) and hunt order settings (for

extensions).Refer to Book 2: Install & Configure Chapter 6, Station Local Settings.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . M G C P P H O N E I N S T A L L A T I O N T E S T

To test for successful extension and stat ion conf igurat ion1 Make calls between MGCP phones.2 Make calls between MGCP phones and other types of phones or external calls.3 Test MGCP phone keys and softkeys.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I P P H O N E T R O U B L E S H O O T I N G

Most troubleshooting issues may arise during the initial deployment of SoundPoint IP phones versus when the system is operational. It is essential to remember that three key elements are necessary to deploy the SoundPoint IP phones:• IP address

• Either assign a static IP address or have an operational DHCP server with a properly defined DHCP scope

• FTP server address• You MUST have an operational and properly configured FTP server

• SNTP server address• The phone MUST get its date and time from an operational FTP server


. . .

. .M G C P I P P H O N E SIP Phone Troubleshooting

N O R M A L B O O T S C R E E N SDuring the boot sequence of the IP phone, the following normal messages may be shown.

Table 2.2 IP Boot Screen Messages

I N S T A L L A T I O N I S S U E SDuring the boot sequence of the IP phone, the following error messages may be shown.

Table 2.3 Boot Sequence Errors

Screen Message Readout

1. Rebooting...Please Wait.

2. WelcomeInitializing Phone

3. [Seconds count down] to Auto Reboot

4. Updating Configuration...

5. Loading Application...

6. Running:App=gmx.id

7. WelcomeProcessing ConfigurationThis may take a few seconds.[IP address may appear]

8. Initialing...Ready[soft keys will now appear]

Error Message Description Resolution

Failed to get boot parameters via DHCP.Tell your SysAdmin, 500 seconds until autoboot.

This error message indicates that DHCP is enabled on the phone, and the DCHP server could not be reached. The IP phone is unable to boot without the DHCP parameters.

1. Verify your DHCP server is opera-tional and connected to the network.

2. If you are using static IP addressing, disable DHCP Client on the phone and enter the appropriate IP address and Default Gateway.

Error loading <MAC>.cfgNote: After seeing this message the phone will reboot after a 5 second timeout.

There are two scenarios when you might see this error:1. The FTP server was found and logon

successful. The <MAC>.cfg was not found on server. The 00000000000.cfg file was not found on server. The <MAC>.cfg was not found in FLASH.

2. The FTP server was NOT found or login was unsuccessful, and the <MAC>.cfg was not found in flash.

Scenario One1. Verify that the <MAC>.cfg or

00000000000.cfg file is in the root directory of the FTP user account.

Scenario Two1. Verify the FTP settings in the DHCP

scope are correct.2. Verify that the FTP server is opera-

tional and connected to the network.3. Verify the FTP user and password

match on both the phone and the FTP server.


M G C P I P P H O N E SIP Phone Troubleshooting

Note: PING is a useful tool when trying to determine connectivity of DHCP, FTP, or SNTP server using the PC at the location of the IP phone.

R E S T A R T I N G T H E I P P H O N EOccasionally, it may be necessary to restart the IP phone to clear a problem or reinitialize it.

SoundPoint phone—to restar t the IP phone• IP501 & IP 600 Press the four following keys simultaneously until the IP phone restarts:

• Both Audio Path Volume keys (+ -)• Hold key• Voice Mail key simultaneously until the IP phone restarts.

• IP301 Press the four following keys simultaneously until the IP phone restarts:

• Both Audio Path Volume keys (+ -)• Hold key• Directories key simultaneously until the IP phone restarts.

Spher ical l Manager—to restar t the IP phone remotelyFrom the Stations tab on the Sphericall Administration application:

1 Expand the Stations tree in this window view.2 Click to highlight the IP phone unit.3 Right-click.4 Select Restart.

Spher ical l Manager—to sync IP Phone Fi lesIf the occasion arises when the PBX.mdb and the MGCP or SIP phone configuration files are not synchronized, the Sphericall Administrator application has a feature that allows the PBX.mdb files to override the configuration files.From the Sphericall Administrator application:

1 Click Tools\Configure MGCP Phones\Polycom\Sync IP Phone Files.2 Confirm that you want to complete this task by clicking YES.

Spher ical l Manager—to Revert to Defaul t Conf igurat ion1 Click Tools\Configure IP Phones\Polycom\Revert to Default Configuration.

Failover/LAN port down! This error message indicates when there has been a loss of connection to the IP switch or the switch has lost its network connection.

1. Verify your IP cable connection to the IP switch.

2. Verify the switch connection to the rest of the LAN.

Error Message Description Resolution


. . .

. .M G C P I P P H O N E SIP Phone Troubleshooting

2 Confirm that you want to complete this task by clicking YES.

Spher ical l Manager—to v iew Defaul t Parameters1 Click Tools\Configure IP Phones\Polycom\Default Parameters.2 Click OK when finished.

I P P H O N E C O N F I G U R A T I O N F I L E U P G R A D E SIf there is an update available for the IP phone, the Sphericall Administrator application will notify of such an update. Here is how the IP phone upgrade works:• During a Sphere system install or upgrade, the script loads the IP phone files to:

...Program files\Sphere\ftproot\PlcmSpIp\Update directory. If this FTP directory is not being used, the Sphere system administrator will have to move these files to the appropriate directory on the FTP server.

• Upon starting the Sphericall Administrator application, if the FTP server is running and defined in the PBX Properties window, the Sphericall Administrator application checks for files in the Program Files\Sphere\ftproot\PlcmSpIp\Update directory. If these files exist, the Sphere system administrator is prompted to upgrade the IP phone configuration files.

Figure 2.4 Admin window

• The Sphericall Administrator application will backup the current PlcmSpIp files in the PlcmSplp\backup directory.

• The Sphericall Administrator application will check to see if the SPHERICALL_VERSION tags within the Sphericall.cfg and 000000000000.cfg files are newer than the current file tags. If so, the PlcmSpIp\Update\000000000000.cfg file is copied into PlcmSpIp\000000000000.cfg. The PlcmSpIp\Update\Sphericall.cfg is copied into PlcmSpIp\Sphericall.cfg. It is then updated (information gathered from the database) with the Sphere system’s current default Primary and Secondary Sphericall Manager(s).

• If 000000000000.cfg was changed, the MAC-specific.cfg (for example, 0004f20011110.cfg) files are updated.

• All other files remaining in the PlcmSpIp\Update directory are copied into the PlcmSpIp directory unchanged.

• The update directory contents are deleted.


M G C P I P P H O N E SIP Phone Troubleshooting

Figure 2.5 Admin window

C O N V E R T P O L Y C O M M G C P T O S I P C O M M A N DA new command,"Convert Polycom MGCP to SIP," is part of the toolbar on the Stations page of the Sphericall Administrator application. This command aids in converting existing Polycom MGCP phones to Polycom SIP phones. To activate the toolbar button, the user must select one or more Polycom MGCP phones from the Station page. The user can optionally select a Polycom SIP phone to be used as a template during the conversion process. The non-phone specific Polycom template parameters will be copied to the converted phone (i.e. the AoR, line label, phone-managed overrides and custom file parameters are not copied).

Note: The SIP2.2.0 software release will no longer include new feature support for the IP 300 and IP 500 products. These products were discontinued in May 2006 (Refer to Product Bulletin Number 532.PB available from The Polycom Resource Centre (http://extranet.polycom.com). They were replaced with the SoundPoint IP 301 and SoundPoint IP 501 products which have additional internal memory.

Note: Installations that have a mixture of IP 300 and/or IP500 phones deployed along with other models will require changes to the phone configuration files to continue to support IP 300 and IP 500s when software releases SIP 2.2 and newer are deployed. Please refer to Polycom knowledge base articles for further and updated information.


. . .

. .M G C P I P P H O N E SIP Phone Upgrades

Figure 2.6 Polcyom MGC to SIP Command

I P P H O N E F A I L O V E RIP phones within a Sphere system can be configured, via the Sphericall Administration application, to fail over to the appropriate Sphericall Manager (from their Primary Sphericall Manager). If a failover occurs, the IP phone tries to register with the Primary Sphericall Manager. When the phone discovers that the Primary Sphericall Manager is not available, the phone fails over to the Secondary Sphericall Manager.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I P P H O N E U P G R A D E S

All upgrades related to IP phones generate from Sphericall software upgrades. If there is a full system upgrade, IP phones will be upgraded as a part of this process. Refer to the Sphere System Release Notes for full upgrade procedures.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . U S I N G

The Quick Reference Guide for this version of IP phone is located in the Appendix of this manual.


M G C P I P P H O N E SII. Aastra 480i IP Phone

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I I . A A S T R A 4 8 0 I I P P H O N E

Sayson and Aastra Technologies Ltd., have developed a versatile, feature-rich IP screenphone that supports current and emerging industry standards. The 480i has been designed to work with the Sphere system. Sayson’s large, backlit display, with 5 programmable "Softkeys," allow for Sphericall and Sayson technology integration. A 10/100 Ethernet switch eliminates the need for additional wiring to the desktop.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

The following information needs to be considered before installing an Aastra 480i phone:Each IP phone needs three addresses in order to begin its configuration:

• IP address• FTP server address• SNTP server address

Note: Microsoft Windows Server has built-in DHCP and FTP servers which ease the installation process. Sphere recommends that the FTP server is not housed on the Primary Sphericall Manager.

Make sure an FTP server is set up on your network. For notes related to configuration of FTP, SNTP or DHCP see Appendix A of this manual, or refer to Microsoft documentation.

Note: Every system requires adherence to the Sphere System Requirements.

When an IP phone initializes, DHCP is enabled by default. The DHCP server passes information to the IP phone so that it can configure itself for subsequent Sphericall address assignment and normal runtime operations.

• If you are planning on using Dynamic IP addresses, make sure a DHCP server is set up on your network.


. . .


• If you are not planning on using a Dynamic IP address, refer to the section Configuring the 480i IP Phone on how to set up an IP Address manually on each phone.

• Caution: If you choose to use static IP addresses with your 480i implementation, you will not have designed your system for failover capability. Failover requires the use of Dynamic IP Addresses.

Final P lanning1 Review Sphere System Requirements for domain password issues.2 Verify that the rest of the system is operational prior to starting voice mail services.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P R E P A R I N G

Veri fy permissions requiredTable 2.4 Third-party permissions required



Group Policy Support for Microsoft Windows Installer and Group Policy Snap-in for installing, updating or uninstalling Sphericall Desktop or Sphericall Desktop Softphone clients.See Microsoft product documentation for full permission information.

Domain Administrator permissions required for Group Policy Administration

FTP ServerFTProot

Directory

Organizations using a separate FTP server must do the following:• Copy the FTProot directory from the Sphericall Manager to the FTP Server.• Grant FULL CONTROL Security access to the FTProot directory, based on the

type of phone(s) you are using (i.e. if using SoundPoint IP phones, you must cre-ate that account; if using Aastra 480i phones, you must create that account; if using both, you must create both accounts).

Required Folder(s):If you are installing Windows FTP server, the ftproot folder will be located by default at: c:\inetpub\ftproot. This default setting needs to be changed as follows:• For systems with the FTP Server on the Primary Sphericall Manager, the following

folder is required for the location of IP phone resource files:<drive>:\\Program Files\Sphere\ftproot


See below based on manufacturer



Ac••

)

•

-

Aastra 480i:Used for IP

phones

Aastra480i


Accounts NotesBelow

NEW AastraSIP 9112i &

9133i IP Phone

Aastra 480i FTP login: Login: Sayson; Password: Aastra480iEither of the following two:1. Must be a Domain User if FTP server is also the Domain/Active Directory server.2. Local User if on the FTP server and FTP server is not the Domain/Active Direc-

tory server. Create a user account with username Sayson; create password: Aastra480i

Default Administrative Passcode on 480i phone: 22222Configurable Administrative Passcode on 480i phone only (not web interface):1) Options2) Option #9 - MGCP Settings3) Option #9 - Admin PasswordAdministrative Passcode & web interface login on 480i phones: Login: admin Password: 22222

Default Administrative Passcode on 9112i and 9133i phone: 22222

Administrative Passcode & web interface login on 9112i and 9133i phones: Login: admin Password: 22222



Entered manually on the web interface

Entered manually on the phone Entered manually on the web interface

GrandstreamBT series&

GXP2000

Default administrative password: admin Entered manually on the web interface

Polycom: Usedfor IP phonesPlcmSpIp

(case sensitive)


Accounts NotesBelow

Login & Password: PlcmSpIp (case sensitive)Either of the following two:1. PlcmSpIp Must be a Domain User if FTP server is also the Domain/Active

Directory server.2. PlcmSpIp Local User if on the FTP server and FTP server is not the

Domain/Active Directory. Create a user account with username PlcmSpIp; create password PlcmSpIp

Administrative Passcode on the Polycom SoundPoint IP phone and SoundStation 4000 IP Conference Phone: 456More detailed instructions for this account are contained in Book 4.



Quintum SSG Username: adminPassword: admin

UTStarcomF1000/G

DNS Domain name requiredMore detailed instructions for this account are contained in Book 4.User Name: userPassword: 888888

Manually on the web interface

tive Directory User Accounts: Sphere-MS & Sphere-DB Accounts MUST be set for: “User cannot change password” and “Password never expires”.Microsoft Password Complexity (affecting IP phones, Sphere-DB and Sphere-MS passwords):New Windows 2003 Domain/Active Directory servers (versus Windows 2000 Domain/Active Directory that has been upgraded to W2003by default enables password complexity. Default passwords for IP phones do not satisfy the complexity requirement. It is recommendedthat password complexity be disabled at the domain level on new Windows 2003 Domain/Active Directory.Options: An alternative solution to disabling password complexity on the Windows 2003 Domain/Active Directory would be to redefine the pass-word on all IP phones. Redefining the password on all IP phones also requires that you redefine the password on the Sphericall Administrator-> System - View Properties -> IP Phones tab.




. . .


C R E A T E T H E L O C A L U S E R A C C O U N T ( W I T H P A S S W O R D ) F O R T H E A A S T R A 4 8 0 I I P P H O N E S O N T H E F T P S E R V E R

FTP server—to create login and password for IP phones on the FTP Server1 Create a local user account on the FTP server with username Sayson.2 Create password Aastra480i.3 Deselect “User must change password at next logon.”4 Select “User cannot change password.”5 Select “Password never expires.”

FTP—to change the IP Phone User AccountIf you wish to change the login name and password used by the IP phone(s), you must change:• The user account defined on the local machine • You must also change the login and password in the Sphericall Administrator

application so that it can use this new account to communicate with the FTP service.

• Finally, you must also change the default login and password on the phoneset (this may also be done on the MGCP Call Client Provisioning web browser window).

FTP Server—to add required Sayson f i les to FTP server1 Install 480i specific XML file for screen management on FTP server.

Note: Installing an FTP server on your Primary Sphericall Manager is not recommended. If the FTP server resides on one of the Sphericall Managers within your system, you do not have to place these files on the FTP server. The Sphericall Manager will discover them automatically.

The appropriate path to the FTP site directory for a Sphere system integrating the Aastra 480i phone is:[Sphericall Manager hard drive]:\program files\sphere\ftproot\sayson\SaysonDeck1.xml

This is the directory in which all XML configuration files will be stored within the Sphere system for use by the 480i phones. If this file is newer than the file the phone is currently using, the phone will download this file from the server.

2 If you have a separate FTP server, paste the SaysonDeck1.xml file into the following FTP directory on the FTP server:[FTP Server hard drive]:...\sayson\SaysonDeck1.xml

TFTP Server—to copy f i les to the TFTP server1 Files stored in the Sphericall Manager for the TFTP server are:



c:\program files\sphere\images\appmgcp.bin.gzFile 1. 480i.imageFile 2. 480i.update-flash.imageEvery Sphericall Manager is a TFTP server and is updated with these files during installation or upgrade.

Spher ical l Manager—to add FTP server locat ion for use by Aastra phones1 Right-click on the System utility from the General tab of the Sphericall Administrator

application.2 View Properties.3 Select the General tab.4 Enter or browse to select the FTP Server name in the FTP Server section.

Spher ical l Manager—to add Aastra FTP login and passwordFrom the Sphericall Administrator application.

1 Right-click on System from the General tab.2 View Properties.3 Select IP Phones tab.4 Verify the following login/password:

Login: SaysonPassword: Aastra480i

M G C - T O - M G C P P H O N E C O N N E C T I O N C O N T R O LBefore defining Media Gateway properties, it is necessary to configure the connections between MGCP phones and the Sphericall Manager.With the Polycom MGCP phones, this step is required. With the Sayson MGCP phones, this step is optional since the MGCP phone automatically discovers the MGC and is assigned its MGC by load.

Unique Load Balancing for Aastra MGCP PhoneThe following procedure is optional for Sayson 480i phone users. Administrators may take the extra time to assign unique or specific MGCs as Primary, Secondary or Tertiary to the 480i phone. The 480i phone is designed to automatically find the MGC that has the appropriate load for that phone. Administrators may want to override the systems auto-balancing of the 480i phone additions based on the location of the phone in relation to its MGC (i.e. via a WAN link), or perhaps the location of a Secondary Sphericall Manager for manager failover.

Spher ical l Manager—to conf igure MGC-to-MGCP phone connect ions on a Sphere systemFrom the Sphericall Administrator:

1 Click the General tab.2 Expand Media Gateway Controllers from the tree list.3 Highlight the Media Gateway Controller you wish to configure.4 Right-click to View Properties.5 Click the MGCP Phones tab.


. . .


From the Properties for Media Gateway Controllers window:In the Set As Default area:

6 Select the Primary MGC check box if this Sphericall Manager is to be considered the default Primary Sphericall Manager that THIS MGCP phone will use to obtain its configuration information upon initialization.


7 Select the Secondary MGC check box if this Sphericall Manager is to be considered the default Secondary Sphericall Manager THIS MGCP phone will use to obtain its configuration information upon initialization if the Primary Sphericall Manager is unavailable.


8 Click ADD under each category to add this MGCP phone to its Primary, Secondary or Tertiary MGC.

9 Click Apply.

You will receive a message asking if you want to restart the MGCP phones.Select the appropriate option based on your system. The MGCP phone will need to be restarted to apply all these settings.

10 Click OK.

You may also choose the MGC to which each MGCP phone will be assigned via their Station Properites.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I N S T A L L I N G

480i IP Phone—to assemble and power the phonePlease use the Installation Guide provided with your phone for phone assembly and cable connections.

1 Complete all cable and cord connections for operability.2 Refer to the Troubleshooting section of the chapter for an overview of the phone key

setup options.

A A S T R A 4 8 0 I I P P H O N E I N S T A L L A T I O N W I T H D H C P I P A D D R E S S I N GFor systems requiring dynamic IP addresses for their IP phones, this procedure covers how to configure a dynamic IP address assignment.

480i IP Phone—to conf igure the 480i phoneAs the phone powers up:

1 Enable DHCP.(you will be required to login)



Options: select 8 NetworkSelect 1 DHCP (press change until Yes appears under DHCP)

2 Determine IP AddressOptions: select 8 NetworkSelect 2 IP Address (IP address will be displayed)Make note of IP address

Once an IP address has been assigned to a phone, you may now configure with the Options keys on the phoneset, or via the Sayson Web Client from any web browser.

3 The factory defaults on the phone are now in place.Important: Once an IP address has been assigned to the phone, you may see the time is not correct. Restarting the phone at this juncture allows the 480i to check-in with the Sphericall Manager and receive its time stamp from that server.

Aastra 480i factory installation guide within the packaging covers how to reset the “offset time” from Eastern time to your timezone. Sphere also provides this manual on the Documentation folder on the DVD.• Select the Options button on the phone.• Choose Option 2 - Time and Date.• Choose Option 6 - Time Zones.• Choose the appropriate time zone.

4 Return to the Sphericall Manager to assign this phone an extension.

A A S T R A 4 8 0 I I P P H O N E I N S T A L L A T I O N W I T H S T A T I C I P A D D R E S S I N GFor systems requiring static IP addresses for their IP phones, this procedure covers how to configure a static IP address assignment.

To insta l l the Aastra IP phone wi th a stat ic IP address you wi l l need to:After connecting phone to network:

On the phone:1 Select Options.2 Scroll to Network (Option 8).3 You will be required to login.4 Select Enter.5 Select Show for DHCP (Option 1).6 Press change to select “No” for DHCP.7 Select Done.8 Scroll to select Option 2 (IP Address).9 Press Show and enter IP Address.

10 Press Done.11 Press Done again until you return to the No Service Screen.

Open Internet Explorer :1 Enter http://<ip address given to phone>>.2 FIRST, Click Status “Network”.


. . .


3 THEN, Click Admin “Network”.

• User Name: Admin• Password: 22222

4 Complete the following fields based on your IP addresses for your system:

• Subnet Mask• Gateway• DNS1• DNS2

5 Click Set Values.6 Click Admin “Firmware”.7 Enter the IP address if the FTP Server.8 Click “Call Client”.9 Enter information:

• Call Agent Address 1 = Primary MGC• Call Agent Address 2 = Any Additional Managers• Call Agent Address 3 = Any Additional Managers• Call Agent Port: 2727• Client Port: 2427• FTP User Name: Sayson• FTP Password: Aastra480i• CTIP: Enter information from Sphericall Administrator Configuration.• Discovery: 239.193.0.0

10 Click “Set Values”.11 Verify that the phone restarts.

O P T I O N A L C O N F I G U R A T I O N S

Web Browser—to cont inue using the web browser for opt ional conf igurat ions1 Enter the IP address of the phone into the web browser address field.

Example: 192.168.0.1002 Press enter.3 Logon as administrator:

Login: adminPassword: 22222

4 View default configuration of 480i phone.

S P H E R E D I S C O V E R Y P R O T O C O LThe 480i IP phone uses the Sphere Discovery Protocol, along with DHCP, to automatically configure all MGCP settings within the phone. If the Sphere Discovery Protocol is not being used, MGCP settings will have to be entered into the phone manually.



If you are using the Sphere Discovery Protocol, there will be minimal changes needed to the defaults. Consider the following changes as per your dial plan and system design:

B U S Y - L A M P - F I E L D C O N F I G U R A T I O NAdministrators may want to configure some phones to display the extensions of monitored lines. These optional fields are configured via the web browser interface.Sphere system administrators may enter users BLF extensions for monitoring, or they may allow users to enter their own extensions for monitoring. If they allow users the right to enter their own extensions, they will need to provide the user with 1) the IP address of the 480i phone, 2) the user login and password: login = user, password = nothing (do not enter anything into the field).Sphere recommends a three step process for preparing users to monitor from Busy-Lamp-Field (BLF):

STEP ONE: Setup Address Group(s) on the Spher ical l ManagerSphere system administrators may want to govern who users monitor and who they may enter into their BLF fields for monitoring. On the Sphere system, all monitoring must be enabled through the use of Address Groups (please see Book 2: Install & Configure the Sphere System for more detailed information on Address Groups). It is recommended that the administrator configure Address Group(s) for this purpose.

1 From the Sphericall Administrator application, right-click on Address Groups from the General tab.

2 Click Add.3 Fill in the name of the Address Group (Ex: 480i phone monitoring, etc.).4 Select the appropriate Multicast Address for this purpose, or create a new address.5 Click Add in the lower section to add users/stations to this Address Group.

Only users who have been added to an Address Group will be able to monitor other users using the 480i BLF feature.

6 Repeat to create any other Address Groups.

STEP TWO: Setup the CTIP Mul t icast IP Address on the Web Browser1 Return to the Web Browser and login as Administrator.

Busy Lamp Field(BLF)

Entries in this field allow users to monitor other phone lines.

CTIP Multicast A CTIP Multicast IP Address and Port is needed in the Address Group on the Sphere system in order for users to monitor other users with Sphericall Desktop, Administrator, or with the 480i BLF field.If an administrator wants to enable monitoring by Zone, the CTIP Multicast IP Address from the Zone will need to be entered into this field on the Aastra Web Client or via options on the phone.

Reset UserPassword

System administrators may reset the user password to allow individual users to enter their own BLF fields. This is used on a case-by-case basis as system administrators deem appropriate.

DHCP If changing from DHCP to static configuration of IP addresses, the system administrator will need to enter each phone’s IP Address, Subnet Mask and Gateway information.

FTP All systems must configure the FTP server location.


. . .


Only those with admin login privileges may change the CTIP Multicast IP Address field.

2 Select Call Client.3 Enter the appropriate CTIP Multicast IP Address from the Address Group on Sphericall

into this field. 4 If using more than one CTIP Multicast IP Address for different users, logon to the Aastra

Web Browser as Administrator and enter a new users IP address in the browser address. Enter a new/different CTIP Multicast IP Address for this user.

5 Repeat for all users or for all unique Address Groups.

STEP THREE: Web Browser—to conf igure Busy Lamp Field Keys1 Select Soft Keys from the Web Browser User field.2 Enter extensions and names of users this phone will monitor using the BLF feature.

Figure 2.7 Soft Keys - Busy Lamp Field

In addition to listing the monitored extensions, the following states may shown on the phone:

Table 2.5 Busy Lamp Field States

State Icon

Idle

Ringing


M G C P I P P H O N E SIP Phone Installation Test

Spher ical l Manager—to complete conf igurat ion1 Return to the Properties for Station and Properties for Extension windows to configure

settings including

• Forwarding conditions• Emergency group• Telephony area association (for stations) and hunt order settings (for

extensions).Refer to Book 2: Install & Configure Chapter 6, Station Local Settings.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I P P H O N E I N S T A L L A T I O N T E S T

To test for successful extension and stat ion conf igurat ion1 Make calls between IP phones.2 Make calls between IP phones and other types of phones or external calls.3 Test IP phone keys and softkeys.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . T R O U B L E S H O O T I N G

For overview of phone key opt ions:Administrators have two options for setting up 480i phones. They may use the Options buttons on the phoneset or they may use the web browser.

Connected

Do Not Disturb

Call Forwarded

State Icon


. . .

. .M G C P I P P H O N E STroubleshooting

,

1 “Options” refers to the recessed button on the 480i phone used for configuration.

• From the phoneset, there are Administrator level options that are secured by a passcode. The default administrative passcode is: 22222. However, you have the ability to change the default administrative passcode from:

• Option 9: MGCP Settings• Option 9: Admin Password

The following Options are those which may be configured at install:Please note each of these options may also be configured, verified or changed from the web browser as well.

Press Options to enter the Options list.

Use the up or down arrow buttons to scroll through the list of options

Press Show softkey and arrow buttons to select an option.

Press Goodbye at any time to exit without saving changes.

Show Press Show softkey and arrow buttons to select an option.

Done Press the Done softkey at any time to exit the option and save the change.

Cancel Press the Cancel softkey at any time to exit without saving changes.

Option 8 Network Settings 1. DHCP• Turns DHCP on or off.• IP Address, Subnet Mask and Gateway options are read only

when DHCP is on.• Default: DHCP is on.

2. IP Address

3. Subnet Mask

4. Gateway

5. DNS

6. TFTP• TFTP server address is where the configuration files will be

download from.

Option 9 MGCP Settings If DHCP is turned on and the Sphere Discovery Protocol is enabledall the MGCP settings will be automatically configured. You shouldonly have to set these if the Discovery Protocol has not been enabled.

1. Call AgentsThis option contains up to 3 Call Agent IP Addresses. The Discovery Protocol automatically configures this setting. If the Discovery IP option is configured correctly, you shouldn't have to input these IP Addresses.

2. Call Agent PortThe Call Agent port should be 2727. If the Discovery Protocol is working, this option will automatically be set to port 2727.

Options

Goodbye


M G C P I P P H O N E STroubleshooting

P

g

y n.

e

a

2 “Browser” refers to accessing the phone configuration via a web browser.To access the Aastra 480i Web Client, open your web browser (i.e. Internet Explorer or Netscape, etc) and enter the phone's IP address into the address field, starting with the web prefix http://.In the side menu of the Aastra 480i Web Client, there are 3 main categories: Status, User and Admin.

• "The Status category contains read only status information for subcategories Network, Hardware and Firmware.

• "The User category contains user configurable subcategories Reset, Password, and BLF Softkeys. This section is accessed through either the user level or the administrator level user name and password. For more information, refer to the 480i User Guide provided with the phone.

• "The Admin category contains administrator only configurable subcategories: Network, Firmware and Call Client. This section is accessed through the admin level user name and password.

3. Client PortSet this port to 2427.

4. FTP User NameThis is the username used to access the Sphere FTP server. The XML decks for the phone's screen are located on this server.

5. FTP PasswordPassword used to access the Sphere FTP server.

6. CTIP Multicast IPThe CTIP Multicast IP address should match the multicast IP address configured in the Sphere server software. Please see CTIMulticast section.

7. CTIP PortThe CTIP port should match the port configured in the Sphere server software.

8. Discovery IPThe default setting for the Discovery IP is 239.193.0.0. This settinis very important because many of the MGC IP Addresses are automatically set when this is configured. To change the DiscoverIP address, please see Changing the Discovery IP Address sectio

9. Admin PasswordThe default Admin Password is 22222. This can be changed by performing the following steps from the phone:1.)Options2.)9.MGCP Settings3.)9.Admin Password

Option 10 Phone Status 1. Network StatusShows the network status of the 2 Ethernet ports on the back of thphone.

2. Firmware Status

3. Reset Phone

4. Factory DefaultSets the phone back to the factory default settings.

Note: For information on other settings in the options list of the phone, please refer to the Aastr480i Installer Guide provided with your phone.


. . .


For the administrator, the default user name is "admin" (all lowercase) and password is "22222", and for the user, the default user name is "user" (all lowercase) and password field is left blank.

Web Browser—to Reset the 480i phone1 Verify the phone’s IP address in the web browser.2 Click Reset from the Browser window.

You may be required to logon as administrator.

Spher ical l Manager—to Reset the 480i phone f rom the Spher ical l AdministratorFrom the Stations tab:

1 Highlight the IP phone and right-click and choose Restart.2 Repeat for any 480i IP phones that need restarted.

480i phoneset—to Reset the 480i phone back to factory defaul ts1 Press Options on the 480i phone.

You may be required to login as administrator.2 Scroll to choose Option 10. Phone Status and press Show softkey, or press 0 to jump

directly to this option.3 Scroll down the Phone Status to 4. Factory Defaults and press Show softkey, or press 4

to jump directly to this option.4 Press the Set Default softkey to set the phone back to factory default settings.5 Press Cancel if you wish to exit without resetting to factory defaults.

O V E R V I E W O F W E B B R O W S E R F I E L D SPlease note, all fields are informational except where noted.

Figure 2.8 STATUS - Network



Figure 2.9 STATUS - Hardware

Figure 2.10 STATUS - Firmware


. . .


Figure 2.11 USER - Reset

Figure 2.12 USER - Password

Allows Administrators to resetusers passwords.



Figure 2.13 USER - Soft Keys

Figure 2.14 ADMIN - Network

May be edited by or forusers.


. . .


Figure 2.15 ADMIN - Firmware


M G C P I P P H O N E SChanging the Discovery IP Address

Figure 2.16 ADMIN - Call Client

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C H A N G I N G T H E D I S C O V E R Y I P A D D R E S S

STEP ONE: Spher ical l Manager—to change the Discovery IP AddressFrom the Sphericall Administrator:

1 General tab.2 Right-click on System.3 View Properties.4 Select the System Initialization Settings tab.5 Click Add.6 Select MG Poll Multicast Address from the pull down. 7 Change the default from 239.192.0.0 to your choice.

STEP TWO: From the Phone—to change the Discovery IP AddressIf the Discovery IP address has been changed in the Sphericall Administrator, this setting also needs to be changed within the phone via the phone options or web browser. From the Aastra 480i Web Client:

1 Under the Admin section, click on Call Client.

May be edited for AddressGroup Multicast Address.


. . .

. .M G C P I P P H O N E SIP Phone Upgrades

2 If prompted, enter the 480i Administrator user name and password and press ok. 3 In the Discovery Multicast IP field, enter the new Discovery IP address. The default is

239.192.0.0.

From the Options List in the 480i phone:4 Press the "Options" button on the phone.5 Select option "9. MGCP Settings" and enter 480i Administrator password if prompted.6 Under option "8.Discovery IP", enter the new Discovery IP address. The default is

239.192.0.0.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I P P H O N E U P G R A D E S

All upgrades related to IP phones generate from Sphericall software upgrades. If there is a full system upgrade, IP phones will be upgraded as a part of this process. Refer to the Sphere System Release Notes for full upgrade procedures.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . U S I N G

The Quick Reference Guide for this version of IP phone is located in the Appendix of this manual.


M G C P I P P H O N E SUsing


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SIP PHONES 3

Sphericall supports the SIP standard and are compatible with the Sphere system.

Section Manufacturer Phone Topic

SIP Configuration Notes Information and overview of SIP parameters

I Configuring SIP User Agents

II SIP Failover: Configuring DNS Record for multiple Sphericall Managers

III Aastra 9112i9133i480i480i CT

Installation & Configuration

IV Grandstream GXP-2000GXV-3000


V Polycom IP30xIP430IP50xIP550IP601IP650IP4000


VI UTStarcom F1000G/F3000 Installation & Configuration

VII SIP Phone Sphericall Desktop Compatibility Table

VIII SIP Phone Star Codes Table


S I P P H O N E SSIP Configuration Notes

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I P C O N F I G U R A T I O N N O T E S

G E N E R A L S I P C O N F I G U R A T I O N I N F O R M A T I O NPhone ParametersVarious models of phones support different configuration parameters. Here are the most common. Outbound Proxy (Cisco, ClearOne, Grandstream, Polycom)—The Outbound Proxy field specifies the destination for all SIP requests. If a phone supports both Outbound Proxy and Proxy, Outbound Proxy overrides Proxy as the destination for SIP requests. Most phones allow an IP address, host name or DNS domain name in this field.Proxy (Aastra, Cisco)—The Proxy field specifies the destination for SIP requests if the Outbound Proxy is not supported by the phone or is left unspecified. The Proxy field also specifies the host portion of the SIP URI the phone places into the From header of outbound SIP requests and the To header of REGISTER requests. Most phones allow an IP address, host name or DNS domain name in this field.Server/SIP Server (Grandstream, Polycom)—Server and SIP Server are aliases for Proxy.Registrar (Aastra)—The Registrar field specifies the destination for SIP REGISTER requests. Most phones assume the registrar is collocated with the proxy or outbound proxy.RFC 3263 – Locating SIP Servers—RFC 3263 specifies procedures that SIP User Agent Clients (UAC) can use to locate SIP servers. Given a domain name and transport protocol, a UAC can locate an ordered list of SIP servers capable of handling SIP requests by using DNS NAPTR and SRV queries. RFC 3263 requires the UAC to send SIP requests to the first server in the list. Should that server be unavailable, the UAC may send the request to subsequent servers in the list until it finds an available server or the list is exhausted.

• Many phones do not yet support RFC 3263 but some do (Cisco, Polycom).• Not all DNS servers support NAPTR records.

User Name/User ID (Aastra, Cisco, Grandstream)—The User Name field specifies the user portion of the SIP URI the phone places into the From header of outbound SIP requests and the To header of REGISTER requests.Authentication Name/Password (Aastra, Cisco, Grandstream, Polycom)—The Authentication Name and Password fields specify the credentials used by the phone to authenticate itself when challenged. Sphericall does not yet challenge SIP phones, so these fields are not currently relevant.SIP Phone MAC AddressWhen a phone checks into a Sphericall system for the first time, the MGC automatically creates a station for the phone. As part of the station creation process, the MGC assigns a unique name to the phone and optionally assigns an extension. For Sphericall MGs and MGCP phones, the MGC uses the phone’s MAC address as the unique name. For SIP phones, the MGC creates the station when the phone registers with the system for the first time. The MGC uses the SIP URI of the REGISTER request To header as the phone’s unique name.


. . .

. .S I P P H O N E SSIP Configuration Notes

As mentioned above, the REGISTER request To header SIP URI is composed of the User Name and Proxy fields. It is important to note that if either of these fields is modified in the phone, the MGC will create a new station and extension for the phone the first time the phone sends a REGISTER request with the new value(s).SIP DomainThe SIP Domain (SD) is specified in the SIP Domain field of the System Properties dialog box. When the MGC sends a SIP request to a SIP phone, it populates the host portion of the SIP URI of the From header with the contents of the SD. INVITE, ACK, BYE, CANCEL, OPTIONS and SUBSCRIBE are examples of SIP requests that use the SD in their From header.

• The SD does not apply to SIP trunks.

R E C O M M E N D E D C O N F I G U R A T I O N SHere are some recommended configurations based upon the capabilities of the SIP phones and DNS infrastructure.IP Address vs. DNS Name—DNS host or domain names should be used rather than IP addresses wherever possible for items such as Outbound Proxy and Proxy.Outbound Proxy vs. Proxy—Outbound Proxy should be left unspecified if the phone supports both Outbound Proxy and Proxy. Proxy provides greater flexibility since Outbound Proxy specifies the destination for all SIP requests from the phone. For example, the Cisco 7960 is a six line SIP phone. Since each line has a unique Proxy field, it’s possible to have a line on six different Sphericall systems. If configured with an Outbound Proxy, all six lines must be on the same Sphericall system.SIP Domain—The SD can be an essentially arbitrary string but it should be a valid DNS domain name that can be published on the public Internet.Full RFC 3263 Support—If the phones and DNS infrastructure support NAPTR queries, the following configuration is recommended:• Choose a valid DNS domain name for the SD

• Example: spherecom.com.• Create an NAPTR DNS record for the SD that points to an SRV record for SIP over

UDP• Example: SIP+D2U _sip._udp.spherecom.com

• Note that since Sphericall currently only supports UDP, it is not necessary to create records for SIP over TCP or TLS

• Create an SRV record for SIP over UDP and populate it with the fully qualified domain names of at least two MGCs• Example: priority 0, weight 0, voip_manager.spherecom.com• Example: priority 10, weight 0, ritas.spherecom.com

• Set the phones’ Proxy or Outbound Proxy to the SD• Example: spherecom.com.

This configuration will allow the phones to locate an alternate MGC should the MGC they are currently registered with fail.


S I P P H O N E SSIP Configuration Notes

P A R T I A L R F C 3 2 6 3 S U P P O R TIf the phones and DNS infrastructure support SRV queries but not NAPTR queries, the following configuration is recommended:• Choose a valid DNS domain name for the SD

• Example: spherecom.com.• Create an SRV record for SIP over UDP and populate it with the fully qualified

domain names of at least two MGCs• Example: priority 0, weight 0, voip_manager.spherecom.com• Example: priority 10, weight 0, ritas.spherecom.com

• Set the phones’ Proxy or Outbound Proxy to the SD• Example: spherecom.com.

This configuration will allow the phones to locate an alternate MGC should the MGC they are currently registered with fail.

N O R F C 3 2 6 3 S U P P O R TIf the phones do not support RFC 3263, the following configuration is recommended:• Choose a valid DNS domain name for the SD

• Example: spherecom.com.• Set the phones’ Proxy or Outbound Proxy to the host name of an MGC

• Example: voip_manager.spherecom.com.This configuration will bind the phones to a single MGC.


. . .

. .S I P P H O N E SSection I - SIP User Agents

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S E C T I O N I - S I P U S E R A G E N T S

To conf igure SIP User AgentsSphere has opened the User Agents interface to the system administrator for administration.

• Required: All SIP endpoints must be listed in this window prior to configuration. Once entered here, the endpoint will be accepted into the full system. AGAIN, this step is required prior to any SIP endpoint being added to the system—trunk or station. The Sphere system requires this information in order to know how to treat the endpoint or to know what features to apply per endpoint.

• All widely-used, tested and approved SIP endpoints, for this version of software, are listed in this SIP properties window.

• All defaults are automatically configured for your convenience. Defaults and supported User Agents are indicated by the check mark.

• User Agents are also available for “adding” a new, untested SIP endpoint. If a site has a new untested SIP endpoint, they must add it to the system themselves and verify its operability through their own testing (this testing is not supported by Sphere support personnel).

• There are generic Agents listed that can suffice for unknown SIP endpoints.• Properties of the User Agents can be viewed for appropriate overrides of the

default settings for some deployments.All SIP devices of a make, model and firmware version have same attributes, for example, all Polycom IP601 SIP phones running 2.1.0.2708 firmware support "talk" event in the NOTIFY request to answer an incoming call. Instead of assigning these attributes on each SIP endpoint individually, the attributes are assigned to the User-Agent/Firmware-Version binding and gets applied to all corresponding SIP devices. Several new database tables have been added to support this feature. Sphere-supported SIP devices (stations and trunks) are pre-configured in the database.

1 Select the SIP tab on the System Properties window.


S I P P H O N E SSection I - SIP User Agents

TIFY /un-

Web

Figure 3.1 SIP Properties

2 Review the User Agents listed in the window.3 If the SIP endpoint you are using is not listed in this window, you must add it.4 Click Add.5 Enter the User Agent name.6 Select Endpoint Type: REQUIRED. The Sphere system requires this information in order to

know how to treat it or what features to apply per endpoint.

7 Enter an Agent Description that is appropriate for the Name & Endpoint Type.8 Click Apply.

Note: 1) Those User Agents listed in the SIP dialog window that also have the Default checked, are those User Agents created into the system by default. These will remain in the system. User Agents added by system administrators are not indicated with a check in the Default column.

Note: 2) If the name of a non-default user agent matches the name of a default user agent, the Agent Name, Agent Description and Endpoint Type fields cannot be edited. Conversely, if multiple user agent entires exist that have the same name, but none are marked as default, changing any field (other than Version) will change the same corresponding field in the other same-named User Agent entries.

The following fields are customizable for entering a non-default user agent:

Table 3.1 User Agent Profile Descriptions

User Agent Parameter Possible Values Description

‘talk’ Event (Notify Request) Based 3PCC

Supported Unsupported (default)

The MGC sends the NOrequest to answer/holdhold a call remotely (Sphericall Desktop or Services).


. . .

. .S I P P H O N E SSection I - SIP User Agents

a ith eived

d.

alue of swer, st to o

SIP in the 1+ all re pre-

he SIP ility nes

GC

com

to an iately

with a 00 ime

ration t es

in TER

iate inal oming e

o

ER

)

ited the

‘to-tag’ (SUBSCRIBE REQUEST) In New Subscription

AllowedDisallowed (default)

MGC sends a NOTIFYrequest with terminated(reason=timeout) subscription-state whenSUBSCRIBE request wnon-empty to-tag is recand the corresponding subscription is not foun

Click-To-Dial Ring Caller’s Phone First (default)Use ‘answer-after’ ??param (INVITE:: Call-Info Header)Use ‘auto-answer’ value (INVITE:: Call-Info Header)

MGC sends a special vthe parameter (Auto Ananswer-after etc) in theoutgoing INVITE requeinform the SIP station tanswer immediately.

Endpoint Created By AdministratorCall Manager (default)

Call Manager creates aphone when not found database. Since in 5.2.Polycom SIP phones acreated by the system administrator (just like tsoft trunks), this capabensures that these phoare not created by the Mwhen not found in the database. When a Polyphone sends its first REGISTER to check inMGC, the MGC immedsends a "503 Service Unavailable" response Retry-After timeout of 3seconds. In the mean tMGC obtains phone configuration from the database. If the configufrom the database is noavailable, MGC continusending "503 Service Unavailable" message response to the REGISrequests.

Find Terminal Method Authentication InfoDefault (default)DID MappingFrom Header URIOutbound Contact URIP-Asserted-Identity Header URI (currently supported)

MGC uses the approprmethod to find the termassociated with the incINVITE request from thendpoint.

Hardware Address Available (REGISTER:: User-Agent Header)Unavailable (default)

SIP has the capability tsearch for a hardware address in the REGISTrequest.

INVITE Request-URI Source Invite Request To-URIOutbound Contact-URI (default)

(Quintum products only

MWI NOTIFY Request Supported (default)Unsupported

MGC sends an unsolicNOTIFY request when MWI state changes.



S I P P H O N E SUser Agent Maintenance

o ill not t to the s a us. If

send (out there status s not uest.

r nd.

SIP not est.

FER call.

initiate

Y ion.

er RI) in

the

in the

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . U S E R A G E N T M A I N T E N A N C E

U S E R A G E N T S A N D U P G R A D E SThese tables essentially provide the functionality provided by the MGSetting table, but at a much less overhead. These tables also provide the following benefits:• When a Sphericall system is upgraded (pre-6.0 to 6.0+), Sphericall automatically

creates UserAgentVersionHubAssociation for all Outbound/None registration type devices. It also creates a binding in UserAgentVersionHubAssociation table for those SIP devices (irrespective of registration type) that have an entry in the deprecated UserAgentHubAssociation table. This ensures that the SIP devices keep functioning after the upgrade.

• When a Sphericall system is upgraded (6.0 to 6.0+), Sphericall will not change the existing UserAgentVersionHubAssociation bindings. Therefore, the SIP devices overriding the default behavior will continue to work the same way after the

MWI SUBSCRIBE Request Supported (default)Unsupported

When the value is set tSupported, the MGC wsend a NOTIFY requesendpoint even if there ichange in the MWI statthe value is set to Unsupported MGC will the unsolicited NOTIFYof dialog) request whenis a change in the MWIeven if the endpoint hasent a SUBSCRIBE req

MediaServer Max Packetization (ms)

10 ms - 160 ms range80 ms (default)

Setting specifies the maximum packet size Sphericall Media Serveshould send to the far-e

OPTIONS Request Supported (default)Unsupported

Setting specifies whichendpoints support or dosupport OPTIONS requ

REFER Based 3PCC SupportedUnsupported (default)

MGC does not send REto initiate a click-to-dial

REFER Based Transfer Supported (default)Unsupported

MGC sends REFER to a transfer.

Remote Reboot SupportedUnsupported (default)

MGC sends the NOTIFrequest to reboot a stat

URI ‘qheaders’ Parameter Supported (default)Unsupported

MGC sends the QHead(Question header in a Uthe Refer-To header in REFER request.

Video SupportedUnsupported (default)

MGC sends Video SDPoutgoing INVITE.



. . .

. .S I P P H O N E SSIP Failover

upgrade. However, Sphericall may change the default values of the Parameters configured in the Parameter table (this is similar to that of the MGSetting defaults which may infrequently get changed on an upgrade).

• When the firmware of a SIP device is upgraded, if the SIP device has a forced binding in the UserAgentVersionHubAssociation table, the DbServer/MGC do not update the binding for the new firmware and the SIP device continue the work same way in the MGC. For Outbound/None registration type endpoints DbServer and MGC never update the UserAgentVersionHubAssociation binding even if the SIP device is reporting a completely different User-Agent/Firmware-Version than what is configured in the UserAgentVersionHubAssociation table.

• A new User-Agent/Firmware-Version can be created and assigned to a custom ParameterProfile and then this User-Agent/Firmware-Version can be bound to the UserAgentVersionHubAssociation and set to "forced" type binding. This feature provides a similar level of control provided by the MGSetting table.

• SIP station upgrades—Upgrades to User Agents with firmware are automatic and apply to all the endpoints on the system.

• SIP trunk upgrades—Upgrades to User Agents are not automatic.

U S E R A G E N T C L E A N U POver time, a number of older User Agents Parameter versions will accummulate in the dialog of User Agents. It is the Sphere system administrator’s responsibility to remove old versions of the User Agents that are no longer used on the system.

To remove a User Agent Parameter1 From the Sphericall Administrator application:2 Open the General System properties.3 Select the SIP tab.4 Scroll to view the User Agents.5 Click the far right column of the User Agent to be removed.6 Click Remove.7 Repeat for other User Agents.8 Click OK to exit.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I P F A I L O V E R

C R E A T E A D N S R E C O R DCreating a DNS Record is necessary to determine the Sphericall Manager with which the phones register.Book 1: Plan & Prepare the Sphere System has an extensive discussion of what considerations are important for systems with multiple Sphericall Managers or MGCs.


S I P P H O N E SSIP Failover

In those systems, in order to provision for failover, it is required for any SIP device to have a DNS record. This affords the phone device a single point of connection, even if the local MGC fails.

On DNS Server—To create A DNS RecordOn a Microsoft Windows DNS Server:

1 Click Start\All Programs\Administrative Tools\DNS.2 Expand the Forward Lookup Zones tree.3 Locate your organization’s Domain Name.4 On the Domain Name, right-click New Domain.5 Type a name to designate the DNS Record. For example, “spheresip.”6 Right-click on the newly-created “spheresip” folder, and select Other New Records.7 Select Service Location (SRV).8 Click Create Record.

Figure 3.2 New Resource Record window

9 Key in the following DNS Record information:a. Domain: This is the name of the Sphericall Manager with which phones will register.b. Service: Type _sip. Note: This must be typed. There is no _sip option in the list.c. Protocol: Select _udp.d. Priority: Set value to 0 for the highest priority. e. Weight: In a multi-MGC environment, this option prioritizes which Sphericall Manager

you want phones to check into. Set value to 0 for all Sphericall.


. . .

. .S I P P H O N E SSIP Failover

If your organization has multiple Sphericall Managers, it is recommended that you establish the following values:

f. Port Number: Set value to 5060. The Sphericall Manager uses port 5060, which is the standard port for SIP traffic.

g. Host offering this service: The fully qualified domain name of the Sphericall Manager.10 Click Done.11 Right-click on the Record name. In this case, “spheresip,” and select New Host(A).12 Leave Name field blank.13 Key in the Sphericall Manager’s IP address.14 Click Add Host.15 Click OK.

Refer to Book 1 for other configuration options required per device.

Primary Sphericall Manager

Secondary Sphericall Manager

Priority: 0 Priority: 1000

Weight: 0 Weight: 0


S I P P H O N E SSection III - Aastra SIP Phone Configuration

S E C T I O N I I I - A A S T R A S I P P H O N E

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C O N F I G U R A T I O N

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W

Aastra’s line of SIP phones, with minimal configuration, are interoperable with the Sphere system.This document supports the following Aastra phones:• 9112i• 9133i• 480i• 480i CT

Note: For a description of configuration settings specific to each phone, please refer to the Aastra SIP IP Phones Administrator Guide at www.aastra.com.

To integrate the Aastra phone wi th the Sphere systemAn administrator can configure the IP Phone Network and SIP options from the phone user interface, from the Aastra Web user interface, or the configuration files. Administrator level options are password protected in both the IP phone user interface and the Aastra Web user interface.

1 YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints.

2 Plug the phone in and allow the phone to discover an IP address.a. Select Optionsb. Arrow down to Option #8 - Network Settingsc. Enter the default Admin Password = 22222d. Arrow down to Option #2 - IP Address

3 From a web browser, use the IP address to open a web page for the phone.4 Select SIP Settings from the lower left-hand side of the web interface and enter the

following information:Login: admin Password: 22222


. . .

. .S I P P H O N E SOverview

Figure 3.3 Aastra Web User Interface page

5 Key in the following Basic SIP Authentication Settings:a. Screen Name - enter the screen name that displays on the idle screen.b. Phone Number - enter the phone number of the IP phone.c. Caller ID - enter the phone number of the IP phone.

6 Key in the following Basic SIP Network Settings:a. Proxy Server - enter the DNS address of one or more MGCs that accept SIP

registrations.

This may be an IP address, host name, or FQDN DNS address. For redundancy and failover, it is recommended that this field be an FQDN DNS address that contains a prioritized list of SRV records. Systems that want to share MGCs across SIP phones can create multiple DNS addresses, each with a different prioritized SRV record list.If you have multiple Sphericall Managers, it is recommended that you use the DNS SRV.b. Outbound Proxy Port - the MGC uses port 5060, which is the standard port for SIP

traffic.c. Registrar Server - enter the name/address of the MGC with which the endpoint will

register.7 Test the phone for dial tone.8 Use the Sphere Administrator application to modify the station properties as

appropriate.

To enable MWI on an Aastra SIP phoneFrom the Advance Settings\Global SIP page:


S I P P H O N E SOverview


1 Select the Explicit MWI Subscription checkbox to Enabled.2 Enter the requested duration, in seconds, before the MWI subscription times out.

The phone re-subscribes to MWI before the subscription period ends.

S O F T K E Y S / P R O G R A M M A B L E K E Y SYou can configure the softkeys (480i/480i CT) and programmable keys (9112i/9133i) to perform specific functions on the IP phones.The following table provides the number of sofkeys and programmable keys you can configure, and the number of lines available for each type of phone.

Table 3.2 Aastra Phone Operational Features

H A R D K E Y SThere are hard keys on your phone, such as Hold, Redial, Xfer, Icom and Conf (Hold and Icom not available on the 9112i and 9133i) that are configured for specific call-handling features. (See the product-specific User Guide for more information about the hard key functions).

IP Phone Model Softkeys Available Programmable Keys Available Lines Available Handset Keys

Available

480i 20 9

480i CT 20 9 15

9112i 2 1

9133i 7 9


. . .


You can configure your programmable keys in two ways; either by saving a listing from your Callers, Redial or Directory list to a programmable key through your phone, or by using the Aastra Web user interface.

To save programmable keys f rom the phone1 Press the desired hard key.2 Scroll through the selected list to find the name and number you wish to save to your

speed dial.3 Press Save.4 Press the selected speed dial key.

If the selected name is displayed with the number, both are saved to the speed dial.

P R O G R A M M I N G K E Y S F R O M T H E W E BYou can also program keys by accessing the Aastra Web user interface.


To program keys f rom the Aastra web inter face1 Access the Aastra Web user interface, according to the instructions above.2 Select Programmable Keys.3 Select the Type from the following options:

4 In the Value field, enter a value to associate with the softkey or programmable key.

For example, for a speed dial value, enter *1.5 For the 480i/480i CT and 9133i, in the Line field, select the line for which you want to

associate the softkey or programmable key.

none line BLF XML spre pickup

speed dial do not disturb BLF/List Flash park lcr

directory callers list intercom services empty



6 Click on Save Settings to finish.

C O N F I G U R A T I O N F I L E SIf your organization intends to deploy hundreds of Aastra phones, the use of configuration files may be more beneficial than the use of the Aastra Web user interface. The Aastra configuration file is supplied on the Sphere CD at Server\Data\Sayson\aastra.cfgA system administrator can enter specific parameters in the configuration files to configure the IP phones. All parameters in configuration files can only be set by an administrator. For a description of each configuration file parameter, refer to the Aastra Administrator’s Guide at www.Aastra.com.

U P G R A D E SAastra phones can use either TFTP or through the Web interface for upgrades. For use with Sphericall Managers the phones and ease of use administrators will probably want to use the TFTP server.

Note: Sphericall Managers are equipped with TFTP servers.

On reboot, each phone looks for a firmware file in the TFTP directory. If the firmware file exists and the version is different from the installed firmware the Aastra phone performs a TFTP get on the firmware and either performs a upgrade or downgrade depending on the installed firmware version and the firmware version in the TFTP directory.The Aastra phone attempts to locate the following file (located in Program Files\Sphere\Images):• 9112i.st• 9133i.st• 480i.st• 480i CT.st

Note: Occurring on 9112i, 9133i and 480i phones, Aastra has documented that their phones will only reboot if the version of firmware has changed/updated (old or new). Do not expect Aastra phones to reboot immediately after invoking the reboot command.


. . .


To upgrade the Aastra PhoneFigure 3.6 Aastra Web User Interface page

• Select the "Firmware Update" tab and enter the IP address of the Sphericall Manager that is running the Sphericall TFTP server.

For ease of management, it is probably easiest to have all phones use the same IP address for image downloads, although potentially any phone could be configured with the IP address of any manager if the administrator wants to implement load sharing across the TFTP servers.In order to accomplish an upgrade, the Sphere administrator will be required to reboot each Aastra phone, via the phone keypad, web interface or Sphere Administrator application.Aastra phones reboot only if there is an updated (old or new) firmware is available at the TFTP server. If the firmware is unchanged, or if the TFTP server is not running, the Aastra phone will not restart when MGC sends a NOTIFY request to remote-restart the phone.

S P H E R E S Y S T E M U P G R A D E SUnlike MGCP IP phones, SIP phones need to be upgraded manually. The Sphere Administrator application does not prompt the Sphere administrator to restart the SIP phones after the Sphericall upgrade (even if a newer SIP firmware is available). Most SIP phones have TFTP clients to download the firmware (to the phone). The Sphericall upgrade program copies the latest firmware for all these phones in to the \sphere\images directory. Only a manual reboot of the SIP phones is required to upgrade them to their respective latest firmware. The Sphere Administrator should pay close attention to both Sphere Release Notes and Sphere System Requirements which mention the firmware revision of each supported SIP phone (and if the version has changed from the previous release).


S I P P H O N E SSection IV - Grandstream GXP-2000 SIP Phone Configuration

S E C T I O N I V - G R A N D S T R E A M G X P - 2 0 0 0 S I P

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P H O N E C O N F I G U R A T I O N

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W

The Grandstream GXP-2000 is an enterprise IP telephone based on open industry standards

Note: This product can be utilized with the Sphericall Desktop.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• Verify Sphere System Requirements for Grandstream GXP-2000 and interoperability with Sphericall.

• Verify that the Sphere system is installed, configured and tested as fully functional.• Refer to the Grandstream GXP-2000 User Manual for installation planning, setup,

package contents, safety and conditions of use.


. . .

. .S I P P H O N E SInstalling

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I N S T A L L I N G

To integrate the Grandstream GXP-2000 wi th the Sphere systemFigure 3.7 GXP-2000 Back Panel


2 Using the supplied power adapter, power the phone up.3 Connect the phone to the network using the supplied ethernet cable.4 Allow the phone to discover an IP address.5 Locate the IP address on the phone’s LCD display.

Figure 3.8 GXP-2000 Menu Buttons

To locate the IP address, press the circular MENU button to enter menu mode. Use the down arrow button to locate menu item #3 - “IP Address.”

6 Open a web browser.7 Enter the IP address of the phone into the web browser address field.

Example: //192.168.0.1008 Enter the default password: admin

Note: Only the Administrator can change the password for entry to the Advanced Settings page. Otherwise, the end user password can be changed on the Basic Settings page.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . W E B C O N F I G U R A T I O N

A series of web pages allow you to configure the GXP-2000’s many features.

PCLANPOWER


S I P P H O N E SWeb Configuration

Table 3.3 Grandstream GXP-2000 Web Device Configuration table

A C C O U N T P A G EThe GXP-2000 has four configurable lines, each uniquely configurable through the Accounts tab on the Device Configuration web page. Lines that are not configured use the Account 1 configuration. Each uniquely defined account appears as a separate device in the Sphericall Administrator application.

Figure 3.9 Account page

9 Key in the following SIP setting information in the Account 1 page:a. SIP Server: The SIP domain name that all endpoints on the Sphere system must use

in order to utilize SIP.b. Outbound Proxy: the name/address of the Sphericall Manager to which the endpoint

will register.c. SIP User ID: the user name portion of the SIP address for this endpoint. In this case,

the MAC address of this phone. In some instances, the SIP extension is used.

Web Page Required Entries

Status Information is not configurable from this page.

Basic Settings • IP Address dynamically assigned via DHCP• Speed Dial Configuration• Time Zone Configuration

Advanced Settings • No Key Entry Timeout• Layer 2 and Layer 3 QOS tagging• Local RTP Port• TFTP Server Address for Firmware Upgrade

Account 1, 2, 3, and 4 • SIP Server• Outbound Proxy• SIP User ID• Subscribe for MWI• Voice Mail UserID


. . .

. .S I P P H O N E SWeb Configuration

There are side effects to filling in certain values under the Account(s) tab:• Accounts not configured use the values set in Account 1.• The creation of a unique account on the phone requires, at a minimum, a value in

the SIP Server field. However, the account and line will not be usable until the SIP User ID is also set.

• If values are set in an account tab, the minimum information, specified above, is required. The values need not be unique for each account.

• If the SIP User ID field is entered, but the SIP Server field is not entered, the values from ACCOUNT 1 are used.

10 Reboot the phone from the Device Configuration web browser.

The phone will appear in the Sphere Administrator application as a SIP phone with a MAC address of <phone’s mac address>@SIP Server.

11 Test the phone for dial tone.12 Use the Sphere Administrator application to modify the station properties as

appropriate.

At this point, the phone will have dial tone, you should be able to place calls, and the device will appear in the Sphere Administrator application. Further configuration is needed to ensure the full functionality of the phone.

U S E D N S S R VRecommended for SIP stations failover/failback using RFC 3263 procedures. If you have multiple Sphericall Managers, it is recommended that you use the DNS SRV.


V O I C E M A I L U S E R I DWhen configured, the user will be able to dial voice mail by pressing the MSG button.


• Type the extension of the Voice Mail server.

S U B S C R I B E F O R M W I


• Set to Yes.



P U B L I S H F O R P R E S E N C E


• Verify that setting is No.

A L L O W A U T O A N S W E R B Y C A L L - I N F OThis setting enables click-to-dial functionality.


• Set to Yes.

S P E C I A L F E A T U R EThis setting enables the third party call control functionality.


• Set to Broadsoft.

B A S I C S E T T I N G S P A G EThe Basic Settings page allows you to configure the speed dial feature.There are seven speed dial fields that can be configured.

To conf igure the GXP-2000 basic set t ings1 Click the Basic Settings page.2 Verify that dynamically assigned via DHCP is selected.

Figure 3.16 Basic Settings page


. . .



To conf igure speed dia l1 Type a name in the Name field which is used to identify the person. It will be displayed

on the LCD when pressing the corresponding key.2 Type a phone number or extension in the UserID filed.3 Select the appropriate account in the Account field. This is the SIP account associated

with the number.

To conf igure the t ime zoneFigure 3.18 Basic Settings page

• Select the appropriate time zone to control how the date/time is displayed on the LCD.

A D V A N C E D S E T T I N G S

Q O SLayer 2 and Layer 3 QoS tagging can be set. The values are applied to RTP and SIP packets, but cannot be set independently for each type of packet.IPv4 provisions two methods for the tagging of packets in the IP header, IPv4 Type of Service (TOS) and Differentiated Services (DiffServ). Although they partition the bits differently, both methods use the same octet in the IP header. Tags are provided merely so that the network infrastructure can prioritize traffic based on the interpretation of these tags.Previously, Sphere’s System Initialization Settings supported Layer 3 only. Sphericall now supports both Layer 2 and Layer 3 prioritization. These QoS parameters may be set on the QoS tab of the System Properties.Sphere’s QoS settings are system wide in scope, but please note: any Layer 2 settings that are set on the Sphericall Manager, must also be set at the driver level of the Sphericall Manager, and must also be set on each and every endpoint on the system in order for it to effectively work throughout the system.



The Layer 3: QoS field defines the Layer 3 parameter which can be used for IP Precedence or Diff-Serv or MPLS. The default value is 48.The Layer 2: QoS field contains the value used for layer 2 VLAN tagging. The default setting is blank.

Figure 3.19 Advanced Settings page

N O K E Y E N T R Y T I M E O U TIf Send is not entered after dialing a phone number, the GXP-2000 will push the call out.


L O C A L R T P P O R TThis parameter defines the local RTP-RTCP port pair the GXP-2000 will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.


U P G R A D E SGrandstream phones can be configured to use either TFTP or HTTP for upgrades. For use with Sphericall Managers the phone should be configured to use TFTP.

To upgrade the Grandstream GXP-2000Using the web interface:• Select the Advanced Settings tab and enter the IP address of the Sphericall

Manager that is running the TFTP server. Note: Sphericall Managers are equipped with TFTP servers.

Figure 3.22 Grandstream GXP-2000 Advanced Setting for Firmware Upgrades

For ease of management, it is probably easiest to have all phones use the same IP address for image downloads, although potentially any phone could be configured


. . .


with the IP address of any manager if the administrator wants to implement load sharing across the TFTP servers.The Grandstream GXP-2000 attempts to locate the following files (located in Program Files\Sphere\Images):• boot55.bin• gxp2000.bin• ring1.bin• ring2.bin• ring3.binIf the phone is unable to find the TFTP server or one of the image files, the phone aborts the TFTP transfer and simply uses the image stored in flash. The phones will take slightly longer to boot when image files are located in the TFTP server directory. Depending on the connection speed and size of the image file, Grandstream states that the boot process may take up to 5 minutes. If the boot times becomes an issue the administrator may remove the files from the Images directory. However, if the files are removed any new Grandstream phones added to the system may not boot up with the image shipped and compatible with the Sphere system.In order to accomplish an upgrade. the Sphere administrator is required to reboot each Grandstream phone either via the web interface, phone keypad or, power cycle.

S P H E R E S Y S T E M U P G R A D E SUnlike MGCP IP phones, SIP phones need to be upgraded manually. The Sphere Administrator application does not prompt the Sphere administrator to restart the SIP phones after the Sphericall upgrade (even if a newer SIP firmware is available). Most SIP phones have TFTP clients to download the firmware (to the phone). The Sphericall upgrade program copies the latest firmware for all these phones in to the \sphere\images directory. Only a manual reboot of SIP phones is required to upgrade them to their respective latest firmware. Grandstream phones will restart only if the Special feature option in Account page is set to "Broadsoft." Only BT100, GXP2000 and GXV3000 phones will be allowed to remote-reboot from the Sphericall Administrator application.The Sphere Administrator should pay close attention to both Sphere Release Notes and Sphere System Requirements which mention the firmware revision of each supported SIP phone (and if the version has changed from the previous release).


S I P P H O N E SGrandstream GXV-3000 Phone

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . G R A N D S T R E A M G X V - 3 0 0 0 P H O N E

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W

The Grandstream GXV-3000 is a next generation enterprise IP telephone based on open industry standards.

Note: This product is intended to operate as an independent device, and therefore, can not be utilized with the Sphericall Desktop.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• Verify Sphere System Requirements for Grandstream GXV-3000 and interoperability with Sphericall.

• Verify that the Sphere system is installed, configured and tested as fully functional.• Refer to the Grandstream GXV-3000 User Manual for installation planning, setup,

package contents, safety and conditions of use.


. . .

. .S I P P H O N E SInstalling

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I N S T A L L I N G

To integrate the Grandstream GXV-3000 wi th the Sphere systemFigure 3.23 GXV-3000 Back Panel


2 Using the supplied power adapter, power the phone up.3 Connect the phone to the network using the supplied ethernet cable.4 Allow the phone to discover an IP address.5 Locate the IP address on the phone’s video display.6 Open a web browser.7 Enter the IP address of the phone into the web browser address field.

Example: //192.168.0.1008 Enter the default password: admin

Note: Only the Administrator can change the password for entry to the Advanced Settings page. Otherwise, the end user password can be changed on the Basic Settings page.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . W E B C O N F I G U R A T I O N

A series of web pages allow you to configure the GXP-3000’s many features.

Table 3.4 Grandstream GXP-3000 Web Device Configuration table


Status Information is not configurable from this page.



A C C O U N T P A G EThe GXP-3000 has four configurable lines, each uniquely configurable through the Accounts tab on the Device Configuration web page. Lines that are not configured use the Account 1 configuration. Each uniquely defined account appears as a separate device in the Sphericall Administrator application.


9 Key in the following SIP setting information in the Account 1 page:a. SIP Server: The SIP domain name that all endpoints on the Sphere system must use

in order to utilize SIP.b. Outbound Proxy: the name/address of the Sphericall Manager to which the endpoint

will register.c. SIP User ID: the user name portion of the SIP address for this endpoint. In this case,

the MAC address of this phone. In some instances, the SIP extension is used.

There are side effects to filling in certain values under the Account(s) tab:• Accounts not configured use the values set in Account 1.

Basic Settings • IP Address dynamically assigned via DHCP• Speed Dial Configuration• Time Zone Configuration

Advanced Settings • No Key Entry Timeout• Layer 2 and Layer 3 QOS tagging• Local RTP Port• TFTP Server Address for Firmware Upgrade

Account 1, 2, 3, and 4 • SIP Server• Outbound Proxy• SIP User ID• Subscribe for MWI• Voice Mail UserID



. . .


• The creation of a unique account on the phone requires, at a minimum, a value in the SIP Server field. However, the account and line will not be usable until the SIP User ID is also set.

• If values are set in an account tab, the minimum information, specified above, is required. The values need not be unique for each account.

• If the SIP User ID field is entered, but the SIP Server field is not entered, the values from ACCOUNT 1 are used.

10 Reboot the phone from the Device Configuration web browser.

The phone will appear in the Sphere Administrator application as a SIP phone with a MAC address of <phone’s mac address>@SIP Server.

11 Test the phone for dial tone.12 Use the Sphere Administrator application to modify the station properties as

appropriate.

At this point, the phone will have dial tone, you should be able to place calls, and the device will appear in the Sphere Administrator application. Further configuration is needed to ensure the full functionality of the phone.

U S E D N S S R VRecommended for SIP stations failover/failback using RFC 3263 procedures. If you have multiple Sphericall Managers, it is recommended that you use the DNS SRV.


V O I C E M A I L U S E R I DWhen configured, the user will be able to dial voice mail by pressing the MSG button.


• Type the extension of the Voice Mail server.

S U B S C R I B E F O R M W I


• Set to Yes.

P U B L I S H F O R P R E S E N C E




• Verify that setting is No.

A L L O W A U T O A N S W E R B Y C A L L - I N F OThis setting enables click-to-dial functionality.


• Set to Yes.

S P E C I A L F E A T U R EThis setting enables the third party call control functionality.


• Set to Broadsoft.

B A S I C S E T T I N G S P A G EThe Basic Settings page allows you to configure the speed dial feature.There are seven speed dial fields that can be configured.

To conf igure the GXP-3000 basic set t ings1 Click the Basic Settings page.2 Verify that dynamically assigned via DHCP is selected.




. . .


To conf igure speed dia l1 Type a name in the Name field which is used to identify the person. It will be displayed

on the LCD when pressing the corresponding key.2 Type a phone number or extension in the UserID filed.3 Select the appropriate account in the Account field. This is the SIP account associated

with the number.

To conf igure the t ime zoneFigure 3.33 Basic Settings page

• Select the appropriate time zone to control how the date/time is displayed on the LCD.

A D V A N C E D S E T T I N G S

Q O SLayer 2 and Layer 3 QoS tagging can be set. The values are applied to RTP and SIP packets, but cannot be set independently for each type of packet.IPv4 provisions two methods for the tagging of packets in the IP header, IPv4 Type of Service (TOS) and Differentiated Services (DiffServ). Although they partition the bits differently, both methods use the same octet in the IP header. Tags are provided merely so that the network infrastructure can prioritize traffic based on the interpretation of these tags.Previously, Sphere’s System Initialization Settings supported Layer 3 only. Sphericall now supports both Layer 2 and Layer 3 prioritization. These QoS parameters may be set on the QoS tab of the System Properties.Sphere’s QoS settings are system wide in scope, but please note: any Layer 2 settings that are set on the Sphericall Manager, must also be set at the driver level of the Sphericall Manager, and must also be set on each and every endpoint on the system in order for it to effectively work throughout the system.The Layer 3: QoS field defines the Layer 3 parameter which can be used for IP Precedence or Diff-Serv or MPLS. The default value is 48.The Layer 2: QoS field contains the value used for layer 2 VLAN tagging. The default setting is blank.


N O K E Y E N T R Y T I M E O U TIf Send is not entered after dialing a phone number, the GXV-3000 will push the call out.




L O C A L R T P P O R TThis parameter defines the local RTP-RTCP port pair the GXV-3000 will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.


U P G R A D E SGrandstream phones can be configured to use either TFTP or HTTP for upgrades. For use with Sphericall Managers the phone should be configured to use TFTP.

To upgrade the Grandstream GXV-3000Using the web interface:• Select the Advanced Settings tab and enter the IP address of the Sphericall

Manager that is running the TFTP server. Note: Sphericall Managers are equipped with TFTP servers.

Figure 3.37 Grandstream GXV-3000 Advanced Setting for Firmware Upgrades

For ease of management, it is probably easiest to have all phones use the same IP address for image downloads, although potentially any phone could be configured with the IP address of any manager if the administrator wants to implement load sharing across the TFTP servers.The Grandstream GXV-3000 attempts to locate the following files (located in Program Files\Sphere\Images):• boot64.bin• load64.bin• gxv3000a.bin• ring1.bin• ring2.bin• ring3.binIf the phone is unable to find the TFTP server or one of the image files, the phone aborts the TFTP transfer and simply uses the image stored in flash. The phones will take slightly longer to boot when image files are located in the TFTP server directory. Depending on the connection speed and size of the image file, Grandstream states


. . .


that the boot process may take up to 5 minutes. If the boot times becomes an issue the administrator may remove the files from the Images directory. However, if the files are removed any new Grandstream phones added to the system may not boot up with the image shipped and compatible with the Sphere system.In order to accomplish an upgrade. the Sphere administrator is required to reboot each Grandstream phone either via the web interface, phone keypad or, power cycle.

S P H E R E S Y S T E M U P G R A D E SUnlike MGCP IP phones, SIP phones need to be upgraded manually. The Sphere Administrator application does not prompt the Sphere administrator to restart the SIP phones after the Sphericall upgrade (even if a newer SIP firmware is available). Most SIP phones have TFTP clients to download the firmware (to the phone). The Sphericall upgrade program copies the latest firmware for all these phones in to the \sphere\images directory. Only a manual reboot of SIP phones is required to upgrade them to their respective latest firmware. Grandstream phones will restart only if the Special feature option in Account page is set to "Broadsoft." Only BT100, GXP2000 and GXV3000 phones will be allowed to remote-reboot from the Sphericall Administrator application.The Sphere Administrator should pay close attention to both Sphere Release Notes and Sphere System Requirements which mention the firmware revision of each supported SIP phone (and if the version has changed from the previous release).


S I P P H O N E SSection V - Polycom SIP Phone Configuration

S E C T I O N V - P O L Y C O M S I P P H O N E

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C O N F I G U R A T I O N

Please verify Sphericall and Polycom version compatibility in the Sphere System Requirements document. Verify with the System Requirements document the compatability of Polycom phones (IP30x, IP430, IP50x, IP550, IP601, IP650, and the IP4000). The following documentation can be applied to any of these SIP phones.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• Verify Sphere System Requirements for the appropriate Polycom phone and interoperability with Sphericall.• If you deploy SIP2.2.0 software release for Polycom or higher, the 300 and 500

will not have feature support. Most 300 and 500 phones are running MGCP, but those who may be converting from MGCP to SIP could run into this problem if/when the phones are flipped to SIP.

• Sphere system should be installed, configured and tested as fully functional.• Refer to the Polycom User Manual for installation planning, setup, package

contents, safety and conditions of use.• Verify the Polycom phone keycap labels that are appropriate for this application.• Customization of microbrowser information on some units can be made by

developers. Sphere has a document available for developers in a Software Development Kit (SDK) that specifically addresses support for microbrowsers.

• The Polycom SIP IP650 can support the wide band CODEC G.722. G722 should be higher priority than PCMU in both the LAN and WAN preferred CODEC lists.


. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W

Sphericall IP phone support is being divided into tiers. For tier 1 phones, Sphericall will support native features of the phone. In addition, Sphericall will develop tools that aid in the commissioning, configuration and maintenance of the phone. Each tier 1 phone family will have their own test plan created specifically to test features supported by the phone.Tier 2 phones are not supported to the same depth as tier 1. Tier 2 phones are functionally tested in a Sphericall environment to verify compatibility. Tools for commissioning and configuration are supplied by the phone vendor and are not integrated within Sphericall.


. . .


This document describes the commissioning, configuration and maintenance of the tier 1 Polycom SIP phone family which includes the IP320, IP330, IP430, IP501, IP550, IP601, IP650, and the IP4000.

S T R U C T U R E O F P O L Y C O M C O N F I G U R A T I O N F I L E SThe Polycom SIP release of software is shipped with two binary images (bootrom.ld, sip.ld) and four default configuration files (sip.cfg, 000000000000_sip.cfg, phone1.cfg and directory-000000000000~.cfg). The image files are to remain untouched but the configuration files can be copied, renamed and massaged.Polycom allows configuration files to be renamed from their defaults. Specific phone configuration files will have a naming convention of <MAC_address>[-_]<config_type>.cfg or xml. Having the MAC address as a file prefix allows for better sorting within the configuration directory.

Figure 3.38 Polycom SIP Phone Files

D E S C R I P T I O N O F P O L Y C O M F I L E S

000000000000.cfgThe 000000000000.cfg file is the default master configuration file. If a MAC-specific master configuration file does not exist, the phone will download this file. The 000000000000.cfg file contains a list of other files (application image and configuration files). No configuration parameters are to be entered into this file.


Polycom has two versions of this file: one for SIP and the other for MGCP. The Sphericall MGCP implementation currently relies on this file being the MGCP version. Since each Polycom SIP phone must have a unique registration file, the 000000000000.cfg file will be left as the MGCP version.The Polycom SIP version of the 000000000000.cfg file is being renamed to 000000000000_sip.cfg so it is not confused with the MGCP version.

<MAC>.cfgThe <MAC>.cfg file (e.g. 0004f205b4f6.cfg) is the master configuration file for a specific phone. If a MAC-specific master configuration file does not exist, the phone will download the 000000000000.cfg file. The <MAC>.cfg file contains a list of other files (application image and configuration files). No configuration parameters are to be entered into this file. The <MAC>.cfg file is created by first using the SIP version of the 000000000000.cfg file (000000000000_sip.cfg) as a template. The file list is then updated to include configuration files that override some of the parameters from the default files. E.g.<?xml version="1.0" standalone="yes"?>







<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone_cust.cfg, 0004F211CE9E_phone_overrides.cfg, Spanish_Spain.cfg, phone1.cfg, sip_overrides.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/>

The order of the file list is important. The IP phone parses the file from the beginning of a line to the end. The first occurrence of a parameter is used, thereby, overriding any other occurrence of the same parameter.This file and the <MAC>_phone_overrides.cfg file can be recreated from information stored in the configuration database.

sip.cfgThe sip.cfg file contains basic operation parameters that are common to all Polycom SIP phones. This file is provided by Polycom and will not be modified by Sphere.

sip_overrides.cfgThe sip_overrides.cfg file is a Sphericall-created file that contains sip.cfg override parameters. This file has parameters that may either be administratively controlled, defaulted to something different than the values in sip.cfg or both. The parameters in sip_overrides.cfg that are administratively controlled are also stored in the Sphericall configuration database.

<language_country>.cfgThe <language_country>.cfg file is a Sphericall-provided file that contains call progress parameters unique for a specific country. The <language_country> portion of the name will correspond to the sub-directory names of the Polycom SoundPointIPLocalization folder.

. . .


phone1.cfgThe phone1.cfg file is a Polycom-provided template that contains parameters unique for a particular phone. This file will not be modified by Sphere.

phone_overrides.cfgThe phone_overrides.cfg file is a template file from which <MAC>_phone_overrides.cfg files are created. Phone_overrides.cfg contains all the phone-specific parameters that can be manipulated using Sphericall via the <MAC>_phone_overrides.cfg files. This file contains elements of both sip.cfg and phone1.cfg parameters.

<MAC>_phone_overrides.cfgThe <MAC>_phone_overrides.cfg file contains all the parameters for a particular phone that can be manipulated using Sphericall. The parameters in <MAC>_phone_overrides.cfg are also stored in the Sphericall configuration database.

phone_cust.cfgThe phone_cust.cfg is the repository for settings made by the customer that should be applied to all Polycom SIP phones. Except for an XML header, this file will be empty on new installations. Sphericall upgrades will not modify this file.A reference to phone_cust.cfg will be placed in all <MAC>.cfg files in such a way as to override parameter definitions in other configuration files.

<MAC>_phone_cust.cfgThe <MAC>_phone_cust.cfg is the repository for settings made by the customer for a particular phone. Since the majority of the time customer overrides won't be used/ needed, this file is only created and deleted on-demand. This file's existence and contents can be determined from the Sphericall Administrator application.

<MAC>-phone.cfgThe <MAC>-phone.cfg is created by the phone and uploaded to the server whenever the user makes a configuration change using the phone's menu key or web interface. Changes made via the phone or web interface override all other matching setting within the configuration files.Past Sphericall implementations for configuring the Polycom MGCP phone manipulated this file.The Polycom phone provides a facility ("Reset to Default") through its Menu key to reset user changes. The Sphericall Administrator application displays the contents of this file. It also provides a command for deleting the file's contents.Note: When upgrading a Polycom SIP phone from v5.1 to v6.0, each phone-specific file must be deleted before the upgrade. Once this is done, the phone can be configured in the Sphericall Administrator application (in Sphericall v6.0).



000000000000-directory.cfgThe 000000000000-directory.cfg file contains the default local contact directory. If a <MAC>-directory.cfg doesn't exist the phone will use this file. This is an optional file and will be managed by the customer.

<MAC>-directory.cfgThe <MAC>-directory.cfg file contains the local contact directory for a specific phone. Polycom provides a "Directory" button to access and edit local contacts. Changes made by the phone can be used immediately and saved to this file without requiring a reboot. If this file is changed outside of the phone, the phone must be rebooted to load the changed file. This is an optional file and will be managed by the customer.

S Y S T E M O V E R R I D E SSystem-wide parameters that can be managed by Sphericall are stored in sip_overrides.cfg. This file also contain parameters meant to override Polycom defaults, but are not managed by Sphericall.

D I G I T M A PDigit maps are used to eliminate the need to press the send button on the Polycom SIP phone. A digit map override allows extension dialing without having to wait for a digit timeout. Polycom phones follow RFC 3435 "Media Gateway Control Protocol (MGCP)" for its digit map syntax. The default digit map assumes the following::• The North American PSTN numbering plan• A minimum of 3 digit extensions• A PSTN access digit of '8'• That 0 is a valid extension

The digit map can be accessed/overridden from the phone_cust.cfg file. The default digit map for Polycom SIP phones is: digitmap dialplan.digitmap="8[2-9]11|80T|8011xxx.T|8[0-1][2-9]xxxxxxxxx|8[2-9]xxxxxxxxx|0|[1-7]xx.T|9xx.T|*7[246]xxx.T|*7[3789]|*75x.#x.T|*8x|*91x|*9[25]|*9[346]xxx.T|*971[01]|*99x.#xxx.T"The default digit map segments have the following meaning:

8[2-9]11 Outside sevice -X11 e.g. 411, 911

80T Outside operator

8011xxx.T International calls

8[0-1][2-9]xxxxxxxxx Long distance calls

8[2-9]xxxxxxxxx 10 digit dialing

0 Inside operator

[1-7]xx.T Extensions - Minimum 3 digits


. . .


Refer to Polycom Technical Bulletin 11572 for dial plan configuration information. Any changes in the digit map will require all affected Polycom SIP phones to be rebooted.The digit map defaults can be tailored per country and stored in the <language_country>.cfg files. Polycom doesn't provide any support.

Q O S P A R A M E T E R S802.1p/Q Ethernet user priority settings and IP Type of Service (TOS) settings are supported. These settings will control both Polycom SIP and MGCP endpoints.

R T P P A Y L O A D F O R D T M D I G I T S ( R F C 2 8 3 3 )A common RTP payload type for DTMF digits setting is supported between Polycom SIP and MGCP endpoints. If RFC 2833 functionality is enabled, the payload type will be 101.

L O C A L I Z A T I O N

L A N G U A G EPolycom phones can be configured to display other languages besides English. The implementation is extensible where other languages can be added by third parties. Sixteen different languages are currently being shipped which include: English / US, English/UK, French/France, German/Germany, Italian/Italy, and Spanish/Spain. The other languages that Sphere is interested in, but not currently supported, are French/ Canadian and Spanish/Mexico.• The bootrom messages are not localized and are displayed in English.• The language of the default localization setting is set as the language for each new

phone.

|9xx.T| If 911 is setup as an emergency number for the telephony area of the phone then dialing 911 will progress as an outbound emergency call.If internal numbers can start with a 9 then this rule allows 3 or more digit extensions to be dialed.

*7[246]xxx.T Star code - forwarding, transfer to VM, intercom

*7[3789] Star code - deactivate forwarding, drop last call, DND on, DND off

*75x.#x.T| Star code - assign account code

*8x Star code - park at zone

*91x Star code - unpark at zone

*9[25] Star code - group pickup, conference

*9[346]xxx.T Star code - pickup at extension, blind transfer

*971[01] Star code - MWI on, MWI off

*99x.#xxx.T Star code - user authentication



• The system override for Polycom SIP phone language will be configured via the Sphericall Administrator application’s Localization Settings.

C A L L P R O G R E S S T O N E SCall progress tones, for a specific phone, are determined from the country template chosen from the Sphericall Administrator application’s Localization Settings facility. A single set of call progress tones are supported per country.

I D L E D I S P L A Y S C R E E NThe idle display is the area of the phone where call information is presented during a call. When no calls are in progress, Polycom SIP phones have the option displaying web browser-based content, an animated bitmap or the date and time. When a call is not in progress the default idle display will contain the following web browser-based information: • Station's line name (50 chars max)• Organization name (50 chars, 2 lines max)• Idle display message (100 chars, 2 lines max) - This is an optional message that

can convey a variety of information such as daily or important events (meetings, visitors, holidays), company or customer data (sales figures or goals), or advertisements (cafeteria specials).

The URL for accessing the idle screen defaults will be setup as http://PrimaryServerFQDN/polycom/home.html. The administrator may change this or delete it. If the Idle Screen URL is deleted, a bitmap of the SPHERE is displayed.

S E R V I C E S B U T T O N ( M I C R O B R O W S E R )The Services button provides a means to access web services. Pressing the Services button will invoke a URL defined in the <MAC>_phone_overrides.cfg configuration file. This URL can point to any service that returns “properly limited” HTML, but unless the writer of a web service understands the limitations and nuances of supporting the Polycom micro-browser it is recommended that only the preconfigured Sphericall web services is used. By default, Sphericall web services URL presents on the browser screen a list of call control and data viewing options. Sphericall provides default web services that include:• Re-direct a call to voice mail• Record a connected call• Display and join a conference call• Directory lookup• Display national news• Display stock indices

Note: The Web Services/Microbrowser feature is supported on the Polycom IP30x, IP430, IP50x, IP550, IP60x, and IP650 SIP phones


. . .


C O N V E R T P O L Y C O M M G C P T O S I P C O M M A N DA new command, "Convert Polycom MGCP to SIP," is part of the toolbar on the Stations page of the Sphericall Administrator application. This command aids in converting existing Polycom MGCP phones to Polycom SIP phones. To activate the toolbar button, the user must select one or more Polycom MGCP phones from the Stations page. The user can optionally select a Polycom SIP phone to be used as a template during the conversion process. The non-phone specific Polycom template parameters will be copied to the converted phone (i.e. the AoR, line label, phone-managed overrides and custom file parameters are not copied).

Figure 3.39 Polycom MGC to SIP Command

A D D I N G A P O L Y C O M S I P P H O N E T O A S T A T I O NIn order for Polycom SIP phones to register with the Sphere system, they must be configured in the Sphericall Administrator application.Sphericall Administrators must pay careful attention to the required procedures and mechanisms to support and configure the Polycom XXX. These procedures will help ensure that the configuration files are maintained properly.• YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT

TO THE SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints.

To add a Polycom SIP phone to the Sphere system1 Using the supplied power adapter, power the phone up.2 Connect the phone to the network using the supplied ethernet cable.

From the Sphericall Administrator application:3 Click the Stations tab.4 Highlight any Sphere hub (MG).



5 Right-click and select Add\Add Polycom SIP Phone.

Figure 3.40 Add Polycom SIP Phone window

6 Enter the phone’s MAC address in the MAC Address field.

The MAC Address can be filled in using the keyboard or bar code reader. It is limited to 12 hexadecimal characters.

7 Enter the extension that will become primary for this phone in the Extension field. If necessary, click Add to create a new extension for this phone.

The Extension entered must already exist within the system. 8 Select from the available, supported Polycom SIP phones in the drop down box.

The phone type choices are derived from the supported user agent strings. Any supported user agent that contains "Polycom" will be listed.If you select an incorrect Polycom phone, the system corrects your choice.

9 Click OK.

The Station Properties window appears.10 Click the Polycom tab for further configuration.


. . .


Figure 3.41 Station Properties window

In the Polycom SIP Phone Settings area:11 Verify the Address of Record.

This is auto-generated from the station’s primary address and the system’s DNS domain.

12 Type the text that appears next to the line key on the phone.

This label is limited to 12 characters and defaults to the primary extension number.13 Type the line key for the number of phone keys that should be bound to the phone’s

registration.

If Line Keys is set to greater than 1, the phone will be set to limit the calls per line to 1.In the Server area:

14 Enter the DNS address of one or more MGCs that accept SIP registrations.

This may be an IP address, host name, or FQDN DNS address. For redundancy and failover, it is recommended that this field be an FQDN DNS address that contains a prioritized list of SRV records. Systems that want to share MGCs across SIP phones can create multiple DNS addresses, each with a different prioritized SRV record list.If you have multiple Sphericall Managers, it is recommended that you use the DNS SRV.

15 Select the appropriate transport settings in the Transport and Port fields

This is the interaction between these fields and the server address.

Note: Refer to the Polycom administration guide for settings.

Note: Do not enter a setting in the Port field if NAPTR or SRV records are used.

turned off by defaultoverrides systemforwarding

<MAC>-phone.cfgUpdated in

Updated in

<MAC>_phone_cust.cfg



In the Phone Managed Override Parameters area:16 Verify that the appropriate file is being used.

Parameters that are set at the phone or using the Polycom web interface are saved in an XML configuration file (<MAC>-phone.cfg). The Phone Managed Override Parameters field displays the contents of this file, if it exists. Pressing the Reset button will clear the contents of this file.In the Custom File Parameters area:

17 Verify that the appropriate file is being used.

This field displays the contents of the <MAC>_phone_cust.cfg file, if it exists. The file may also be deleted by pressing the Delete File button.In the VM Address field:

18 Enter the address for contact when the Voice Mail Messages button is pressed.

In the Idle Screen URL field:19 Enter a URL for accessing the phone’s micro-browser feature.

The Idle Screen URL is the address the phone’s micro-browser periodically contacts to update what is on the idle display. The Refresh Rate (in seconds) is used to determine how often the idle display is updated. The Refresh Rate defaults to 15 minutes (900 seconds).In the Services URL field:

20 Enter a URL for contact when the Services button is pressed.

The phone changes its display to act as a browser, using the arrow keys to navigate.21 Press the DNS Test button to verify the DNS address of the Server and Outbound Proxy.

DNS Test will query DNS resolving NAPTR records. The DNS "Server Address" and "Outbound Proxy Address" will always be checked for A records and SRV records (records along with priorities and weights), as well as NAPTR records. It will be up to the administrator to interpret the results and what affect, if any, it has on the Polycom SIP phone.The DNS NAPTR query is not attempted when the Sphericall Administrator application is run under Vista.

22 Click OK.

A warning message window will remind you to bind a user to the primary extension when a Polycom SIP phone is created.

D I R E C T O R Y A N D S I D E C A R S E T U PPolycom IP phones have directory and sidecar/expansion module (compatible with the IP601 and IP650) features. The <MAC>-directory.xml file contains the local contact directory for a specific phone. Changes made by the phone can be used immediately and saved to this file without requiring a reboot. If this file is changed outside of the phone, the phone must be rebooted to load the changed file.

To add directory informat ion to the Polycom phone1 Copy/paste <MAC>directory.xml and rename as

<the phone’s MAC address>-directory.xml.2 Open the file in Notepad and make modifications to the fields described in the following

table.


. . .


Table 3.5 Directory Information table

Element Permitted Values Interpretation

fn UTF-8 encoded string of up to 40 bytes first name

ln UTF-8 encoded string of up to 40 bytes last name

ct UTF-8 encoded string containing 8 digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL

contact

Cannot be null or duplicated; is used by the phoneto address a remote party in the same way that astring of digits or a SIP URL are dialed manuallyby the user. This element is also used to associateincoming callers with a particular directory entry.

sd Null, 1 to 9999 speed-dial index

Associates a particular entry with a speed dial bin for one-touch dialing or dialing from the speed dialmenu.

rt Null, 1 to 21 ring type

When incoming calls can be associated with a directory entry by matching the address fields, this field is used to specify ring type to be used.

dc UTF-8 encoded string containing 8 digits (the user part of a SIP URL) or a string that constitutes a valid SIP URL

divert contact

When incoming calls can be associated with a directory entry by matching the address fields, this field is used

ad 0,1 auto divert

If 1, automatically diverts callers that match the directory entry to the address specified in divert contact.

ar 0,1 auto reject

If 1, automatically rejects callers that match the directory entry.

bw 0,1 buddy watching

If 1, add this contact to the list of watched phones.



E X P O R T P H O N E D I S T R I B U T I O N M A PThe Sphericall Administrator has a Export report that allows an adminstrator to distribute newly commissioned phones to the appropriate users.If a system administrator purchases 25 phones, commissions 25 phones by creating line, adding address and adding the user, then the next step is to install the 25 phones on the appropriate desks. The only information on the phone box is the MAC address. Therefore, the system administrator either needs to provide a mapping between MAC address and extension/user in order for the phones to be delivered to the correct offices and or the correct user(s).

To export phone distr ibut ion map1 From the Sphericall Administrator application.2 Click on File/Export.3 Select Phone Distribution Map from the Export dropdown field.

Figure 3.42 Phone Distribution Map

4 Select the fields of information you require or this report.5 Select Export.6 Enter the name of the file and location for the .csv file that will be generated.7 Save.8 Open the file using Excel.

Optimally the report would provide First Name and Last Name and possibly Username along with the MAC address and primary address. The MAC address will appear in the Phone S/N (Mac) field.

bb 0,1 buddyblock

If 1, block this contact from watching this phone.

Element Permitted Values Interpretation


. . .


U P G R A D E S

B A C K U PDuring a Sphericall upgrade, the Polycom root directory is backed up into "backup" directory.

C O N F I G U R A T I O N F I L E U P D A T E SDuring a Sphericall upgrade the following file manipulation occurs:• The phone_overrides.cfg and <language_country>.cfg files will be replaced.• The <MAC>_phone_overrides.cfg files are recreated by using the

phone_overrides.cfg file as a template and then overlaying phone-specific database data.

• The <MAC>.cfg files are recreated from Sphericall data.• The sip_overrides.cfg file will be replaced with the latest Sphericall version and

then updated with system-specific Sphericall controlled data.• The sip.cfg and phone1.cfg files will be overridden with the latest Polycom version.• The <MAC>-directory.cfg files are not modified• The 000000000000-directory.cfg file is not modified• The <MAC>-phone.cfg files for Polycom SIP phone will not modified.• The <MAC>_phone_cust.cfg and phone_cust.cfg files will not be modified by

Sphericall.

U P G R A D I N G T O A N E W P O L Y C O M R E L E A S EIt is not recommended for customers to upgrade Polycom phone application software outside of a Sphericall release for the following reasons:• New Polycom software should be system tested at Sphere to insure compatibility.• Configuration file changes may be incompatible with current administration tools.For new Polycom software that is released between Sphericall releases, Sphere will make a statement of support within four weeks.

F T P S E R V I C EThe Polycom phones require a local user account on the FTP server to have access to images and configuration files. The default FTP user account name is "PlcmSpIp" with a password of "PlcmSpIp”. If the FTP server resides on a Microsoft server, the password will not be compliant with Microsoft's default password complexity rules.To help in debugging phone issues, a single file directory is used to hold images, configuration and log files.

L O G F I L E SPolycom SIP phones will create or rewrite a boot log (e.g. 0004f21114e4-boot.log) whenever the phone is restarted. Boot log files are useful when debugging image and configuration file issues.



Polycom also provides uploading of application log files (e.g. 0004f21114e4-app.log). Empirical evidence has indicated the phone does not create the application file but it must exist before the phone will write to it. During commissioning of the phone, the Sphericall Administrator will create a blank application log file if optioned to do so. Application log files are useful in debugging content issues within configuration files.The FTP server must be configured to allow write access for log files to be uploaded.

S P H E R E S Y S T E M U P G R A D E SUnlike MGCP IP phones, SIP phones need to be upgraded manually. The Sphere Administrator application does not prompt the Sphere administrator to restart the SIP phones after the Sphericall upgrade (even if a newer SIP firmware is available). Most SIP phones have TFTP clients to download the firmware (to the phone). The Sphericall upgrade program copies the latest firmware for all these phones in to the \sphere\images directory. Only a manual reboot of SIP phones is required to upgrade them to their respective latest firmware. The Sphere Administrator should pay close attention to both Sphere Release Notes and Sphere System Requirements which mention the firmware revision of each supported SIP phone (and if the version has changed from the previous release).


. . .

. .S I P P H O N E SSection VI - UTStarcom F1000G/F3000 SIP Phone Configuration

S E C T I O N V I - U T S T A R C O M F 1 0 0 0 G / F 3 0 0 0 S I P

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P H O N E C O N F I G U R A T I O N

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W

The F1000/F3000 Wi-Fi handset is a device that expands the Sphere system over the enterprise wireless IP network for enhanced mobility of IP communications.Consider the F1000’s potential as an effective way to communicate by allowing mobile users, within the enterprise, access to the IP PBX functionality. For example, the flexibility to take the phone “out on the shop floor.”The F1000/3000 includes the following features:• Caller ID• Blind Transfer• Attended Transfer• 3 Party Conferencing• Sphericall Voice Mail Access

Note: The G in the UTStarcom F1000/3000 represents an 802.11g radio. All other aspects of the phone’s form, fit and function remains the same as the F1000/3000.

T H I N G S T O C O N S I D E R Before attempting to install and configure a UTStarcom F1000/3000 phone, you will need to consider the following:• Do you have rights to the DNS Server?• Are you familiar with your organization’s wireless security standards?



• Do you have Admin rights to access the Sphericall Administrator application on the Sphericall Manager?


This document is divided into three sections. Each is necessary for successful installation and configuration of the F1000/3000 phone.

A . W I F I A N D N E T W O R K S E T T I N G SYou must define certain network parameters in order for the wireless network to recognize the F1000/3000.

Note: Ensure that the F1000/3000’s battery is adequately charged before proceeding with configuration.

On the F1000/3000—To conf igure WiFi and Network Set t ings1 Press the Menu key.2 Using the WiFi down arrow key, scroll to WiFi settings.3 Click OK.4 Go to WiFi Config and click OK.5 Select AP Profile. SSID1 is one of the default profiles associated with the F1000/3000.

You will need to rename this to the name of the wireless network access point with which the F1000/3000 will connect.

Your organization may have multiple wireless network access points. You may need to add multiple profiles to ensure phone connectivity throughout the network.

6 Select SSID1 and click Edit.7 Click OK.8 From the phone keypad, enter the name of the appropriate wireless network access

point.9 Click OK.

10 Scroll to Security Mode and click OK.11 Select WEP Key and click OK.

Short for Wired Equivalent Privacy, a security protocol for wireless local area networks (WLANs) defined in the 802.11b/g standard. WEP is designed to provide the same level of security as that of a wired LAN. WEP aims to provide security by encrypting data over radio waves so that it is protected as it is transmitted from one end point to another.Your organization’s wireless security standards may not require WEP.

12 Select the appropriate security mode for your organization and click OK.

Create a DNS Record on the DNS server

A. Configure WiFi and network settings from each F1000/3000 phone

B. Access and configure phone settings from a web interface

C. Configure initialization settings for F1000/3000 and Sphericall Voice Mail

D. Firmware Updates


. . .


13 If necessary, enter the WEP Key password and click OK.

Your wireless network administrator may need to supply you with the WEP KEY password.

14 Click OK on the Activate menu.15 After completing WiFi settings, switch off the phone and then switch it on, in order for

the changed WiFi settings to take effect.16 To retrieve the F1000/3000’s IP Address, press Menu\WiFi Settings\Network Info.

Make note of the phone’s IP address, as you will need it for web configuration.

B . F 1 0 0 0 / 3 0 0 0 W E B C O N F I G U R A T I O N

Web Browser—To conf igure the F1000/3000 f rom the web1 From a web browser, use the F1000/3000’s IP address to open a web page for the phone.2 At the prompt, enter the following information:

a. User Name: userb. Password: 888888

3 From the navigation pane on the left side, select the User Menu\SIP and RTP Config option.

Figure 3.43 F1000/3000 SIP Web Configuration Web Page



4 Key in the following SIP setting information:a. SIP Terminal Use Outbound Proxy: Select the Yes radio button.b. SIP Terminal User Register: Select the Yes radio button.c. SIP Outbound Server Domain Name: This is the Sub Domain d. SIP Outbound Server IP Address: The address of the Sphericall Manager with which

the endpoint will register.e. SIP Outbound Server Port: The Sphericall Manager uses port 5060, which is the

standard port for SIP traffic.f. SIP User Name: The user name portion of the SIP address for this endpoint.g. SIP Terminal Port: Again, the Sphericall Manager uses port 5060, which is the

standard port for SIP traffic.h. SIP Terminal Use Null Packet: Select the No radio button.i. SIP Terminal Use DNS: Select Both Register And SIP Proxy Servers radio button.j. DNS Query Type: Select the SRV radio button. This queries the previously-created

DNS Record.k. Set Registration Duration: Set value to 3600 seconds. Standard SIP values range

between 1800 and 3600 seconds.l. Terminal Audio RTP Port: Set value to 11110.m. Terminal Audio Packetize Time: Set value to 20 milliseconds.

5 Click Submit.6 Click Reboot.

Note: Upon reboot, the F1000/3000 phone will initialize and the phone will appear in the Sphericall Administrator application.

A D D I T I O N A L W E B C O N F I G U R A T I O NUser settings that include how time, password setup, and display name appear on the F1000/3000 can be configured in this window.


. . .


Web Browser—To conf igure F1000/3000 user set t ings1 From the navigation pane on the left side, select the User Menu\User Settings option.2 Key in the following user information:

Figure 3.44 F1000/3000 User Settings Web Configuration page

a. Display Name: Type in the name as you want it displayed on the LCD of the F1000/3000.

b. Autoscan Flag: Scans the wireless network for seamless transition from one network access point to another.

c. Autoscan Interval: The frequency at which the F1000/3000 scans for a wireless network access point.

d. Time Zone: Select the time zone for its appearance on the F1000/3000.e. DST Setting: Option to enable/disable Daylight Saving Time.f. User Password: Ability to change the default password.g. Re-enter User Password to Confirm: Confirmation of new password.

3 Click Submit.

Web Browser—To conf igure wire less access point set t ingsInitially, you establish the AP profile (see WiFi and Network Settings section) and security standards from the F1000/3000’s keypad. However, once you have access to the web interface, you have the ability to edit and add network and security standards from the web. Upon submission and reboot, you can see the changes on the F1000/3000.

1 From the navigation pane on the left side, select the User Menu\Wireless AP Config option.

2 Make the necessary changes to wireless settings.3 Click Submit.



Figure 3.45 F1000/3000 Wireless Access Web Configuration page

C . F 1 0 0 0 / 3 0 0 0 A N D S P H E R I C A L L V O I C E M A I LIn order for the F1000/3000 to integrate properly with Sphericall Voice Mail, a setting must be added in the Sphericall Administrator application.

Spher ical l Manager—To conf igure Spher ical l Voice Mai l set t ingsFrom the Sphericall Administrator application:


2 Click F5 or select File\Refresh to make the recently-installed F1000/3000 viewable on the Sphericall Administrator application.

3 Click the Stations tab.4 Highlight the F1000/3000’s station.

Hint: If you use the “Sort By: Device Type” feature, the phone will reside under the SIP Phones folder.

5 Right-click on the F1000/3000 and select View Properties.6 Click the Settings tab.7 Click Add.8 Select RTP receive packet size maximum from Sphericall Media Server.9 Set value to 20.

10 Click OK.

D . F I R M W A R E U P D A T E SNote: Refer to the System Requirements documentation and the Software

Compatibility table to ensure the F1000/3000’s firmware version is compatible with the Sphere system.

To update the F1000/3000 f i rmware vers ion1 Open the following folder from the Sphericall software CD:

\Server\Data\UTStarcom


. . .


2 Unzip the F1000/3000 TFTP Package.zip file.3 Run fwupgrade.exe to push the latest image down to each individual phone.

Note: This executable must be run for each F1000/3000 phone that requires an update.

Figure 3.46 F1000/3000 Upgrade Utility

4 Enter the IP Address and click UpLoad.5 When the utility completes the upload process, the F1000/3000 should reboot and

recognize the new image.

S P H E R E S Y S T E M U P G R A D E SUnlike MGCP IP phones, SIP phones need to be upgraded manually. The Sphere Administrator application does not prompt the Sphere administrator to restart the SIP phones after the Sphericall upgrade (even if a newer SIP firmware is available). Most SIP phones have TFTP clients to download the firmware (to the phone). The Sphericall upgrade program copies the latest firmware for all these phones in to the \sphere\images directory. Only a manual reboot of SIP phones is required to upgrade them to their respective latest firmware. The Sphere Administrator should pay close attention to both Sphere Release Notes and Sphere System Requirements which mention the firmware revision of each supported SIP phone (and if the version has changed from the previous release).


S I P P H O N E SSection VII - SIP Phone Compatibility and Capability with Spherciall Desktop

S E C T I O N V I I - S I P P H O N E C O M P A T I B I L I T Y A N D C A P A B I L I T Y W I T H S P H E R C I A L L

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D E S K T O P

The following SIP phones have features that are compatible with the Sphericall Desktop.

Table 3.6 SIP Phone/Sphericall Desktop Compatibility table

Desktop Feature ManufacturerPolycom IP4000

ManufacturerPolycom

IP301, IP501IP601, IP430IP550, IP650

ManufacturerAastra

9112i, 9133i480i, 480iCT

Controlled by Sphericall Desktop

Yes Yes Yes

Click-to-Dial Yes Yes Yes

Click-to-Answer Yes Yes Yes

Transfer (Blind: connected and ringing)

Yes Yes Yes

Transfer (Attended) Yes6 Yes6 Yes6

Hold Yes Yes Yes

Pick-up Yes Yes Yes

Park Yes Yes Yes

Do Not Disturb Yes5 Yes5 Yes5

Three Way Call No4 No4 No4

Call Recording Yes Yes Yes

Drag-n-drop MeetingHub Conference (blind transfer to a conference extension)

Yes Yes Yes

Precedence Dialing (CallNOW/MLPP)

Yes4 Yes4 Yes4

Phonebook Directory Yes Yes Yes

Recent Calls List Yes Yes Yes

Dual Mode (Softphone and Computer Telephony)

No1 No1 No1

Media Switching (PC to Phone)

No2 No2 No2

Video Calling No3 No3 No3

Instant Messaging - Send & Receive

Yes1 Yes1 Yes1


. . .

. .S I P P H O N E SSection VIII - SIP Phone Administrative Star Codes

S E C T I O N V I I I - S I P P H O N E A D M I N I S T R A T I V E

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S T A R C O D E S

Table 3.7 Administrative Star Codes

Presence Agent Yes3 Yes3 Yes3

Manage Forwarding Profiles

Yes Yes Yes

Star Code Support No Yes 480i Yes; Others No

PIN Code Support No No No

Yes1 Requires that IM be disabled on the IP phoneYes2 Requires that IM be disabled on the IP phoneYes3 Requires that Presence be disabled on the IP phoneYes4 Note: MLPP is not supported on the SIP phones.Precedence receiving is not possible using a SIP phone Yes5However, the local DND indicator will not be reflected on the IP phone. If DND is set at the phone, the Sphericall Desktop will not reflect this.Yes6However, the first call must be placed on hold before initiating the consult call. Simply dialing the second call will not currently work.No1When the Sphericall Desktop is associated with a SIP phone, it can not be used as a softphone to terminate media.No2Due to No1, media cannot be terminated at the PC.No3SIP phones require that media only be terminated in one place (video and audio together).No4Three way conferences can not be established by the Sphericall Desktop that is associated with a SIP phone.

Administrative Action Sphericall Star Code Explanation

Forward to user’s voice mail (IP and and analog phone)

*74 + extension Allows users the ability to forward phone calls directly to another user’s voice mail, thus bypassing the ringing of the user’s extension.

Call detail information sent to call detail record

*75+ PIN (client-assigned account code)+*

The Sphericall Manager sends call detail information to the call detail record.

Intercom from non-intercom phone to intercom enabled phone

*76 + extension Callers without intercom can enter this star code to intercom a phone equipped with intercom.

Desktop Feature ManufacturerPolycom IP4000

ManufacturerPolycom

IP301, IP501IP601, IP430IP550, IP650

ManufacturerAastra

9112i, 9133i480i, 480iCT


S I P P H O N E STroubleshooting SIP Connections

Table 3.8 Diagnostic Star Codes

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . T R O U B L E S H O O T I N G S I P C O N N E C T I O N S

S I PSession Initiation Protocol, SIP, is a protocol for transporting call setup, routing, authentication and other feature messages to endpoints within the IP domain. Within the Sphere system, SIP is used to allow external systems to participate in calls with the Sphericall Manager. The Manager targets the use of SIP for integration with some specific third-party products for integration with the following: two-way calls (Sphericall Manager and external system), calls between two systems placed “on hold,” transfer of calls between the two systems, passing of DTMF digits into the third-party system during a call, and notification of message waiting.The Terminal location logic to accommodate gateways that register multiple trunks from the same IP address. Understanding this logic will help enormously when troubleshooting SIP connection issues.The Sphericall Manager (MGC) uses the following logic to locate a SIP terminal in 6.0. and later:

From Header • Userinfo is compared to the Account field of the Service Provider information. If no

match is found, the Sphericall Manager moves to step 2.

Commission Station *98+ primary extension number

Allows administrators to assign an extension number to a station within a Sphere system.Use the *97 administrative star code to confirm the station’s assigned extension number.Note: A confirmation tone will sound after applying this star code command.

Diagnostic Action Sphericall Star Code Explanation

Forcibily turn MWI lamp “ON”

*9711 This turns on the message waiting indicator on the phone set (analog & IP phones with MWI lamp)

Forcibily turn MWI lamp “OFF”

*9710 This turns off the message waiting indicator on the phone set (analog & IP phones with MWI lamp)

*973-*979 Reserved for future use.

Administrative Action Sphericall Star Code Explanation


. . .

. .S I P P H O N E STroubleshooting SIP Connections

Note: this is the most common way stations are identified, but does not help for trunks since the FROM field contains the caller ID of the incoming call.

To Header • Userinfo is compared to the DID maps configured for SIP trunks. If two trunks have

overlapping DID maps, the Sphericall Manager moves to step 4. If no match is found, the Sphericall Manager moves to step 3.

Contact Header • hostname:port is compared to the Outbound Proxy if configured, otherwise the

Service Provider Domain of the Service provider information.• If the hostname:port is an IP address it is compared exactly to what is configured.• If the hostname:port is not an IP address, a partial compare is performed against

the Service Provider information. For example, the hostname "horatio.rndlab.spherecom.com" would match the Service Provider information "rndlab.spherecom.com".

Note: in both cases the port must match.• If more than one User Agent matches this criterion, or no match is found, the

Sphericall Manager moves to step 4.

Authorization Header • If the request contains an Authorization header, the credentials included in the

Authorization Header are compared against the credentials configured in the Sphericall Admin Authorization window. If no match is found, the Sphericall Manager moves to step 5.

The Sphericall Manager (MGC) challenges the sender to obtain credentials via the Authorization Header.


S I P P H O N E STroubleshooting SIP Connections


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .ATI RG6XX OR IMG6XX 4

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• Verify Sphere System Requirements for RG613 and interoperability with Sphericall.• Verify the required firmware version in the Sphere System Requirements.• The RG configuration is also compatible with the iMG. Wherever you see RG in

these instructions (except for command line interface entries) you may substitution iMG.

Note: ATI manufactured serial cable is required for configuration:Model number: AT-RGCONSOLECABLE-00Part number: 990-11748-00

• Sphere system should be installed, configured and tested as fully functional.• Refer to the AT-RG613/623TX Residential VoIP Gateway User Manual for

installation planning, setup, package contents, safety and conditions of use.

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Figure 4.1 RG613 Back Panel

1 Along with ordering RG613, you must order at least ONE of the Allied Telesyn proprietary serial cables for coding the residential gateway:

Voice PortsTEL 2 TEL 1

10BASE-T/100BASE-TX NETWORK PORTSLAN 3 LAN 2 LAN 1 WAN 1 POWER


AT I R G 6 X X O R I M G 6 X XInstalling

Model number: AT-RGCONSOLECABLE-00Part number: 990-11748-00

2 Prepare network according to Sphere System Requirements.3 Verify that a SNTP (Simple Network Time Protocol) time server is configured.4 Know the required login for working with this gateway:

Login: managerPassword: friend

5 Connect appropriate network cable to appropriate port on RG613.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I N S T A L L I N G

To ver i fy f i rmware vers ion of RG or iMGThe RG configuration is also compatible with the iMG. Wherever you see RG in these instructions (except for command line interface entries) you may substitution iMG.

1 Connect serial cable on RG613 to lap top computer for configuring device.2 Connect power cable to the RG613 unit. Verify it powers up.3 Connect all network connections, there will be a firmware check by the unit later.4 Open a HyperTerminal Session:

Serial Port Info:Bps: 38400Data: 8Parity: NoneStop Bits: 1Flow Control: None

5 Verify that HyperTerminal has an open session.6 Power the unit off and back on again.

You will now see the RG boot process and the firmware will check in.7 Verify the MG version using HyperTerminal.

Note: If the version verified is v1.0, you will need to upgrade the Flash. If the version verified is v2-3 or greater, you may proceed to RG configuration (please verify the current supported version of firmware with the Sphere System Requirements).

To update f lash of RG1 Login: manager

Password: friend2 --> ip set interface ip0 dhcp enabled3 --> dhcpclient update4 --> system config create <filename>.cfg

wait for configuration saved message5 --> system config set <filename>.cfg (as created above)6 --> console enable ati [enter] [enter] 7 [RG ipaddress]> tftp


. . .

. .AT I R G 6 X X O R I M G 6 X XInstalling

8 [RG ipaddress]\ tftp> connect [ip address of sphericall manager to connect to]

9 getflash rg6xx-image-2-2_06-2-4_58.bin (This is an example of the syntax. Verify the current flash image required for this in the Sphere System Requirements)

Wait for the RG to get the image and reboot itself (3-5) minutes.NOTE: do NOT reboot the RG manually at this time. This will corrupt the flash.After the RG has rebooted:

10 Do not close HyperTerminal at this time--it must remain open.11 Open the following folder on the Sphericall DVD:

Server\Data\AlliedTelesyn12 Run the Loader_RG600_2-5_55_10_04.exe to push the latest image down to the RG or

iMG (This is an example of the syntax. Verify the current executable required for this in the Sphere System Requirements).

13 When the window opens during this process, enter the IP address you retrieved from the ip list interfaces step (above).

14 Enter password: friend.15 Click Start.16 Watch the HyperTerminal window to verify the reboot of the RG. This indicates that the

image download is complete (check Sphere System Requirements for current version number).

17 Continue with RG configuration as follows.

To conf igure RGAt the login prompt:

1 Login: managerPassword: friend

2 Verify confirmation that the upload is complete.

From the serial port connection, configure the following:3 ip set interface ip0 dhcp enabled

4 sntp set server ipaddress [SNTP Server IP Address]

5 sntp set timezone cdt(or edt, mdt, pdt, CST, EST, MST PST--choose appropriate time zone)

6 voip mgcp protocol enable

7 voip mgcp protocol set netinterface ip0

8 voip mgcp protocol set profile sphere

9 voip mgcp callagent create sphere contact [mgc ip address to which you want this device to check in]

10 voip ep analogue create tel2 type al-fxs-del physical-port tel2

11 voip ep analogue create tel1 type al-fxs-del physical-port tel1

12 voip ep analogue enable tel2

13 voip ep analogue enable tel1

14 voip ep analogue set tel2 country usa

15 voip ep analogue set tel1 country usa


AT I R G 6 X X O R I M G 6 X XInstalling

16 voip ep analogue set tel2 clip BELL

17 voip ep analogue set tel1 clip BELL

18 voip ep analogue set tel2 codecs g711u,g711a,g729ab,t38

19 voip ep analogue set tel1 codecs g711u,g711a,g729ab,t38

20 system config create <filename>.cfg

This will create the file.21 system config list

This will show file names that were created.22 system config set <filename>.cfg

This will set this file to active.

To set the f lash-hook and on-hook t imingThe following settings should be set at this time to ensure synchronization with the rest of the Sphere media gateways:

1 voip ep analogue set tel1 onhook 1100

2 voip ep analogue set tel2 onhook 1100

3 voip ep analogue set tel1 flashhook 600

4 voip ep analogue set tel2 flashhook 600

5 --> system config create <filename>.cfg

wait for configuration saved message6 --> system config set <filename>.cfg (as created above)

To restar t device and apply changes1 system restart (to restart the device and apply all the changes)

2 Logon to the Sphericall Manager to verify the RG device has checked in under the Stations tab.

3 If you have not already assigned extensions or users to this device, follow configuration with Book 1: Plan and Prepare Sphericall Installation and Book 2: Install and Configure Sphericall.

To update f i rmware af ter upgrading to Spher ical l v6.0 .0 .6Note: this requires RG firmware of 2-5_55_10_04 or greater and only needs to be performed once, after the Sphericall v6.0.0.6 upgrade. Please always consult the Sphere System Requirements for version compatibility.

1 The following commands must be entered in the RG configuration console (or via Telnet) for the analog ports to work propertly :

2 voip mgcp protocol set wildcard enable

A new config with a different file name must be created for this command to take effect (cannot have the same config file as before).

3 system config create <file_name.cfg>

4 system config set <file_name.cfg>

5 system config restart

6 Repeat this sequence for every RG on the system.


. . .

. .AT I R G 6 X X O R I M G 6 X XUsing

Once this command is entered and set, this sequence of commands will not be required again for the RG.

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The unit is now ready for phone support. Either:• Plug in analog phone telephone port.

OR• Plug in IP phone to LAN port.Refer to IP phone or analog phone guides for phone usage.


AT I R G 6 X X O R I M G 6 X XUsing


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. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .USB DEVICES 5

Eutectics USB Handsets are ideal for integrating with Sphere softphones. With minimal configuration, these handsets are ready for use with the Sphere system.

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P L A N N I N GPlanning your softphone installation will involve the following:

• Installation or Sphericall Desktop• Installation of phone• Installation of drivers for phone, if necessary• Reference of the phone’s user manual for operation information

P R E P A R I N GBe sure to the have following components ready for the installation:• Drivers to handle the plug-n-play device.• DirectX 8.1 is required on the PC.• Any additional executables for the device.

In the case of the IPP200, there is an additional set of APIs required for the function of “on hook” and “off hook” telephone functions. These files will be included in the final release of Sphericall.


U S B D E V I C E SEutectics IPP520

I N S T A L L I N G1 Read the instructions with the USB device first. 2 Determine whether you install device or drivers first. Then proceed.3 Plug in the phoneset (either handset or headset).4 Update drivers and APIs as required. Sphere software installs will include needed DLLs

for some of the Eutechtic phones.5 Verify Sphericall Desktop software is also installed.6 Test for functionality. Be sure to read product documents to understand equipment

functionality.

U S I N GRefer to the Sphericall Desktop Manual and Help files for more user information and quick reference guides.Note: You must use the Windows system volume controls to change volume settings.

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P L A N N I N G• Verify system requirements for Eutectics IPP520 and Sphericall.• The Sphere system should be installed, configured and tested as fully functional.• Refer to the IPP520 User Manual for installation planning, setup, package

contents, safety and conditions of use.


. . .

. .U S B D E V I C E SEutectics IPP520

O V E R V I E W O F O P E R A T I O NThe Eutectics IPP520 phone is a USB phone that can used with the Sphericall Desktop when running as a softphone. Communication between the phone and the Sphericall Desktop is done via the USB interface. A proprietary interface is supplied by Eutectics which allows the Sphericall Desktop to receive messages from the phone as well as send messages to the phone.

I N S T A L L A T I O NThe drivers for the IPP520 are available at Eutectics’ web site and are not installed as part of the Sphericall Desktop installation. The drivers are also shipped with the phone. These drivers should be installed when the user first plugs the IPP520 into the USB port on the PC. The dll's that are needed for the interface include the eusbcontrol.dll, eusbevent.dll and eusbi2chook.dll. These files are installed as part of the Sphericall Desktop installation and provide the interface between the Sphericall Desktop and the IPP520. They are installed into the system32 folder.

T E S T T H E I N S T A L L A T I O N

To test the insta l lat ion and make sure the IPP520 is proper ly recognized by your PC.

C O N F I G U R A T I O NThe IPP520 does not require any special configuration to associate itself with the Sphere system. A softphone line should be created through the Sphericall administrator application and assigned to the user that will be using the phone. When the user starts the Sphericall Desktop, they should verify that the IPP520 is configured as their PC Phone device. This is done through the Configure\Options\PC Phone widow.

WINDOWS 98, ME and 2000 WINDOWS XP

• Move your cursor over the “Volume” icon in the systray (looks like a small speaker), and click it with your right mouse button.

• Choose “Adjust Audio Properties”.• Select the Audio tab.• Ensure that the “Preferred Device” drop-down

windows for “Sound Playback” and “Sound Recording” are both set to “USB Audio Device”.

• Ensure that “Use only preferred devices” is selected.

• Move your cursor over the “Volume” icon in the systray (looks like a small speaker), and click it with your right mouse button.

• Choose “Adjust Audio Properties”.• Ensure that the “Preferred Device” drop-down

windows for “Voice Playback” and “Voice Recording” are both set to “Eutectics IPP520”.

• Select the Audio tab.• Ensure that the “Preferred Device” drop-down

window for “Sound Playback” is set to your soundcard device.

• Ensure that “Use only preferred devices” is selected.


U S B D E V I C E SEutectics IPP520

Figure 5.1 Desktop Options window

At this time the Sphericall Desktop does not get phone specific parameters from the database, such as ring cadences, stutter dial tone, and flash times.

F U N C T I O N A L I T Y

Function Description

Off Hook When the IPP520 goes off hook the Sphericall Desktop will answer an incoming call or it will display the Dial dialog and play dialtone. If there is already a call in progress it will put the current call on hold before playing dialtone. The user can enter digits through the keypad on the phone or through the dialog. The user can close the dial dialog which will stop dialtone from being played and a call cannot be established until the user goes off hook again.

On Hook When the user goes on hook the Sphericall Desktop will hang up the current call.

Dialing The Sphericall Desktop will use the digit map to determine whether it has a valid number before initiating the call. When the user enters digits from the phone the Sphericall Desktop will check the number to see if it can be dialed or if its an invalid number. If it can be dialed the Desktop will initiate a call, if its invalid the Desktop will play fast busy. If the user dials digits over a connected call the digits will be sent over the media stream.

Progress Tones The following tones will be played with the IPP520: dial tone, congestion, busy and ringback. Tones not currently supported are outside dial tone, stutter dial tone, call waiting and on-hold reminder.


. . .

. .U S B D E V I C E SEutectics IPP520

Hold The user can initiate a hold from the phone or from the Sphericall Desktop. Pressing the hold button while a call is connected will put the call on hold, pressing the hold button while a call is on hold puts the call into the connected state.

Mute The phone provides a mute button which stops audio locally to the phone. The phone does not send a message to the Desktop so it does not know the phone is on mute. If the user selects Mute from the Sphericall Desktop it will mute the call at the Desktop and the phone will not know the call has been muted.

Transfer A transfer can be initiated by pressing the Transfer key on the phone. When the button is pressed the Sphericall Desktop will display the Transfer dialog if the call is connected or on hold. After the dialog is displayed the Desktop will play dial tone and the user can enter the number using the keypad or using the Desktop. As digits are entered the Desktop will check if the number is valid using the digit map and start the transfer once a valid number has been entered. If an invalid number is entered the transfer will be initiated and congestion will be played.

Conferencing Setting up of a 3 part conference can only be done from the Sphericall Desktop.

Redial The phone has a redial button which will send the last number you dialed. This is stored locally.

Flash A flash can be initiated using the flash button or by flashing the hook switch for a time less than X seconds. If the user has a call and triggers a flash the Desktop will put the first call on hold and start a new call, another flash will hang up the new call and the first call will be active, another flash and the first call will be reconnected. If the user has more than 1 call the flash will cycle through the calls, creating a new call when a flash is done and the last call is active. If the user then does another flash the first call will be made active. A flash does not initiate a transfer.

Volume There are two volume buttons located on the phone just above the microphone. These buttons are active when a call is in the connected state. When these buttons are pressed a message is sent to the Sphericall Desktop which then changes the volume settings for the audio device.

Speaker The IPP520 has a speaker and external microphone which allows the user to operate the phone as a speaker phone.

Multiple Calls

Memory Keys The phone has memory keys which allows for storage of frequently used numbers. Since these numbers are stored in the phone the Sphericall Desktop has no knowledge of these numbers.

Headset A headset can be connected to the IPP520.

Star Codes Star codes are not supported with the IPP520. The Sphericall Desktop must be running for the IPP520 to function so the Desktop is used rather than the star codes.

Function Description


U S B D E V I C E SPlantronics Wireless set CS50-USB

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P L A N N I N G• Verify system requirements for Plantronics CS50-USB and Sphericall.• The Sphere system should be installed, configured and tested as fully functional.• Refer to the CS50 User Manual for installation planning, setup, package contents,

safety and conditions of use.• Verify necessary firmware for base and ear piece.

O V E R V I E W O F O P E R A T I O NThe CS50-USB offers wireless, hands-free headset convenience and long range workspace mobility.

I N S T A L L A T I O NConsult the manual shipped with the CS50-USB unit.

1 Install the hardware.2 Install the software.3 Charge your headset battery.4 Choose your headset wearing style.5 Carry out the initial setup recommended by the manufacterer.6 Open the Sphericall Desktop application.7 Go to Configure, Options.8 Select the Plantronics tab.9 Make your usability choices.


. . .

. .U S B D E V I C E SPlantronics Wireless set CS50-USB

Figure 5.2 Desktop Options Plantronics window

Test The Installation

OperationThe Sphericall Desktop interface opens the Place New Call window when the Plantronics CS50 headset is taken out of the charging cradle.The user is able to get dialtone, answer calls, or drop calls all with 1 click. The user can click the button once when there is an incoming call and the call would be answered. Also, when the user takes the headset out of the cradle, the Sphericall Desktop obtains a radio link established message and handles it accordingly, it provides dial tone if there is no call, or answers an active incoming call.


U S B D E V I C E SPlantronics Wireless set CS50-USB


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. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .MICROSOFT WINDOWS MESSENGER CLIENT 6

Theory of OperationThe Windows Messenger application has gained such a popular reputation for “instant” communication that some organizations are incorporating other protocols within the messenger interface. Sphere Communications supports Microsoft Windows Messenger v5.1.0701 on Windows 2000 or Windows XP application integration to its communication platform. In addition to the convenience of full-text messaging offered by Windows Messenger, users can add the operability of voice communications from the same user interface.Windows 2000Windows 2000 clients may be required to uninstall versions of MSN Messenger in order for the Windows Messenger to operate. Only voice calls are supported; not video.Windows XPVoice & Video are supported.Windows Messenger supports SIP (RFC3261 and 3428) and the SIMPLE extensions for presence and instant messaging. The SIP protocol is used to communicate with Sphericall for establising and maintaining voice and video connections with other Sphericall endpoints.The Windows Messenger user must configure a SIP communications service account to:

• Identify the Sphericall Manager that will act as a SIP server for this Messenger client.

• Identify the Sphericall Manager name and/or IP address.• Identify the appropriate connection protocol (UDP).

Once configured properly, Windows Messenger can be used as a softphone interface as well as a text messaging interface. No other application or program is needed.

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The following items are needed in planning for a system using Windows Messenger clients for the softphone:


M I C R O S O FT WI N D O W S M E S S E N G E R C L I E N TPreparing

• As always, refer to the Sphere System Requirements for version compatibility and interoperability notes.

• Messenger clients are limited to handling a single voice or video call at a time. An integrated environment with Messenger and Sphericall Desktops allows growth for some users past the limit of a single call.

• Messenger clients can be configured with a Class of Service (CoS) profile. • Permissions can also be used with Allow and Disallow on the Sphericall

Administrator.• All Messenger users must be assigned a primary numeric extension number.

This allows a number to which hardware phone users may call.• Forwarding is not configurable at the Messenger client level, rather, it can only

be configured by the Sphere system administrator on Sphericall Administrator.• For a Sphericall Desktop user to be able to text message a Windows Messenger

user, the administrator must set up privileges for the Desktop user to monitor the line of the Instant Messenger recipient.

• Video is not enabled by default for SIP Windows Messenger clients. You must adjust an initialization setting to enable this. Please refer to the Sphere System Requirements for information regarding SIP and video.

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REQUIRED:• Computer/Processor (as specified by Microsoft, see their site for the most up-to-

date requirements): • Computer with 300 megahertz or higher processor clock speed recommended.• 233 MHz minimum required (single or dual processor system).• Intel Pentium/Celeron family, or AMD K6/Athlon/Duron family, or compatible

processor recommended.• Additional requirements for video and/or application sharing

• Windows Messenger Version 5.1.0701 on Windows XP or download for Windows 2000.

• Speaker and microphone device for voice communciations OR USB headset or handset.

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Note: The installation procedure for Windows Messenger should be followed by the Sphere system administrator and not an end user.


. . .

. .M I C R O S O FT W I N D O W S M E S S E N G E R C L I E N TInstalling

Spher ical l Manager—to prepare for Windows Messenger endpoints1 Logon to Sphericall Manager.2 Open the Sphericall Administrator application.3 Click on the General tab.4 Highlight the top tree level System [SERVER NAME].5 Double-click to open properties or right-click and select View Properties.6 Enter the SIP Domain name into open field: Session Initiation Protocol.7 Create extensions and fill in the user information (First Name, Last Name, etc.).

Cl ient PC—to insta l l Windows Messenger to run in the Spher ical l environement1 Install or download Windows Messenger v5.1.0701 or higher version.

As you open the Messenger application for login: 2 Choose Tools/Options/Accounts.

Figure 6.1 Windows Messenger Accounts

3 Enter the user information in the SIP Communications Service Account field.4 Click the Advanced tab in that field.


M I C R O S O FT WI N D O W S M E S S E N G E R C L I E N TInstalling

Figure 6.2 SIP Communications Service Connection Configuration

5 Select Configure Settings.6 Enter the name or IP address of the Sphericall Manager to which this Windows

Messenger will check-in.7 Click on UDP in the Connect Using area.8 Click OK.9 Click OK again to exit Options.

10 Sign-in with Windows Messenger connecting to the Sphericall Manager.

Cl ient PC—to a l low any c l ients to text message Messenger c l ientsThis will allow either Messenger clients to text message each other or the Sphericall Desktop to text message with a Messenger client.

1 Set Windows Messenger security policy to “Low Security.”2 The following registry key must be added to each PC running Messenger

HKLM\Software\Policies\Microsoft\Messenger\Client\{83D4679F-B6D7-11D2-BF36-00C04FB90A03}\_Default\EnableSIPHighSecurityModeDWORD= 0 -- Low Security 1 -- High Security 2(Default, same as not set) --- Medium security"

For ease of deployment, this registry setting can be wrapped in an executable registry file. This file is available on the Sphericall software DVD under: \Client\Messenger\MessengerPolicy.reg. The system administrator may configure his network so that this file is executed when users log into the domain.

3 You will need to return to the Sphericall Manager to configure this Messenger Clients’ primary Extension number at this time.

Spher ical l Manager—to conf igure a pr imary extension for the messenger cl ient1 Return to the Sphericall Manager.2 Open the extension number which you have assigned this user.


. . .

. .M I C R O S O FT W I N D O W S M E S S E N G E R C L I E N TInstalling

3 Add the SIP address, which has now checked-in on the Sphericall Manager, to the Numbers area of the Properties for this Station.

This step must also be completed for regular phones without messenger to make calls to this Windows Messenger client.

4 Verify that the Sphere system assigns this extension as the primary extension.

If Windows Messenger client will have video enabled:5 Click on the Settings tab.6 Click Add to add a new Initialization Setting for this Messenger client.7 Select SIP > Video.8 Change the Value from Disabled to ENABLED.9 Click OK.

Cl ient PC—to a l low Messenger c l ients to moni tor presence stateIn order for other endpoints (including other Messengers running in Sphericall) to monitor a Messenger's presence states, Messenger needs to allow [email protected] (pseudo endpoint of the Sphericall Manager itself) to monitor its presence states.

Figure 6.3 Windows Messenger

This is the customer message users will see when users are added to their Windows client.This dialog only appears the first time the Messenger line checks into a Sphericall Manager. If the Messenger line is deleted from Sphericall and then re-added through another check-in, this dialog box may not appear.

Note: You may need to log off and log back on in order to get this message.1 Click OK.

Cl ient PC—to add phone equipment to the Windows Messenger cl ient• See IPP200 USB Handset chapter of this manual. Or install USB device of choice.


M I C R O S O FT WI N D O W S M E S S E N G E R C L I E N TTesting

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . T E S T I N G

After installing the Windows Messenger client, the Sphere system administrator should test and verify that the application is consistent in the following areas:• Always verify from the Sphere System Requirements that the correct version of

Microsoft Windows Messenger is being used the the Sphericall software.• Send and receive an audio call to/from another extension in the Sphere system.• Set up a conference call where the Windows Messenger client is a member.• Conduct blind and attended transfers from the Windows Messenger client to

another station.• Have the Windows Messenger client send and receive video.• Have the Windows Messenger client send and receive an Instant Message.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . U S I N G W I N D O W S M E S S E N G E R

Note: When calling another station from Windows Messenger, the caller does not hear ringback.


. . .

. .M I C R O S O FT W I N D O W S M E S S E N G E R C L I E N TUsing Windows Messenger

Figure 6.4 Windows Messenger window

Figure 6.5 Notification of change

Click to initiate a call

Click to combine video andvoice on call

Click to answer a call


M I C R O S O FT WI N D O W S M E S S E N G E R C L I E N TUsing Windows Messenger


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .MUSIC ON HOLD 7

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . M U S I C - O N - H O L D

This chapter is divided into three sections:• Section I - Hardware-Based Music-on-Hold

• Section II - Music-on-Hold Installation Test

• Section III - Music-on-Hold and Zones

S E C T I O N I - H A R D W A R E - B A S E D M U S I C - O N - H O L DHardware-based Music-on-Hold (MOH) lines are PhoneHub or BranchHub lines attached to a third-party music source. When a line is designated as MOH, it has none of the attributes of a normal line such as outbound calling or call waiting. When calls are put on hold, a one-way media stream from the MOH is set up to the party on hold. This media stream plays the music source.Each Sphere system provides support for the MOH feature functionality. Certain requirements must be met before your organization’s Sphere system can utilize this feature:

A N O T E O N H A R D W A R E - B A S E D M U S I C - O N - H O L D A N D F A I L E D C A L L A N N O U N C M E N T SThe Sphere system provides five call completion warning announcements for MLPP. Of those five announcements, two can be used by all customers for regular usage on non-MLPP calls. Sphere recommends re-recording these two announcements to be of a more general informational nature to your callers.Media Server-based MOH (and not Hardware-based MOH) must be enabled for Call Failure Announcements to function.For more information on Media Server-based MOH and Call Failure Announcements, refer to the Media Server Options chapter in Book 2: Intall & Configure.

M U S I C - O N - H O L D R E Q U I R E M E N T S• Hardware-based Music-on-Hold sources are supported. However, both

hardware and media server MOH will not be supported at the same time.• You must configure a station as a MOH input port via the Sphericall Administrator

application.• Your organization must secure a music source with a standard audio output jack

(such as a CD/DVD player).


M U S I C O N H O L DMusic-on-Hold

• A line within a zone must be designated as the MOH source for the zone.• All zones with independent MOH must have individual physical equipment for their

audio source.• Each zone’s physical audio source must correspond to a distinct station port on a

PhoneHub or BranchHub.• All re-broadcast rights must be purchased according to FCC statutes.• Your organization must secure a third-party interface device that will connect the

standard audio output jack at the music source to a station (i.e. PhoneHub port). An example of this type of interface device is the Bogen WMT-1A Telephone Line Matching Transformer.

MOH Requirements• Multicast (wherever available)• Unicast (as necessary)• MOH must be assigned to a Sphere system zone.• Be sure to refer to the Sphere System Requirements for UDP port information

for system-wide MOH multicast address.

Figure 7.1 EXAMPLE: Bogen WMT-1A Telephone Line Matching Transformer

Refer to the following table for a short list of distributors that manage an inventory of these requisite, third-party interface devices. Contact your organization’s Sphere Certified Channel Partner or Sphere Communications sales representative, if you require further assistance in locating a local supplier.

Table 7.1 Third-Party Interface Device Distributors

R E - B R O A D C A S T I N G R I G H T SFCC statutes define a corporation or other organization as a place where a substantial number of persons outside the normal circle of a family and its social acquaintances is gathered (17 U.S.C. SECTION 101). Corporations and

Distributor Telephone Number Internet Address

Jenne Distributors +1 (800) 422-6191 www.jenne.com

ALLTELL Supply +1 (800) 725-5835 www.alltell.com

Sprint North Supply www.sprintnorthsupply.com

BogenWMT-1A

Punchdown Block

Sphere MG

Audio Source


. . .

. .M U S I C O N H O L DMusic-on-Hold

organizations are considered public places that must adhere to copyright guidelines regarding the performance of recorded audio and video materials.Whether the performance is the use of a Music-on-Hold device (which constitutes a public performance by virtue of its being a transmission to the public), the playing of a CD/DVD, MP3 or tape, or the tuning of a radio to a particular station, permission must first be obtained in order for the use of that performance to be considered lawful.Once an organization purchases a Music-on-Hold source, they have full rights to re-broadcast music via that device and no other royalties are due the vendor. Radio broadcasts throughout a Sphere system are permissible if licenses have been purchased for each incoming telephone (i.e. trunk) line.

U N D E R S T A N D I N G M U S I C - O N - H O L D O P T I O N SNote: This overview takes into consideration only MOH’s relationship to hardware

station ports.

Sphericall’s hardware-based MOH feature can support both a single music source for an entire system, as well as individual music sources for multiple zones within the system. Prior to Sphericall v3.4.4.4, all MOH sources were based in the COHub’s Zone, meaning the inbound call determined what MOH music was heard. With v3.4.4.4, the MOH is determined by the person placing the call on hold from within the system; MOH for that zone will reflect that station’s zone (i.e. MOH could include time of day, weather, etc. reflecting that called station’s zone rather than the COHub from which the call entered the system).All zones must have independent physical resources for MOH. If a system requires three unique zones of MOH and a default zone with Music-on-Hold for fallback, there must be four physical MOH sources to meet this need as well as a separate station port for each MOH source. If a system needs only one MOH source, only one physical source is needed. All MOH music sources may be located on one PhoneHub or BranchHub, however each source must be assigned to a separate zone.One default zone may serve all MOH needs. If individual MOH needs are present, then the system must be designed for MOH by Zones. In order to design your system for optimal MOH service to meet all zone needs, you will need to consider all physical requirements. If a specific MOH source is not configured for a zone, the MOH audio source of the default zone will be used. Systems with a large number of zones may only require a few specific MOH sources. Rather than force MOH to be configured for each zone, the fallback to the default zone will ease configuration and minimize cost.Use the following instructions for configuration of one MOH per zone, as needed for the system you are configuring.

To Conf igure Music-on-Hold SourcesA line within a zone must be designated as the MOH source for the zone. A line selected as the MOH source in the system’s default zone will be used as the default MOH source for the system.From the Sphericall Administrator application window:

1 Click the General tab.2 Select Music On-Hold Sources from the tree.



Figure 7.2 Media Server Music-on-Hold window

• The Music On-Hold Sources branch will contain either a list of hardware station lines, established from a pre v4.1 version of Sphericall or a Sphericall Media Server(s) entry.

• By default, a new media server is initialized to be a source for MOH . Configuration or loading may necessitate barring a media server from sourcing MOH. Media servers can be barred from sourcing MOH.

Assuming that this is a new installation and there are no previously-established Music On-Hold Sources:

3 DELETE the MOH Media Server listed here (you will not need it if you are using hardware based MOH).

4 Click OK to exit this window.5 Highlight Music On-Hold Sources.6 Right-click to Add.7 Click Add.

Adding a new MOH Source when the list is blank will display a dialog box asking the user to choose either media server or hardware-based MOH.


. . .


Figure 7.3 Choose Music-on-Hold Source window

8 Select Hardware MOH.9 Click OK.

From the Select Station window10 Select the zone with which to associate this station from the Zone list.11 Expand the MG(s) listed in the display area to view all available stations within your

organization’s Sphere system.12 Highlight the station you wish to configure as the MOH line.13 Click OK.

The selected station appears in the Music On-Hold Line display area.14 Highlight the station.15 Click Properties.



Figure 7.4 Properties for Station window

16 Type the name of this Music-on-Hold line in the Line Name field.17 Select the In Service check box if you wish to classify this station as active within your

organization’s Sphere system.18 Clear the Log Calls check box.19 Select the Drop Loop Current check box.

Drop loop current is primarily used for paging and voice mail lines. For every station on a PhoneHub or BranchHub that is connected to a paging system, that station’s properties must be configured in this manner.

20 Clear the Stutter Dial Tone check box.

E X T E N S I O N A S S I G N M E N T T O A M U S I C - O N - H O L D S T A T I O N

To assign an extension to a Music-on-Hold stat ionIn the Numbers area:

1 Verify that the Music-on-Hold extension, moh1, is listed in the Numbers area.2 Click Add Extension.3 Expand the folders listed in the file tree to view all available extensions within your

organization’s Sphere system.ORClick New Extension to create a new extension for this purpose.

Note: Because the moh1 group extension cannot be made primary for this station, you must configure a separate extension—with a Single Line hunt order—that


. . .


will serve as a primary extension for the Music-on-Hold station and associate it with this station.

4 Highlight the extension you wish to assign to this station.5 Click OK.

The newly-assigned extension appears in the Numbers area in the Properties for Station window.

6 Highlight any assigned extension number (the one you just created) besides the moh1 extension.

7 Click Make Primary for this new extension number (do not make MOH numbers primary).

Primary extensions appear in bold type in the Numbers area as well as in the Stations tab.CAUTION! GROUP EXTENSIONS AND VOICE MAIL EXTENSIONS ARE NEVER

TO BE MADE PRIMARY EXTENSIONS WITHIN A SPHERE SYSTEM.

8 Click Apply.9 Click OK.

The configured Music-on-Hold station appears in the Music Hold Line display area.

S E C T I O N I I - M U S I C - O N - H O L D I N S T A L L A T I O N T E S T

To test funct ional i ty of the Music-on-Hold feature1 Place an external call from your organization’s Sphere system to an address located

within the Sphere system.

Table 7.2 Music-on-Hold Installation Test Procedures

S E C T I O N I I I - M U S I C - O N - H O L D A N D Z O N E S

M U S I C - O N - H O L D B E H A V I O R W I T H T R U S T E D Z O N E S

Table 7.3 Music-on-Hold Behavior with trusted zones

At the Auto Attendant prompt: If you do not access the Auto Attendant prompt:

1. Dial the number of the appropriate extension.

1. Instruct a Sphericall user to transfer you to another address on the Sphere system.

2. Verify Music-on-Hold functionality while the call is pending consultation by the dialed extension.

2. Verify Music-on-Hold functionality while the call is pending consultation by the dialed extension.

Zone trust level Music-on-Hold behavior example

Music-on-Hold between trusted zones Three zones (Z1, Z2, Z3) all trusted. There is Music-on-Hold configured for each zone. Station Z1 puts caller from Z2 on hold. Z2 call hears Z1 Music-on-Hold.



M U S I C - O N - H O L D B E H A V I O R W H E N M U S I C - O N - H O L D S O U R C E I S U N A V A I L A B L EIf the address of the zone specific Music-on-Hold source is unresolved, during a possible MG timeout or MGC failover, the system default Music-on-Hold source will be utilized. This default Music-on-Hold source must be configured and operational.

M U S I C - O N - H O L D A N D U P G R A D E SFor systems first upgrading, the following table will help to identify the upgrade path procedure that will be taken during the software upgrade script:

Table 7.4 Music-on-Hold Upgrade considerations

Music-on-Hold between un-trusted zones

Three zones (Z1, Z2, Z3) with the following trusts: • Z1 <--> Z2• Z1 <--> Z3There is a Music-on-Hold line configured in each zone. A station from Z2 calls a station in Z1 and is transferred to a station in Z3. If the Z3 station puts the Z2 station on hold, the Z2 station hears Z3 Music-on-Hold.

Music-on-Hold for zones without a specific Music-on-Hold source

Three zones (Z1, Z2, Z3) all trusted: There is a Music-on-Hold line configured for zones Z1 and Z2. Z1 is the default zone. Station in Z3 puts caller from Z2 on hold. Since Z3 does not have a specific Music-on-Hold source, Z2 call hears Z1 Music-on-Hold, the default Music-on-Hold source. If both Z1 and Z3 did not have a Music-on-Hold source configured, Z2 would not have heard any Music-on-Hold.

Music-on-Hold Existing System Music-on-Hold Upgrade path taken

No previous Music-on-Hold • System will default to Server-based MOH. • Recommended use of system-wide multicast when-

ever available.• System requires the use of Zones for MOH.

v4.0 Media Server-based Music-on-Hold

• System will default to Server-based MOH. • Recommended use of system-wide multicast when-

ever available.• System requires the use of Zones for MOH.

One Existing Hardware-based Music-on-Hold source and more than one zone

• Hardware-based MOH will still be the default after the upgrade.

• System requires the use of system-wide multicast.• If more than one zone, the first zone found in the

zone table will be configured as the default zone.

Zone trust level Music-on-Hold behavior example


. . .


More than one zone and more than one existing Hardware-based Music-on-Hold source


• System requires the use of system-wide multicast.• When there is more than one line encountered with

the Music-on-Hold address associated with it, the line located in the default zone is made the system default as well as the Music-on-Hold source for that zone. All other Music-on-Hold sources remain spe-cific to the zone they were in prior to upgrade.

Hardware-based MOH: more than one zone with more than one Music-on-Hold and with no Music-on-Hold source located in the default zone


• System requires the use of system-wide multicast.• The first line encountered will be moved into the

default zone to become the system default Music-on-Hold source as well as the Music-on-Hold source for the default zone. The zone that the line was moved from will continue to use the same Music-on-Hold source only via the system default association rather than a specific association. All other Music-on-Hold sources remain specific to the zone they were in.

Music-on-Hold Existing System Music-on-Hold Upgrade path taken


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .PAGING 8

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P A G I N G L I N E S

O V E R V I E WCreating a paging line provides users with an opportunity to dial an extension and page their party over your organization’s intercom or public address system. These paging lines are used to integrate separately-purchased, third-party systems with a Sphere system. The paging line is configured, similar to any other station, via the Sphericall Administrator application and is connected to a station interface on either a PhoneHub or BranchHub.

R E C O M M E N D E D P R O D U C T SAll paging station adapters must have the capability to disconnect on dialtone. A product that has been Sphere-tested is the V-9970 Station-Level Page Adapter from Valcom, Incorporated:• +1(540)427-3900

orhttp://www.valcom.com

The V-9970 can be enabled for this disconnect capability by setting its dipswitch #1 to the “On” position. This prevents inadvertent dialtone broadcasts over your organization’s paging system.

Common Paging System Features

Table 8.1 Paging System Features

Feature Description

Talkback If your organization’s Sphere system has been integrated with the appropriate paging system hardware, you can configure paging for two-way voice communication over that system.Note: This configuration is commonly used for security and entry-management purposes.

Music Many paging systems offer background music inputs that allow the paging system to play music over the PA when not in use.

Night Ringing Some paging systems can announce incoming calls to any address throughout the entire system via ring tones played over the PA.Note: This configuration often requires additional system hardware. Consult the documentation included with the third-party paging system for more information on this feature functionality.


P A G I N GPaging Lines

Note: Some paging systems do not support a station-level interface. If your organization intends to integrate such a paging system with their Sphere system, you must secure a station-level paging adapter to provide the necessary call disconnect functionality.Call disconnect functionality can be provided with either an open-loop detector (i.e. drop loop current), a time-out release (per the time frame specified for the paging device), or an audio-sense release (i.e. the paging system disconnects the call when it detects no sound on the line).

C O N F I G U R I N G A P A G I N G L I N ENote: The following instructions detail adding a station-level interface. However, an

analog loop-start trunk interface may be used.

To add a paging l ine to a Sphere systemFrom the Sphericall Administrator window:

1 Click the Number Plan tab.2 Click Add Extension Number.3 Type the paging extension number in the Number field.4 Select Single Line from the Hunt Order list.5 Type the name of this paging extension in the Last Name field.6 Select the Show in Phonebook check box if this extension is to appear in your

organization’s Sphere system extension list.

This check box is selected by default.7 Select the Paging Device icon type to associate with this extension.

By selecting the Paging Device icon type, the extension/station combination will play a different dial tone than a normal station.

8 Click Apply.9 Click New.

andRepeat the previous steps to add another paging extension to your organization’s Sphere numbering plan.

10 Click OK when you are finished adding paging extensions to your organization’s numbering plan.

To conf igure a paging l ine for a Sphere systemNote: You must add and configure a paging device extension to your organization’s

Sphere numbering plan before you can add and configure a paging line (i.e. station).

From the Sphericall Administrator window:

All Call Additional extensions can be connected to the paging system to enable the simultaneous paging of all zones within your organization’s Sphere system.Note: This configuration is commonly used with multi-zone paging systems utilizing multiple paging extensions.

Feature Description


. . .

. .P A G I N GPaging Lines

1 Click the General tab. 2 Select Paging Lines from the list.3 Click Add.

In the Paging Device Address area:4 Select Paging Line from the list.

In the Station area:5 Expand the MG(s) listed in the display area to view all available stations within your

organization’s Sphere system.6 Highlight the station or BranchHub line you wish to designate as the paging line.7 Click OK.

The paging line instance appears in the display area.8 Highlight the paging line instance.9 Click Properties.


10 Type the name of this paging line in the Line Name field.11 Select the zone with which to associate this station from the Zone list.12 Select None from the Pickup Group list.13 Select the telephony area with which to associate this station from the Telephony Area

list.14 Select the emergency group with which to associate this station from the Emergency

Group list.



15 Select the paging line profile from the Default Profile list.

Note: Sphere Communications strongly suggests that you create a separate paging line profile with Max Calls set to 1 (to prevent more than one call to access the PA system at one time) and with call waiting disabled (to negate the ability to interrupt an in-progress page).

16 Select the In Service check box if you wish to classify this station as active within your organization’s Sphere system.

17 Clear the Log Calls check box.18 Select the Drop Loop Current check box.

Drop loop current is primarily used for paging and voice mail lines. For every station on a PhoneHub or BranchHub that is connected to a paging system, that station’s properties must be configured in this manner.


In the Numbers area:20 Verify that the paging extension number appears in the display area.21 Highlight the paging extension.22 Click Make Primary.23 Click Apply.24 Click OK.

I N S T A L L I N G A N D I N T E G R A T I N G A P A G I N G D E V I C E

To integrate a Sphere system wi th a paging device1 Connect the tip-and-ring of the 6-position modular jack to the tip-and-ring inputs for the

adapter on the punchdown block (or breakout box/patch panel) serving the PhoneHub (or BranchHub).

Note: The RJ-11 wiring arrangement is:Pin3 = RingPin4 = Tip

2 Refer to the documentation included with the third-party paging system for information and instructions regarding the connection and configuration of speakers and other associated hardware.

P A G I N G S Y S T E M Z O N E C O N F I G U R A T I O N SPublic address systems can be configured as single- or multi-zone paging systems depending upon the paging hardware and the interface(s) available for integration with a Sphere system.


. . .

. .P A G I N GPaging Lines

Single-Zone Paging System

Figure 8.2 Single-Zone Paging System

Multi-Zone Paging System Connected to Single Extension

Figure 8.3 Multi-Zone Paging System, Single Extension

Zone 1Station-Level

PagingAdapter

Single-ZonePaging System

Punchdown Block

Sphere MG

Zone 1

Zone 2

Zone 3

Multi-ZonePaging System

Station-LevelPagingAdapter

Punchdown Block

Sphere MG



Multi-Zone Paging System Connected to Separate Extensions

Figure 8.4 Multi-Zone Paging System, Separate Extensions

I N S T A L L A T I O N T E S T

To ver i fy successful paging system insta l lat ion and conf igurat ion1 Confirm the appropriate hardware installation for the third-party paging system.2 Dial the paging extension configured for your organization’s Sphere system.3 Announce the page for the appropriate paging zone.

Zone 1

Zone 2

Zone 3

Zone 1 Station-Level Paging

Adapter



Adapter



Adapter


All Call Station-Level Paging

Adapter

Punchdown Block

Sphere MG


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SMDI PLANNING & PREPARATION 9

A B O U T S M D I I N T E G R A T I O N W I T H

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S P H E R I C A L L V O I C E M A I L

S M D I A N D S P H E R I C A L L V O I C E M A I LSMDI and Sphericall Voice Mail are independent of one another. Each voice mail system can be accessed independently by using unique addresses. A Sphere system can use either its native Sphericall Voice Mail, SMDI integration of voice messaging with another application, or both.A station can receive message waiting indications from either an SMDI or Sphericall Voice Mail system. Stations that are using both Sphericall and SMDI voice mail systems may get their MWI state out of synchronization. SMDI may clear the MWI regardless of any unread Sphericall messages. The preferred configuration is to limit a station to single voice mail system.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S M D I O V E R V I E W

A Sphericall system’s integration with an SMDI platform utilizes the Simplified Message Desk Interface and conforms to an industry-standard method of voice messaging platform integration (via the RS-232 serial connection found on any personal computer).Because the information about the telephone calls travels along a different channel than the calls themselves, the Sphericall/SMDI integration is considered an out-of-band integration. This differs from an in-band or DTMF integration where telephone call information is passed via the same channel as the call.Any service utilizing SMDI integration can be used with the Sphere system, for example, all of the following can integrate with Sphere via SMDI:

• Voice Messaging• Call Center• Fax Center


S M D I P L A N N I N G & P R E PA R A T I O NSMDI Integration Requirements

S U P P O R T E D V O I C E M E S S A G I N G F E A T U R E SSphericall supports the following voice messaging integration features:

Sphericall also supports the transmission(s) of the following messages within an integration:

Table 9.1 Sphericall/Voice Mail Integration Message Transmission

M A N U A L SDetailed manuals for Sphericall and other third-party voice application products may be necessary to determine full integration capabilities as well as instructions for installation.All Sphere System Requirements as well as the SMDI requirements are also contained in the Sphere System Requirements manual.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S M D I I N T E G R A T I O N R E Q U I R E M E N T S

Voice messaging-related portions of the Sphere system adhere to the Telcordia (formerly Bellcore) standard. Refer to Telcordia’s web site at http://www.telcordia.com for more information.Sphericall integrates with third-party SMDI capable voice mail applications.

Table 9.2 Sphericall Integration Requirements

• SMDI packet format • Caller ID

• Line ID • Destination ID

• Message Waiting Indicator • Station hunt groups

Messages Sent to Voice Mail System Messages Received from Voice Mail System

Direct call Operate Message Waiting Indicator

Forward all calls Remove Message Waiting Indicator

Forward on no answer

Forward on busy

Forward with unknown reason

Invalid message waiting request

Blocked message waiting request

Requirement Description

Sphere SMDI Hardware Requirements


. . .

. .S M D I P L A N N I N G & P R E PA R A T I O NSMDI Integration Requirements

Sphericall Manager The Sphericall integration relies heavily upon the communication between the two server types, Sphericall Manager and the other third-party voice server.The Sphericall Manager houses the Sphericall software, the engine that ultimately drives the integration of the systems.Note: A single Sphere system can support one voice messaging platform per SMDI link. Each Sphericall Manager must have its own set of voice messaging ports and its own RS-232 serial port link. As such, a separate Sphericall Manager will be used for each voice messaging platform connection.

Sphere PhoneHub or BranchHub Each MG must have the appropriate number of available ports.The Sphericall integration passes information between servers (Sphericall Manager and other voice server), and this information is ultimately passed to the endpoints (i.e. telephones attached to MGs) on a Sphere system.

Voice Messaging Platform (i.e. third-party voice server) Requirements

Analog Station Ports One per-station appearance on the voice server

Line Speed and Baud Rate Recommended is 9600 bps.Minimum required is 1200 bps.2400 bps and 4800 bps are optional and supported by the Sphericall system.Note: Sphericall supports the maximum baud rate supported by the messaging platform.

Data Bits Telcordia recommended is 7-bit word length.

Parity Telcordia recommended is Even parity.

Stop Bits Telcordia recommended is 1 mark stop bit.

Hardware Flow Control The Sphericall Manager should be configured to match the flow control settings of the third-party voice server.

Link Operations Minimum required is Half Duplex.Full Duplex (which supports MWI) is optional and supported by the Sphericall system.

Digit Length The Sphericall Manager must be configured to send and receive the number of digits expected by the third-party voice server (i.e. 7 or 10).Note: If this number is not indicated in product documentation, you must locate the message packet length in the SMDI coding.

Cabling Requirements

Sphere PhoneHub and BranchHub Refer to the PhoneHub Installation Manual or the BranchHub Installation Manual for complete MG cabling requirements.

Cables 1 (one) RJ-11 two-wire or RJ-14 four-wire telephone cord is required:• Per station port• Per 50-pin connectordepending upon the platform connection requirement.

RS-232 Compliant Serial Port 1 (one) available RS-232 compliant serial port is required per SMDI integration.The serial port may be either 9- or 15-pin. However, you must have the appropriate cable adapters if you plan to mix the two types.

RS-232 Cable The serial ports on the Sphericall Manager and the third-party voice server are both configured as DTE (Data Terminal Equipment) devices. The RS-232 (i.e. serial-to-serial) cable, otherwise known as a Null Modem or cross-over cable, is utilized to connect the Sphericall Manager to the third-party voice server.The cable should be of sufficient length to reach between the Sphericall Manager and thethird-party voice server but should not exceed a 50-foot distance limitation for the RS-232connection.

Requirement Description


S M D I P L A N N I N G & P R E PA R A T I O NSMDI Integration Requirements

S M D I P L A T F O R M H A R D W A R E C O N S I D E R A T I O N SSphere recommends strongly that you verify your hardware requirements with the voice mail vendor prior to installation. Third-party voice server hardware requirements vary depending upon the following parameters:• Number of voice ports or possibly fax portsThird-party voice systems may be implemented on a single server or on multiple servers depending on the system and network configuration, which may include:

Table 9.3 Platform recommendations/considerations

Component Description

SMDI Platform Minimum Requirements

Processor Type and/or Third-party voice processor platform requirements vary depending upon system size and the selected applications.

Memory Memory requirements vary based upon the application and system size. The minimum memory requirement for any third-party voice servers is 128 MB of RAM; the maximum memory requirement is 512 MB of RAM.

Hard Disk Storage Capacity The third-party voice system uses the system hard drive storage for five purposes:Storage of operating system and voice message softwareStorage of voice and fax messagesStorage of reporting data source filesStorage of message buffer file (i.e. when a system incorporates DMM for Lotus Notes as well as Microsoft Outlook or Microsoft Exchange)The minimum storage requirement on a third-party voice system is 2 GB.Note: Each hour of voice message storage on the third-party voice server uses 10.4 MB of hard disk space. To determine the maximum number of hours for voice message storage on your organization’s third-party voice server, divide the server’s hard disk capacity by 10.4 (i.e. Total storage on third-party voice server / 10.4 = Maximum number of hours for voice message storage).

Expansion Slot Types Two types of card slots are available in PC-based server platforms: ISA slots and PCI slots.Third-party voice systems, able to use both of the available card slots on a single server, support Dialogic and Brooktrout PCI and ISA cards.Note: All voice cards (and, separately, fax cards) must be located in only one type of slot, either PCI or ISA. The cards (voice and fax) cannot be distributed between different types of card slots on a server.

Available Card Slots The number of required expansion card slots is determined by system size, applications running on the system, and the type of telephone system integration. Each voice card and each fax card needs a slot on the server.Note: In some telephone system integrations, there may be integration cards that require one or more slots.


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. .S M D I P L A N N I N G & P R E PA R A T I O NSMDI Operation

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S M D I O P E R A T I O N

O V E R V I E WA Sphere system is integrated with third-party voice messaging systems in the following manner:

Figure 9.1 SMDI-to-Sphericall Connections

In order to process an incoming call, a voice messaging system needs certain pieces of information about the call:• The voice messaging system needs to know the called party ID. This is the

extension number to which the call was forwarded by the voice mail system.• The voice messaging system can also use the calling party ID. This is the

extension or telephone number of the party placing the call. • The voice messaging system needs a way to communicate information to the

telephone system (i.e. the correct status of the message waiting indicators that correspond to the subscribers’ mailboxes).

S Y S T E M I N T E G R A T I O NThe Sphericall/SMDI integration is considered an out-of-band integration. In such an integration, information is sent between the telephone system and the voice messaging system on a dedicated data link, generally an RS-232 format link. When the voice messaging system answers a forwarded call from the telephone system, the telephone system has also sent the relevant integration information to the voice messaging system across the data link. When the voice messaging system wants to notify the telephone system of a change in MWI status for a specific user, it uses the same data link.

RS-232

SMDI Voice Mail Server

Sphericall Manager (Primary or Secondary) Switch

ManagerSphericall

Software ServiceDialogic Service

RJ-11 or 50-pin

Switch

BranchHub

PSTN

PhoneHub

COHub

PSTN

Voice Line Cards

LAN via TCP/IP or NetBEUI SMDI Admin Client

Program

SMDI Configuration

NT Control Panel, startup and shutdown to Telephony Server etc

NT Services

Hardware


S M D I P L A N N I N G & P R E PA R A T I O NSMDI Operation

C A L L P R O C E S S I N G

T H E C O M P L E T E I N T E G R A T I O NBecause the Sphericall/SMDI integration is an out-of-band integration, some form of synchronization must occur to link the out-of-band message packet to the appropriate telephone call. This synchronization occurs as a result of the interaction of two separate entities, the Message Storage Retrieval Identifier and the Message Storage Retrieval Line ID.• The MSRID is the unique identifier for the voice mail platform within a Sphere

system.• The MSRLineID is the name given to an individual analog station port connecting

the Sphericall Manager and the voice mail server.

Note: You must obtain all of the port numbers from the voice mail server prior to configuring the Sphericall Softswitch.

Examples and FunctionalityRefer to the following examples to determine the relationship of Sphericall addresses to MSRLineIDs.

Table 9.4 Address Correlation to MSRLineIDs

If an analog station port identified as MSRLineID 0001 on a Sphericall Manager was connected to a port that the voice mail server identifies as MSRLineID 0002, integrated calls will not travel between the two servers.Calls will traverse the RS-232 serial cable connection, and the voice mail server will be notified that it is to pick up the call at the port identified as 0001. However, the voice mail server will be unable to retrieve the correct call because the Sphericall Softswitch sent the call to the station port identified as 0002.These matching requirements hold true for the MSRID as well. All IDs must match or the voice mail server will be unable to process the telephone calls.

VM Address MSRLineID VM Group Number AA Group Number

6000 N/A (Group Number) 6000 *

6001 N/A (Group Number) * 6001

6002 0001 6000 6001

6003 0002 6000 6001

6004 0003 6000 6001

6005 0004 6000 6001

6006 0005 6000 6001

6007 0006 6000 6001

6008 0007 6000 6001

6009 0008 6000 6001


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. .S M D I P L A N N I N G & P R E PA R A T I O NSMDI Operation

S P H E R I C A L LA Sphere system presents two general types of calls to voice mail. The first call type is an auto attendant call. Auto attendant calls are directed to voice mail via telephone system programming: Voice mail voice ports are programmed into a hunt group within Sphericall, and specific trunks are programmed to ring that hunt group. Sphericall then finds an available voice mail port and presents the call to the voice messaging system.The second type of call presented to voice mail is a call forwarded from a station. Individual stations within Sphericall can be programmed to forward to the voice mail hunt group if the station does not answer within a specific number of rings (a Ring No Answer Forward).Regardless of call type, the Sphericall Softswitch delivers the call to voice mail along with the required integration information. The voice mail system uses this integration information to determine how to handle the call.

V O I C E M A I L S Y S T E MWhen a voice messaging systsem answers the incoming call from Sphericall and processes the integration information, the voice mail system first determines what type of call it is being presented. The voice mail system divides calls into five basic types:

Table 9.5 Voice Mail Call Types

A voice mail systems’ integrated messaging function will process each of these call types differently:• In the case of a forwarded station or forwarded trunk call, voice mail routes the call

to the Subscriber mailbox associated with the called party ID extension number.• In the case of a direct trunk call, voice mail acts as an auto attendant and routes

the caller to the appropriate mailbox. • If callers dial an extension number via one of the call processor menu options,

voice mail attempts to transfer the call to the appropriate extension. If that extension returns Busy or RNA CP tones, a voice mail system, acting as a voice messaging system, prompts the caller to record a message for the subscriber.

• In the case of a direct station call, voice mail prompts the caller to leave a voice or fax message for the subscriber or, if the caller is a subscriber, prompts the caller to enter the security code for the mailbox.

Call Type Description

Direct Station Call An internal telephone extension calls directly into voice mail

Forwarded Station Call An internal telephone extension calls another internal telephone extension and is forwarded to voice mail

Direct Trunk Call An outside call is sent directly to voice mail

Forwarded Trunk Call An outside call is directed to an internal station and is forwarded to voice mail

System Attendant Call The telephone system operator initiates a call to voice mail


S M D I P L A N N I N G & P R E PA R A T I O NSMDI and System Planning

• If the caller leaves a message, voice mail routes the message to the appropriate mailbox and uses the telephone system to light the MWI on the telephone set associated with the appropriate subscriber.

• If the caller accesses the mailbox, the voice mail system provides a prompt for mailbox administration and maintenance. Subscribers can listen to, delete, and send messages as well as record personal greetings and name recordings for their mailbox.

T H E S U B S C R I B E RWhen voice mail subscribers notice that their MWI is lit, they may pick up the telephone and dial into voice mail to retrieve messages. When the voice mail system recognizes this call type (a direct station call), it uses the calling party ID to identify the mailbox associated with the extension placing the call and prompts subscribers to enter their security code. After they enter their security code, voice mail places them into their mailbox. Once in their mailbox, subscribers are presented with a menu that, among other choices, allows them to process any new messages. If subscribers choose to delete the messages after listening to them, voice mail again uses the telephone system integration to extinguish the MWI on the appropriate telephone sets.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S M D I A N D S Y S T E M P L A N N I N G

P L A N N I N G F O R V O I C E P O R T SUtilizing the Erlang model, system integrators must estimate the number of IVR ports an organization needs in order to connect the Sphericall Manager(s) to a third-party voice messaging system. This analysis and subsequent planning prevents unacceptable call blocking based on a system’s lack of IVR ports.

E R L A N G M O D E L T E R M I N O L O G Y

Table 9.6 System Sizing Terminology

Recall Factor The percentage of calls immediately retried because the original call encountered blocking.The effect of this traffic is to increase the traffic load offered to a trunk group.For example, if half of the callers who encounter blocking immediately try to call again, the Recall Factor is 50%.

Busy Hour Traffic The amount of traffic offered to a trunk group or IVR port during the busiest hour of the day of the busiest day of business.Busy hour traffic is used as the basis for calculations using the Erlang B traffic model.


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. .S M D I P L A N N I N G & P R E PA R A T I O NSMDI and System Planning

Cal l Traf f ic Calculat ion:1 Call loggers

Many PBXs have call loggers connected to them. These record and analyze details of calls made using the PBX and can provide traffic figures for particular routes.To ensure that the figures used are busy hour figures:• Verify the busy hour figures as provided by the call logger itself• Print hourly traffic figures over a period of time then use the highest hourly figure

over that period• Convert call figures from minutes to Erlangs

• To convert from hourly call minutes to Erlangs, divide the figure by 60.• Remember that call loggers are aware only of calls that were successfully

completed (i.e. call loggers provide carried figures rather than offered figures, which your calculations require); this is only an issue if the blocking on the trunk groups is already a problem

2 PBX statistics

Some PBXs (i.e. Nortel’s Meridian PBX) have software modules provided that provide the required traffic figures.

Note: The same considerations as for call logging figures apply here.3 Telephone bills

Busy Hour Call Multiplier Indicator of the amount of call volume depending on an organization’s work flows.An organization with flexible hours would have a smaller multiplier than the business with set hours.An organization with a heavy call volume would have a lower multiplier than an organization with light traffic. Typical values are 1.5 to 2.5.

Blocking A fraction representing the calls that cannot be completed because all lines are busy.If blocking = 0.1, 10% of calls are blocked.Depending upon the application, reasonable figures for blocking are between 0.01 and 0.05. Callers experiencing blocking usually hear a busy signal.

Erlang A measurement of telephone traffic.An Erlang is equal to one full hour of use (e.g. conversation), or 60 * 60 = 3600 seconds of phone conversation.CCS is converted into Erlangs by multiplying by 100 then dividing by 3600 (i.e. dividing by 36).Numerically, traffic on the trunk group (when measured in Erlangs) is equal to the average number of trunks in use during the hour in question.Note: In the US, the unit commonly used for traffic measurement is CCS (centi-call second). The following conversion factor can be applied between the two units:• 1 Erlang = 36 CCS1 CCS = 0.0278 Erlangs

Trunks Common term referring to the physical communication line between two switching systems such as a local PBX and the telephone company’s CO switch.The number of trunks is a figure represented by the number of lines in a trunk group.

IVR Port An IVR system’s electrical interface through which a PBX sends information to a voice mail system.The system design needs to account for the potential volume requirements of port interfaces between the PBX and the voice mail server.The number of IVR ports is a figure represented by the number of station ports necessary on the system.



You can use telephone bills, sometimes, to discover the traffic carried by a trunk group.

4 Carrier traffic studies

For a fee, your existing carrier may be prepared to complete a study of your organization’s call traffic and may be able to use historical data.

5 Estimating

If you do not have any of these luxuries, you must make a reasonable estimate based on what you know about the way the organization uses its telephones; you may also be able to use information from other similar sites. A table is provided concerning the estimation of Erlangs for trunk and IVR usage.

E S T I M A T I N G B U S Y H O U R T R A F F I C• Assume that 50% of all calls are blocked and must immediately recall• BHT can be calculated by multiplying the average length of calls by the average

number of calls during the busy hour• BHT (1%) is the Busy Hour Traffic in Hours with 1% blocked; 1% blockage for

the Busy Hour is traditionally lowest acceptable without over provisioning for trunks

• BHT (5%) is the Busy Hour Traffic in Hours with 5% blocked; 5% blockage for the Busy Hour is traditionally highest acceptable without having excessive blocking

Figure 9.2 Trunk Sizing Chart for Busy Hour Traffic


. . .


Table 9.7 Busy Hour Blocking

C O N F I G U R I N G S P H E R E S Y S T E M P O R T SOnce your organization has planned for, secured, and configured a message server of sufficient or greater-than-necessary capability, you must set aside the appropriate number of PhoneHub or BranchHub ports within the Sphere system. These ports will be used for the necessary hunt groups, for the auto attendant, as well as for all voice mail functionality within the integration.

Note: Sphere Communications recommends that you configure contiguous groups of ports for the integration. For example, if your organization’s voice message system requires 8 ports (or a multiple of 8 ports), Sphere system setup is simplified and troubleshooting, minimized, if you designate for service an entire PhoneHub or a portion of a PhoneHub.

I N T E G R A T I O N N O T E SUtilize the information in the following sections as appropriate during configuration of your organization’s Sphericall and voice messaging integration.

C A L L P R O G R E S S T O N E SThe Sphericall Softswitch provides the following CP tones to the message server:

Table 9.8 Sphericall Call Progress Tones

Trunks BHT (1% blocking) BHT (5% blocking)

8 3.1 4.4

16 8.8 11.2

24 15.2 18.5

32 21.9 26.0

40 28.8 33.7

48 35.9 41.4

Call Progress Tones Frequency Cadence Amplitude

Internal Dial Tone 400 Hz Continuous -27 dBm

External Dial Tone 350 Hz + 440 Hz Continuous -24 dBm

Busy Tone 480 Hz + 620 Hz On = 0.5 sOff = 0.5 s

-24 dBm

Ringback Tone 440 Hz + 480 Hz On = 2.0 sOff = 4.0 s

-24 dBm

Reorder Tone 950 Hz, 1400 Hz, 1750 Hz

0.25 s, 0.4s, 0.4 s -24 dBm

Disconnect Tone None NA NA



A C C E S S H A N G U P D E T E C T I O NThe minimum duration for a drop in loop current is 550 ms. The Sphericall Softswitch sends 600 ms for a disconnect (i.e. a drop in loop current).

T R A N S F E R I N I T I A T I O N A N D T R A N S F E R R E L E A S EThe Sphericall Softswitch has the ability to manage call transfers in one of two ways. Both push the call to the intended recipient on the Sphere system, yet each handle caller ID presentation differently.

FP*96You can configure signalling over the serial port connection between the Sphericall Manager and the voice messaging platform as FLASH/PAUSE + *96. When FP*96 is used within an integration, the caller ID of the transferring party, not the original calling party, is displayed to the recipient party (as depicted in the following figure).

Figure 9.3 FP*96 and Caller ID Presentation

Note: This setting is configured on a message server platform under the Configuration\Hardware\Switch Protocol tab.Set the value for the “Transfer Init” field to FP*96.

The ivrline Line SettingYou can also configure signalling over the serial port connection as a terminal setting—ivrline with a value of yes—for each of the IVR lines configured within the Sphere system.Beginning with Sphericall v3.2, the Sphericall Softswitch uses the correct caller ID for calls regardless of whether FP *96 is passed between the voice messaging platform

Other Tones NA NA -24 dBm

Amplitude levels: Measured on the PhoneHub with HP 3558 Transmission & Noise Measuring Set relative to 600 Ohms.Acceptable Flashtime (range): 350-1100 ms

Call Progress Tones Frequency Cadence Amplitude

IVR Line

Caller ACaller ID = +1 8475551234 Caller B

Caller ID = 5011

Caller CCaller ID = +1 8475551234 Initial Caller ID = 5011

Attended Transfer to Caller C

Caller ID is that of Caller B until Caller B hangs up to connect the call.Then, caller ID information passed is that of Caller A.


. . .


and the Sphere system. This terminal setting notifies the MGC that a particular line is used for voice messaging functionality within the integration.Now, when an IVR line places an attended transfer call to a recipient party, the caller ID that is displayed is the caller ID of the original calling party, not the caller ID of the transferring station (as depicted in the following figure).

Figure 9.4 The ivrline Line Setting and Caller ID Presentation

IVR Line

Caller ACaller ID = +1 8475551234 Caller B

Caller ID = 5011

Caller CCaller ID = +1 8475551234 Initial Caller ID = +1 8475551234

Attended Transfer to Caller C

Caller ID is that of Caller A. W hen Caller B hangs up to connect thecall, the caller ID information passed is still that of Caller A.

ivrline term inalsetting here


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. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SMDI INSTALLATION 10

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B E F O R E Y O U B E G I N

• Have you installed, configured, and ensured network connectivity to all of the appropriate Sphere MGs and IP phones for your organization’s Sphere system?

• Have you configured the general PBX properties and functionality for your organization’s Sphere system?

• Have you defined the numbering plan for your organization’s Sphere system?• Have you defined all of the global and local system settings for the Sphericall

Manager?• Have you installed, configured, and integrated the Sphericall Desktop on

workstations throughout your organization’s Sphere system?

This chapter at a glance1 Prepare a Sphere system for SMDI integration with the voice mail platform2 Configure the Sphere system to support voice mail functionality

a. Configure Sphericall extensions as voice mail extensionsb. Configure voice mail and auto attendant hunt groups

3 Enable a message store that will serve as the intermediary between a Sphere system and voice mail

4 Test and troubleshoot the SMDI integration5 Interpret SMDI messages passed within an integration

N O T E SMuch of the information contained in this chapter is intended to serve as a quick reference point for integrating a Sphere system with a voice messaging platform. The more detailed, site- and system-specific information is contained in specific product manuals, for example refer to the your third-party voice message server documentation AND Sphericall manual for specifics SMDI integration.


S M D I I N S TA L L A T I O NPreparing the Sphere system

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P R E P A R I N G T H E S P H E R E S Y S T E M

Before installing and configuring the voice mail server as your organization’s voice messaging solution, you need to prepare your organization’s Sphere system for integration with the new voice mail server.

P R E P A R I N G T H E S P H E R E S Y S T E M

O B J E C T I V EAfter completing this lab you will be able to prepare a Sphere system for integration via SMDI by creating “phantom” voice mail extensions (extensions that will have numbers on the Sphericall end and will associate with the ports on the voice message server on the other end), configuring the voice mail and auto attendant extensions, as well as creating a message store on the Sphericall Manager that represents the SMDI voice messaging platform.

To set f low control set t ings1 Log on to the Sphericall Manager as Administrator.

From the Microsoft Windows Taskbar:2 Click Start\Settings\Control Panel.3 Double-click Ports.4 Highlight COM<X> where X is the number of the appropriate serial port to be used for

platform integration.5 Click Settings.


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. .S M D I I N S TA L L A T I O NPreparing the Sphere system

Figure 10.1 Settings for COM1 window

6 Select 9600 from the Bits per second drop-down list box.7 Select 7 from the Data bits drop-down list box.8 Select Even from the Parity drop-down list box.9 Select 1 from the Stop bits drop-down list box.

10 Select Hardware from the Flow control drop-down list box.11 Click Advanced.

Figure 10.2 Advanced Settings for COM1 window

12 Verify that the Use FIFO buffers check box is selected.


S M D I I N S TA L L A T I O NPreparing the Sphere system

Note: FIFO should always be enabled.The other settings should not be altered.

13 Click OK.14 Click OK.15 Exit Computer Management.

To enable the SMDI process1 Log on to the Sphericall Manager as SphereSupport.

From the Sphericall Administration application:2 Click Tools\Commission Sphericall Manager.

Figure 10.3 Sphericall Manager Commissioning window

3 Click the Primary Sphericall Manager or Secondary Sphericall Manager radio button to declare if this Sphericall Manager is to be considered a Primary or Secondary Sphericall Manager.

There can be only one Primary Sphericall Manager in the Sphere system.

Note: If you selected the Secondary Sphericall Manager radio button, you will be asked to enter the Primary Sphericall Manager for your organization’s Sphere system. Otherwise, the commissioning process is the same.


. . .

. .S M D I I N S TA L L A T I O NPreparing the Sphere system


4 Select the appropriate check box if this Sphericall Manager will be connected to a voice mail system through a serial port (i.e. an SMDI link).

5 Click Next.


6 Click Finish.


S M D I I N S TA L L A T I O NConfiguring a Sphericall Numbering Plan for Voice Mail

C O N F I G U R I N G A S P H E R I C A L L N U M B E R I N G

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N F O R V O I C E M A I L

When integrating voice messaging and auto attendant systems with your organization’s Sphere system, you should establish a numbering plan that incorporates two lead numbers (one for the voice mail and one for the auto attendant) as well as the extensions used with the appropriate voice mail ports:

In accordance with your system integration design, you should create and configure the requisite extensions and stations only after you enable the SMDI service via the Sphericall Manager Configuration utility.Once you enable the SMDI service, you should then create and configure all of the “phantom extensions” that will serve as station ports for voice mail lines (i.e. 6000, 6001, 6002, 6003, etc.). The number of extensions required for your organization’s Sphericall/SMDI integration depends upon the number of ports to be connected to the voice mail server.

To conf igure a voice mai l extensionFrom the Sphericall Administrator application window:

1 Click the Number Plan tab.2 Click Add Extension Number.

Voice Mail Extension the Sphere system uses for voice mail (example: 4000 or 500).

Auto Attendant Extension the Sphere system uses for auto attendant (example: 4001 or 501).

Hunt Extensions Primary extensions the Sphere system uses to enable the voice messaging ports (example: 4005, 4006, 4007, etc. or 504, 505, 506, etc.).


. . .

. .S M D I I N S TA L L A T I O NConfiguring a Sphericall Numbering Plan for Voice Mail

Figure 0.1 Properties for New Extension window

3 Type the number of the phantom extension in the Number field.

A voice mail extension can never be the primary extension on any station. If only one extension exists on a station, that extension, by default, is made the primary extension. Phantom extensions are created to be the primary extensions associated with the stations.

4 Select Single Line from the Hunt Order list.5 Type the name of the extension in the Last Name field.

For example, VM Port1.6 Clear the Show in Phonebook check box to prevent this phantom extension from

appearing in your organization’s phonebook.7 Select the Unknown icon type.8 Click Apply.9 Click OK.

To conf igure a voice mai l s tat ionFrom the Sphericall Administrator application:

1 Click the Stations tab.2 Expand the MG(s) listed in the display area to view all available stations within your

organization’s Sphere system.3 Highlight the station you wish to configure.4 Click Station Properties.




5 Type the name of this station in the Line Name field.6 Select the zone with which to associate this station from the Zone list.7 Select None from the Pickup Group list.8 Select the telephony area with which to associate this station from the Telephony Area

list.9 Select the emergency group with which to associate this station from the Emergency

Group list.10 Select VM Port from the Default Profile list.

The VM Port default profile assigns the following settings for the line:

11 Select the In Service check box if you wish to classify this station as active within your organization’s Sphere system.

12 Clear the Log Calls check box.13 Select the Drop Loop Current check box.

Drop loop current is primarily used for paging and voice mail lines. For every station on a PhoneHub or BranchHub that is connected to a port on the voice mail platform, that station’s properties must be configured in this manner.


• Max Calls = 1 • Calls Allowed

• Outside Calls Allowed • Outside Forward Allowed

• No Call Waiting


. . .


To assign an extension to a stat ionIn the Numbers area:

1 Click Add Extension.2 Expand the folders listed in the file tree to view all available extensions within your

organization’s Sphere system.3 Highlight the appropriate phantom extension you wish to assign to this station.4 Click OK.5 Click Add Extension.6 Expand the folders listed in the file tree to view all available extensions within your

organization’s Sphere system.7 Highlight the appropriate voice mail extension to be assigned to this station.8 Click OK.9 Click Add Extension.

10 Expand the folders listed in the file tree to view all available extensions within your organization’s Sphere system.

11 Highlight the appropriate auto attendant extension to be assigned to this station.12 Click OK.

The three voice mail, auto attendant, and phantom voice mail extensions are now associated with this station and appear in the Numbers area.


To conf igure the voice mai l l ine set t ingThe Sphericall Manager uses the correct Caller ID for calls regardless of whether FP *96 is passed between the voice messaging platform and the Sphere system.This change requires the configuration of a terminal setting for each voice mail line within the Sphere system. This terminal setting notifies the Sphericall Manager that a particular line is used for voice messaging functionality within the integration.From the Properties for Station window:

1 Click the Settings tab.




2 Click Add.3 Click the highlighted line under Name.4 Select Set As Voice Mail from the list.5 Click the adjacent line under Value.6 Select true from the list.7 Click Apply.8 Click OK.

To re-conf igure the VM and AA hunt orderOnce you associate all phantom extensions with their appropriate stations, you need to re-configure the hunt order for the voice mail and auto attendant extensions.From the Sphericall Administration window:

1 Click the Number Plan tab.2 Expand the folders listed in the file tree to view all available extensions within your

organization’s Sphere system.3 Highlight the voice mail extension you wish to configure.4 Click Properties.5 Select Round Robin from the Hunt Order list.6 Click OK.

Repeat this process to configure the hunt order for your organization’s auto attendant extension.


. . .


To ver i fy Moni tor pr iv i leges for the SMDI instanceThe SMDI instance is automatically created and configured in the User Rights tab of the Sphericall Administrator application when you enable the SMDI process within the Sphericall Configuration utility. This instance has monitor privileges to all lines within a Sphere system in order to integrate the voice mail/telephone system integration between the Sphere system and the voice messaging system.If the SMDI instance was not created, you must contact your organization’s Sphere Certified Partner and notify them that you require an SMDI instance rebuilt for voice messaging system integration.From the Sphericall Administration window:

1 Click the User Rights tab.2 Expand the folders listed in the file tree to view all available user accounts within your

organization’s Sphere system.3 Highlight the SMDI instance.4 Click Properties.

Figure 10.8 Properties for User window

5 Click OK.


S M D I I N S TA L L A T I O NConfiguring a Media Server

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C O N F I G U R I N G A M E D I A S E R V E R

Media servers within a Sphere system define the MSRIDs and MSRLineIDs required for integration with a voice mail server. Configuration settings defined within each media server must mirror the appropriate, corresponding settings on the voice messaging server.

To create a media server wi th in a Sphere systemFrom the Sphericall Administrator application:

1 Click the General tab.2 Select Media Servers from the list.3 Right-click and select Add.

Figure 10.9 New Message Server window

4 Type the name of this Media Server in the Name field.

Note: You must configure a unique message store for each voice messaging server that is to be integrated with your organization’s Sphere system.

5 Type the name of the Sphericall Manager running the SMDI process in the Server field.

The name in this field may or may not be the name of the server upon which voice mail is installed. However, at least one Sphericall Manager throughout your organization must run the SMDI process if you wish to integrate the Sphere system with voice mail.

6 Type the 3-digit MSRID in the ID field.

This is a unique identifier that designates a particular Message Server Retrieval system. The number (between 000 and 999) is prepended with zeros to pad the length.This entry is typically 001 and is matched on the voice mail platform.


. . .

. .S M D I I N S TA L L A T I O NConfiguring a Media Server

7 Select the COM port that will be connecting the Sphericall Manager to the voice mail server from the Serial Port list.

Note: The specified COM port must be the COM port on the server that is running the SMDI process.

8 Select the number of digits (7 or 10) that the MSR is expecting to receive from the Number of Digits list.

This 7- or 10-digit number is the caller ID information sent to the voice mail server. This number should be consistent with the settings in the Properties for Trunk window\Inward Routing tab and the settings on the voice mail server.

Note: Sphere Communications recommends that you select 10 (which equates to the area code plus the 7-digit telephone number) as the caller ID digit length.

9 Select how unknown numbers are to appear as caller ID information from the Unknown Number list.

• Select Blank if caller ID information is to display nothing when no caller ID information is available.

• Select Zero if caller ID information is to display all zeros when no caller ID information is available.

Note: Sphere Communications recommends that you select Zero from the Unknown Number list. The presence or absence of these zero digits aids in the verification of message receipt during troubleshooting scenarios.

In the Lines area:10 Click Add to add the appropriate stations to the MSR.

These stations connect directly from the PhoneHub and/or BranchHub ports to the voice mail server’s Dialogic card(s). These stations are those that were associated with the phantom extensions you created for voice mail functionality.

11 Expand the MG(s) listed in the display area to view all available stations within your organization’s Sphere system.

12 Highlight the station(s) you wish to associate with this MSR.13 Click OK.

The associated stations appear in the Lines area.14 In the MSR Line ID column, type the appropriate MSR Line ID, in the appropriate field, for

each line assignment.

The MSR Line ID is an integer prepended with zeroes to create a 4-digit integer. It identifies the appropriate station to the associated message store.

Note: The MSR Line ID must match the associated port number on the voice mail server.

CAUTION! ALTHOUGH THE MSR LINE ID DOES NOT HAVE TO BEGIN WITH 0001 AND INCREMENT BY 1, A DIRECT CORRELATION MUST EXIST BETWEEN THE SETTINGS ON THE SPHERE SYSTEM AND THE SETTINGS ON THE VOICE MESSAGING PLATFORM.SPHERE COMMUNICATIONS STRONGLY RECOMMENDS THAT, FOR EASE OF INSTALLATION AND ADMINISTRATION, THE MSR LINE ID SERIES BEGINS WITH 0001 AND INCREMENTS BY 1 FOR EACH SUBSEQUENT LINE.

15 Click OK.

The newly-configured media server appears in the display area.


S M D I I N S TA L L A T I O NExport User Info from Sphericall

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . E X P O R T U S E R I N F O F R O M S P H E R I C A L L

User information may be exported from the Sphere system to be imported the SMDI connect voice messaging system. The information is delivered in a CSV file.

To export to CSVOn the Sphericall Administrator application:

1 Select File\Export.2 In the Export field, drop down to select Media Server CSV.3 Choose the correct Zone and/or Media Server.4 Optionally you may choose Template Mailbox.5 Select the name order for your exported data: Lastname, Firstname or Fristname

Lastname.6 Click Export.7 Enter the folder location and name the file.

Copy the file and load on the Media Server of the 3rd party voice messaging server.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . T E S T I N G T H E V O I C E M E S S A G I N G P L A T F O R M

Once the integration of the Sphere system and voice mail is complete, calls can be sent from the addresses within the Sphere system to the appropriate ports on the voice mail server in order to verify connection status. If your organization’s integration has been designed with breakout boxes and RJ-11 cables, you can test all of the voice messaging port assignments in this manner.When you are finished with the testing, you should replace all RJ-11 cables and ensure full connection between the Sphere system and the voice mail server.To complete a full range of testing for the integration, you should verify the functionality of the following features:• Direct subscriber access of mailboxes• Direct calls into voice mail by the operator• Set MWI and Cancel MWI (with MWI-capable telephones only)• Call forward to a personal greeting• Call disconnect


. . .

. .S M D I I N S TA L L A T I O NTesting the Voice Messaging Platform

To test d i rect subscr iber access of mai lboxes1 Verify that the Sphere system is programmed appropriately to pass integration

information to voice mail.

To test d i rect ca l ls into voice mai l by the operatorOn the voice mail server:

1 Create a Subscriber mailbox for an operator.a. Match the alternate extension for the Subscriber mailbox with the operator’s actual

extension number on the Sphere system.b. Set the type for the alternate extension to Operator.c. Select the Set MWI check box in the Subscriber Mailbox Options window\Features

tab.d. Select None from the Subscriber Mailbox Options window\Answering tab\Busy

Action drop-down list box.e. Click the First radio button in the Subscriber Mailbox Options window\Features

tab\Clear MWI group box.

On the Sphericall Manager:2 Open the Sphericall Administrator application.3 Assign the appropriate user rights to the operator in the User Rights tab.

On the operator’s client workstation:4 Open the Sphericall Desktop.5 Open the appropriate line to be used by the operator.

Using one of the test telephones:6 Dial 0 to call the operator.

From the Sphericall Desktop window (open on the operator’s client workstation):7 Transfer the active call to voice mail.

The caller whom the operator wished to transfer is now connected, after a 2-second delay, to voice mail.

To test the set t ing and cancel l ing of MWINote: You can only test MWI on telephones with message waiting indicators.

This test involves the use of two test telephones.From the first test telephone:

1 Place a call to the lead number of the SMDI integration.

In the training lab, the lead number of the integration is 5000.2 Enter the appropriate security code to access the Subscriber mailbox.3 Record a message to be delivered to the first test Subscriber mailbox.

From the second test telephone:4 Place a call to the lead number of the SMDI integration.5 Enter the appropriate security code to access the Subscriber mailbox.6 Listen to the test message.7 Delete the message.

Once voice mail ends the call session:


S M D I I N S TA L L A T I O NTesting the Voice Messaging Platform

8 Confirm that the MWI has been canceled on the test telephone.

If the MWI is not canceled on the test telephone:9 Manually cancel MWI for the affected station by using the Administrative Star Code

*9710 to disable the MWI.

Note: if a user has more than one extension assigned to that station, the MWI may continue to flash until all the extension messages have been received.

To test forwarding to voice mai l and to test ca l l d isconnectOn the Sphericall Manager:

1 Open the Sphericall Administrator application.2 Click the Number Plan tab.3 Configure fowarding conditions for the test telephone stations so that calls will be

covered to voice mail after 3 rings.

With coverage enabled, the Sphericall Manager will send any calls to your extension(s) that remain unanswered after 3 rings to voice mail.From the first test telephone:

4 Place a call to the second test telephone.5 Do not answer the test call.

After 3 rings, the Sphericall Manager should send the call to voice mail.6 Use the Monitor status utility to verify that voice mail completes the following actions:

a. Answers the call sent from the Sphericall Manager.b. Forwards it to the appropriate Subscriber mailbox.

Once voice mail answers the call, you should hear the personal greeting recorded for this Subscriber mailbox.

7 Record a message for this mailbox.8 Hang up the telephone.9 Use the Monitor utility to verify that voice mail disconnects the call once it is finished

processing the recorded message for the appropriate subscriber.

To test voice messaging integrat ion1 Dial the extension of the voice messaging system.2 Record name(s) and personal greeting(s) for a sampling of the voice mailboxes and their

corresponding extensions.3 Dial one (ore more) of the configured-for-voice mail extensions and listen to the

greeting.4 Leave a voice message to the extension’s voice mailbox.5 Verify that the MWI light flashes on the recipient’s telephone set once you are finished

leaving the sample voice message.6 Dial the extension of the voice messaging system.7 Enter the appropriate security code for the intended recipient’s extension/voice mailbox.8 Listen to the voice message.9 Forward the voice message to another voice mailbox on the system.

10 Verify that the MWI light flashes on the second recipient’s telephone set once you are finished with forwarding the sample message.

11 Delete the voice message.


. . .


12 Verify that the MWI light turns off on the recipient’s telephone.

To test auto at tendant integrat ion1 Dial the extension of the auto attendant for the system.2 Verify your organization’s greeting(s).3 Dial the extension of the auto attendant.4 Dial 0 (or the appropriate operator extension) to test utilization of the auto attendant to

reach the operator.

R E S T A R T S & R E F R E S H E SIf, for any reason, the Message Waiting Indicators lose their synchronization with the Sphericall Manager or any of the MGs, you must refresh the MWI manually on the voice messaging platform.

Note: You can refresh the MWI utilizing the SMDI Process window.

S M D I C A L L R E C O R D SRefer to the following table to interpret standard SMDI messages when utilizing another voice messaging vendor’s equipment to troubleshoot your organization’s Sphericall and SMDI integration.

Table 10.1 SMDI Codes

Code Description

MD Message desk (incoming call record)

MWI Message waiting indication (error record)

ggg Message desk number or MSRID

mmmm Message desk terminal number or MSR Line ID

a Any of the following letters where:A = forwarded all calls ("is unavailable")N = forwarded on ring-no-answer ("doesn’t answer")B = forwarded on busyD = direct call to the voice messaging platform ("to enter your mailbox")U = unknown reason code (call sent to auto attendant)

xxxxxxx Called party station number followed by a space. If no called party, then just a space

yyyyyyy Calling party station number (if known) followed by a space.If no called party, then just a space (depending on voice application platform.If the number is 7 digits and the platform requires 10 digits, it will prepend the number with zeros.

— Indicates a space separating information.



C O M M O N S M D I M E S S A G E C O D I N G

Direct Call Message<cr><lf>MDgggmmmmD<sp>yyyyyyyyyy<sp><cr><lf><ctrl y>• ggg is the Message Desk Number or MSRID. This is a three-digit identifier that

specifies which voice messaging system the message is intended. If a voice messaging system gets a message with an MSRID that is not its own, it ignores the message.

• mmmm is the Message Desk Terminal (MSR Line ID). This is a four-digit number that specifies upon which phone line the call will be presented to the voice messaging system. This is how the system matches the incoming call with the integration message.

• yyyyyyyyyy is the Calling Station Number. In the case of direct dial, it is the phone whose mailbox will be accessed.

• Calling number will be either 7 or 10 digits.

Forward All Calls<cr><lf>MDgggmmmmAxxxxxxxxxx<sp>yyyyyyyyyy<sp><cr><lf><ctrl y>• ggg is the Message Desk Number or MSRID.• mmmm is the Message Desk Terminal (MSR Line ID).• xxxxxxxxxx is the forwarding from station or called station number. This is the

mailbox to which the voice messaging system will link this message.• yyyyyyyyyy is the calling station number.• The calling station may be omitted, in which case xxxxxxxxxx is replaced with a

<sp>.• Calling and forwarding number will be either 7 or 10 digits.

Forward On Busy<cr><lf>MDgggmmmmBxxxxxxxxxx<sp>yyyyyyyyyy<sp><cr><lf><ctrl y>• ggg is the Message Desk Number or MSRID.• mmmm is the Message Desk Terminal (MSR Line ID).• xxxxxxxxxx is the forwarding from station or called station number. This is the

mailbox to which the voice messaging system will link this message.• yyyyyyyyyy is the calling station number.• The calling station may be omitted, in which case xxxxxxxxxx is replaced with a


Forward On No Answer<cr><lf>MDgggmmmmNxxxxxxxxxx<sp>yyyyyyyyyy<sp><cr><lf><ctrl y>• ggg is the Message Desk Number or MSRID.• mmmm is the Message Desk Terminal (MSR Line ID).• xxxxxxxxxx is the forwarding from station or called station number. This is the

mailbox to which the voice messaging system will link this message.


. . .


• yyyyyyyyyy is the calling station number.• The calling station may be omitted, in which case xxxxxxxxxx is replaced with a


Forward On Unknown Reason<cr><lf>MDgggmmmmUxxxxxxxxxx<sp>yyyyyyyyyy<sp><cr><lf><ctrl y>• ggg is the Message Desk Number or MSRID.• mmmm is the Message Desk Terminal (MSR Line ID).• xxxxxxxxxx is the forwarding from station or called station number.• yyyyyyyyyy is the calling station number.• The calling station may be omitted, in which case xxxxxxxxxx is replaced with a

<sp>.• Calling and forwarding number will be either 7 or 10 digits.• This message is typically used to trigger the auto attendant menu linked to the

called party, xxxxxxxxxx.

Invalid Message Waiting<cr><lf>MWIxxxxxxxxxx<sp>INV<cr><lf><ctrl y>• xxxxxxxxxx is the forwarding from station or called station number. This is the

mailbox to which the voice messaging system will link this message.• Calling and forwarding number will be either 7 or 10 digits.• This message is sent to the voice messaging system when an MWI light request is

sent for an extension that does not exist. This causes the voice messaging platform to cease transmitting the messages for this mailbox.

Note: This is currently not implemented in the Sphere system as it causes no harm with our current loading.

Blocked Message Waiting<cr><lf>MWIxxxxxxxxxx<sp>BLK<cr><lf><ctrl y>• xxxxxxxxxx is the Forwarding From station or called station number. This is the

mailbox to which the voice messaging system will link this message.• Forwarding number will be either 7 or 10 digits.• This message is sent to the voice messaging system when a message waiting light

request could not be acted upon.• This is currently not implemented in the Sphere system. All message waiting

requests will be acted upon if received in the Sphere system.• Testing is underway to see if this can be used to sync the message waiting lights

on Sphericall Manager re-initializations.

Message Waiting Indicator MessagesOP:MWI<sp>xxxxxxxxxx!<ctrl D>• Turns on (operates) the message waiting light for the extension at xxxxxxxxxx.RMV:MWI<sp>xxxxxxxxxx!<ctrl D>



• Turns off (removes) the message waiting light for the extension at xxxxxxxxxx.


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SMDI VOICE MAIL TROUBLESHOOTING 11

SMDI Troubleshooting Information

T R O U B L E S H O O T I N G S M D I V O I C E M E S S A G I N GInstances may occur when calls bound for a voice messaging platform never reach their voice mailbox destination(s). If calls are sent only to the auto attendant and never to the voice mail server, it is likely that the serial cable connection between the Sphericall and voice mail systems has been compromised. Issues of this sort are, at times, difficult to detect in that the Sphericall system will continue to function normally even though voice messaging sessions are never offered to calls within the integration.Sphere recommends the use of the Sphere Troubleshooting Methodology to ascertain the source of any problems in the system. This problem solving methodology is found in the Sphericall Maintenance and Troubleshooting Manual.

To t roubleshoot a voice messaging plat form1 Verify a secure connection at the Sphericall Manager(s).2 Verify a secure connection at the voice messaging platform(s).

If connection disruptions persist:3 Resolve serial cable or COM port operability issues within the Sphericall/SMDI

integration.

H Y P E R T E R M I N A L M O N I T O R I N GSphere Communications recommends the use of HyperTerminal to monitor the status of information processed through the appropriate COM port(s) within a Sphericall/SMDI integration.

To t roubleshoot COM port set t ingsFrom the Microsoft Windows Taskbar:

1 Click Start\Programs\Accessories\Communications\HyperTerminal.


S M D I VO I C E M A I L TR O U B L E S H O O T I N G

Figure 11.1 Connection Description window

2 Type the name of this connection in the Name field.3 Select an icon for the connection.4 Click OK.

Figure 11.2 Connect To window

5 Select the appropriate COM port for connection to the voice messaging platform from the Connect using list.

6 Click OK.


. . .

. .S M D I VO I C E M A I L TR O U B L E S H O O T I N G

Figure 11.3 COM Properties window

7 Configure the COM port as necessary for the appropriate connection to the voice messaging platform.

8 Click OK.

Figure 11.4 HyperTerminal window

You can now view the SMDI packets as they pass through the selected COM port.

To ver i fy the sending of informat ion f rom the Spher ical l Manager1 Detach the RS-232 cable from the voice mail server’s serial port.



2 Connect the cable to another COM port on the Sphericall Manager or on another network resource in order to place the Sphericall Manager into loopback state.orGenerate calls to voice mail.

3 Monitor the HyperTerminal utility to verify that the Sphericall Manager is attempting to send information to the voice mail server.

4 Reconnect the RS-232 (Null Modem) cable to the voice messaging platform.

To ver i fy the sending of informat ion f rom the voice mai l server1 Dial an extension within your organization’s Sphericall numbering plan.2 Leave a voice message in the Subscriber mailbox associated with that extension.3 Monitor the appropriate station to ensure that the telephone’s MWI light flashes.

orMonitor the appropriate station to ensure that the stutter dial tone sounds in the telephone handset, indicating a waiting voice message.

C O M M O N R E S O L U T I O N S C E N A R I O S

Table 11.1 Voice Mail Problem/Resolution Scenarios

T R O U B L E S H O O T I N G S M D I V O I C E M E S S A G I N GInstances may occur when calls bound for a voice messaging platform never reach their voice mailbox destination(s). If calls are sent only to the auto attendant and never to the voice mail server, it is likely that the serial cable connection between the

Case # Issue Resolution

Q4008 Auto attendant and voice mail do not answer.Also, a power outage had occurred and both servers were at the NT logon.

1. Log on to both servers.2. Ensure that the Sphere services have started on Spheri-

call Manager and that the voice mail services have started on the voice mail server.

3. Open the Line Status utility on the voice mail server.4. Test the integrationIf the test is unsuccessful:5. Check the message stores configured on the Sphericall

Manager (The In Service check box must be selected).6. Restart the SMDI and voice mail services.7. Retest the integration.

Q3870 Users receive an error message each time they attempt to access voice mail.

1. Check the MSR Line ID in the Message Store Properties window.

2. Remove the extension number in the MSR Line ID table.3. Enter 0001 through 0004 in the MSR Line ID table (corre-

sponding to the number of voice mail ports in use on the system).

4. Remove the IP address from the Server field.5. Type the actual name of the server in the Server field.

Q3838 Callers hear the “Name Not Recognized” message when they enter letters of a subscriber's last name within the company directory.Mailboxes are set up and names have been recorded.

The name entry in voice mail was First Name, Last Name.1. Change the naming conventions used for the Subscriber

mailboxes to Last Name, First Name (without the comma).


. . .


Sphericall and voice mail systems has been compromised. Issues of this sort are, at times, difficult to detect in that the Sphericall system will continue to function normally even though voice messaging sessions are never offered to calls within the integration.Sphere recommends the use of the Sphere Troubleshooting Methodology to ascertain the source of any problems in the system. This problem solving methodology is found in the Sphericall Maintenance and Troubleshooting Manual.

To t roubleshoot a voice messaging plat form1 Verify a secure connection at the Sphericall Manager(s).2 Verify a secure connection at the voice messaging platform(s).

If connection disruptions persist:3 Resolve serial cable or COM port operability issues within the Sphericall/SMDI

integration.

H Y P E R T E R M I N A L M O N I T O R I N GSphere Communications recommends the use of HyperTerminal to monitor the status of information processed through the appropriate COM port(s) within a Sphericall/SMDI integration.

To t roubleshoot COM port set t ingsFrom the Microsoft Windows Taskbar:

1 Click Start\Programs\Accessories\Communications\HyperTerminal.

Figure 11.5 Connection Description window

2 Type the name of this connection in the Name field.3 Select an icon for the connection.4 Click OK.



Figure 11.6 Connect To window

5 Select the appropriate COM port for connection to the voice messaging platform from the Connect using list.

6 Click OK.

Figure 11.7 COM Properties window

7 Configure the COM port as necessary for the appropriate connection to the voice messaging platform.

8 Click OK.


. . .


Figure 11.8 HyperTerminal window

You can now view the SMDI packets as they pass through the selected COM port.

To ver i fy the sending of informat ion f rom the Spher ical l Manager1 Detach the RS-232 cable from the voice mail server’s serial port.2 Connect the cable to another COM port on the Sphericall Manager or on another network

resource in order to place the Sphericall Manager into loopback state.orGenerate calls to voice mail.

3 Monitor the HyperTerminal utility to verify that the Sphericall Manager is attempting to send information to the voice mail server.

4 Reconnect the RS-232 (Null Modem) cable to the voice messaging platform.

To ver i fy the sending of informat ion f rom the voice mai l server1 Dial an extension within your organization’s Sphericall numbering plan.2 Leave a voice message in the Subscriber mailbox associated with that extension.3 Monitor the appropriate station to ensure that the telephone’s MWI light flashes.

orMonitor the appropriate station to ensure that the stutter dial tone sounds in the telephone handset, indicating a waiting voice message.



C O M M O N R E S O L U T I O N S C E N A R I O S

Table 11.2 Voice Mail Problem/Resolution Scenarios

R E S T A R T S A N D R E F R E S H E SIf for any reason, the Message Waiting Indicators lose their synchronization with the Sphericall Manager or any of the MGs, you must refresh the MWI manually on the voice messaging platform.

Note: You can refresh the MWI utilizing the SMDI Process window.

V O I C E M A I L T R O U B L E S H O O T I N G S T E P SIf Sphericall is integrated with Microsoft Exchange, voice mail is dependent on many factors for correct operation:• MAPI must be installed on the Sphericall Manager in order to integrate with

Exchange.• The version of Exchange must be at least Microsoft Exchange Server 5.5.• The telephony service must be running on the Sphericall Manager as this service is

used for the call control of voice mail.• Networking must be in good working order and name resolution must be available

in both directions. This means that you must be able to PING by name from the

Case # Issue Resolution

Q4008 Auto attendant and voice mail do not answer.Also, a power outage had occurred and both servers were at the NT logon.

1. Log on to both servers.2. Ensure that the Sphere services have started on Spheri-

call Manager and that the voice mail services have started on the voice mail server.

3. Open the Line Status utility on the voice mail server.4. Test the integrationIf the test is unsuccessful:5. Check the message stores configured on the Sphericall

Manager (The In Service check box must be selected).6. Restart the SMDI and voice mail services.7. Retest the integration.

Q3870 Users receive an error message each time they attempt to access voice mail.

1. Check the MSR Line ID in the Message Store Properties window.

2. Remove the extension number in the MSR Line ID table.3. Enter 0001 through 0004 in the MSR Line ID table (corre-

sponding to the number of voice mail ports in use on the system).

4. Remove the IP address from the Server field.5. Type the actual name of the server in the Server field.

Q3838 Callers hear the “Name Not Recognized” message when they enter letters of a subscriber's last name within the company directory.Mailboxes are set up and names have been recorded.

The name entry in voice mail was First Name, Last Name.1. Change the naming conventions used for the Subscriber

mailboxes to Last Name, First Name (without the comma).


. . .


l

f

t

. :

Sphericall Manager to the Exchange server and receive a reply, and you must be able to do this in reverse order as well.

• The RPC service must be running. RPC calls are used for a majority of TAPI and MAPI functions as well as some functionality from the vmail.exe process to the pbx.exe process.

If all of these factors are functioning as specified, the voice mail service will work.

H A S I T E V E R W O R K E D ?When troubleshooting a problem with many complex interactions, such as voice mail, follow the Sphere methodology. The first question you should ask is “Has it ever worked?” If it has never worked most likely the integration was not set up correctly. If it has worked in the past then something has changed. This is the most difficult question to correctly answer. The SMDI standard is a simple standard and has very few requirements. The majority of SMDI integration problems will fall into the “No” category. If it has worked, you should then ask, “What has changed?” As before, there are conditions that affect the operation of the voice mail should they change. Once you know the symptoms, you can ask the correct questions to determine the possible solution.

C O M M O N V O I C E M A I L I S S U E STable 11.3 SMDI Causation Table

Problem Symptom Possible Cause

A user can only reach the auto attendant during configuration.

During configuration of the voice mail system, every time a call is made, there is a long pause and the auto attendant plays the main greeting.

The serial packets are not reaching the voice mail server or thepackets are not 100% accurate. This can be caused by an incorrect cable, non-matching serial port configurations, or incorrect serial port usage. Whenever a voice mail system hasreceived no packet or a packet that it does not recognize, it wiltend to wait some time while it attempts to retrieve a recognizable packet. If it does not receive this packet, it will time out and travel to the main auto attendant.

If the cable is plugged in to an incorrect serial port in the back oeither machine, the packet will not be recognized by the voice mail system.The communication port on your organization’s server might not be set appropriately. Each of the serial ports on the motherboard has the capability to be any of four differencommunication ports. It is frequently assumed that COM1 is thetopmost port on the back of the server. This is not always the case. Note: The system BIOS provides port settings. Upon re-initializing the server, press the delete key or F2 to enter the system setup. Under the integrated peripherals option you cansee the com port setting for the serial ports on the motherboardThe first serial port is typically the top or the furthest to the left.These settings relate to the operating system view of the portsCOM1: Address 3f8 IRQ 4COM2: Address 2f8 IRQ 3COM3: Address 3e8 IRQ 4COM4: Address 2e8 IRQ 3



t

t

f

A caller accesses the wrong mailbox when trying to leave a message.

After a caller is transferred to the voice mail system from coverage conditions, the caller is given the wrong voice mailbox greeting from the person with whom they wished to speak.

The MSR Line ID is not configured with the correct values. TheMSR Line ID is a value that corresponds to the actual physicalline number that is entering the voice mail server. The field is a4-digit value that is pre-pended with zeroes.

For example, the first line into the voice mailbox would be 0001.This number has to match up with the line ID on the voicemail server; it is standard to make the first line 0001, the secondline 0002, etc. This line identifier is sent over the serial port with the call packeto tell the voice mail that the call that is coming over line one has X packet. The system will see a phone call coming in on aspecific line and wait for the related packet until it answers thecall. If two lines are switched and they each get a call, both callers will get an incorrect voice mailbox.

A caller accesses the auto attendant on a busy system.

After a caller is transferred to the voicemail system from coverage conditions, the caller is given the wrong voice mailbox greeting from the person with whom they wished to speak.

The MSR Line ID is not configured with the correct value on thelast few lines. This will only show itself during peak calling hours if the hunt group is set to Linear. If the last few lines are reversed, you may either get an incorrect voice mailbox or thesystem will time out to the auto attendant.

Serial port is unavailable for use.

The error message “SerialPort Open Failed for COM1:” appears.

The serial port is no longer available from the system. MicrosofWindows NT does not allow programs to share system devicessuch as parallel ports. If the device is used by another application (such as a modem) when SMDI has started, this error message will appear. Either choose a different serial port or remove the other user othe appropriate serial port.

Message waiting notification is not working.

Users receive no stutter dial tone on a phone without a message waiting light.

The affected line does not have stutter dial tone enabled. Fromthe Sphericall Administration application, you must enable stutter dial tone under the Properties For Multiple Stations window.

Users receive no stutter dial tone on a phone without a message waiting light; they just hear a crackling every half second.

User perception of stutter dial tone is incorrect. Most people expect to hear what is called Message Waiting dial tone so theyare usually confused by true “stutter” dial tone. Change the system setting to message waiting dial tone.

Users receive no message waiting indication at all.

The cable is only a simplex cable with one pair from Transmit on the PBX side to Receive on the voice mail side. Change thecable to a duplex cable.The voice mail system is not set to send message waiting notification. This is typically a configured option and will need tobe set correctly on the voice mail side. Another possible cause is incorrect serial port configuration. Certain combinations allow one-way communications to have few errors while the other way might have more. Confirm that the configurations are the same.



. . .

. .S M D I VO I C E M A I L TR O U B L E S H O O T I N GSummary

f

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S U M M A R Y

Upon completing this chapter, you should now have a better understanding of SMDI. You will also increase your ability to recognize voice mail issues and how to resolve them.The Sphericall system integrates with voice mail platforms utilizing SMDI, an industry-standard method of integrating a voice messaging platform with a PBX using the RS-232c serial connection found on any personal computer.

Unable to retrieve voice mail from the telephone.

When a user dials in from the outside, he hears the auto attendant but is unable to retrieve voice mail.

When integrated with Microsoft Exchange Server 5.0, name resolution is faulty. If the Sphericall Softswitch can not resolve the name of Exchange Server, the auto attendant will still answer calls andtake voice mail messages even though you cannot retrieve them. This is because all greetings are cached on the Sphericall Softswitch after first connection to Microsoft Exchange Server 5.0. If name resolution changes between the time the Sphericall Softswitch first starts and the attempt to retrieve messages, voice mail messages might be unretrievable through the telephone user interface.

When a user dials in from the outside he is able to attempt voice mail dialing but is told that he has entered an invalid extension.

When integrated with Microsoft Exchange Server 5.0, the mailbox is no longer associated with the extension. This is possibly caused by the Administrator removing the mailbox from the extension in the Sphericall Administration application.

Dialed phone number incorrectly (user error).

The voice mail auto attendant does not answer at the main number.

The user gets an invalid password message.

The user is entering an invalid password. Change the passwordfrom Microsoft Outlook 98.

Greetings are known to have been changed but have not taken effect.

The old greeting is playing after the greeting was changed.

There may be a timer for refreshing the greetings. If you change your voice mail greetings, allow approximately five minutes for the system to update changes before confirming changes.

The user is unable to transfer to voice mail or reach the auto attendant.

The user dials the main number but gets a busy tone.

Voice mail ports may be busy. This will happen at peak times othe day when everyone is checking voice mail from the phone or when customers are calling into the system. This usually occurs around an organization’s opening and closing time.



S M D I VO I C E M A I L TR O U B L E S H O O T I N GSummary


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SNMP INTEGRATION 12

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B E F O R E Y O U B E G I N

• Have you installed, configured, and ensured network connectivity to all of the appropriate Sphere MGs and IP phones for your organization’s Sphere system?

• Have you configured the general System properties and functionality for your organization’s Sphere system?

• Have you defined the numbering plan for your organization’s Sphere system?• Have you defined all of the global and local system settings for the Sphericall

Manager?• Have you installed, configured, and integrated the Sphericall Desktop on

workstations throughout your organization’s Sphere system?

In th is chapter , you wi l l learn how to• Monitor a Sphere system utilizing the functionality inherent with SNMP

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I M P L E N E T W O R K M A N A G E M E N T P R O T O C O L

O V E R V I E WAdministrators have the ability to monitor IP network information throughout a Sphere system via the Simple Network Management Protocol, a software-based management service that accepts information messages from endpoints using UDP.This information is essentially the same information that can be obtained via tell commands. With SNMP, you can now use an SNMP manager (in conjunction with point-and-click functionality) to gather the same information formerly obtained by using the MG command line interface.SNMP relies upon the functionality of two types of devices within a network, an SNMP manager and the SNMP agents.• The SNMP manager resides on a network resource and serves as the collection

point for all data generated by the traps. An SNMP manager’s main functions are as follows:• To gather information from SNMP agents using the Get and Get-Next requests• To modify settings on the appropriate SNMP agents using the Set request• To accept traps from SNMP agents


S N M P I N T E G R A T I O NSimple Network Management Protocol

• The SNMP agents are client machines or devices such as workstations, network hubs, resource servers, etc. located throughout an organization’s network. All Sphere MGs are built with the SNMP agent installed. Once the SNMP agents have been configured, they are responsible for the following tasks:• To respond to queries from the SNMP manager• To monitor hardware devices for any service faults and notify the SNMP

manager if a fault is detected

Note: Information that is passed from the SNMP agent to the SNMP manager is called a “trap” (i.e. a specific error condition). You must configure each device to send traps to the SNMP manager via a trap destination address. This is a System Initialization Setting configured via the Sphericall Administrator application (SNMP Trap Destination setting).

An organization may decide to use the SNMP functionality upon initial installation of the Sphere system. Sphericall’s Management Information Bases must be installed on the designated SNMP manager utilizing standard installation mechanisms. Once administrators specify the location of the Sphericall MIBs, which reside on the Sphericall software DVD-ROM, they can use the SNMP manager to query Sphere hardware by selecting the appropriate OID defined in one of the MIBs.

R E Q U I R E M E N T SThe Sphere system adheres to the following SNMP standards and compatibility:• SNMP Version: MIBs-II Version 1• SNMP Specification: RFC 1213, as stated in IETF definitions

S N M P C O M M U N I T Y N A M EThe SNMP access community name is the password for Telnet and grants either read or read/write privileges. There is a default SNMP access write community and a default SNMP access read community defined on the MG. The following community names are defined on the MG:• The default community name for SNMP access write is private.• The default community name for SNMP access read is public.

To change the SNMP access wr i te community name1 Open a serial connection or a Telnet connection to the MG.2 Enter the following command:

prompt> snmp access write <community>

where <community> is the new password.Exampleprompt> snmp access write bigcomputer

Note: Changing the SNMP write community name does NOT overwrite the default password. It adds an additional password which can be used to gain access to the MG. You can also restrict access by entering the IP address of the only device to have access to the MG.


. . .

. .S N M P I N T E G R A T I O NSimple Network Management Protocol

• To view all the SNMP community names, use the snmp access list command.• To remove a community name from the access list, use the snmp access delete

<community> command.• To remove all community names, use the snmp access flush command.

S N M P M A N A G E R A N D A G E N T C O N F I G U R A T I O NSNMP managers and agents reside on the same network as the Sphere system. You must configure these manager(s) and agent(s) in order to establish networking parameters for all subsequent addressing queries.

Figure 12.1 SNMP Manager and Agent Network Design

The Manager- to-Agent- to-Manager Moni tor ing Process1 MIBs store all available OIDs.2 The administrator selects an OID and specifies the device to monitor by its IP address.3 The SNMP manager builds the SNMP request.4 The SNMP manager transmits the request to the selected device.5 The SNMP agent responds to the request.6 The SNMP manager displays the information received from the SNMP agent.

The Agent- to-Manager Trap Funct ion1 An event occurs at the SNMP agent.2 The SNMP agent sends a trap, as defined in the MIBs, if the event falls within a

predefined set of possible events.3 The SNMP manager receives the SNMP agent’s trap.4 The SNMP manager references the MIBs.5 The SNMP manager displays the trap with other error condition severity information.

Network

MIB

Network

SNMP Manager (server)

ManagerSoftware:

SNMPUDP

IPPHY

SNMP Agent (client)

Agent Software

Network


S N M P I N T E G R A T I O NSphere SNMP Traps

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S P H E R E S N M P T R A P S

C R I T I C A L T R A P I N F O R M A T I O NRefer to the following table for the specific trap information relative to a Sphere system.

Table 12.1 Sphere SNMP Traps

Sphere SNMP Traps

-24 volt supply out of specification

-12 volt supply out of specification - VBX only

+2.5 volt supply out of specification

+3.5 volt supply out of specification

+5 volt supply out of specification

+12 volt supply out of specification

DSP programming failed (per board basis)

Fan fault detected

FPGA programming failed (per board basis)

HV supply voltage out of specification on board 0

HV supply voltage out of specification on board 1

Media stream connection failures (per board basis)

MGC connection lost

MGC connection gained

Temperature too high on engine board

Temperature too high on LM78 - VBX only

Trunk came up

Trunk went down

Unknown board type detected (per board basis)


. . .

. .S N M P I N T E G R A T I O NSphere SNMP MIBs

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S P H E R E S N M P M I B S

O V E R V I E WOnce you have installed and configured the SNMP manager on a network resource server within your organization’s existing network, you must load the SNMP manager with the appropriate Sphere SNMP MIBs. These MIBs can be found on the Sphericall software DVD-ROM in the \server\data\vbx folder.

I N S T A L L A T I O NNote: Because several SNMP manager applications currently exist, the following

information regarding the installation of Sphere SNMP MIBs is rather general in nature.You should integrate the Sphere SNMP MIB files into the SNMP manager according to the SNMP manager’s application installation instructions.

R E Q U I R E M E N T SYou must load the following two files before proceeding with the installation of the remaining Sphere SNMP MIBs:• sphere-reg.mib• sphere-tc.mib

To insta l l Sphere SNMP MIBs on an SNMP manager1 Open the SNMP manager software.2 Load the sphere-reg.mib file.3 Load the sphere-tc.mib file.4 Load the following MIBs to complete the Sphere SNMP MIBs installation procedure:

5 Exit the SNMP manager software.

• sphere-board.mib

• sphere-board-trap.mib

• sphere-channel-trap.mib

• sphere-channel.mib

• sphere-e2prom.mib

• sphere-hlog.mib

• sphere-lm78.mib

• sphere-lm78-trap.mib

• sphere-lm80.mib (new with VG3)

• sphere-lm80-trap.mib (new with VG3)

• sphere-mgcfinder-trap.mib

• sphere-system.mib

• sphere-version.mib


S N M P I N T E G R A T I O NSphere SNMP MIBs

C O N T E N T SRefer to the following table for a list of the information provided and available within each Sphere SNMP MIB.

Table 12.2 SNMP SNMP MIBs Contents

Name Contents

Table of version objects • Telephony subsystem build date and time• Telephony controller version• Media stream version• Station version• Trunk version• Telephony driver version• Multi-media version• MGC interface version• CPLD version - VBX units only

Table of system objects • Amount of free memory• Port type• MGC address• Number of installed channels• Number of connected channels• Loop current loss threshold• Number of active calls• Total number of calls

Table of telephony board description objects

• Board type• Board ID• Board revision• DSP bootstrap image description, date, and time• DSP image description, date, and time• FPGA image description, date, and time• Echo canceller type• Line termination impedance

Table of channel description objects • Channel installed status• MGC connection status• Hook state status• Call in progress status• Volume setting• Gain setting

Table of e2prom objects • Scroll rate• Watchdog cycles


. . .

. .S N M P I N T E G R A T I O NSphere SNMP Object IDs

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S P H E R E S N M P O B J E C T I D S

Object IDs comprise the base of data within Sphere’s SNMP MIBs.Refer to the following table for an index of the Sphere SNMP OIDs that are able to be monitored via SNMP within a Sphere system.

Table 12.3 Sphere SNMP Object IDs

Table of voltage/temperature objects • 60 volt sense reading for board 0• 60 volt sense reading for board 1• 5 volt supply reading - VBX units only• 12 volt supply reading• -12 volt supply reading• -24 volt supply reading• Lm78 chip temperature

VG3 Units:• -58 volt sense reading for board 0• -58 volt sense reading for board 1• +2.5 volt supply reading• +3.3 volt supply reading• +5 volt supply reading• +12 volt supply reading• +24 volt supply reading

OID Name OID Number Related Tell Command

sphereRoot 1.3.6.1.4.1.4613 N/A

sphereRegistration 1.3.6.1.4.1.4613.1 N/A

spherePlatformReg 1.3.6.1.4.1.4613.1.1 N/A

sphereVimReg 1.3.6.1.4.1.4613.1.1.1 N/A

sphereHubReg 1.3.6.1.4.1.4613.1.1.2 N/A

sphereCohubReg 1.3.6.1.4.1.4613.1.1.3 N/A

spherePhonePortReg 1.3.6.1.4.1.4613.1.1.4 N/A

sphereBranchReg 1.3.6.1.4.1.4613.1.1.5 N/A

sphereSystemsReg 1.3.6.1.4.1.4613.1.2 N/A

sphereGeneric 1.3.6.1.4.1.4613.2 N/A

spherePlatformGen 1.3.6.1.4.1.4613.2.1 N/A

spherePfmGenObjs 1.3.6.1.4.1.4613.2.1.1 N/A

historicLogTable 1.3.6.1.4.1.4613.2.1.1.1 N/A

Name Contents


S N M P I N T E G R A T I O NSphere SNMP Object IDs

historicLogEntry 1.3.6.1.4.1.4613.2.1.1.1.1 N/A

sequenceNumber 1.3.6.1.4.1.4613.2.1.1.1.1.1 N/A

notificationId 1.3.6.1.4.1.4613.2.1.1.1.1.2 N/A

notificationNumber 1.3.6.1.4.1.4613.2.1.1.1.1.3 N/A

affectedEntityId 1.3.6.1.4.1.4613.2.1.1.1.1.4 N/A

severity 1.3.6.1.4.1.4613.2.1.1.1.1.5 N/A

timestamp 1.3.6.1.4.1.4613.2.1.1.1.1.6 N/A

ipAddress 1.3.6.1.4.1.4613.2.1.1.1.1.7 N/A

macAddress 1.3.6.1.4.1.4613.2.1.1.1.1.8 N/A

phoneNumber 1.3.6.1.4.1.4613.2.1.1.1.1.9 N/A

dataId 1.3.6.1.4.1.4613.2.1.1.1.1.10 N/A

versionTable 1.3.6.1.4.1.4613.2.1.1.2 N/A

versionEntry 1.3.6.1.4.1.4613.2.1.1.2.1 N/A

versionIndex 1.3.6.1.4.1.4613.2.1.1.2.1.1 N/A

versionTelBuildInfo 1.3.6.1.4.1.4613.2.1.1.2.1.2 telcon info

versionTelController 1.3.6.1.4.1.4613.2.1.1.2.1.3 telcon info

versionMediaStream 1.3.6.1.4.1.4613.2.1.1.2.1.4 telcon info

versionStation 1.3.6.1.4.1.4613.2.1.1.2.1.5 telcon info

versionTrunk 1.3.6.1.4.1.4613.2.1.1.2.1.6 telcon info

versionTelDriver 1.3.6.1.4.1.4613.2.1.1.2.1.7 telcon info

versionMultiMedia 1.3.6.1.4.1.4613.2.1.1.2.1.8 telcon info

versionMgcInterface 1.3.6.1.4.1.4613.2.1.1.2.1.9 telcon info

versionCpld 1.3.6.1.4.1.4613.2.1.1.2.1.10 engine hw

systemTable 1.3.6.1.4.1.4613.2.1.1.3 N/A

systemEntry 1.3.6.1.4.1.4613.2.1.1.3.1 N/A

systemIndex 1.3.6.1.4.1.4613.2.1.1.3.1.1 N/A

systemFreeMemory 1.3.6.1.4.1.4613.2.1.1.3.1.2 chips mem

systemSyncSource 1.3.6.1.4.1.4613.2.1.1.3.1.3 N/A

systemPortType 1.3.6.1.4.1.4613.2.1.1.3.1.4 N/A

systemMgcAddress 1.3.6.1.4.1.4613.2.1.1.3.1.5 telcon status

systemInstalledChannels 1.3.6.1.4.1.4613.2.1.1.3.1.6 telcon info

systemConnectedChannels 1.3.6.1.4.1.4613.2.1.1.3.1.7 N/A



. . .


systemLclThreshold 1.3.6.1.4.1.4613.2.1.1.3.1.8 telcon trunk setting

systemVoiceOnThreshold 1.3.6.1.4.1.4613.2.1.1.3.1.9 telcon trunk setting

systemVoiceOffThreshold 1.3.6.1.4.1.4613.2.1.1.3.1.10 telcon trunk setting

systemHysteresisTimer 1.3.6.1.4.1.4613.2.1.1.3.1.11 telcon trunk setting

systemActiveCalls 1.3.6.1.4.1.4613.2.1.1.3.1.12 telcon all calls

systemTotalCalls 1.3.6.1.4.1.4613.2.1.1.3.1.13 N/A

boardTable 1.3.6.1.4.1.4613.2.1.1.4 N/A

boardEntry 1.3.6.1.4.1.4613.2.1.1.4.1 N/A

boardIndex 1.3.6.1.4.1.4613.2.1.1.4.1.1 N/A

boardType 1.3.6.1.4.1.4613.2.1.1.4.1.2 engine <board number> dump

boardId 1.3.6.1.4.1.4613.2.1.1.4.1.3 engine <board number> dump

boardRevision 1.3.6.1.4.1.4613.2.1.1.4.1.4 engine <board number> dump

boardDspBootstrapDesc 1.3.6.1.4.1.4613.2.1.1.4.1.5 engine <board number> dump

boardDspImageDesc 1.3.6.1.4.1.4613.2.1.1.4.1.6 engine <board number> dump

boardFpgaImageDesc 1.3.6.1.4.1.4613.2.1.1.4.1.7 engine <board number> dump

boardEchoCancellerType 1.3.6.1.4.1.4613.2.1.1.4.1.8 engine <board number> dump

boardLineTermImpedance 1.3.6.1.4.1.4613.2.1.1.4.1.9 telcon trunk settings

channelTable 1.3.6.1.4.1.4613.2.1.1.5 N/A

channelEntry 1.3.6.1.4.1.4613.2.1.1.5.1 N/A

channelIndex 1.3.6.1.4.1.4613.2.1.1.5.1.1 N/A

channelInstalled 1.3.6.1.4.1.4613.2.1.1.5.1.2 telcon <start channel> <end channel> dump

channelConnectedToMgc 1.3.6.1.4.1.4613.2.1.1.5.1.3 telcon <start channel> <end channel> dump

channelHookState 1.3.6.1.4.1.4613.2.1.1.5.1.4 telcon summary

channelCallInProgress 1.3.6.1.4.1.4613.2.1.1.5.1.5 telcon <start channel> <end channel> calls

channelVolumeSetting 1.3.6.1.4.1.4613.2.1.1.5.1.6 telcon <start channel> <end channel> dump

channelGainSetting 1.3.6.1.4.1.4613.2.1.1.5.1.7 telcon <start channel> <end channel> dump

e2promTable 1.3.6.1.4.1.4613.2.1.1.6 N/A

e2promEntry 1.3.6.1.4.1.4613.2.1.1.6.1 N/A

e2promIndex 1.3.6.1.4.1.4613.2.1.1.6.1.1 N/A

e2promScrollRate 1.3.6.1.4.1.4613.2.1.1.6.1.2 engine list menus

e2promPeakCellRate 1.3.6.1.4.1.4613.2.1.1.6.1.3 N/A

e2promTempUnits 1.3.6.1.4.1.4613.2.1.1.6.1.4 N/A




e2promVpiNumber 1.3.6.1.4.1.4613.2.1.1.6.1.5 N/A

e2promWatchdogCycles 1.3.6.1.4.1.4613.2.1.1.6.1.6 N/A

lm78Table 1.3.6.1.4.1.4613.2.1.1.7 N/A

lm78Entry 1.3.6.1.4.1.4613.2.1.1.7.1 N/A

lm78Index 1.3.6.1.4.1.4613.2.1.1.7.1.1 N/A

lm78Neg60VoltBoard0 1.3.6.1.4.1.4613.2.1.1.7.1.2 engine list menus

lm78Neg60VoltBoard1 1.3.6.1.4.1.4613.2.1.1.7.1.3 engine list menus

lm78Pos5VoltSupply 1.3.6.1.4.1.4613.2.1.1.7.1.4 engine list menus

lm78Pos12VoltSupply 1.3.6.1.4.1.4613.2.1.1.7.1.5 engine list menus

lm78Neg12VoltSupply 1.3.6.1.4.1.4613.2.1.1.7.1.6 engine list menus

lm78Neg24VoltSupply 1.3.6.1.4.1.4613.2.1.1.7.1.7 engine list menus

lm78ChipTemperature 1.3.6.1.4.1.4613.2.1.1.7.1.8 engine list menus

Im80Table 1.3.6.1.4.1.4613.2.1.1.8 engine list menus

Im80Entry 1.3.6.1.4.1.4613.2.1.1.8.1 engine list menus

Im80Index 1.3.6.1.4.1.4613.2.1.1.8.1.1 engine list menus

Im80Neg58VoltBoard0 1.3.6.1.4.1.4613.2.1.1.8.1.2 engine list menus

Im80Neg58VoltBoard1 1.3.6.1.4.1.4613.2.1.1.8.1.3 engine list menus

Im80Pos2Pt5VoltBoard 1.3.6.1.4.1.4613.2.1.1.8.1.4 engine list menus

Im80Pos3Pt3VoltBoard 1.3.6.1.4.1.4613.2.1.1.8.1.5 engine list menus

Im80Pos5VoltSupply 1.3.6.1.4.1.4613.2.1.1.8.1.6 engine list menus

Im80Pos12VoltSupply 1.3.6.1.4.1.4613.2.1.1.8.1.7 engine list menus

Im80Neg24VoltSupply 1.3.6.1.4.1.4613.2.1.1.8.1.8 engine list menus

spherePfmGenTraps 1.3.6.1.4.1.4613.2.1.2 N/A

spherePfmBoardTraps 1.3.6.1.4.1.4613.2.1.2.1 N/A

boardTypeTrap 1.3.6.1.4.1.4613.2.1.2.1.1 N/A

boardDspProgramTrap 1.3.6.1.4.1.4613.2.1.2.1.2 N/A

boardFpgaProgramTrap 1.3.6.1.4.1.4613.2.1.2.1.3 N/A

spherePfmChannelTraps 1.3.6.1.4.1.4613.2.1.2.2 N/A

channelMsConnectionTrap 1.3.6.1.4.1.4613.2.1.2.2.1 N/A

spherePfmLm78Traps 1.3.6.1.4.1.4613.2.1.2.3 N/A

lm78TempHighEngTrap 1.3.6.1.4.1.4613.2.1.2.3.1 N/A

lm78Neg60vBoard0Trap 1.3.6.1.4.1.4613.2.1.2.3.2 N/A



. . .


lm78Neg60vBoard1Trap 1.3.6.1.4.1.4613.2.1.2.3.3 N/A

lm78Pos5vSupplyTrap 1.3.6.1.4.1.4613.2.1.2.3.4 N/A

lm78TempHighChipTrap 1.3.6.1.4.1.4613.2.1.2.3.5 N/A

lm78Pos12vSupplyTrap 1.3.6.1.4.1.4613.2.1.2.3.6 N/A

lm78Neg12vSupplyTrap 1.3.6.1.4.1.4613.2.1.2.3.7 N/A

lm78Neg24vSupplyTrap 1.3.6.1.4.1.4613.2.1.2.3.8 N/A

lm78FanFaultTrap 1.3.6.1.4.1.4613.2.1.2.3.9 N/A

spherePfmMgcFinderTraps 1.3.6.1.4.1.4613.2.1.2.4 N/A

mgcFinderConLostTrap 1.3.6.1.4.1.4613.2.1.2.4.1 N/A

Note: This trap is network connection-based, not channel-based. For example, if a Sphere PhoneHub loses its connection to a Sphericall Manager, only one trap will be sent to the SNMP manager from the SNMP agent rather than one trap for each affected channel on that MG.

mgcFinderConGainTrap 1.3.6.1.4.1.4613.2.1.2.4.2 N/A

Note: This trap is network connection-based, not channel-based. For example, if a Sphere PhoneHub loses its connection to a Sphericall Manager, only one trap will be sent to the SNMP manager from the SNMP agent rather than one trap for each affected channel on that MG.

mgcFinderTrunkDownTrap 1.3.6.1.4.1.4613.2.1.2.4.3 N/A

mgcFinderTrunkUpTrap 1.3.6.1.4.1.4613.2.1.2.4.4 N/A

sphereSystemsGen 1.3.6.1.4.1.4613.2.2 N/A

sphereSysGenObjs 1.3.6.1.4.1.4613.2.2.1 N/A

sphereSysGenTraps 1.3.6.1.4.1.4613.2.2.2 N/A

sphereProduct 1.3.6.1.4.1.4613.3 N/A

spherePlatformProd 1.3.6.1.4.1.4613.3.1 N/A

sphereSystemsProd 1.3.6.1.4.1.4613.3.2 N/A

sphereCapabilities 1.3.6.1.4.1.4613.4 N/A

sphereRequirements 1.3.6.1.4.1.4613.5 N/A

sphereExperimental 1.3.6.1.4.1.4613.6 N/A



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. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .MICROSOFT WINDOWS INSTALLER 13

W I N D O W S I N S T A L L E R A N D S P H E R I C A L L

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D E S K T O P I N S T A L L A T I O N

Previously, Sphericall Desktops were installed using either an installation CD/DVD or shared network drive containing a copy of the CD/DVD. Installers were required to have administrative privileges for the PC on which the Sphericall Desktop is being installed. The initial installation of larger Sphere systems required several installers to walk around logging into client PCs and loading the Sphericall Desktop. This process is time consuming.Sphericall introduces support for Microsoft Windows Installer. Windows installation packages can be managed using the Active Directory’s Group Policy thereby centralizing installation.

Note: Sphere Communications assumes that the Sphere system administrator has experience with Microsoft networking and Group Policy.

Microsoft Windows Installer is an installation and configuration service that reduces the total cost of ownership. The Installer ships with Windows Vista, the Windows Server 2003 family, Windows XP, and Windows 2000.

I . G R O U P P O L I C YUsing Windows Installer, the Sphere administrator has a way of installing, upgrading or uninstalling the Sphericall Desktop on PCs of users who do not have local administrative privileges for their PCs.Instmsi.exe is the redistributable package for installing or upgrading Windows Installer. To manage an “MSI” package, use the following Group Policy: User Configuration (or Computer Configuration)->Software installation->New package. Sphericall does not provide zap installation files.Group Policy is not supported on Windows 98, ME or Windows NT Workstation 4.0.The Sphericall_Desktop.msi and its supporting files are placed into a shared directory. This directory does not have to be on a Sphericall Manager. A true file server can be used for better performance.

G R O U P P O L I C Y S E T T I N G S

A. Elevated PrivilegesThe Sphericall installation program writes entries into the registry at HKEY_LOCAL_MACHINE. It also copies files into the system32 directory. Both actions require administrative privilege to the target PC. If users do not have


M I C R O S O FT WI N D O W S I N S TA L L E RWindows Installer And Sphericall Desktop Installation

administrative privileges, the following Group Policy setting should be enabled: User Configuration (or Computer Configuration)->Administrative Templates->Windows Components->Windows Installer-> "Always install with elevated privileges".

B. Allowing User InputIf the administrator wants users to input installation parameters as when the Maximum user interface option is used, the following Group Policy setting should be enabled: Computer Configuration->Administrative Templates->Windows Components->Windows Installer->"Enable user control over installs".

D E P L O Y M E N T S E T T I N G SThe deployment type defines how the Sphere installation process is initiated:

A. PublishedYou publish an application when you want the application to be available to people managed by the Group Policy object, should a user want the application. With published applications, it is up to each person to decide whether or not to install the published application. When the Sphericall Desktop is published, the program is available in the client's Add/Remove Programs -> Add New Programs list within the client's Control Panel. Pressing "Add" will start the Sphericall Desktop installation.

B. AssignedYou assign an application when you want everyone to have the application on his or her computer. When the Sphericall Desktop is assigned a Sphericall Desktop icon will appear in each user's Start menu. The first time the application is invoked, the installation process will begin.C. Run at LogonThis setting allows the installation to run automatically when the user logs onto the system.

U S E R I N T E R F A C E O P T I O N

A. BasicThe "Installation User Interface Option" of "Basic" does not allow the user to configure any attribute of the installation. The Basic option should be used with the AutoInstall.ini configuration file to create sensible defaults for language and primary server name.

Note: The installation location is at Program Files\Sphere

B. MaximumThe "Installation User Interface Option" of "Maximum" allows the user the opportunity to set the install path, language and primary server name. The Maximum option should not be used with the AutoInstall.ini configuration file because the user-entered values will be overridden by values from the file.Using the Maximum user interface option presents the following dialog boxes to the user:• License Agreement


. . .

. .M I C R O S O FT W I N D O W S I N S TA L L E RWindows Installer And Sphericall Desktop Installation

• Language Selection• Destination Folder• Primary Sphericall Manager• Ready to Install Confirmation• Installation Status /Installation Complete

Table 13.1 U/I Option Comparison

U N I N S T A L LUsers that have been granted permissions to uninstall software may use the Control Panel’s “Add or Remove Programs” to uninstall the Sphericall Desktop.Group Policy can be configured to uninstall the Sphericall Desktop from client PC. The uninstall must be set to run with elevated security privileges.

T A P I S E R V I C E P R O V I D E RThe Sphericall TAPI service provider is installed with the Desktop and initialized when the user runs the Desktop. This is different from Sphericall v4.0 where the TAPI service provider initialization was initiated using a Configure button within the Control Panel’s “Phone and Modem Options” facility.

I N T E R A C T I O N B E T W E E N C L I E N T U P D A T E R A N D M S I I N S T A L L A T I O N SClient Updater’s function is to upgrade existing Sphericall Desktop installations. Client Updater is an application that runs as a separate executable on each client PC. It has the same lifespan as the Sphericall Desktop in that it starts/stops automatically when Desktop starts/stops. Its purpose is to receive and act on upgrade notifications sent from the Desktop Manager. The Desktop Manager’s function is to manage the Sphericall Desktop upgrades. When a new Sphericall Desktop is available, the files Update.exe and ClientUpdate.exe are placed in the DesktopUpdate folder. The Desktop Manager will notify Client Updaters of the new files. A client will get notified of a new update even if it already has upgraded (manually or using Group Policy) to the same version.Client Updater and the Desktop Manager are supported in Sphericall v4.1. If Group Policy is going to be used to install or upgrade Sphericall clients, the Desktop Manager should be disabled. The administrator should choose not to place the update.exe desktop file in the …\Sphericall\DesktopUpdate folder during the server installation or upgrade.

No AutoInstall.ini settings AutoInstall.ini settings

Basic User I/F Not supported – There is no way to configure the server name.

Recommended choice – Administrator defines server name & language. Install path is hard coded.

Maximum User I/F Supported – Users are prompted for server name, language and install path.

Not recommended – The user input will be ignored in favor of the AutoInstall.ini parameters. This may be confusing for users.



I I . I N S T A L L A T I O N C O M P O N E N T SThere are several installation components built or managed by the Sphere system.

C O M P O N E N T S• Sphericall_Desktop.msi is an InstallShield built installation program. It does the

bulk of the file copying and registry setup. The registry keys and values that are added are the same for each installation.

• InstallConfig.exe is an MFC application used for loading the TAPI service provider and for creating the more dynamic registry settings. InstallConfig.exe uses AutoInstall.ini to load administrator-defined preferences. Preferences include definition of the primary Sphericall Manager and Desktop language. AutoInstall.ini will be shipped with no preferences defined.

I I I . D I F F E R E N C E S B E T W E E N M S I A N D T H E D E S K T O P M A N A G E R

A D V A N T A G E S• Group Policy allows the administrator to centrally install and uninstall the Sphericall

Desktop. Previous Desktop versions required an installer to go to each client PC to perform the initial installation.

• Group Policy can be configured so that users are not required to have explicit administrative privileges over their PCs.

• The Sphericall Desktop MSI package doesn’t have to be on the Sphericall Manager for it to be installed. The administrator can place the files into a file server that is capable of handling a large amount of downloads.

• The Desktop Manager runs exclusively on the Primary Sphericall Manager. Therefore, it can place a substantial burden on the Sphericall Manager and Sphere systems with a large number of Sphericall Desktops.

• Administrators are familiar with Active Directory and Group Policy tools.• Active Directory and Group Policy allow upgrades to be managed in the same way

as installs & uninstalls.

D I S A D V A N T A G E S• The Desktop Manager has an integrated display that offers an inventory of all the

PC clients that have ever updated. Each entry includes: operating system, date of last upgrade, whether or not an upgrade is needed, and logon account of user who did the last upgrade.

To insta l l the Spher ica l l Desktop Insta l la t ionFrom the MSIInstall folder (located under the Client folder on the Sphericall DVD):

1 Copy all files/folders to the appropriate distribution point.2 Based upon the distribution point, modify the AutoInstall.ini file accordingly.

• The lines that have a ; at the beginning are comments and will be ignored.


. . .

. .M I C R O S O FT W I N D O W S I N S TA L L E RWindows Installer And Sphericall Desktop Installation

• Server = must contain the primary Sphericall Manager server name• Language = 0 is english (default), 1 is Mexico-Spanish, 2 is German, 3 is Italian, 4

is French-Canada, 5 is French (France), 6 is Spanish (Spain).• The Sphericall Desktop MSI installation script (Sphericall Desktop.msi) will use this

AutoInstall.ini file If it is located in the sub directory \Program Files\Sphere of the root directory from where the Sphericall Desktop.msi is located.

• AutoInstall.ini is typically used when the Sphericall Desktop is installed through group policy using the ‘basic’ user interface option.

• The use of this file is not recommended when the Sphericall Desktop installation is run from group policy using ‘maximum’ user interface option. It’s also not recommended when executing “Sphericall Desktop.msi directly. The field values defined here takes precedence over input provided by the user/installer.

3 Set up the Group Policy.

Instruct ions for upgrading a user ’s Spher ical l Desktop1 Click Start\Control Panel.2 Click Add or Remove Programs.3 Click Add New Programs.

The available Sphericall Desktop program will appear.4 Click Add.

The installation will install.


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. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .AUDIOCODES 14

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A U D I O C O D E S M P 1 1 X A C C E S S & S E T U P

• Verify system requirements for AudioCodes MP11X and Sphericall.• The Sphere system should be installed, configured and tested as fully functional.• Refer to the MP11XFXO Analog Gateway User Manual for installation planning,

setup, package contents, safety and conditions of use.

O V E R V I E W O F O P E R A T I O NThe AudioCodes MP11X Analog Media Gateway is a multi-port analog gateway that connects analog terminals, PBXs or key systems to the IP network using FXO connectivity. Using AudioCodes Analog Media Gateways, Sphere system Administrators can effectively deliver carrier-hosted converged services as well as enterprise-based applications.MP112 - 2 Port FXO GatewayMP114 - 4 Port FXO GatewayMP118 - 8 Port FXO Gateway

Note: The AudioCodes MP11X series spans ranging from 2 to 8 analog ports. The X in the following documentation refers to any (2, 4, 6, 8) of the multi-port gateways

Note: It is important to note that setup for all of the MP11X series gateways is identical.

Choose this method i f the insta l ler doesn’ t have access to the DHCP lease l is t . 1 Choose a computer that will be used for configuring the AudioCodes gateway. The

computer will have its network disrupted during configuration so do not choose an in-service Sphericall Manager.


A U D I O C O D E SAudioCodes MP11X Access & Setup

2 Make sure the AudioCodes configuration files are accessible from the configuration computer (copied onto or accessible via the Sphericall DVD).

3 Power up the AudioCodes FXO MP11X gateway. A new AudioCodes FXO MP11X gateway uses a statically bound IP address of 10.1.10.11.

4 Change the configuration computer’s subnet mask to 255.255.0.0. Change its IP address to an address on the same subnet as the AudioCodes gateway (ex. 10.1.10.12).

5 Either connect the gateway directly to a computer using a cross-over cable OR connect both the computer and gateway together using a switch / hub.

6 Access the gateway Quick Setup screen using a web browser such as Internet Explorer. The URL is http://10.1.10.11.

Choose this method i f the insta l ler has access to the DHCP lease l is t . 1 Choose a PC for configuring the AudioCodes gateway that is in the same subnet as the

gateway. 2 Make sure the AudioCodes configuration files are accessible from the configuration

computer (copied onto or accessible via the Sphericall DVD).3 Power up the AudioCodes FXO MP11X gateway. When the AudioCodes FXO MP11X

gateway is connected on a network that supports DHCP, it will be given an IP address within the DHCP address scope.

4 Determine the gateway’s new IP address. The DHCP provided IP address can be determined by searching for the MAC address of the gateway within the IP address lease list of the DHCP server. The MAC address of the gateway is on a sticker located on the bottom side of the unit.

5 Access the gateway Quick Setup screen using a web browser such as Internet Explorer. Use the IP address of the gateway as the URL (ex. http://172.16.15.35).

To update and conf igure: independent of the method of access 1 The gateway’s embedded web server will prompt for a username and password. Enter

“Admin” for both (case sensitive).2 Download the software image. Go to Software Updates and browse to the AudioCodes

file directory. Download MP118_SIP_F4.80A.020.001.cmp to the AudioCodes gateway. After 2 minutes, refresh the browser window. A new looking configuration pane will be displayed.

3 From the Software Updates -> Load Auxiliary Files, load the following files:a. INI file: SampleSIP.ini

Note: Review the SampleSIP.ini file for specific configuration information.b. FXO Coefficient file: MP1xx12-1-16khz-fxo.datc. Call Progress Tone: spherecp.dat


. . .

. .A U D I O C O D E SAudioCodes MP11X Access & Setup

Using the AudioCodes FXO MP-11X Quick Setup ScreenFigure 14.1 AudioCodes MP-11X Quick Setup screen

I P C O N F I G U R A T I O N1 From the Quick Setup screen, enter either a valid static IP address & subnet mask that is

appropriate for the target network.2 In the “Default Gateway IP Address” field, configure the IP address of the default

gateway for the subnet..

S I P P A R A M E T E R S3 Enter the name of the AudioCodes gateway in the Gateway Name field.4 When working with a Proxy server, set the ‘Working with Proxy’ field to ‘Yes’.5 Enter the Proxy Name of the primary Proxy server in the ‘Proxy IP Address’ field. When

no Proxy is used, the internal routing table is used to route the calls.6 Enter the Proxy Name in the ‘Proxy Name’ field. If Proxy name is used, it replaces the

Proxy IP address in all SIP messages. This means that messages are still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead.

7 Set Enable Registration to “Enable”. ‘Disable’ = the MediaPack does not register to a Proxy server/Registrar (default). ‘Enable’ = the MediaPack registers to a Proxy server/Registrar at power up and every ‘Registration Time’ seconds; The MediaPack sends a REGISTER request according to the ‘Authentication Mode’ parameter.


A U D I O C O D E SAudioCodes MP11X Access & Setup

8 Click the Reset button and click OK in the prompt; the MediaPack applies the changes and restarts.

S P H E R I C A L L A D M I N I S T R A T O R A P P L I C A T I O N9 From the Sphericall Administrator Application Trunk panel: verify the gateway has been

recognized by the system.10 Access the Properties for Hub for the gateway. Select the Gateway Admin button. An

Internet Explorer will appear and attempt to access the gateway’s embedded web server.11 If this gateway is to be configured with a secondary (redundant) MGC, go to the Protocol

Management screen followed by the Protocol Definition screen and enter an MGC IP address other than the primary in the Redundant Call Agent IP field.

Using the Automat ic Dial ing TableUse the Automatic Dialing Table to define telephone numbers that are automatically dialed when an incoming call is received.

1 Open the Automatic Dialing page: Protocol Management\EndpointSettings submenu\Automatic Dialing.

Figure 14.2 Automatic Dialing Table window

2 In the Destination Phone Number field for a port, enter the telephone number to dial.

Incoming calls may be directed to a specific extension by specifying a valid extension. It is possible to specify the same extension for every port (e.g. the Auto Attendant) or a different extension for each port on this page (e.g. DID lines).It is possible to use the default routing specified in the Sphericall Admininstrator application by specifyng an invalid extension for one or more of the ports on the Automatic Dialing page. For example, if the nonexistant extension "xyz" is specified on the Automatic Dialing page and the domain is "spherecom.com", when an incoming call is received, the MP-11x will send an INVITE to "[email protected]". Since "xyz" does not exist on the system, the MGC will apply the default route to the call.

3 In the ‘Auto Dial Status’ field, select one of the following:


. . .

. .A U D I O C O D E SAdvanced Configuration

• Enable [1] – When a port is selected, when making a call, the number in the Destination Phone Number field is automatically dialed if ring signal is applied to port.

• Disable [0] – The automatic dialing option on the specific port is disabled (the number in the Destination Phone Number field is ignored).

• Hotline [2] – When a phone is offhook and no digit is pressed for HotLineDialToneDuration, the number in the Destination Phone Number field is automatically dialed.

4 Repeat step 3 for each port you want to use for Automatic Dialing.5 Click the Submit button to save your changes.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A D V A N C E D C O N F I G U R A T I O N

It is strongly recommended that your organization’s Sphere Administrator change these settings for proper operation of the device.

C H A N N E L S E L E C T M O D E *Menu: Protocol Management\Protocol Definitions\General ParametersDisplay: Channel Select ModeRecommended Value: DescendingComments: Channel 8 will be used for the first outbound call, then channel 7 and so on.

I S P R O X Y U S E DMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Enable ProxyRecommended Value: Use ProxyComments: A proxy must be used with Sphericall.

P R O X Y N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Proxy NameRecommended Value: Host of your primary MGCComments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP-11X to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.


A U D I O C O D E SAdvanced Configuration

P R O X Y I PMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Proxy IP AddressRecommended Value: Host of your primary MGCComments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP-11X to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.

S I P G A T E W A Y N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Gateway NameRecommended Value: A value unique to each MP-11X, for example, MP-11X-1Comments: A gateway name should be specified. If the gateway name is left unspecified, the MP-11X will use its IP address instead. If the IP address ever changes, example due to DHCP, the MP-11X will appear as a new gateway to Sphericall.

I S R E G I S T E R N E E D E DMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Enable RegistrationRecommended Value: EnableComments:

R E G I S T R A R N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Registrar NameRecommended Value: Hostname of your primary MGCComments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP-11X to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.

R E G I S T R A T I O N T I M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Registration TimeRecommended Value: 600 secondsComments:


. . .


U S E R N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: User NameRecommended Value: Must be the same as the gatway name

A U T H E N T I C A T I O N M O D EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Authentication ModeRecommended Value: Per GatewayComments: Setting the authentication mode to per gateway forces the MP-11X to register once for the entire gateway. More sophisticated configurations may require per endpoint.

C O D E R N A M EMenu: Protocol Management\Protocol Definition\CodersDisplay: Coder NameRecommended Value: g711Ulaw64k,20,0,$$,0Comments: This value enables G.711 U law. Other CODECs, such as G.711 A law and G.729 may also be enabled.

E N A B L E C U R R E N T D I S C O N N E C TMenu: Protocol Management\Advanced Parameters\General ParametersDisplay: Enable Current DisconnectRecommended Value: EnableComments: This setting causes the MP-11X to drop the call if loss of loop current is detected. This typically occurs when the party on the other side of the PSTN hangs up.

D I S C O N N E C T O N B R O K E N C O N N E C T I O NMenu: Protocol Management\Advanced Parameters\General ParametersDisplay: Disconnect on Broken ConnectionRecommended Value: NoComments: If set to yes, the MP-11x will disconnect a call when it has not received any RTP packets for a configurable length of time. This causes premature disconnection of calls into Sphere voice mail and other devices that do not sent RTP packets when they are muted.



C U R R E N T D I S C O N N E C T D U R A T I O NMenu: None (INI file parameter)Display: None (INI file parameter)Recommended Value: 300 millisecondsComments: This setting sets the loss of loop current detection window to 200 to 600 milliseconds. This setting is not accessible via the web interface. It must be manually added to the INI file.

T I M E T O S A M P L E A N A L O G L I N E V O L T A G EMenu: None (INI file parameter)Display: None (INI file parameter)Recommended Value: 100 millisecondsThis setting forces the MP-11X to check the trunks for loss of loop current every 100 milliseconds. This settingis not accessible via the web interface. It must be manually added to the INI file.

E N A B L E C A L L E R I DMenu: Protocol Management\Advanced Parameters\Supplementary ServicesDisplay: Enable Caller IDRecommended Value: EnableComments: Enables the delivery of Caller ID to the SIP Proxy in the Display name field of the INVITE From header.

P S T N P R E F I XMenu: Protocol Management\Routing Tables\IP to Hunt Group RoutingDisplay: Dest Phone PrefixRecommended Value: *,1,*,xxx.xxx.xxx.xxx,1Comments: This directs all outbound calls to the PSTN to the first hunt group.

T R U N K G R O U P _ 1Menu: Protocol Management\Endpoint Phone NumbersDisplay: 1Recommended Value: 1-8, MP-11X, 1Comments: This places all channels in the first hunt group.

E N A B L E C A L L E R I D _ < P O R T >Menu: Protocol Management\Endpoint Settings\Detect Caller ID from Tel


. . .


Display: Port <Port>Recommended Value: EnableComments: This is a per channel setting. Enable it if your line delivers caller ID.

T A R G E T O F C H A N N E L < P O R T >Menu: Protocol Management\Endpoint Settings\Automatic dialingDisplay: Port <Port>Recommended Value: <Auto Attendant Extension> EnableComments: This is a per channel setting that determines where calls inbound from the PSTN on that channel are routed to.

I S T W O S T S T A G E D I A LMenu: Protocol Management\FXO SettingsDisplay: Dialing ModeRecommended Value: One StageComments: Use one stage dialing if the MP-11X is connected directly to the PSTN. Use two stage dialing if the MP-11X is connected to a PBX.

I S W A I T F O R D I A L T O N EMenu: Protocol Management\FXO SettingsDisplay: Waiting for Dial ToneRecommended Value: YesComments: Enabling this setting forces the MP11X to detect dialtone before dialing.

D T M F T R A N S P O R T T Y P EMenu: Advanced Configuration\Media Settingss\Voice SettingsDisplay: DTMF Transport TypeRecommended Value: Transparent DTMF Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.

M F T R A N S P O R T T Y P EMenu: Advanced Configuration\Media Settingss\Voice SettingsDisplay: MF Transport TypeRecommended Value: Transparent MF


A U D I O C O D E SAudioCodes MP104 Access & Setup

Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A U D I O C O D E S M P 1 0 4 A C C E S S & S E T U P

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• Verify system requirements for AudioCodes MP104 and Sphericall.• The Sphere system should be installed, configured and tested as fully functional.

O V E R V I E W O F O P E R A T I O NThe AudioCodes MP104 Analog Media Gateway is a 4 port analog gateway that connects analog terminals, PBXs or key systems to the IP network using FXO connectivity. Using AudioCodes Analog Media Gateways, Sphere system Administrators can effectively deliver carrier-hosted converged services as well as enterprise-based applications.The AudioCodes MP104 SIP gateway is supported in Sphericall V5.1. Sphericall requires the AudioCodes version ID to be at or greater than 4.6. The following procedure outlines how to upgrade and configure an AudioCodes MP104 to version 4.60A.036.005 and allow it to check into a Sphere MGC.

Choose this method i f the insta l ler doesn’ t have access to the DHCP lease l is t . 1 Choose a computer that will be used for configuring the AudioCodes gateway. The

computer will have its network disrupted during configuration so do not choose an in-service Sphericall Manager.

2 Make sure the AudioCodes configuration files are accessible from the configuration computer (copied onto or accessible via the Sphericall CD).

3 Power up the AudioCodes FXO MP104 gateway. A new AudioCodes FXO MP104 gateway uses a statically bound IP address of 10.1.10.11.


. . .

. .A U D I O C O D E SPlanning

4 Change the configuration computer’s subnet mask to 255.255.0.0. Change its IP address to an address on the same subnet as the AudioCodes gateway (ex. 10.1.10.12).

5 Either connect the gateway directly to a computer using a cross-over cable OR connect both the computer and gateway together using a switch / hub.

6 Access the gateway Quick Setup screen using a web browser such as Internet Explorer. The URL is http://10.1.10.11.

Choose this method i f the insta l ler has access to the DHCP lease l is t . 1 Choose a PC for configuring the AudioCodes gateway that is in the same subnet as the

gateway. 2 Make sure the AudioCodes configuration files are accessible from the configuration

computer (copied onto or accessible via the Sphericall CD).3 Power up the AudioCodes FXO MP104 gateway. When the AudioCodes FXO MP104

gateway is connected on a network that supports DHCP, it will be given an IP address within the DHCP address scope.

4 Determine the gateway’s new IP address. The DHCP provided IP address can be determined by searching for the MAC address of the gateway within the IP address lease list of the DHCP server. The MAC address of the gateway is on a sticker located on the bottom side of the unit.

5 Access the gateway Quick Setup screen using a web browser such as Internet Explorer. Use the IP address of the gateway as the URL (ex. http://172.16.15.35).

To update and conf igure: independent of the method of access1 The gateway’s embedded web server will prompt for a username and password. Enter

“Admin” for both (case sensitive).2 Download the software image. Go to Software Updates and browse to the AudioCodes

file directory. Download 4.60A.036.005.cmp to the AudioCodes gateway. After 2 minutes, refresh the browser window. A new looking configuration pane will be displayed.

3 From the Software Updates -> Load Auxiliary Files, load the following files:a. INI file: SampleSIP.ini

Note: Review the SampleSIP.ini file for specific configuration information.b. FXO Coefficient file: MP1xx12-1-16khz-fxo.datc. Call Progress Tone: spherecp.dat


A U D I O C O D E SPlanning

Using the AudioCodes FXO MP104 Quick Setup ScreenFigure 14.3 AudioCodes MP104 Quick Setup screen

I P C O N F I G U R A T I O N1 From the Quick Setup screen, enter either a valid static IP address & subnet mask that is

appropriate for the target network.2 In the “Default Gateway IP Address” field, configure the IP address of the Sphericall

Manager that is considered primary for this device. It may not necessarily be the primary Sphericall Manager for the system.

S I P P A R A M E T E R S3 Enter the name of the AudioCodes gateway in the Gateway Name field.4 When working with a Proxy server, set the ‘Working with Proxy’ field to ‘Yes’.5 Enter the Proxy Name of the primary Proxy server in the ‘Proxy IP Address’ field. When

no Proxy is used, the internal routing table is used to route the calls.6 Enter the Proxy Name in the ‘Proxy Name’ field. If Proxy name is used, it replaces the

Proxy IP address in all SIP messages. This means that messages are still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead.

7 Set Enable Registration to “Enable”. ‘Disable’ = the MediaPack does not register to a Proxy server/ (default). ‘Enable’ = the MediaPack registers to a Proxy server/Registrar at power up and every ‘Registration Time’ seconds; The MediaPack sends a REGISTER request according to the ‘Authentication Mode’ parameter.


. . .

. .A U D I O C O D E SPlanning

8 Click the Reset button and click OK in the prompt; the MediaPack applies the changes and restarts.

Fax Signal ing Method Using T.38T.38 is a standard for transmitting FAX across IP networks in real-time mode. And it is the most reliable standard for transmitting FAX via an AudioCodes gateway. T.38 is disabled by default on AudioCodes gateways. The “Fax Signaling Method should be set to “Fax Fallback.” The gateway negotiates a T.38 call. If this fails, it falls back to G.711.You will need to change to settings to ensure that FAX transmit using the T.38 standard.

1 Protocol Management.2 Protocol Definition.3 General Parameters tab.4 Change Fax Signaling Method to Fax Fallback.5 Advanced Configuration.6 Channel Settings.7 Fax/Modem/CID Settings tab.8 Change Fax Transport Mode to T.38 Relay.

Using the Automat ic Dial ing TableUse the Automatic Dialing Table to define telephone numbers that are automatically dialed when an incoming call is received.

1 Open the Automatic Dialing page: Protocol Management\EndpointSettings submenu\Automatic Dialing.

Figure 14.4 Automatic Dialing Table window

2 In the Destination Phone Number field for a port, enter the telephone number to dial.

Incoming calls may be directed to a specific extension by specifying a valid extension. It is possible to specify the same extension for every port (e.g. the Auto Attendant) or a different extension for each port on this page (e.g. DID lines).



It is possible to use the default routing specified in the Sphericall Administrator application by specifying an invalid extension for one or more of the ports on the Automatic Dialing page. For example, if the nonexistent extension "xyz" is specified on the Automatic Dialing page and the domain is "spherecom.com", when an incoming call is received, the MP11x will send an INVITE to "[email protected]". Since "xyz" does not exist on the system, the MGC will apply the default route to the call.

3 In the ‘Auto Dial Status’ field, select one of the following:

• Enable [1] – When a port is selected, when making a call, the number in the Destination Phone Number field is automatically dialed if phone is off hook (for FXS gateways) or ring signal is applied to port (FXO gateways).

• Disable [0] – The automatic dialing option on the specific port is disabled (the number in the Destination Phone Number field is ignored).

• Hotline [2] – When a phone is offhook and no digit is pressed for HotLineDialToneDuration, the number in the Destination Phone Number field is automatically dialed (applies to FXS and FXO gateways).

4 Repeat step 3 for each port you want to use for Automatic Dialing.

Click the Submit button to save your changes.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A D V A N C E D C O N F I G U R A T I O N

While your settings may vary, Sphere Communications has changed the following Advanced MP104 settings for integration with the Sphere system.

C H A N N E L S E L E C T M O D E *Menu: Protocol Management\Protocol Definitions\General ParametersDisplay: Channel Select ModeRecommended Value: DescendingComments: Channel 8 will be used for the first outbound call, then channel 7 and so on.

I S P R O X Y U S E DMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Enable ProxyRecommended Value: Use ProxyComments: A proxy must be used with Sphericall.

P R O X Y N A M EMenu: Protocol Management\Protocol Definition\Proxy & Registration


. . .


Display: Proxy NameRecommended Value: Host of your primary MGCComments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP104 to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.

P R O X Y I PMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Proxy IP AddressRecommended Value: Host of your primary MGCComments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP104 to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.

S I P G A T E W A Y N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Gateway NameRecommended Value: A value unique to each MP104, for example, MP104-1Comments: A gateway name should be specified. If the gateway name is left unspecified, the MP104 will use its IP address instead. If the IP address ever changes, example due to DHCP, the MP104 will appear as a new gateway to Sphericall.

I S R E G I S T E R N E E D E DMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Enable RegistrationRecommended Value: EnableComments:

R E G I S T R A R N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Registrar NameRecommended Value: Hostname of your primary MGCComments: Due to problems with the way AudioCodes uses the results of DNS SRV lookups, Sphere recommends binding the MP104 to a single MGC by specifying a hostname in this field. This recommendation may change with future releases of AudioCodes firmware.



R E G I S T R A T I O N T I M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Registration TimeRecommended Value: 600 secondsComments:

U S E R N A M EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: User NameRecommended Value: Must be the same as the gateway nameComments:

A U T H E N T I C A T I O N M O D EMenu: Protocol Management\Protocol Definition\Proxy & RegistrationDisplay: Authentication ModeRecommended Value: Per GatewayComments: Setting the authentication mode to per gateway forces the MP104 to register once for the entire gateway. More sophisticated configurations may require per endpoint.

C O D E R N A M EMenu: Protocol Management\Protocol Display: Coder NameRecommended Value: g711Ulaw64k,20,0,$$,0Comments: This value enables G.711 U law. Other CODECs, such as G.711 A law and G.729 may also be enabled.

E N A B L E C U R R E N T D I S C O N N E C TMenu: Protocol Management\Advanced Parameters\General ParametersDisplay: Enable Current DisconnectRecommended Value: EnableComments: This setting causes the MP11X to drop the call if loss of loop current is detected. This typically occurs when the party on the other side of the PSTN hangs up.

C U R R E N T D I S C O N N E C T D U R A T I O NMenu: None (INI file parameter)


. . .


Display: None (INI file parameter)Recommended Value: 300 millisecondsComments: This setting sets the loss of loop current detection window to 200 to 600 milliseconds. This setting is not accessible via the web interface. It must be manually added to the INI file.

T I M E T O S A M P L E A N A L O G L I N E V O L T A G EMenu: None (INI file parameter)Display: None (INI file parameter)Recommended Value: 100 millisecondsThis setting forces the MP104 to check the trunks for loss of loop current every 100 milliseconds. This setting is not accessible via the web interface. It must be manually added to the INI file.

E N A B L E C A L L E R I DMenu: Protocol Management\Advanced Parameters\Supplementary ServicesDisplay: Enable Caller IDRecommended Value: EnableComments: Enables the delivery of Caller ID to the SIP Proxy in the Display name field of the INVITE From header.

P S T N P R E F I XMenu: Protocol Management\Routing Tables\IP to Hunt Group RoutingDisplay: Dest Phone PrefixRecommended Value: *,1,*,xxx.xxx.xxx.xxx,1Comments: This directs all outbound calls to the PSTN to the first hunt group.

T R U N K G R O U P _ 1Menu: Protocol Management\Endpoint Phone NumbersDisplay: 1Recommended Value: 1-8, MP104, 1Comments: This places all channels in the first hunt group.

E N A B L E C A L L E R I D _ < P O R T >Menu: Protocol Management\Endpoint Settings\Detect Caller ID from TelDisplay: Port <Port>Recommended Value: Enable



Comments: This is a per channel setting. Enable it if your line delivers caller ID.

T A R G E T O F C H A N N E L < P O R T >Menu: Protocol Management\Endpoint Settings\Automatic dialingDisplay: Port <Port>Recommended Value: <Auto Attendant Extension> EnableComments: This is a per channel setting that determines where calls inbound from the PSTN on that channel are routed to.

I S T W O S T S T A G E D I A LMenu: Protocol Management\FXO SettingsDisplay: Dialing ModeRecommended Value: One StageComments: Use one stage dialing if the MP104 is connected directly to the PSTN. Use two stage dialing if the MP11X is connected to a PBX.

I S W A I T F O R D I A L T O N EMenu: Protocol Management\FXO SettingsDisplay: Waiting for Dial ToneRecommended Value: YesComments: Enabling this setting forces the MP104 to detect dialtone before dialing.

D T M F T R A N S P O R T T Y P EMenu: Advanced Configuration\Media Settings\Voice SettingsDisplay: DTMF Transport TypeRecommended Value: Transparent DTMF Comments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.

M F T R A N S P O R T T Y P EMenu: Advanced Configuration\Media Settings\Voice SettingsDisplay: MF Transport TypeRecommended Value: Transparent MFComments: This setting disables RFC2833 relay of DTMF digits and forces the digits to be carried in-band. This setting is required unless you have configured your Sphere system to use RFC2833.


. . .



. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SPECTRALINK WIRELESS TELEPHONES 15

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• The Sphere system should be installed, configured and tested as fully functional.• Refer to the (1)Spectralink Facility Preparation and (2)Installation and Operation

Manuals for installation planning, setup, package contents, safety and conditions of use.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . R E Q U I R E D M A T E R I A L S

• Cross Connect Block - required to connect the telephone switch ports and the base stations to the MCU.

• 25 Pair Cables - RJ-21 male at MCU end, required to connect the MCU to the cross-connect blocks.

S P E C T R A L I N K O V E R V I E WThe Link Wireless Telephone is a durable and feature-rich handset for workplace applications. The Link Wireless Telephone integrates with the Sphere system to provide advanced calling features throughout the workplace.

Note: To Sphere-related hardware, the Spectralink phone takes on the characteristics of an analog phone. However, the Spectralink phone adheres to the definitions of voice-over-wireless technology and is subject to the requirements of said technology. As previously stated, refer to Spectralink’s documentation for all operational issues.


SP E C T R A L I N K W I R E L E S S TE L E P H O N E SRequired Materials

Figure 15.1 SpectraLink System

M A S T E R C O N T R O L U N I TThe Master Control Unit (MCU) is a modular unit that connects the VG3 PhoneHub to the wireless telephone system using digital line interfaces, or the VG3 BranchHub using analog line interfaces.The front panel contains the connections to the Sphere system, switches to control system administration, and Status LEDs.

VG3 PhoneHub

Spectralink Wireless

Telephones

Master Control Unit

Base StationsMODE SELECT

10/100ETHERNET STATION LINES

12345678910111224 23 22 21 20 19 18 17 16 15 14 13

25 FAILO

VER

25 FAILO

VER

VG3-PB2430

LAN

Switch


. . .

. .SP E C T R A L I N K W I R E L E S S TE L E P H O N E SRequired Materials

Figure 15.2 Master Control Unit Front Panel diagram

Table 15.1 Spectralink MCU Front Panel

B A S E S T A T I O N SBase Stations act as radio transceivers to provide the communications signal between the wireless telephone and the MCU. Base Stations are slightly larger than a smoke detector and are typically mounted on the ceiling, in strategic locations throughout the facility. A single Base Station can provide radio coverage for an area of 5,000 to 50,000 square feet depending on building obstructions. Base Stations may be located up to 2,200 cable feet from the MCU. When a Wireless Telephone user makes or receives a call, the Wireless Telephone and Base Station establish a digital radio communication link. As the user moves around the coverage area, calls are “handed off” to the Base Station that is able to provide the best radio signal (typically the closest Base Station). These handoffs involve the Wireless Telephone

Front Panel Definition

IPC OUT Port Not Used in integration with Sphere system

IPC IN Port Not Used in integration with Sphere system

Ethernet Network Interface Port Connect Spectralink MCU to organization’s network

Error LED Flashes when the system has detected an error. When flashing, check the Status LEDs for an error code.

Status LEDs Indicate system error messages and status.

Power Jack Connects to the AC adapter to supply power to the system.

Conn A RJ-21 connector to the cross-connect demarc block.

RS-232 9-Port Serial Connector used to interface with Spectralink MCU for device configuration.

Spectralink

RS-232

OUT IN

IPC

LINKOK

ACT

COL

Network ERROR

1 2 3 4 5

STATUS CONN A

PWR

IPC Out Port

IPC IN Port

Error LED

Status LEDs

Power Jack

Station Lines 50-Poistion Connector

9-Port Serial Connector

Ethernet Network Interface Port


SP E C T R A L I N K W I R E L E S S TE L E P H O N E SRequired Materials

establishing a communication link with another Base Station and dropping the previous link.

Note: Spectralink provides a list of access points that are compatible with the Spectralink Wireless Telephone system. This list can be accessed at www.spectralink.com.


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .VISUAL BASIC TYPE LIBRARIES 16

I N T R O D U C T I O N The Sphericall Phone type library exposes Sphericall objects that enable the programmer to create their own Graphical User Interface(s) for their telephones.The Sphericall Phone Type Library has built-in Help files on how to use the Sphericall Phone Objects, but it does not teach programming skills. Each Help file shows an example, gives properties, methods and events, and indicates Applies To. This online Help file will be your main resource for using the Sphericall Phone Type Library (the help file name is: Phonetlb.hlp; the contents file name is: Phonetlb.cnt).Consult your programming manuals and on-line Help for programming ideas. If you wish to integrate this Help with the Visual Basic Help, you must add the following line to the Visual Basic contents file (i.e. VB.cnt) after the final :Index statement. :include phonetlb.cnt

• Use the SCPhone Object to create an instance of the Sphericall Phone object, dial numbers, pick up extensions, retrieve an individual call object, respond to phone events, and access line and title properties.

• Use the SCCalls Object to step through all active calls on the SCPhone Object.• Use the SCCall Object to retrieve information about a single call on the SCPhone

Object. This object also handles answering, transferring, hanging up, placing a call on hold, playing sounds, and dialing additional digits after the call has been identified.

C O N S T A N T S

C A L L S T A T E C O N S T A N T SThe CallStateConstants are enumerated values representing the status of a call. They are typically used with the State and PreviousState properties of SCCall. When the State property of the current call matches a CallStateConstants constant, the StateName of the call matches the indicated description.

Constant Value Description

callStateUnknown 0 Unknown

callStateRinging 1 Ringing

callStateDialTone 2 Dial Tone

callStateDialing 3 Dialing

callStateProceeding 4 Proceeding

callStateRingback 5 Ringback

callStateBusy 6 Busy

callStateSpecialInfo 7 Special Info


VI S U A L B A S I C TY P E L I B R A R I E SGetting Started

Be sure to take advantage of the example code fragments within the Help files provided by Sphere. Before long, you will create your own integration of Sphericall with your most commonly performed tasks within the application of your work.Microsoft publishes a printed VBA reference manual, but it doesn't come with most VBA applications. The manual is included with the Developer's Edition of Office.

S A M P L E A P P L I C A T I O N S F O R S P H E R I C A L L T Y P E L I B R A R I E SSphere has included a few sample applications for demonstration of using the Sphericall Type Libraries in customizing applications. You can find the files in the Samples subdirectory on the Sphericall DVD-ROM. Instructions for each sample type can be found in the individual sample directory as a readme.txt file.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . G E T T I N G S T A R T E D

The Sphericall COM type library provides an API where developers can create custom applications utilizing call information and control. Using COM allows the developer to choose the programming language such as Visual Basic, VB script, javascript , or C++.

To insta l l and ver i fy the Spher ical l COM API1 Login into the PC with a user account that has local administrative privileges .2 Install the Sphericall Desktop.3 Using the Sphericall Administrator, configure user rights so that the user account has

full privileges to open one or more stations.4 Start the Sphericall Desktop and open a station line.5 Start an Internet Explorer. 6 Go to a web site that provides a phone number or use a name lookup web site such as

people.yahoo.com or www.switchboard.com. 7 Highlight a number, right-mouse-click and Select “Sphericall Dial”.

The Sphericall Desktop will receive the number via the COM interface and then dial it. When the Sphericall Desktop is started, it will load and register the phone.tlb type library. Registration requires write access to HKEY_CLASSES_ROOT registry key, which in turn requires the user to have administrative access to the PC.

callStateConnected 8 Connected

callStateConferenced 9 Conferenced

callStateOnHold 10 On Hold

callStateTransferring 11 Transferring

callStateConferencing 12 Conferencing

callStateDisconnected 13 Disconnected


. . .

. .VI S U A L B A S I C TY P E L I B R A R I E SGetting Started

In a future version of Sphericall (timeline is not yet available), you can anticipate a features such as a status message indicating the result of loading and registering the type library will be printed in the Sphericall Desktop log file. A status message will also be added to the log files for every command invoked using the COM API. Desktop log files are stored in C:\Documents and Settings\Logged_In_Username\Application Data\Sphere\Desktop\Logs.If the Sphericall Desktop is not running when a COM call control command is invoked, the Sphericall Desktop will be started. It is possible for applications written using the COM interface to block thereby keeping control of the Sphericall Desktop. The Sphericall Desktop will release control if the application is blocked for more than 5000 milliseconds. This value can be changed through [HKEY_CURRENT_USER\Software\Sphere\Phone\COMInterface]"MessagePendingDelay"The sample applications (Access, Visual Basic, Web, DesktopCOMTest ) are located in …\Server\Data\samples\Desktop COM API\install. Visual Basic applications require the VB runtime. Installing the sample applications will install the VB runtime. The MS Access sample requires MS Access to run.The descriptions of the Sphericall Objects and their methods is located in C:\Program Files\Sphere\phonetlb.hlp.


VI S U A L B A S I C TY P E L I B R A R I E SGetting Started


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .SIP TRUNKING 17

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W O F S I P

Session Initiation Protocol (SIP) is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users. Such sessions include voice, video, chat, interactive games, and virtual reality.The Sphericall Manager is designed to optimize the use of Session Initiation Protocol for communications via SIP station devices as well as SIP trunk facilities, or “soft trunks.”Created in 1996 by IETF (Internet Engineering Task Force—the same organization that created TCP/IP and HTTP), SIP was originally intended to enable the establishment of media (audio or video) sessions between users. Since that time, it has evolved to cover a wide range of real-time collaboration functionalities.In the simplest of terms, SIP provides a mechanism for setting up generic “sessions” of information exchanged among disparate endpoints across the IP network. Within the Sphericall Manager, SIP allows external systems to participate in calls with the system. Early on, the Sphericall Manager interacted with the Windows Messenger client using the SIP standard protocol to “connect” their application and signaling needs. SIP has achieved wide adoption throughout the telecommunications and enterprise markets for its ability to streamline communication session control and provide cross-application interoperability. Although there are other protocols designed to have similar functionality, SIP is said to be simpler and consumes fewer resources, the protocol of choice for truly converged networks. Further, SIP can be harnessed to do various tasks that are currently accomplished by multiple protocols, many of which are proprietary.Voice over IP (VoIP) networks use SIP as a “call control” functionality to connect phones so the phones can exchange media information through a media exchange protocol, like RTP. In this context, "connect" includes resolving the address of the destination and negotiating the types of media to be exchanged. For address resolution, SIP provides functionality similar to what Domain Name System (DNS) provides for URLs. DNS maps human-friendly, fully qualified domain names, like www.spherecom.com, to numeric IP addresses required for communication over an IP network. Similarly, SIP maps a URL or a phone number (xxx-xxx-xxxx in the United States) to an IP address to which the phone can send media. SIP also provides additional functionality that DNS does not provide. For example, people can have multiple IP phones that they use in different locations. A SIP server can dynamically change the IP address it returns for a specific URL or phone number depending on whether the recipient is in the office, car, home or other location. Because the IP address changes, calls are routed to the phone that the person is using. In addition to resolving addresses of destination phones, SIP uses Session Description Protocol (SDP) to negotiate media formats that both phones support. The content of the SIP invitation message that a phone generates during call setup


S I P TR U N K I N GSIP Terminal Location in Sphericall

contains an SDP message that defines the media formats that the source phone supports. The destination phone examines the SDP message and creates a SIP response that says which media formats the call should use.In terms of communication between SIP endpoints and the structure of SIP packets, SIP resembles HTTP in that it is a request-response protocol. However, unlike HTTP, SIP allows for additional responses to a single request. In response to an invitation, a phone can initially return a "ringing" response, and then an "ok" response when the phone is picked up.VoIP networks that use SIP generally provide a few common services that phones use to carry out SIP conversations. These services provide much of the additional functionality for which SIP was designed, like robust address resolution to multiple locations for a single URL or phone number. These services refer to server functionality, not to where or how the functionality is implemented. An example: in many cases the same physical server provides all of these services.The following additional services are commonly provided by SIP:

• Registrar and Location. One of the first SIP operations an IP phone performs is to register with a registrar server, by providing a URL or phone number and a corresponding IP address. The registrar server stores this information and often provides it to proxy and redirect servers to resolve URLs and phone numbers to IP addresses.

• Proxy and Redirect. Proxy and redirect servers assist IP phones in routing SIP messages to a destination. The mechanisms they use to accomplish this goal are slightly different. A proxy server accepts SIP requests and forwards them to another SIP device, which can be an endpoint like a phone, or another server (which can then forward to a phone or another server, and so on). As far as the original client is concerned, the proxy server handles everything required to get the request to the destination. In contrast, a redirect server does not forward requests. Instead, it returns an IP address to the IP phone; the IP phone then uses the returned address to submit the request to the correct location (which can be a proxy server or the actual phone).

For More Information:Additional reading of the following documents may help you to understand further the SIP environment and goals:

• Documents, wiki and white papers at: www.sipforum.org or

• sip.edu Cookbook http://mit.edu/sip/sip.edu/index.shtml

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I P T E R M I N A L L O C A T I O N I N S P H E R I C A L L

The Sphericall system needs to identify the terminal that inbound calls are being placed from in order to apply policy such as inbound routing, capacity management, etc.This following is the logic the Sphericall Manager uses to locate a SIP terminal:Definition: SIP URI = userinfo@hostname:port


. . .

. .S I P TR U N K I N GNEC Sphere SIP Trunking

1 FROM Header

Userinfo is compared to the Account field of the Service Provider information.

Note: This is the most common way stations are identified but does not help for trunks since the FROM field contains the caller ID of the incoming call.

2 Contact Header

hostname:port is compared to the Outbound Proxy if configured, otherwise the Service Provider Domain of the Service provider information.If the hostname:port is an ip address it is compared exactly to what is configured.If the hostname:port is not an IP address, a partial compare is performed against the Service Provider information. For example, the hostname "horatio.Nlab.spherecom.com" would match the Service Provider information "Nlab.spherecom.com".

Note: In both cases the port must match.3 To Header

Userinfo is compared to the DID maps configured for SIP trunks. First match is found is used if there is overlapping DID configuration.

4 Authorization Header

The credentials included in the Authorization Header are compared against the credentials configured in the Authorization window for the trunk.The Sphericall Manager will challenge the sender to obtain credentials via the Authorization Header if no match is found using the above tests.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . N E C S P H E R E S I P T R U N K I N G

The following three configuration scenarios define a number of the installation scenarios that might be seen in the initial stages of NEC Sphere SIP trunking implementation. SCENARIO 1: Sphericall to SIP carrier - softtrunking

• This is a pure SIP softtrunking to an outside service provider.• This implementation will routinely use Outbound registration type.

SCENARIO 2: Sphericall to Sphericall - softtrunking tie line• SIP trunking as a tie line between existing Sphere systems.• This implementation will routinely use None registration type.

SCENARIO 3: Sphericall to Centurion Call Center - softtrunking with tie line to call center

• SIP trunk as a tie line to a call center.• This implementation will routinely use Outbound registration type.


S I P TR U N K I N GNEC Sphere SIP Trunking

B E F O R E Y O U B E G I N A N Y S I P I N T E G R A T I O N




• All defaults are automatically configured for your convenience. Defaults and supported User Agents are indicated by the check mark.

• User Agents are also available for “adding” a new, untested SIP endpoint. If a site has a new untested SIP endpoint, they must add it to the system themselves and verify its operability through their own testing (this testing is not supported by Sphere support personnel).


default settings for some deployments.All SIP devices of a make, model and firmware version have same attributes, for example, all Polycom IP601 SIP phones running 2.1.0.2708 firmware support "talk" event in the NOTIFY request to answer an incoming call. Instead of assigning these attributes on each SIP endpoint individually, the attributes are assigned to the User-Agent/Firmware-Version binding and gets applied to all corresponding SIP devices. Several new database tables have been added to support this feature. Sphere-supported SIP devices (stations and trunks) are pre-configured in the database.

1 Select the SIP tab on the System Properties window.


. . .

. .S I P TR U N K I N GNEC Sphere SIP Trunking

TIFY /un-

Web



know how to treat it or what features to apply per endpoint.7 Enter an Agent Description that is appropriate for the Name & Endpoint Type.8 Click Apply.

Note: 1) Those User Agents listed in the SIP dialog window that also have the Default checked, are those User Agents created into the system by default. These will remain in the system. User Agents added by system administrators are not indicated with a check in the Default column.

Note: 2) If the name of a non-default user agent matches the name of a default user agent, the Agent Name, Agent Description and Endpoint Type fields cannot be edited. Conversely, if multiple user agent entires exist that have the same name, but none are marked as default, changing any field (other than Version) will change the same corresponding field in the other same-named User Agent entries.

The following fields are customizable for entering a non-default user agent:

Table 17.1 User Agent Profile Descriptions


‘talk’ Event (Notify Request) Based 3PCC

Supported Unsupported (default)

The MGC sends the NOrequest to answer/holdhold a call remotely (Sphericall Desktop or Services).


S I P TR U N K I N GNEC Sphere SIP Trunking

a ith eived

d.

alue of swer, st to o

SIP in the 1+ all re pre-

he SIP ility nes

GC

com

to an iately

with a 00 ime

ration t es

in TER

iate inal oming e

o

ER

)

ited the

‘to-tag’ (SUBSCRIBE REQUEST) In New Subscription

AllowedDisallowed (default)

MGC sends a NOTIFYrequest with terminated(reason=timeout) subscription-state whenSUBSCRIBE request wnon-empty to-tag is recand the corresponding subscription is not foun

Click-To-Dial Ring Caller’s Phone First (default)Use ‘answer-after’ ??param (INVITE:: Call-Info Header)Use ‘auto-answer’ value (INVITE:: Call-Info Header)

MGC sends a special vthe parameter (Auto Ananswer-after etc) in theoutgoing INVITE requeinform the SIP station tanswer immediately.

Endpoint Created By AdministratorCall Manager (default)

Call Manager creates aphone when not found database. Since in 5.2.Polycom SIP phones acreated by the system administrator (just like tsoft trunks), this capabensures that these phoare not created by the Mwhen not found in the database. When a Polyphone sends its first REGISTER to check inMGC, the MGC immedsends a "503 Service Unavailable" response Retry-After timeout of 3seconds. In the mean tMGC obtains phone configuration from the database. If the configufrom the database is noavailable, MGC continusending "503 Service Unavailable" message response to the REGISrequests.

Find Terminal Method Authentication InfoDefault (default)DID MappingFrom Header URIOutbound Contact URIP-Asserted-Identity Header URI (currently supported)

MGC uses the approprmethod to find the termassociated with the incINVITE request from thendpoint.

Hardware Address Available (REGISTER:: User-Agent Header)Unavailable (default)

SIP has the capability tsearch for a hardware address in the REGISTrequest.

INVITE Request-URI Source Invite Request To-URIOutbound Contact-URI (default)

(Quintum products only

MWI NOTIFY Request Supported (default)Unsupported

MGC sends an unsolicNOTIFY request when MWI state changes.



. . .

. .S I P TR U N K I N GUser Agent Maintenance

o ill not t to the s a us. If

send (out there status s not uest.

r nd.

SIP not est.

FER call.

initiate

Y ion.

er RI) in

the

in the

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . U S E R A G E N T M A I N T E N A N C E

U S E R A G E N T S A N D U P G R A D E SThese tables essentially provide the functionality provided by the MGSetting table, but at a much less overhead. These tables also provide the following benefits:• When a Sphericall system is upgraded (pre-6.0 to 6.0+), Sphericall automatically

creates UserAgentVersionHubAssociation for all Outbound/None registration type devices. It also creates a binding in UserAgentVersionHubAssociation table for those SIP devices (irrespective of registration type) that have an entry in the deprecated UserAgentHubAssociation table. This ensures that the SIP devices keep functioning after the upgrade.

• When a Sphericall system is upgraded (6.0 to 6.0+), Sphericall will not change the existing UserAgentVersionHubAssociation bindings. Therefore, the SIP devices overriding the default behavior will continue to work the same way after the

MWI SUBSCRIBE Request Supported (default)Unsupported

When the value is set tSupported, the MGC wsend a NOTIFY requesendpoint even if there ichange in the MWI statthe value is set to Unsupported MGC will the unsolicited NOTIFYof dialog) request whenis a change in the MWIeven if the endpoint hasent a SUBSCRIBE req

MediaServer Max Packetization (ms)

10 ms - 160 ms range80 ms (default)

Setting specifies the maximum packet size Sphericall Media Serveshould send to the far-e

OPTIONS Request Supported (default)Unsupported

Setting specifies whichendpoints support or dosupport OPTIONS requ

REFER Based 3PCC SupportedUnsupported (default)

MGC does not send REto initiate a click-to-dial

REFER Based Transfer Supported (default)Unsupported

MGC sends REFER to a transfer.

Remote Reboot SupportedUnsupported (default)

MGC sends the NOTIFrequest to reboot a stat

URI ‘qheaders’ Parameter Supported (default)Unsupported

MGC sends the QHead(Question header in a Uthe Refer-To header in REFER request.

Video SupportedUnsupported (default)

MGC sends Video SDPoutgoing INVITE.



S I P TR U N K I N GUser Agent Maintenance

upgrade. However, Sphericall may change the default values of the Parameters configured in the Parameter table (this is similar to that of the MGSetting defaults which may infrequently get changed on an upgrade).

• When the firmware of a SIP device is upgraded, if the SIP device has a forced binding in the UserAgentVersionHubAssociation table, the DbServer/MGC do not update the binding for the new firmware and the SIP device continue the work same way in the MGC. For Outbound/None registration type endpoints DbServer and MGC never update the UserAgentVersionHubAssociation binding even if the SIP device is reporting a completely different User-Agent/Firmware-Version than what is configured in the UserAgentVersionHubAssociation table.

• A new User-Agent/Firmware-Version can be created and assigned to a custom ParameterProfile and then this User-Agent/Firmware-Version can be bound to the UserAgentVersionHubAssociation and set to "forced" type binding. This feature provides a similar level of control provided by the MGSetting table.

• SIP station upgrades—Upgrades to User Agents with firmware are automatic and apply to all the endpoints on the system.

• SIP trunk upgrades—Upgrades to User Agents are not automatic.

U S E R A G E N T C L E A N U POver time, a number of older User Agents Parameter versions will accummulate in the dialog of User Agents. It is the Sphere system administrator’s responsibility to remove old versions of the User Agents that are no longer used on the system.

To remove a User Agent Parameter1 From the Sphericall Administrator application:2 Open the General System properties.3 Select the SIP tab.4 Scroll to view the User Agents.5 Click the far right column of the User Agent to be removed.6 Click Remove.7 Repeat for other User Agents.8 Click OK to exit.

S I P R E G I S T R A T I O N B E H A V I O RThe following patterns have been established for registration of SIP softtrunks on the Sphere system. This reference can help with determining which part of the configuration is generally required.

Table 17.2 SIP Registration

Registration Type Configuration Needed Uses

No Registration Must create trunk on both ends of the softtrunk. These trunks will have the Service Provider tab in the software.

Most SIP softtrunks are generally created in this way.


. . .

. .S I P TR U N K I N GSIP Trunking to SIP Service Provider

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I P T R U N K I N G T O S I P S E R V I C E P R O V I D E R

Figure 17.2 SIP Trunking to SIP Service Provider

The above scenario of a SIP solution has the following characteristics:• This company has one or more individual connection(s) to the SIP Service

Provider (SSP); rather than contracts with an ISP, the local phone services, long distance services, etc. are secured through the SIP service provider.

• Further connections beyond SSP to the PSTN (Public Switched Telephone Network) are the responsibility of the SSP.

• Billing and services are simplified.

Outbound Registration Must create trunk via configuration and softtrunk will be auto-created on the other send. These trunks will have the Service Provider tab in the software.

Tie-lines may be created in this way.Other, non-standard, softtrunks may be created with this method.

Inbound Registration Do nothing with configuration. This softtrunk will just appear in the Sphericall Administrator Trunks section. The far side will have information to send the registration.These trunks will not have the Service Provider tab.

Tie-lines may be created in this way.Call center softtrunks are often created with this method.

Registration Type Configuration Needed Uses


S I P TR U N K I N GSIP Trunking to SIP Service Provider

• Gateway CPE (ALG) (Customer Premise Equipment: Application Layer Gateway or VoIP Aware Gateway) is placed on the premise by SSP and typically maintained by SSP.

• Features and services available through SSP vary by provider.

I N S T A L L A T I O N & C O N F I G U R A T I O N O F S I P T R U N K T O S E R V I C E P R O V I D E R

To plan to insta l l S IP t runking1 YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE

SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints.2 Refer to Book 1: Planning & Preparing the Sphere System for information on ordering

and planning SIP services.3 Once you have your SIP services delivered and the Customer Premise Equipment (CPE)

is onsite (typically an ALG—Application Layer Gateway), connect the CPE to the local area network (straight-through RJ-45 cable from the CPE to the Ethernet network switch).

Application Layer Gateways (NAT traversal)An ALG can allow firewall traversal with SIP. If the firewall has its SIP traffic terminated on an ALG then the responsibility for permitting SIP sessions passes to the ALG instead of the firewall. An ALG can solve another major SIP headache: NAT (network address translation) traversal. Basically a NAT with built-in ALG can re-write information within the SIP messages and can hold address-bindings until the session terminates. This allows for data inside the SIP packet to be rewritten with a public IP address to allow the far end to route calls back to the original party.

Note: ALG hardware: some SIP trunk service providers will provide this piece of hardware as part of their service (like a smartjack). Other providers leave this piece of equipment up to the end customer to purchase. Another piece of equipment that can also be used in this scenario is called a Session Border Controller (SBC). Sphere has worked with the following ALGs in the field: Ingate, Adtran and ATI.

4 During the ordering process of your SIP trunk service, you must secure the following information:

• Account Number• Service Provider Domain name• Outbound Proxy• Registration Type• Contact Domain• Primary MGC (Sphericall Manager)• Secondary MGC (Sphericall Manager)• Password• Realm

5 Be sure to arrange for contact/account holder personnel for your site.

In some cases, the service provider may only make changes or additions to your services or the service order through one authorized individual.


. . .


Before you begin the t runk conf igurat ionWhen integrating with Global Crossing and Sphericall SIP trunking, the following settings must be verified prior to SIP trunk configuration:

1 RFC2833 MUST BE OFF on the Media Streams tab of General/System Properties/Media Streams.

2 ONLY G711 is enabled on General/System Properties/Media Streams. 3 On Media Server Properties, be sure that the Codec Settings are DISABLED from the

system default (be sure to de-select this checkbox).

To conf igure a SIP t runkFrom the Sphericall Administration application:

1 Select the Trunks tab.2 Right-click and select Add.3 Choose Add Softtrunk.

Figure 17.3 Add Softtrunk Properties

Type in the information for each field:4 Description: Any printable ASCII character except double quote or backslash.

This field is a description field that specifies the display-name in the From and To headers of Outbound REGISTER requests. Often it will be the phone number that is assigned to this service.Example: 8477939942

5 Account: Provided to you by SIP carrier (i.e. could also be phone number that is assigned to this service). Example: 8477939942The Account field specifies the userinfo portion of the SIP URI in the From and To headers of outbound REGISTER requests. The Account field may contain any printable character except

Information entered herewill reflect the service provider’sinformation.



the following: space, pound sign #, percent sign %, colon :, less than <, greater than >, at sign @, square brackets [], backslash \, caret ^, brace brackets {}, pipe | and grave accent `.

6 Service Provider Domain: Provided by the SIP service providerIt is the destination identification information (it’s who you are sending TO).Example: 777445.sip.company.net.The Service Provider Domain field (SPD) may be filled with an IP address, DNS host name or DNS domain name. The SPD specifies the hostname portion of the SIP URI in the From and To headers of outbound REGISTER requests. The SPD also specifies the destination for outbound SIP requests if the Outbound Proxy (OP) is blank. A DNS lookup is performed on the contents of the SPD if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below).

7 Outbound Proxy: Provided by the SIP service provider Example: sipstack-5.mnl2.sipcarrier.netThe Outbound Proxy field (OP) may be filled with an IP address, DNS host name, DNS domain name or it may be left empty. If the OP is not empty, all outbound SIP requests are sent to the specified destination. A DNS lookup is performed on the contents of the OP if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). If the OP is empty, the destination for outbound SIP requests is determined by the Service Provider Domain field.

Summary of SPD and OPThe SPD and OP determine where the Sphericall system sends outbound SIP requests. The OP field would typically be used when the administrator wants to force all outbound SIP messages through a specific device, for example, a SIP Session Border Controller (SBC).

Note: The following two System Initialization Settings affect settings for the SPD and OP:

8 Port information: use the default (5060) unless otherwise specified by the SIP service provider.

9 Registration type:Recommended for softtrunk to service provider: OUTBOUNDThe registration type is governed by the service provider.

• Inbound - the SIP trunk sends a register request to the Sphericall system. When the SIP trunk first initiates contact by sending a REGISTER request, the MGC automatically creates a hub, trunk and outside service. The Service

DnsNaptrServiceEnabledThe DnsNaptrServiceEnabled MGC setting specifies whether the MGC will perform DNS NAPTR queries on the SPD/OP. This setting defaults to true. • The administrator may optimize MGC performance by changing this setting to false

if the SPD/OP specifies a host nameor

• if the authoritative DNS server for the SPD does not support NAPTR queries.

DnsSrvServiceEnabledThe DnsSrvServiceEnabled MGC setting specifies whether the MGC will perform DNS SRV queries on the SPD/OP. This setting defaults to true. The administrator must change this setting to false: • if the SPD/OP specifies a host name,• if the SPD is not a valid DNS domain name with an SRV record,

or• if the authoritative DNS server for the SPD does not support SRV queries.


. . .


Provider tab does not appear in the Hub device properties dialog box since the SIP trunk provides all the required information in the REGISTER request.

• Outbound - the Sphericall system sends register request to the SIP trunk; all fields on the Service Provider tab (trunk device properties) are used.

• None - issues no registration; neither the SIP trunk nor the Sphericall system send a register request; all fields on the Service Provider tab except the Contact Domain are used (trunk device properties).

Some SIP trunk service providers require the Sphere trunk to register with their system. This registration can be set to None, Outbound or Inbound depending on the requirement of the SIP trunk Service provider.

10 Primary MGC:

Secondary MGC: The Primary and Secondary MGC fields specify which MGCs are primary and secondary for the SIP trunk. The Secondary MGC assumes control of the trunk should the Primary fail.If the CD specifies an IP address or DNS host name, the Secondary MGC field should be filled with "None." In this case, specifying a Secondary MGC has limited value since the external system will not be able to locate the Secondary MGC to send SIP requests to. In practice, calls inbound to the Sphericall system will fail. Calls outbound from the Sphericall system may fail.If the CD specifies a DNS domain name, the DNS SRV records must match the contents of the Primary and Secondary MGC fields. Specifically, the "first" SRV record must point to the Primary MGC and the "second" SRV record must point to the Secondary MGC. The order in which the SRV records are returned are determined by the priority and weight values assigned to the SRV records in DNS. The external system must attempt to contact the MGC pointed to by the SRV record with the lowest priority first. If multiple SRV records have the same priority, the external system must attempt to contact the MGC pointed to by the SRV record with the highest weight first. Since Microsoft's DNS server returns different orderings for SRV records with the same priority from query to query, we recommend setting the (priority, weight) of the SRV record for the Primary and Secondary MGC to (0, 0) and (1000, 0) respectively.If the CD specifies a DNS domain name, that name must be resolvable from the external system. For example, we cannot use internal_domain.YOURDOMAIN.com as the CD for a SIP Service Provider or carrier since internal_domain.YOURDOMAIN.com cannot be resolved anywhere other than the internal Sphere network.

11 Contact Domain: You provide this informationIt is the identification information YOU publish to the far end (you are identifying who you are).Example: 11.2.27.101The Contact Domain field (CD) specifies the hostname portion of the SIP URI in the Contact header of outbound REGISTER requests. In other words, the CD determines how the external system locates an MGC to send SIP requests to. The CD may be filled with an IP address, DNS host name or DNS domain name. If filled with an IP address or host name, the external system will be bound to a single MGC with no redundancy. The CD should only be filled with a DNS domain name if the external system supports the DNS NAPTR and SRV lookup procedures specified by RFC 3263.Notice that the hostname portion of the SIP URI in the Contact header of all outbound non REGISTER requests is filled with the IP address of the MGC sending the request. This binds the SIP dialog to the MGC.



12 Preferred Transport:Select the appropriate Transport means.

Determines the preferred network transport protocol for SIP signaling (TCP, TLS, UDP (default)). Note: For SIP trunks that use registration types “Outbound” or “None,” this setting specifies what transport is used to send SIP requests such as REGISTER and INVITE to the SIP Trunk. For SIP stations and trunks that use registration type “Inbound,” the MGC sends SIP requests via the transport the SIP endpoint specifies in its REGISTER request.

13 User Agent:Select the appropriate User Agent.

Sphere has included for selection most of the common types of User Agents. If the User Agent your system is using is not offered here, you must create it in the System General Properties (SIP) window, then apply it to the User Agent associated with the trunk.The following vendors specifically should use the “Generic SIP Trunk” User Agent:

14 DNS Test:

DNS test will query DNS resolving NAPTR records. The DNS "Service Provider" and "Outbound Proxy” Addresses will always be checked for A records and SRV records (records along with priorities and weights), as well as Contact Domain records. It will be up to the administrator to interpret the results and what affect, if any, it has on the SIP trunk.The DNS NAPTR query is not attempted when the Sphericall Administrator application is run under Vista.

15 Click OK.

The creation of the softtrunk is complete.

SIP Trunk MAC Address AssignmentWhen a trunk checks into a Sphericall system for the first time, the MGC automatically creates the necessary database records for it. As part of the record creation process, the MGC assigns a unique name to the trunk. For SIP trunks, the MAC address assigned to the trunk depends upon the registration type.For outbound registrations and registrations of type none, the MGC uses "Account@Service_Provider_Domain."For inbound registrations, the MGC uses the SIP URI of the REGISTER request To header. It is important to note that if the URI changes, the MGC will create new records for the trunk the first time it sends a REGISTER with the new URI.

Bandtel Netlogics

Clear SIP Net Vortex

Constant Touch Starvox

Global Crossing Xtra (Spain)


. . .


Figure 17.4 Softtrunk hub properties

Once creation of the SIP softtrunk is complete, the following information will be available on the Hub Properties:

16 Type in the descriptive information as it applies to this device:

• Name: Name of device (can be renamed for user-friendliness).• MAC Address: Assigned to the device by the system.• Stations/Trunks: Information on number of stations or trunks.• LAN: Hub belongs to LAN displayed in this field.• Localization Setting: Country or local language specification as defined in the

Localization Setting profiles.• Description: Information in this field can be edited.• Hub Number: Automatically numbered by the system (recommended that this

number remains unchanged).• Last Check In Time: Information is updated upon each check-in.• Firmware Version: None.• Device Type: Default device type for Softtrunk is “Unknown SIP Agent.”

To conf igure Service Provider informat ion for the sof t t runkDetails for the Service Provider panel can be edited if the registration type is None or Outbound.

• None - there will be no Contact Domain entry automatically filled in.• Outbound - all fields will be filled in; you may edit fields that are not shaded

depending on your system needs.1 Select the Service Provider tab of the softtrunk’s device properties.



Figure 17.5 Service Provider tab

2 Verify the information that appears on this window. It should be the same as the initial “Add SoftTrunk” dialog entries.

This information will vary depending on the service provider, their design and use of the SIP information.You may edit this information if necessary (example: change the Description, Port and Registration Type or other editable fields).

Note: The "Account" and "Service Provider Domain" were left read-only because they make up the line's MAC address (identity).

To conf igure SIP t runk propert iesThe following information on this softtrunk can be found on the trunk properties:

1 Select the Trunks tab from the main Sphericall Administrator application.2 Expand the tree of trunk devices listed in this window.3 Expand the + sign by the trunk device you wish to open.4 Double-click to open the Trunk Properties.


. . .


Figure 17.6 SoftTrunk General properties tab

5 Customize the information about this softtrunk.

• Name: Type name of softtrunk• Hardware ID: Usually assigned dynamically by the system.• Telephony Area: Assign or change Telephony area here.• Zone: Assign or change Zone here.• Total Capacity: Configure based on services provided by SIP provider.• Inbound Capacity: Determine based on services provided, see below.• Outbound Capacity: Determine based on services provided, see below.• Max Duration: Type the number of seconds in the Max Duration field that will

designate the maximum allowable duration of a trunk-to-trunk call.• If Max Duration is configured to 0 (zero), Sphericall places no limit on the call

length.• DEFAULT is 21600 (or 6 hours). The Max Duration edit box accepts integers

between 0 (no limit) and 2147483647 seconds (24855 days). An example of a longer call that may be valid via specific trunks would be an ongoing call to a conference bridge.

• In Service: Default is ENABLED. • Allow Emergency Calls from non-emergency group Stations:

Default is ENABLED.The following information helps you to understand the provisioning of the Total Capacity of the SIP trunks as configured on this window:

Total Capacity • total number of simultaneous calls the softtrunk can support



Guaranteed Outbound = Total Capacity - Inbound CallsGuaranteed Inbound = Total Capacity - Outbound callsExample Configuration: Total Capacity=10, Inbound=8, Outbound=7 would imply the following:Guarantee 3 inbound calls with max of 8Guarantee 2 outbound calls with max of 75 slots can be either inbound or outbound

6 Add any Outbound Caller IDs that are needed for your system and this softtrunk.

Note: Only change these settings if you wish to block certain Caller IDs from being sent out this trunk. This is an area that should not be changed unless there is an advanced reason to configure it.

7 Click Apply.8 Select the Authorization tab.

Figure 17.7 Authorization tab of Softtrunk Properties

Some SIP trunk service providers require that authentication be turned on when communicating over a SIP trunk. This must be configured on the Authentication tab and the details will be provided by the service provider.

9 Complete the fields for this authorization as follows:

NOTE: entries in these fields provide authentication. Contact your service provider for this information.

10 Enabling Use Authorization has the following function:

By enabling this field, the far end SIP service provider authenticates calls.• Account: Provided by the service provider.

Think of this as a “User Name” field.• Password: Provided by the service provider. Verify if adding or changing.

Think of this as a “Password” field.

Inbound Capacity • the number of simultaneous inbound calls the administrator wants to accept.

Outbound Capacity • the number of simultaneous outbound calls the administrator wants to allow.

Information entered here is information fromthe service provider.


. . .


• Realm: Provided by service provider.Think of this as similar to a “domain” field.

• Type: Default type is MD5.Use only MD5 as the type in this field. Only MD5 is currently supported.

• Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism.Together, the account, password and realm form the credentials needed to access a SIP trunk. Individually, these fields are arbitrary.When a SIP trunk challenges a request from the MGC, the trunk sends the realm and a random value called a nonce to the MGC. The MGC must then resubmit the request with an authorization header that contains a hash. The hash is computed using the account, password, realm and nonce. A correct hash proves to the SIP trunk that the MGC knows the account and password.If the administrator is dealing with a SIP service provider, these credentials are specified by the service provider.The Sphericall system needs to have the SIP service provider’s Account (user name), Password (Password) and Realm (domain) in order for the Sphere system to either challenge, respond or both prior to accepting a call into the system.

• Both (Challenge and Respond): This system will challenge and respond to challenges. This authorization credential can be used both to challenge incoming INVITEs and to respond to outgoing INVITE and REGISTER challenges.

• To Challenge: This system will Challenge all incoming calls and will authenticate each call. This Authorization credential is to be used only to challenge the incoming INVITEs (Sphericall Manager is receiving the INVITE).

• To Respond: This system will respond to challenges. This Authorization credential is to be used only to respond to an INVITE or a REGISTER challenge (Sphericall Manager has sent the INVITE/REGISTER and the destination is challenging).

• Unknown: This means no authentication is to be used in either direction (Incoming/Outgoing). Authentication might be set to UNKNOWN to disable the authentication temporarily (or another option is to uncheck the enable/disable box, which loses the authentication details).

11 Click Apply. 12 Select Inward Routing.



Figure 17.8 Inward Routing tab of Softtrunk Properties

13 Select Add to add a DID Mapping to the DID Mappings area (optional).14 Select Add to Schedule the Default Routing (optional).15 Enter a DID in the Input DID and select Test to verify DID (optional).16 Click Apply.17 Select Outward Routing.

Figure 17.9 Softtrunk Properties Outward Routing

18 Click Add Outside Service to add an outside service function to this softtrunk.19 Click Add to add Outside Dialing Rules if required.


. . .


Dialing Rules are covered in more detail in Book 2: Install & Configure the Sphere System.


Only add Emergency Groups or Settings if you are completing the Emergency Groups installation. Refer to Book 6: Emergency Service Installations.

22 Select Settings.

Figure 17.10 Softtrunk Properties Settings

S I P T R U N K S E T T I N G SThese settings are also documented in the System Initialization Settings appendix.

Set as Voice Mail Setting to enable this line for voice mail type services (True or False (default))

DNS NAPTR Service • NAPTR stands for Naming Authority Pointer and is a newer type of DNS record that supports regular expression based rewriting. Many SIP trunk providers do not support NAPTR so disabling this feature on the trunk will decrease the set up time for han-dling calls.

DNS SRV Service • An SRV record or Service record is a category of data in the Internet Domain Name System specifying information on avail-able services. SRV records are important for redundancy and failover with SIP trunks. Some SIP service providers may not support Fully Qualified Domain Names (FQDN) and might prefer to be configured with a single IP address. This could lead to a service disruption under certain failure scenarios.



Vendor Speci f ic SIP Trunking Conf igurat ionOutbound Caller ID: Some SIP trunk service providers (it is known specifically that CBeyond has this limitation) will not accept an Outbound Caller ID that is not assigned to that customer. Within the Sphere system, if a customer performs a trunk to trunk transfer, Sphere will send the original callerID and the call may be rejected by the provider. For example, ACME company has DID numbers from the SIP trunk service provider of 847-555-12XX. A call comes into the system from 847-666-1234 and rings a phone which has a forwarding condition to the end user's cell phone. Sphere will place an outbound call to the cell phone using 847-666-1234 as the Caller ID. The SIP service provider may reject this call since 847-666-1234 is not a part of that customer's DID range of 847-555-12XX. These Outbound Caller ID rules can be configured on the trunk to prevent 847-666-1234 from being sent out the door (located on the Trunks, General tab).Inbound digits: Some SIP service providers may send extra inbound digits or characters in front of the called party number. With a traditional TDM circuit, this character field is usually limited to 4-, 7- or 10-digits, with no special characters. SIP does not have this limitation, therefore some SIP providers will prepend a route code, like 0200, in front of the 11-digit number while others may send a "+" in front of the number. This could have a side effect of this digit or character showing up on the Sphericall Desktop or IP phone screen. Contact Sphere Technical Support if this is your scenario.

SIP > Override Timers B/F (Invite/Non-Invite Client Transaction)

• Setting can be set at the system level and at the trunk level. This setting redefines the SIP Timer B for Invite Transactions. Cur-rently it’s being used only for INVITE client transactions where more than one destination IP address is available (DNS SRV records).

• 8000 milliseconds (default); Range: 5000 - 64000

SIP > Registration Refresh Threshold

• This setting defines the percentage expiry after which the REG-ISTER requests to refresh the registration bindings should be sent. E.G. if the Registration refresh period is 1800 seconds and the threshold is set to 75%, then MGC will send a REGISTER request to refresh the binding every 1350 seconds.

• Value 50 (default)

Use RFC2543 Hold • The following is a description of the method for putting a call on hold as defined in RFC 2543. If a party in a call wants to put the other party "on hold", i.e., request that it temporarily stops send-ing one or more media streams, a party re-invites the other by sending an INVITE request with a modified session description. The session description is the same as in the original invitation (or response), but the "c" destination addresses for the media streams to be put on hold are set to zero (0.0.0.0). This setting might need to be set to False for certain SIP trunk service pro-viders.


. . .

. .S I P TR U N K I N GSIP Trunking Tie Line

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I P T R U N K I N G T I E L I N E

Figure 17.11 SIP Tie-Line Trunking

This scenario of a SIP tie line solution has the following characteristics:• This is the essence of “soft trunking.” When you set up the system appropriately,

and assign a “softtrunk” using only an IP connection, two different organizations may use SIP for control of calls to each other, as well as sending calls out to the PSTN.

• Companies may use SIP for their call control over their WAN calls, but may still use other media gateways to access the PSTN.

I N S T A L L A T I O N & C O N F I G U R A T I O N O F S I P T I E L I N E S E R V I C E

To insta l l S IP t runking or t ie l ines1 YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE

SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints.2 Refer to Book 1: Planning & Preparing the Sphere System for information on ordering

and planning SIP services.3 Once you have your IP/internet services delivered and the Customer Premise Equipment

(CPE) is onsite (typically an ALG—Application Layer Gateway), connect the CPE to the local area network (straight-through RJ-45 cable from the CPE to the Ethernet network switch).

4 During the planning process of your SIP tie line, you must secure the following information:


S I P TR U N K I N GSIP Trunking Tie Line



In some cases, the service provider may only make changes or additions to your services or the service order through one individual.




• All defaults are automatically configured for your convenience.• User Agents are also available for “adding” a new, untested SIP endpoint, the

emphasis being, if a site has a new SIP endpoint, they must add it to the system themselves and verify its operability through their own testing.


default settings for some deployments.1 Select the SIP tab on the System Properties window.


. . .




know how to treat it or what features to apply per endpoint.7 Enter an Agent Description that is appropriate for the Name & Endpoint Type.8 Click Apply.

Note: Those User Agents listed in the SIP dialog window that also have the Default checked, are those User Agents created into the system by default. These will remain in the system. User Agents added by system administrators are not indicated with a check in the Default column.

To conf igure SIP for t ie l ine1 From the Sphericall Administration application:2 Select the Trunks tab.3 Right-click and select Add.4 Choose Add Softtrunk.5 Fill in the fields as follows:




Tie Lines Between Sphericall SystemsSince the time to failover is determined by the registration lifetime of inbound registrations, we recommend that tie lines between Sphericall systems be configured as registration type None. However, it is still possible to use inbound/outbound registration by configuring one end of the tie line for Inbound registration and the other end for Outbound registration. In this case, both ends of the tie line must be configured with the same account and password on the Authorization tab of the Trunk properties dialog box. This is required since either side of the tie line may challenge the other, and we do not support "inbound" and "outbound" credentials.Type in the information for each field:

6 Description: Any printable ASCII character except double quote or backslash. This field is a description field that specifies the display-name in the From and To headers of Outbound REGISTER requests. Often it will be the name that is assigned to this service Example: LincolnWashingtonTieLine (noting the two systems you are tying together).

7 Account: Information regarding this tie line (i.e. could be the same info as entered in the Description field for this tie line). Example: LincolnWashingtonTieLine (noting the two systems you are tying together).The Account field specifies the userinfo portion of the SIP URI in the From and To headers of outbound REGISTER requests. The Account field may contain any printable character except the following: space, pound sign #, percent sign %, colon :, less than <, greater than >, at sign @, square brackets [], backslash \, caret ^, brace brackets {}, pipe | and grave accent `.

8 Service Provider Domain: Information in this field will represent WHO you are sending your registration TO. For the tie line, you fill in the info of the OTHER Sphere system in this field. Example: Lincoln.sip.company.netThe Service Provider Domain field (SPD) may be filled with an IP address, DNS host name or DNS domain name. The SPD specifies the hostname portion of the SIP URI in the From and To headers of outbound REGISTER requests. The SPD also specifies the destination for outbound SIP requests if the Outbound Proxy (OP) is blank. A DNS lookup is performed on the

Information entered herewill reflect the OTHER Sphere system’sinformation.

NOTE:


. . .


contents of the SPD if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below).

9 Outbound Proxy: Recommend using an IP Address of the other Sphere MGC on the other end of the tie line. Example: 66.237.127.53The Outbound Proxy field (OP) used to be called the Host field. The OP may be filled with an IP address, DNS host name, DNS domain name or it may be left empty. If the OP is not empty, all outbound SIP requests are sent to the specified destination. A DNS lookup is performed on the contents of the OP if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). If the OP is empty, the destination for outbound SIP requests is determined by the Service Provider Domain field.

SummaryThe SPD and OP determine where the Sphericall system sends outbound SIP requests. The OP field would typically be used when the administrator wants to force all outbound SIP messages through a specific device, for example, a SIP Session Border Controller (SBC).

Note: The following two System Initialization Settings could affect settings for the SPD and OP:

10 Port information: use the default.11 Registration type:

Recommended for Tie Line installations: NONE

• Inbound - the SIP trunk sends a register request to the Sphericall system. When the SIP trunk first initiates contact by sending a REGISTER request, the MGC automatically creates a hub, trunk and outside service. The Service Provider tab does not appear in the Hub device properties dialog box since the SIP trunk provides all the required information in the REGISTER request.

• Outbound - the Sphericall system sends REGISTER request to the SIP trunk; all fields on the Service Provider tab (trunk device properties) are used.

• None - issues no registration; neither the SIP trunk nor the Sphericall system send a REGISTER request; all fields on the Service Provider tab except the Contact Domain are used (trunk device properties).

Primary MGC:

DnsNaptrServiceEnabledThe DnsNaptrServiceEnabled MGC setting specifies whether the MGC will perform DNS NAPTR queries on the SPD/OP. This setting defaults to true. • The administrator may optimize MGC performance by changing this setting to false if the

SPD/OP specifies a host name,or

• if the authoritative DNS server for the SPD does not support NAPTR queries.

DnsSrvServiceEnabledThe DnsSrvServiceEnabled MGC setting specifies whether the MGC will perform DNS SRV queries on the SPD/OP. This setting defaults to true. The administrator must change this setting to false • if the SPD/OP specifies a host name, • if the SPD is not a valid DNS domain name with an SRV record,

or • if the authoritative DNS server for the SPD does not support SRV queries.



12 Select the Media Server (MGC) name from the drop down box for the Primary.Example: Lincoln1 would be selected

Secondary MGC: 13 Leave the drop down box set to NONE for the Secondary.

• The Primary and Secondary MGC fields specify which MGCs are primary and secondary for the SIP trunk. The Secondary MGC assumes control of the trunk should the Primary fail.

• For tie line installations, with registration type None, and no information in the Contact Domain, set the Primary to reflect the Media Server (MGC) name.

The following information about Primary and Secondary is accurate, but used only if the Contact Domain information field is used, which is rarely used for tie line implementations, but may occasionally be used if the tie is installed with an Outbound REGISTER.If the CD specifies an IP address or DNS host name, the Secondary MGC field should be filled with "None." In this case, specifying a Secondary MGC has limited value since the external system will not be able to locate the Secondary MGC to send SIP requests to. In practice, calls inbound to the Sphericall system will fail. Calls outbound from the Sphericall system may fail.If the CD specifies a DNS domain name, the DNS SRV records must match the contents of the Primary and Secondary MGC fields. Specifically, the "first" SRV record must point to the Primary MGC and the "second" SRV record must point to the Secondary MGC. The order in which the SRV records are returned are determined by the priority and weight values assigned to the SRV records in DNS. The external system must attempt to contact the MGC pointed to by the SRV record with the lowest priority first. If multiple SRV records have the same priority, the external system must attempt to contact the MGC pointed to by the SRV record with the highest weight first. Since Microsoft's DNS server returns different orderings for SRV records with the same priority from query to query, we recommend setting the (priority, weight) of the SRV record for the Primary and Secondary MGC to (0, 0) and (1000, 0) respectively.If the CD specifies a DNS domain name, that name must be resolvable from the external system. For example, we cannot use internaldomain.spherecom.com as the CD for Cbeyond sinceinternaldomain.spherecom.com cannot be resolved anywhere other than the internal Sphere network.

14 Contact Domain: For Tie Line installations, when using NONE for the registration type, Contact Domain information will not be filled in.How do the other side contact you? (IP address or FQDN):The Contact Domain field (CD) specifies the hostname portion of the SIP URI in the Contact header of outbound REGISTER requests. In other words, the CD determines how the external system locates an MGC to send SIP requests to. The CD may be filled with an IP address, DNS host name or DNS domain name. If filled with an IP address or host name, the external system will be bound to a single MGC with no redundancy. The CD should only be filled with a DNS domain name if the external system supports the DNS NAPTR and SRV lookup procedures specified by RFC 3263.Notice that the hostname portion of the SIP URI in the Contact header of all outbound non REGISTER requests is filled with the IP address of the MGC sending the request. This binds the SIP dialog to the MGC. This is necessary because we do not share call state information between MGCs. That is, it is not possible for a second MGC to assume control of a call in the middle of the call in the case of an MGC failure.


. . .



Typically UDP. This determines the preferred network transport protocol for SIP signaling (TCP, TLS, UDP (default)). Note: For SIP trunks that use registration types “Outbound” or “None,” this setting specifies what transport is used to send SIP requests such as REGISTER and INVITE to the SIP Trunk.


Sphere has included for selection most of the common types of User Agents. If the user agent your system is using is not offered here, you must create it in the System General Properties (SIP) window, then apply it to the User Agent associated with the trunk.

17 Click OK.


SIP Trunk MAC AddressWhen a trunk checks into a Sphericall system for the first time, the MGC automatically creates the necessary database records for it. As part of the record creation process, the MGC assigns a unique name to the trunk. For Sphericall MGs, the MGC uses the MG's MAC address as the unique name. For SIP trunks, the MAC address assigned to the trunk depends upon the registration type.For outbound registrations and registrations of type none, the MGC uses "Account@Service Provider Domain."For inbound registrations, the MGC uses the SIP URI of the REGISTER request To header. It is important to note that if the URI changes, the MGC will create new records for the trunk the first time it sends a REGISTER with the new URI.


Once creation of the SIP softtrunk tie line is complete, the following information will be available on the Hub Properties:

18 Type in the descriptive information as it applies to this device.

• Name: Name of device (can be renamed for user-friendliness).• MAC Address: Assigned to the device by the system.• Stations/Trunks: Information on number of stations or trunks.• LAN: Hub belongs to LAN displayed in this field.



• Localization Settings: Country/local language settings based on Localization Settings profile.

• Description: Information in this field can be edited.• Hub Number: Automatically numbered by the system (recommended that this

number remains unchanged).• Last Check In Time: Information is updated upon each check-in.• Firmware Version: None.• Device Type: Default device type for Softtrunk is “Unknown SIP Agent.”

To conf igure SIP t runk propert iesThe following information on this softtrunk can be found on the trunk properties:



• Name: Type name of softtrunk• Hardware ID: Usually assigned dynamically by the system.• Telephony Area: Assign or change Telephony area here.• Zone: Assign or change Zone here.• Total Capacity: Configure based on services provided by SIP provider.• Inbound Capacity: Determine based on services provided.• Outbound Capacity: Determine based on services provided.

Note: Administrators can shape the inbound and outbound activity on this softtrunk.

• Max Duration: Type the number of seconds in the Max Duration field that will designate the maximum allowable duration of a trunk-to-trunk call. Default = 0 = No Limit.

• In Service: Default is ENABLED.


. . .


• Emergency Calls Enabled: Default is ENABLED.2 Add any Outbound Caller IDs that are needed for your system and this softtrunk.3 Click Apply.4 Select the Authorization tab.


Complete the fields for this authorization as follows:

Note: Entries in these fields must match entries in the Authorization tab at the other side of the tie line.


By enabling this field, the far end SIP server authenticates calls. Tie lines currently have two restrictions (please see restrictions below).• Account: Provided by the other Sphericall Manager.

Think of this as a “User Name” field.• Password: Provided by the other Sphericall Manager. Verify if adding or changing.

Think of this as a “Password” field.• Realm: Provided by the other Sphericall Manager.

Think of this as similar to a “domain” field.• Type: Default type is MD5.

Use only MD5 as the type in this field.• Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism.Together, the account, password and realm form the creditials needed to access a SIP trunk. Indiviually, these fields are arbitrary

Entries here must

match the otherend of the tie line.



When a SIP trunk challenges a request from the MGC, the trunk sends the realm and a random value called a nonce to the MGC. The MGC must then resubmit the request with an authorization header that contains a hash. The hash is computed using the account, password, realm and nonce. A correct hash proves to the SIP trunk that the MGC knows the account and password.If the administrator is dealing with a SIP service provider, these credentials are specified by the service provider.Restrictions: If the administrator is setting up a SIP tie line between two Sphericall systems, there are two restrictions:

• The credentials must be identical on both ends since we only support one set of credentials per trunk. That is, we cannot challenge with one set of credentials and respond with a second set.

• Currently, for tie lines, the realm must be set to “Sphericall.” 6 Click Apply. 7 Select Inward Routing.


8 Select Add to add a DID Mapping to the DID Mappings area (optional).9 Select Add to Schedule the Default Routing (optional).

10 Enter a DID in the Input DID and select Test to verify DID (optional).11 Click Apply.12 Select Outward Routing.


. . .



13 Click Add Extension or Tie Line to add this function to this softtrunk.14 Click Add to add Outside Dialing Rules if required.15 Click Apply.16 Click OK.

Only add Emergency Groups or Settings if you are completing the Emergency Groups installation. Refer to Book 6: Emergency Service Installations.

To ver i fy the sof t t runk is registered on the far end of t ie l ineFirst, allow the configured trunk to “register” at the far end (other end of the tie line). You do not need to ADD a softtrunk on the second end of a tie line.From the Sphericall Administrator application:

1 Select the Trunks tab from the main window (this tab should be BLUE to indicate a new device).

2 Double-click to select the new softtrunk device.3 Check the entries for the device properties.

Note: There will be no MGCs tab when using softtrunk tie lines.4 Verify the Service Provider information (optional).5 Double-click to open the softtrunk trunk properties.6 Enter general information as needed on the General tab.7 Select the Authorization tab.8 Enter into these fields the SAME information that you would find in these fields at the

other end of the tie line.9 Complete the fields for this authorization as follows:



NOTE: Authorization is optional for tie line implementations, based on how you wish to communicate with the other Sphere system that you are “tie-ing” to; entries in these fields must match entries in the Authorization tab at the other side of the tie line.


By enabling this field, the far end SIP server authenticates calls.• Account: Provided by the other Sphericall Manager.

Think of this as a “User Name” field.• Password: Provided by the other Sphericall Manager. Verify if adding or changing.

Think of this as a “Password” field.• Realm: Provided by the other Sphericall Manager.

Think of this as similar to a “domain” field.• Type: Default type is MD5.

Use only MD5 as the type in this field.• Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism.This system needs to have the SIP server’s Account (user name), Password (Password) and Realm (domain) in order for the Sphere system to either challenge, respond or both prior to accepting a call into the system.

• Both (Challenge and Respond): This system will challenge and respond to incoming calls. This authorization credential can be used both to challenge incoming INVITEs and to respond to outgoing INVITE and REGISTER challenges

• To Challenge: This system is set to Challenge all incoming calls and will authenticate each call. This Authorization credential is to be used only to challenge the incoming INVITEs (MGC is receiving the INVITE).

• To Respond: This system is set to respond to authentication of the call as it comes in. This Authorization credential is to be used only to respond to an INVITE or a REGISTER challenge (MGC has sent the INVITE/REGISTER and the destination is challenging).


11 Click Apply. 12 Complete the configuration of Inbound, Outbound and Default Routing as desired.


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S I P TR U N K I N GSIP tie line to Third-Party App

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . S I P T I E L I N E T O T H I R D - P A R T Y A P P

Figure 17.19 SIP Tie Line to Third-Party App

This example scenario of a SIP tie line to call center solution has the following characteristics:

• Sphere system administrators are able to connect to a call center for call queuing, processing, etc.

• Calls are directed from the Sphere system to the call center server/manager via DID, Auto Attendant routing, default route, etc.

• This tie-line connection method can be used for a voice mail setup, fax server, call center

I N S T A L L A T I O N & C O N F I G U R A T I O N O F S I P T I E T O C A L L C E N T E R ( E X A M P L E )

To insta l l S IP t runking or t ie l ines1 YOU MUST FIRST VERIFY OR ADD THIS SIP ENDPOINT AS A USER AGENT TO THE

SYSTEM. Refer to the beginning of this chapter for this procedure for all SIP endpoints.2 Refer to Book 1: Planning & Preparing the Sphere System for information on planning

SIP services.3 Once you have your SIP services delivered and the Customer Premise Equipment (CPE)

is onsite (typically an ALG—Application Layer Gateway), connect the CPE to the network (straight-through RJ-45 cable from the CPE to the network switch).

4 During the planning process of your SIP service, you must secure the following information:


. . .

. .S I P TR U N K I N GSIP tie line to Third-Party App



To conf igure SIPFrom the Sphericall Administration application:

1 Select the Trunks tab.2 Right-click and select Add.3 Choose Add Softtrunk.4 Fill in the fields as follows:


Type in the information for each field:5 Description: Any printable ASCII character except double quote or backslash.

This field is a description field that specifies the display-name in the From and To headers of Outbound REGISTER requests. Often it will be the phone number that is assigned to this service.Example: 8477932222

6 Account: Provided to you by call center service provider (i.e. could also be phone number that is assigned to this service). Example: 8477932222The Account field specifies the userinfo portion of the SIP URI in the From and To headers of outbound REGISTER requests. The Account field may contain any printable character except



the following: space, pound sign #, percent sign %, colon :, less than <, greater than >, at sign @, square brackets [], backslash \, caret ^, brace brackets {}, pipe | and grave accent `.

7 Service Provider Domain: Provided by the call center service providerExample: 777445.sip.company.net.The Service Provider Domain field (SPD) may be filled with an IP address, DNS host name or DNS domain name. The SPD specifies the hostname portion of the SIP URI in the From and To headers of outbound REGISTER requests. The SPD also specifies the destination for outbound SIP requests if the Outbound Proxy (OP) is blank. A DNS lookup is performed on the contents of the SPD if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below).

8 Outbound Proxy: Provided by the call center service providerExample: siptrunkinfo.sip.callcenterserver.netThe Outbound Proxy field (OP) used to be called the Host field. The OP may be filled with an IP address, DNS host name, DNS domain name or it may be left empty. If the OP is not empty, all outbound SIP requests are sent to the specified destination. A DNS lookup is performed on the contents of the OP if it does not contain an IP address. The type of DNS lookup performed is specified by the DnsNaptrServiceEnabled and DnsSrvServiceEnabled settings (see Note below). If the OP is empty, the destination for outbound SIP requests is determined by the Service Provider Domain field.

SummaryThe SPD and OP determine where the Sphericall system sends outbound SIP requests. The OP field would typically be used when the administrator wants to force all outbound SIP messages through a specific device, for example, a SIP Session Border Controller (SBC).

Note: The following two System Initialization Settings could affect settings for the SPD and OP:

9 Port information: use the default.10 Registration Type:

Recommended for softtrunk to call center: OUTBOUND

• Inbound - the SIP trunk sends a register request to the Sphericall system. When the SIP trunk first initiates contact by sending a REGISTER request, the MGC automatically creates a hub, trunk and outside service. The Service Provider tab does not appear in the Hub device properties dialog box since the SIP trunk provides all the required information in the REGISTER request.

DnsNaptrServiceEnabledThe DnsNaptrServiceEnabled MGC setting specifies whether the MGC will perform DNS NAPTR queries on the SPD/OP. This setting defaults to true. The administrator may optimize MGC performance by changing this setting to false: • if the SPD/OP specifies a host name

or • if the authoritative DNS server for the SPD does not support NAPTR queries.

DnsSrvServiceEnabledThe DnsSrvServiceEnabled MGC setting specifies whether the MGC will perform DNS SRV queries on the SPD/OP. This setting defaults to true. The administrator must change this setting to false:• if the SPD/OP specifies a host name, • if the SPD is not a valid DNS domain name with an SRV record,

or • if the authoritative DNS server for the SPD does not support SRV queries.


. . .


• Outbound - the Sphericall system sends register request to the SIP trunk; all fields on the Service Provider tab (trunk device properties) are used.

• None - issues no registration; neither the SIP trunk nor the Sphericall system send a register request; all fields on the Service Provider tab except the Contact Domain are used (trunk device properties).

11 Primary MGC:

Secondary MGC: The Primary and Secondary MGC fields specify which MGCs are primary and secondary for the SIP trunk. The Secondary MGC assumes control of the trunk should the Primary fail.If the CD specifies an IP address or DNS host name, the Secondary MGC field should be filled with "None." In this case, specifying a Secondary MGC has limited value since the external system will not be able to locate the Secondary MGC to send SIP requests to. In practice, calls inbound to the Sphericall system will fail. Calls outbound from the Sphericall system may fail.If the CD specifies a DNS domain name, the DNS SRV records must match the contents of the Primary and Secondary MGC fields. Specifically, the "first" SRV record must point to the Primary MGC and the "second" SRV record must point to the Secondary MGC. The order in which the SRV records are returned are determined by the priority and weight values assigned to the SRV records in DNS. The external system must attempt to contact the MGC pointed to by the SRV record with the lowest priority first. If multiple SRV records have the same priority, the external system must attempt to contact the MGC pointed to by the SRV record with the highest weight first. Since Microsoft's DNS server returns different orderings for SRV records with the same priority from query to query, we recommend setting the (priority, weight) of the SRV record for the Primary and Secondary MGC to (0, 0) and (1000, 0) respectively.If the CD specifies a DNS domain name, that name must be resolvable from the external system. For example, we cannot use internaldomain.spherecom.com as the CD for Cbeyond sinceinternaldomain.spherecom.com cannot be resolved anywhere other than the internal Sphere network.

12 Contact Domain: You provide this informationExample: 11.2.27.122The Contact Domain field (CD) specifies the hostname portion of the SIP URI in the Contact header of outbound REGISTER requests. In other words, the CD determines how the external system locates an MGC to send SIP requests to. The CD may be filled with an IP address, DNS host name or DNS domain name. If filled with an IP address or host name, the external system will be bound to a single MGC with no redundancy. The CD should only be filled with a DNS domain name if the external system supports the DNS NAPTR and SRV lookup procedures specified by RFC 3263.Notice that the hostname portion of the SIP URI in the Contact header of all outbound non REGISTER requests is filled with the IP address of the MGC sending the request. This binds the SIP dialog to the MGC. This is necessary because we do not share call state information between MGCs. That is, it is not possible for a second MGC to assume control of a call in the middle of the call in the case of an MGC failure.


Typically UDP. This determines the preferred network transport protocol for SIP signaling (TCP, TLS, UDP (default)). Note: For SIP trunks that use registration types



“Outbound” or “None,” this setting specifies what transport is used to send SIP requests such as REGISTER and INVITE to the SIP Trunk.


Sphere has included for selection most of the common types of User Agents. If the Call Center User Agent your system is using is not offered here, you must create it in the System General Properties (SIP) window, then apply it to the User Agent associated with the trunk.

15 Click OK.


SIP Trunk MAC AddressWhen a trunk checks into a Sphericall system for the first time, the MGC automatically creates the necessary database records for it. As part of the record creation process, the MGC assigns a unique name to the trunk. For Sphericall MGs, the MGC uses the MG's MAC address as the unique name. For SIP trunks, the MAC address assigned to the trunk depends upon the registration type.For outbound registrations and registrations of type none, the MGC uses "Account@Service Provider Domain."For inbound registrations, the MGC uses the SIP URI of the REGISTER request To header. It is important to note that if the URI changes, the MGC will create new records for the trunk the first time it sends a REGISTER with the new URI.


Once creation of the SIP softtrunk tie line is complete, the following information will be available on the Hub Properties:

16 Type in the descriptive information as it applies to this device.

• Name: Name of device (can be renamed for user-friendliness).• MAC Address: Assigned to the device by the system.• Stations/Trunks: Information on number of stations or trunks.• LAN: Hub belongs to LAN displayed in this field.• Localization Settings: Country/local language setting based on the Localization

Setting profile.• Description: Information in this field can be edited.


. . .


• Hub Number: Automatically numbered by the system (recommended that this number remains unchanged).

• Last Check In Time: Information is updated upon each check-in.• Firmware Version: None.• Device Type: Default device type for Softtrunk is “Unknown SIP Agent.”

To conf igure SIP t runk general propert iesThe following information on this softtrunk can be found on the trunk properties:



• Name: Type name of softtrunk• Hardware ID: Usually assigned dynamically by the system.• Telephony Area: Assign or change Telephony area here.• Zone: Assign or change Zone here.• Total Capacity: Configure based on information from the call center application.• Inbound Capacity: Configure based on information from the call center

application.• Outbound Capacity: DConfigure based on information from the call center

application.

Note: Administrators can shape the inbound and outbound activity on this softtrunk.

• Max Duration: Type the number of seconds in the Max Duration field that will designate the maximum allowable duration of a trunk-to-trunk call.

• In Service: Default is ENABLED. • Emergency Calls Enabled: Default is ENABLED.



The following information helps you to understand the provisioning of the Total Capacity of the SIP trunks as configured on this window:

Guaranteed Outbound = Total Capacity - Inbound CallsGuaranteed Inbound = Total Capacity - Outbound callsExample Configuration: Total Capacity=10, Inbound=8, Outbound=7 would imply the following:Guarantee 3 inbound calls with max of 8

2 Add any Outbound Caller IDs that are needed for your system and this softtrunk.3 Click Apply.4 Select the Authorization tab.


5 Complete the fields for this authorization as follows:

NOTE: This field is optional for tie line implementations, based on how you wish to communicate with the call center system that you are “tie-ing” to: entries in these fields indicate who THIS trunk is on this side of the tie line.


By enabling this field, the far end SIP call center authenticates calls.• Account: Provided by the call center system.

Think of this as a “User Name” field.

Total Capacity • total number of simultaneous calls the softtrunk can support

Inbound Capacity • the number of simultaneous inbound calls the administrator wants to accept.

Outbound Capacity • the number of simultaneous outbound calls the administrator wants to allow.

Information entered here is information fromthe call center system.


. . .


• Password: Provided by the call center system. Verify if adding or changing.Think of this as a “Password” field.

• Realm: Provided by call center system.Think of this as similar to a “domain” field.

• Type: Default type is MD5.Use only MD5 as the type in this field.

• Authorization Type: It is important to note that in SIP, registration is used for routing incoming SIP requests and has no role in authorizing outgoing requests. Authorization and authentication are handled in SIP on a request-by-request basis with a challenge/response mechanism.This system needs to have the SIP call center’s Account (user name), Password (Password) and Realm (domain) in order for the Sphere system to either challenge, respond or both prior to accepting a call into the system.

• Both (Challenge and Respond): This system will challenge and respond to incoming calls. This authorization credential can be used both to challenge incoming INVITEs and to respond to outgoing INVITE and REGISTER challenges

• To Challenge: This system is set to Challenge all incoming calls and will authenticate each call. This Authorization credential is to be used only to challenge the incoming INVITEs (MGC is receiving the INVITE).

• To Respond: This system is set to respond to authentication of the call as it comes in. This Authorization credential is to be used only to respond to an INVITE or a REGISTER challenge (MGC has sent the INVITE/REGISTER and the destination is challenging).


7 Click Apply. 8 Select Inward Routing.




9 Select Add to add a DID Mapping to the DID Mappings area (optional).10 Select Add to Schedule the Default Routing (optional).11 Enter a DID in the Input DID and select Test to verify DID (optional).12 Click Apply.13 Select Outward Routing.


14 Click Add Extension to add this function to this softtrunk.


. . .


15 Click Add to add Outside Dialing Rules if required.16 Click Apply.17 Click OK.













S I P TR U N K I N GTroubleshooting SIP Connections

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match is found, the Sphericall Manager moves to step 2.Note: this is the most common way stations are identified, but does not help for trunks since the FROM field contains the caller ID of the incoming call.











. . .

. .S I P TR U N K I N GTroubleshooting SIP Connections



S I P TR U N K I N GTroubleshooting SIP Connections


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .QUINTUM VOIP GATEWAYS 18

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O V E R V I E W

The Quintum Tenor AF series analog VoIP gateways are ideal for small Sphericall systems or small office locations that are part of a larger distributed Sphericall system. The Quintum Tenor AF series analog VoIP gateways will offer enhanced capabilities for Sphericall users in two key areas. First they offer a flexible and cost effective small office and remote office options for both analog trunk and station connections. The Quintum gateways offer a range of options for trunk, station and combined gateway solutions, in a small, cost effective foot print, that is certified to operate with Sphericall. Second, they also offer additional small / remote site survivability options for Sphericall users. By leveraging SIP as an open standard, and the Quintum gateways “built in” proxy and stand-by call agent, Sphericall users can leverage the combined Sphere and Quintum solution to further increase the high availability capabilities of a Sphericall system. The powerful combination of the Sphericall IP PBX and Quintum Tenor gateways provides a complete Unified Communications solution that delivers communications services across your entire enterprise, even at the smallest of office locations.When using the Quintum gateway as part of a survivability solution, the Quintum gateway is configured as an Outbound SIP Proxy on behalf of the SIP telephones at a remote office site. As an Outbound SIP Proxy, the Quintum gateway remains in the “signaling path” between the SIP telephones and a Sphericall Manager that is located across a customer WAN. The Quintum gateway maintains a registration with the Sphericall Manager. If the connection to the Sphericall Manager is interrupted (for instance, a WAN failure), the Quintum gateway will assume responsibility for handling call processing requests for the SIP telephones that are configured to use it as an Outbound SIP Proxy. This includes routing calls between any device (SIP Phone, analog phone, analog trunk) that is “connected” to the Quintum gateway. Following are some examples:Example 1: A local SIP Phone places a local PSTN call during a WAN failure• A SIP phone users dials the outside service digit 8 and the 11 digit PSTN number

1-555-555-5555.


Q U I N T U M VO I P G A T E W A Y SOverview

• The Quintum Outbound SIP Proxy attempts to forward the signaling messages to the Sphericall Manager.

Upon recognizing that the Sphericall Manager is not available, the Quintum gateway will look at the SIP messages and match the dialed number to it’s internal dial plan rules which instructs the Quintum gateway to route calls with an outside service prefix to the local PSTN trunks after removing the prefix from the digit string.Example 2: Inbound call routing from the PSTN during a WAN FailureFor inbound calls over analog trunks, most often these calls are routed by default to a Sphericall Manager auto-attendant. In this environment, the Sphericall Manager is reachable at another location across a WAN. If the WAN fails, and the Sphericall Manager is not reachable, then the Quintum Tenor can route inbound PSTN calls to one of the FXS ports or a SIP phone. The following Quintum Tenor gateway models are certified for use with Sphericall. Note that each Quintum Tenor Survivable gateway listed below can support up to 50 SIP telephones as an Outbound SIP Proxy.

Table 18.1 Quintum Gateway models

Tenor AF Series Gateways AF Classic AF Survivable

AFT200: 2 FXO ports 501-1199-00 501-1199-SG

AFT400:4 FXO ports 501-1202-00 501-1202-SG

AFT800: 8 FXO ports 501-1195-00 501-1195-SG

AFG200: 2 FXO ports 501-1198-00 501-1198-SG

AFG400:4 FXO ports 501-1201-00 501-1201-SG

AFG800: 8 FXO ports 501-1194-00 501-1194-SG

AFE400:4 FXS + 2 FXO ports 501-1203-00 501-1203-SG

AFE600: 6 FXS +2 FXO ports 501-1205-00 501-1205-SG

AFM200: 2 FXS +2 FXO ports 501-1200-00 501-1200-SG

AFM400: 4 FXS +4 FXO ports 501-1204-00 501-1204-SG

Power SupplyPower Cable

UniversalNorth America

520-1033-00501-1190-00


. . .

. .Q U I N T U M VO I P G A T E W A Y SOperation

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . O P E R A T I O N

Figure 18.1 Quintum Solutions

Quintum gateways provide options for supporting analog stations and trunks on the Sphere system. Once configured, the Sphericall Manager (MGC) has full control of calls placed and received on the phones and trunks connected to the Quintum gateway.If using a survivable Quintum gateway, the Quintum gateway knows about the IP and analog phone activities. If at some time the phone sends a signal intended for the Sphericall Manager, and the WAN connection is not available, the Quintum gateway will retry the Sphericall Manager at least 3 times (this is configurable). If there is no reply within the specified time period, the gateway will attempt to complete the call itself. If the call was placed to another extension at that same remote site, it will transfer the call locally. If the call was placed to the PSTN, the gateway will dial the PSTN and pass the call through to the PSTN locally. The Quintum gateway cannot, however, pass the call to another networked location, since the WAN connection is not available.• Example: If you dial 8-1-847-793-9600 from the Quintum side, the Quintum will

offer the call to the Sphericall Manager first. If the Sphericall Manager is down, it consults its internal routing tables. In the hopoff directory you'd have an entry that says anything that starts with 81, strip off the 8 and send it to an FXO port.

• Example: Local PSTN numbers can be routed directly in to the Quintum FXO/FXS unit and configured to first go to the Sphericall Manager auto attendant. If there is a WAN failure, the call would go to one FXS phone on the Quintum side.


Q U I N T U M VO I P G A T E W A Y SPlanning

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P L A N N I N G

• Verify Sphere System Requirements for Quintum gateway’s version for interoperability with Sphericall. Specific versions of Quintum are only compatible with specific Sphericall versions.

• The Sphere system should be installed, configured and tested as fully functional.• Refer to the Quintum VoIP Gateway User Manual for installation planning, setup,

package contents, safety, and conditions of use.• Quintum units can be used on local sites as diagramed above.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P R E P A R I N G

1 Set up Quintum unit as directed by package instructions. Appropriate power supply and power cord for each unit and country should be verified prior to installation.

2 Prepare network according to Sphere System Requirements.3 Ensure that you have the latest Quintum software loaded using the manufacturer’s

instructions.

www.quintum.com/support4 If you are using a survivable gateway, you must update the applicable license file.

The license file will only work on the Quintum device with the serial number that matches the one provided when the license file was generated. The license file is not interchangeable among devices. Consult Quintum at: www.quintum.com/support.

5 Know the required login for working with this gateway:Default Username: adminDefault Password: admin

Be sure to refer to the manufacturer’s specifications for all cabling:6 Connect appropriate LAN cable to the LAN port.7 Connect RJ-45 straight-through pinout cable for PSTN connection to the appropriate

port.8 Connect RJ-11 cable(s) for telephone connection to the appropriate port(s)—analog

only.9 Connect SIP phones to appropriate network port(s).

10 Connect the standard RS-232 serial cable to the Console Port.11 Connect AC Power Adapter jack to the power supply and to the power source.12 Be sure to have the Product Guide for the appropriate Tenor products in CD form.

• There is a version of the Configuration Manager on the CD that is shipped with each product.

• Administrators should be sure to match the version of the Configuration Manager to the unit they are configuring as well as to the firmware version of the software on the gateway unit.

13 Know the dial plan for all the extensions on the remote office node prior to installation of the Quintum gateway unit.


. . .

. .Q U I N T U M VO I P G A T E W A Y SInstalling

To connect analog phone devicesAnalog (only available on FXO/FXS units):

1 Connect one end of the RJ-11 into the phone.2 Connect the other end of the RJ-11 into the FXS port on the Tenor device.

To connect SIP phone devicesSIP:

1 Connect one end of the SIP phone cable into the network/switch.2 Connect the Tenor device to the network.

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . I N S T A L L I N G

There is more than one way to integrate the Quintum VoIP Gateway to the Sphericall system as a remote office communications device depending on the model and the network.

To insta l l wi th Sphere system1 Connect the Quintum gateway to the network. It should obtain a DHCP IP address.

Please see the Quintum manual for assigning a static IP address.2 Load the Quintum Configuration Management software to your Sphere Manager. Once

completed, run the program.3 The Quintum Configuration Management software has the capability to discover (within

the same subnet) all Quintum devices. 4 Once the device is discovered, you can start the configuration by connecting to the

device.

I N S T A L L A T I O N O F T E N O R D E V I C EQuintum gateways may be configured either via a telnet command line interface or via the Quintum Configuration Manager (CM). The CM is a Quintum-provided, custom application that runs on a Windows PC, providing a graphical user interface (GUI) interface to the CLI. When the CM detects that a Quintum gateway is running the factory default configuration, it offers to run the Configuration Wizard. The Configuration Wizard steps the administrator through configuring the Quintum gateway. The wizard dialogs and our recommended settings are:

IP Address Conf igurat ion1 Use DHCP2 Obtain DNS Server Addresses Automatically

Dial P lan Conf igurat ion7-digit Dialing & 10-digit Dialing:


Q U I N T U M VO I P G A T E W A Y SInstalling

1 Dial Plan Country: None2 Progress Tone Country: USA/Canada3 Confirm/OK

11-digit Dialing:4 Dial Plan Country: None5 Progress Tone Country: USA, Canada, Mexico (should match your country)6 Country Code: 17 Area/City Code: Enter area code8 Long Distance Prefix: 19 Int'l Prefix: 0

10 Confirm/OK

Phone Port Conf igurat ionThis dialog configures the FXS ports.

1 Disconnect Generation -> Battery Removal2 Caller ID Generation -> FSK3 Phone Number/Extension -> Assign an extension to each channel. This is the same

extension that is assigned in the Sphericall Administrator application (know your dial plan and extension assignment prior to configuring the Quintum gateway).

4 Adda. Number Pattern: Extension of phone assigned in Sphericall Adminb. Channel: Corresponding connection

5 Confirm/OK

Mult i Path Conf igurat ionWhile Quintum gateways can be configured to route calls without offering them to the Sphericall Manager, this configuration is not certified for use with a Sphere system. Set the following settings for an approved configuration:

1 Pass Through Calls -> No2 Bypass Numbers -> None

Line Port Conf igurat ionThis dialog configures the FXO ports.

1 Disconnect Detection -> Battery Removal2 Caller ID Detection -> FSK or DTMF3 Tone Based Answer Detection -> Disabled

Survivabi l i ty Conf igurat ion (only avai lable for conf igurat ion on survivable uni ts)This dialog enables survivable mode if the Quintum gateway is licensed for it. In survivable mode, if connection to the Sphericall Manager is lost, the gateway will attempt to process calls itself.

1 Listening Port -> 50602 Local Gateway -> Yes3 Default Route Port -> 50614 Confirm/OK


. . .


VoIP Rout ing Conf igurat ionThis dialog configures how the Quintum gateway locates and registers with the Sphericall Manager.Primary SIP Server IP/Domain Name -> In order of preference, this can be a domain name for use with DNS SRV records, a Sphericall Manager host name or Sphericall Manager IP address.Primary SIP Server Port -> 5060Register Expiry Time -> 300User ID/Password -> Create a unique user ID for each channel. For the FXS ports, this must be the extension number assigned to that port.Password is only required on trunks if authorization is enabled on the Sphericall Manager. Sphericall does not use authorization with station ports.NOTES:

• FXS ports must be extension assigned to that port in Sphericall Admin. • FXO ports, enter name of Trunk (Ex - Location, YY)• Gateways w/ FXS/FXO ports > Password: ZZ, Required for trunk authorization

enabled in Sphericall Admin1 Outgoing IP Routing: SIP only2 Confirm/OK

Channel Conf igurat ionDisable Trunk/PSTN-side channels -> No

Conf igurat ion SummaryThis dialog summarizes the configuration before writing it to the Quintum gateway.

Once the Configuration Wizard is finished, further steps are required to configure the Quintum gateway for use with Sphericall. The CM has two modes of operation, basic and advanced.

The fol lowing i tems must be conf igured for SIP phone Reg:• The default setting for Voice Codec-1 is G.723.1 6.3 Kbps with 30 millisecond

packetization and for Voice Codec-2 is G.729AB 8.0 Kbps with 30 millisecond packetization. For best interoperability with other endpoints supported by Sphericall, these must be changed to G.711 Mu-law 64 Kbps (or G.711 A-law for outside of North America) with 20 millisecond packetization and G.729AB 8.0 Kbps with 20 millisecond packetization. The Voice Codec settings are located under VoIP Configuration-Voice Codecs (accessible from advanced mode)

Voice Codecs1 VC-1: G.711 Mu-law2 Codec Payload Size: 20 ms3 VC-2: G.729AB4 Codec Payload Size: 20 ms5 Confirm/OK



• Sphericall classifies SIP User Agents as stations or trunks based upon the User-Agent header in the REGISTER request. Quintum allows the value of the User-Agent header to be specified. The required values for Sphericall are:

To conf igure SIP Signal ing Groups1 Under VOIP Configuration, SIP Signaling groups, SIP signaling Group 12 Go to the Advanced Tab and change the User Agent Header to: Quintum-FXO/1.0.0.

This will allow a match of the name to the one in the Sphere system, and will allow the unit to check into Sphericall.

3 Confirm/OK.4 A second SIP signaling group must be created to assign a different User Agent header

to the FXS ports.

The FXS ports must be moved from the existing SIP signaling group to the new SIP signaling group.

To create a second SIP s ignal ing group and move the FXS por ts:1 Advanced Configuration.2 VoIP Configuration.3 SIP Signaling Groups.4 In the SIP Signaling Group-1 dialog, on the User Agent tab, delete the entries for the FXS

ports.5 Right click on SIP Signaling Groups and select "New". Enter 2 as the group index.6 Select SIP Signaling Group-2 and enter a value for the Primary SIP Server. This will

normally be the same as you configured for SIP Signaling Group-1.7 On the Advanced tab, change the User Agent Header to Quintum-FXS.8 On the User Agent tab, create entries for each FXS port, filling in the Primary User and

Contacts[1] fields with the extension assigned to each FXS port in the Sphericall Administrator application.

Under: Circui t Conf igurat ion/Trunk Rout ing Conf igurat ion/Hopoff Number Directory• The Quintum gateway may be used for toll bypass in a gateway-to-gateway

configuration. To be used as a simple PSTN gateway, a “hopoff number directory” must be configured. This directory tells the Quintum gateway what calls should be placed across the PSTN. The following entries will cover 7- 10- or 11-digit dialing:

• “Quintum-FXS” for FXS ports

• “Quintum-FXO” for FXO ports

If running with Sphericall v5.2 (an early beta), these strings must also be entered into the SIP tab of the Sphericall Administrator System Properties dialog. Sphericall v6.0 will ship with these strings in the database.

EXAMPLE 1 11 digit number [(XXX) 123-4567]

Number Pattern: 1

Replacement: 1

Description: Long Distance

Type: Public


. . .


For 7- and 10-digit dialing, you must create an entry for each digit, 0-9.

To set CAS Signal ing Group- l ineThe Quintum FXO ports do not automatically disconnect when loop current is dropped by the Central Office. To correct this:

1 Under Circuit Configuration/Signaling Configuration – CAS Signaling Groups – CAS Signaling Group-line on the General tab

2 Set Signaling Type to “Loop Start, Forward Disconnect”.3 Set Forward Disconnect Delay to 200 milliseconds on the Signaling tab.4 Confirm/OK

To disable Si lence Suppression on the Quintum:For better compatibility with other endpoints supported by Sphericall, Silence Suppression should be disabled.

1 Advanced Configuration.

Number Pattern: 0

Replacement: 0

Description: International

Type: Public

Number Pattern: 0

Replacement: 0

Description: International

Type: Public

Number Pattern: 911

Replacement: 911

Description: Emergency

Type: Public

EXAMPLE 2 7 digit number

Number Pattern: 7

Replacement: 7

Description: Local

Type: Public

EXAMPLE 3 10 digit number [(xxx) 555-1212]

Number Pattern: 3

Replacement: 3

Description: Local

Type: Public



2 VOIP Configuration.3 IP Routing Group.4 IP Routing Group Default.5 Under the General Tab, uncheck the Silence Suppression box.

To enable Cal ler ID name display for cal ls inbound f rom the PSTN:1 Advanced Configuration.2 VOIP Configuration.3 IP Routing Group.4 IP Routing Group Default.5 Under the ANI/Fax Tab, go to the Relay Calling name field and select “Relay CNAM in

INVITE”.

To adjust the volume of PSTN cal ls :1 Advanced Configuration2 VOIP Configuration3 IP Routing Group 4 IP Routing Group Default5 Under the Advanced tab, adjust the Rx Gain to increase the volume of outside parties as

heard by inside parties6 Under the Advanced tab, adjust the Tx Gain to increase the volume of inside parties as

heard by outside parties

Note: Increasing the gains too much may lead to other undesirable effects such as echo.

To enable Hook Flash and Cal ler ID on the stat ion ports:1 Advanced Configuration.2 Circuit Configuration.3 CAS Signaling Groups.4 CAS Signaling Group – phone.5 Check Detect Flash Hook Signal.6 Set Maximum Flash-Hook Duration to 1100.7 Set Minimum Flash-Hook Duration to 350.8 Set Caller ID Generation to FSK.

To set the Quintum to pass Cal ler ID name to the Quintum FXS portsCaller ID name is not set by default on the Quintum TENOR. Complete the following to change this default to pass Caller ID name.

1 Advanced configuration.2 Circuit Configuration.3 CAS Signaling Groups.4 CAS Signaling Groups - Phone Signaling Tab.5 "Relay Calling Name" should be set to "Relay CNAM"


. . .


To adjust the system for gateways that do not support T.38 Fax RelayT.38 is enabled by default on Quintum gateways, however they do not fall back to G.711 if the other end of the call does not support T.38. If gateways that do not support T.38 are present in the system, the “Fax Relay” setting under Advanced Configuration – VoIP Configuration – IP Routing Groups – IP Routing Group-default must be changed on the Fax/QOS tab from “T.38 w/o fallback” to “T.38 w/G.711 X fallback”, where X is Mu-law or A-law.

1 Advanced Configuration.2 VoIP Configuration.3 IP Routing Groups.4 IP Routing Group-default.5 Fax/QOS tab.6 Change “T.38 w/o fallback” to “T.38 w/G.711 X fallback”

Note: Where X above is either Mu-law or A-law.

The fol lowing i tems may opt ional ly be conf igured:• UTC Offset for the time zone the MG is located in under Time Server Configuration

(accessible from basic or advanced mode). • SNMP trap destination under System-Wide Configuration/SNMP Server

(accessible from advanced mode).• Syslog server under System-Wide Configuration/SysLog Servers/SysLog Server-

1.

The fol lowing set t ing must be ver i f ied in the Spher ical l Administrator appl icat ionThe RFC 2833 payload type must be set to 101 on the Sphericall System Properties – Media Streams tab. Otherwise, hold and transfer will not work on the Quintum FXS ports. Analog phones on the Quintum devices must use alternative star codes documented later in this chapter.

1 Open Sphericall System Properties from the General Tab.2 Select the Media Streams tab.3 At the bottom of the dialog ensure the following setting: DTMF digit payload type RFC

2833: 101.

The Trunk Capaci ty must be set in the Spher ical l Administrator appl icat ion1 Trunk tab.2 Expand and find the trunk under the tree.3 Open the trunk properties.4 Verify that the trunk capacity, inbound and outbound is set to the number of trunks.5 Authorization tab: check Use Authorization.6 Enter Primary User: YY & Primary Password: ZZ7 Realm: Sphericall.8 Type: MD59 Authorization Type: To Respond/To Challenge


Q U I N T U M VO I P G A T E W A Y SConfiguration at the Sphericall Manager

To ver i fy the latest f i rmware vers ion of Quintum GatewayQuintum gateway firmware comes in two varieties, survivable (the firmware version starts with “S”) and nonsurvivable (the firmware version starts with “P”). The gateways may be ordered with survivable firmware or they may be upgraded for an additional fee. If upgraded, the survivable firmware and a license file must be loaded onto the gateway via FTP. If the you do not have a license file, you SHOULD NOT load the survivable firmware, since it will not work, even in nonsurvivable mode, unless a license file is uploaded to the gateway.Please note, Quintum survivable firmware is only compatible on v6.0 or greater Sphere systems. Quintum gateways with a serial number ending in -SG are SSG models and must be integrated with Sphericall v6.0 or greater systems.Ensure you have the latest Sphericall certified gateway firmware loaded using the instructions below. Refer to the most current Sphere System Requirements for firmware compatibility.To enable survivable mode to work with Sphericall, another file, var_config.cfg must be loaded onto the Quintum gateway via FTP. This text file contains a single line, “SecureSSG 0”. By default, Quintum gateways will not send RTP to “unknown” gateways, causing REFER based transfers to break. This is only needed in survivable mode.Refer to Quintum’s release notes and Configuration Manager documents for further configuration information.

1 Copy the firmware from the Sphericall CD and unzip the file (i.e. using a program such as WinZip). Save the files to a new, empty directory on your hard drive.

2 From your PC, select Start> All Programs> Accessories> Command Prompt. The Command Prompt window is displayed.

3 Use the CD command (cd\) to change to the directory containing the unzipped firmware files on your local hard drive.

4 Type ftp followed by the IP address of the unit. Press Enter.5 Login with the username and password. Default for both is admin.

• Type bin <Enter>

• Type hash <Enter>

• Type prompt <Enter>

• Type mput *.* <Enter>

6 Reset the gateway.7 To confirm the upgrade, initiate a telnet session with the unit and use the command

show –v.

The system software should reflect the upgraded firmware version.

C O N F I G U R A T I O N A T T H E S P H E R I C A L L

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . M A N A G E R

Once the Quintum gateway is configured, it can be checked-in to the Sphericall Manager. Trunk and/or Station information should also be configured at the Sphericall


. . .

. .Q U I N T U M VO I P G A T E W A Y SQuintum feature Codes for Analog Phones

Manager to set user address profiles, phone class of service profiles, trunk routing, etc.Performing the above documented configuration settings will assist you with general settings, please refer to Sphere documentation, specifically Book 2: Install & Configure for adding user and extension properties to the stations on the Quintum node.

Q U I N T U M F E A T U R E C O D E S F O R A N A L O G

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . P H O N E S

Analog phones on the Quintum network will have FSK available.Quintum has three preconfigured # codes to enable the features of hold and transfer calls. The default codes are:

Administrators note: The 46, 90 and 48 strings are configurable.

Note: While the rest of the Sphere system may have the use of STAR codes for feature management, any analog phones connected through the Quintum gateway are required to use the above feature # codes for features.

To conf igure codesOpen the advanced mode of the Configuration Manager under:

1 Circuit Configuration/Line Routing Configuration/Line Circuit Routing Group-phone, Call Services tab.

2 Check the following:

• Hold• Blind/Attended Transfer• Call Waiting

Action Feature Codes

Hold a call #46

Unhold a call #46

Blind Transfer #90 + extension

Attended Transfer Flash Hook+Extension

Once connected:Hangup to complete the transferORAnnounce the call and hangupOR#48 to cancel the transfer


Q U I N T U M VO I P G A T E W A Y SConfiguring Inbound routing

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C O N F I G U R I N G I N B O U N D R O U T I N G

To conf igure inbound rout ing of t runk or PSTN cal lsYou must configure each Trunk Circuit Routing Group with a Forced Routing Number:

1 Advanced Configuration2 Circuit Configuration3 Trunk Routing Configuration4 Trunk Circuit Routing Groups5 Trunk Circuit Routing Group-line6 Enter a value in the Forced Routing Number field on the Advanced tab.

• Recommended to enter XX from “DN Channel Map” (The Sphericall MGC will use default routing).

Note that only numeric entries are allowed. The Quintum gateway copies what you enter here into the username portion of the INVITE "To" header.If you want all calls to go to a specified extension, for example the auto attendant, enter that extension here.If you want to use the default routing defined in the Sphericall Administration utility, you must enter a nonexistent extension here. A nonexistent extension is required since if a valid extension is specified, the MGC will forward inbound calls to it. If a nonexistent extension is specified, the MGC will apply the default routing specified in the Sphericall Administration utility.

C O N F I G U R I N G S U R V I V A B L E I N B O U N D R O U T I N G ( S U R V I V A B L E M O D E L S )The Quintum gateway addresses the INVITE to the forced routing number and sends it to the MGC. If the MGC is down, the gateway will enter survivable mode and try to locate a User Agent Client that has registered the forced routing number locally. There are two ways to create a local registration for the forced routing number.A local registration may be created in the Quintum gateway's DN Channel Map and associated with one of the gateway's FXS ports. The registration is called "local" because the gateway knows about it but does not send a REGISTER request to the Sphericall Manager. This is important since the Sphericall Manager cannot accept registrations for the same Address Of Record from multiple devices. Note that this method only allows the forced routing number to be associated with an FXS port, not a SIP phone.

To create a local registrat ion through the DN Channel Map FXS port :1 Advanced Configuration2 VoIP Configuration3 DN Channel Map4 Click Add5 Select the channel that corresponds to the FXS port you wish calls to go to

Extension is assigned to the port in Sphericall Admin application.6 Enter the forced routing number in the DN edit control


. . .

. .Q U I N T U M VO I P G A T E W A Y SConfiguring Inbound routing

7 Uncheck the Register DN check box

A local registration may also be created by creating a static SIP route in the var_config file. This method allows the forced routing number to be associated with a SIP phone, but not an FXS port. The var_config.cfg file is a text file that is loaded onto the Quintum gateway via FTP.

To create a local registrat ion through the DN Channel Map FXO por t :1 Advanced Configuration2 VoIP Configuration3 DN Channel Map4 Click Add5 Select the channel that corresponds to the FXO port you wish calls to go to

Extension is assigned to the port in Sphericall Admin application.6 Enter the forced routing number in the DN edit control7 Uncheck the Register DN check box

A local registration may also be created by creating a static SIP route in the var_config file. This method allows the forced routing number to be associated with a SIP phone, but not an FXO port. The var_config.cfg file is a text file that is loaded onto the Quintum gateway via FTP.

To create a stat ic route , add the fo l lowing two l ines to your exist ing var_conf ig .cfg f i le ( i f any) :

1 SipRoute<x>DN <forced routing number>.2 SipRoute<x>IP <address>.

Where <x>=1-5, <forced routing number>=forced routing number and <address> is an IP address or host name.

To create a var_conf ig.cfg f i leIf you don't have an existing var_config.cfg file:

1 Open notepad (or any other text editor).2 Enter the above lines. 3 Save the file as var_config.cfg.

To load the f i le onto the Quintum gateway1 Connect to the gateway via FTP2 Put the var_config.cfg file into the /cfg folder. 3 After loading the file, the gateway must be rebooted to put the changes into effect.

Note: Note that this method has been verified to work with Polycom SIP phones. It may or may not work with other brands of SIP phones.

C O N F I G U R I N G A U T H E N T I C A T I O N O N T R U N K SIf the Quintum gateway is configured with both FXS and FXO ports, authentication must be enabled on the FXO ports to allow the Sphericall Manager to correctly identify which port inbound calls are coming from. For the FXS ports, the Sphericall Manager relies upon the extension in the SIP INVITE From header. For the FXO


Q U I N T U M VO I P G A T E W A Y SConfiguring Survivable Outbound Routing (Survivable models)

ports, the From header contains the telephone number of the calling party from the PSTN.

To enable authent icat ion for the FXO ports in the Quintum gateway:1 Advanced Configuration2 VoIP Configuration3 SIP Signaling Groups4 SIP Signaling Group-15 User Agent Tab6 Select the trunk UA and click the Edit button.7 Enter a password in the Primary Password field

In the Trunk Properties dialog, Authorization tab of the Sphericall Administrator application:a. Check the Use Authorization check boxb. Enter the user name into the Account field (this must be the same as the Primary User

from the Quintum gateway)c. Enter the password into the Password and Verify Password fields (this must be the

same as the Primary Password from the Quintum gateway)d. Enter Sphericall in the Realm fielde. Set the Type to MD5f. Set the Authorization Type to To Respond/To Challenge

C O N F I G U R I N G S U R V I V A B L E O U T B O U N D

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . R O U T I N G ( S U R V I V A B L E M O D E L S )

To enable survivable outbound rout ing1 The following line must be added to the var_config.cfg file:

SecureSSG 0The var_config.cfg file is a text file that is loaded onto the Quintum gateway via FTP. If you already have a var_config.cfg file, simply add the "SecureSSG 0" line to it. If you do not have a var_config.cfg file, open notepad (or any other text editor) and enter the "SecureSSG 0" line.

2 Save the file as var_config.cfg.

To load the f i le onto the Quintum gateway 1 Connect to the gateway via FTP and put the var_config.cfg file into the /cfg folder. 2 After loading the file, the gateway must be rebooted to put the changes into effect.


. . .

. .Q U I N T U M VO I P G A T E W A Y SUnderstanding the Environment

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . U N D E R S T A N D I N G T H E E N V I R O N M E N T



match is found, the Sphericall Manager moves to step 2.Note: this is the most common way stations are identified, but does not help for trunks since the FROM field contains the caller ID of the incoming call.











Q U I N T U M VO I P G A T E W A Y STesting


. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . T E S T I N G

Once installed and configured, Sphere recommends running user tests for connectivity as follows:• Dial from one phone to another on the Quintum gateway side.• Dial an extension on the main Sphere system from a Quintum side phone.• Dial an extension on the Quintum side from the main Sphere system.• Dial an call hand transfer it from the Sphere side to the Quintum side.• Stop the Sphericall Manager (only during off hours): dial an extension on the

Quintum side from the Quintum side.• Stop the Sphericall Manager (only during off hours): dial an outbound call from the

Quintum side.• Stop the Sphericall Manager (only during off hours): dial an inbound auto attendant

PSTN number to Quintum side, see if it rings the dedicated failover extension.• Test for dial plan.• Test for #-codes.


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .FTP, SNTP & DHCP NOTES A

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . F T P S E R V E R C O N F I G U R A T I O N

For the most current information on FTP server installations, see the Microsoft web site or the product documentation for your FTP service.

C O N F I G U R E F T P S E R V E RIP phones on a Sphere system must download XML configuration files from an FTP server on the network. These files are installed into the FTP root directory upon Sphericall application installation and are responsible for setting the functional parameters of the individual IP phones.

U P G R A D E SPlease refer to Sphere Release Notes for any changes to these procedures or upgrades.

M O V I N G T H E F T P S E R V I C EThe need may arise to move the FTP service from one server to another within your organization’s Sphericall system. If you need to move the FTP service, you must ensure that the Microsoft Windows FTP service is both running and enabled on the new server.Once the FTP service is running and enabled on the new server, you must:• Disable the Microsoft Windows FTP service on the old server via the Windows

Components Wizard.• Transfer all of the files and subdirectories located in the \ftproot directory on the old

server to the \ftproot directory on the new FTP server.• Change the name of the FTP server or IP address of the FTP server on the

Primary Sphericall Manager via the PBX Properties window.

C R E A T E A L O C A L U S E R A C C O U N T ( W I T H P A S S W O R D ) F O R T H E I P P H O N E S O N T H E F T P S E R V E R

FTP server—to create login and password for IP phones on FTP serverThis account must be created first:

1 Create a local user account on the FTP server with username PlcmSpIp OR Sayson.

2 Create password (respectively; case sensitive)PlcmSpIp OR Aastra480i.

Integrate Partner Technologies A-297

F T P, S N T P & D H C P N O T E SFTP Server Configuration

3 Deselect “User must change password at next logon.”4 Select “User cannot change password.”5 Select “Password never expires.”

I N S T A L L A T I O N O F F T P S E R V I C E

To conf igure the FTP service on a Microsof t Windows 2003 ServerFrom the Windows taskbar:

1 Click Start\Settings\Control Panel\Add or Remove Programs.2 Click Add or Remove Windows Components.

From the Windows Components Wizard window3 Select the Application Server.4 Select the Internet Information Services (IIS) check box.5 Click Details and ensure that file transfer protocol (FTP) is checked.6 DISABLE (uncheck) World Wide Web checkbox.7 Click OK.8 Click OK.9 Click Next.

Note: You may need to have the Windows Server CD-ROM available when installing the FTP service.

10 Click Finish.11 Close Add\Remove Programs window.12 Close Control Panel.

C O N F I G U R A T I O N O F F T P S E R V I C EFrom the Microsoft Windows taskbar:

13 Click Start\Programs\Administrative Tools\Internet Information Services Manager (IIS).14 Double-click the appropriate server instance in the Tree pane to view all services

running on that machine.15 Right-click Default FTP Site.

andSelect Properties.

From the Default FTP Site Properties window:In the Identification group box:

16 Type a description for the FTP site in the Description field.17 Select the appropriate IP address for the FTP site from the IP Address drop-down list

box.18 As appropriate for your organization’s network and its connection requirements,

configure the remaining fields and values in the FTP Site tab.19 Click Apply.20 Click the Security Accounts tab.

From the Default FTP Site Properties window21 Clear the Allow Anonymous Connections check box.

A-298 Integrate Partner Technologies

. . .

. .F T P, S N T P & D H C P N O T E SFTP Server Configuration

22 Click Apply.23 Click the Home Directory tab.

Figure A.1 Default FTP Site Properties window

24 Type or browse to the appropriate path to the FTP site directory in the Local Path field.

The appropriate path to the FTP site directory for a Sphere system is:<hard drive>:\program files\sphere\ftproot.This is the directory in which all XML configuration files will be stored within the Sphere system for use by the IP phones.If you are creating FTP service on the Active Directory or file server, for instance, then the configuration files must be copied to that particular computers directory and the Local Path field must be updated accordingly.

Note: If your organization is using an FTP Server for use with Sphericall IP phones, and it is not located on the Sphericall Manager, the FTP Server administrator will need to manually copy the FTProot directory to the FTP Server.

If you are installing Windows FTP server, the ftproot folder will be located by default at: c:\\inetpubs\ftproot. This default setting needs to be changed as follows:• For systems with the FTP Server on the Primary Sphericall Manager, the following

folder is required for the location of IP phpone resource files:<drive>:\\Program Files\Sphere\ftproot


25 Select the Read check box.26 Select the Write check box.27 Select the Log visits check box.

In the Directory Listing Style group box:


F T P, S N T P & D H C P N O T E SFTP Server Configuration

28 Click the appropriate radio button, UNIX or MS-DOS, to determine how the FTP server is to present its directories.

29 Click Apply.30 Click OK.31 Exit Internet Information Services Manager.

To ver i fy FTProot d i rectory secur i ty1 Locate the previously copied FTProot directory on the FTP server.2 Right click the FTProot directory.3 Select properties.4 Select Security tab.5 Click Add button.6 Type PlcmSpIp or Sayson (as appropriate).7 Click Check Names.8 Click OK.9 Highlight account.

10 In permissions area: select Full Control.11 Click OK.

To ver i fy FTP serv ice funct ional i ty1 Start an FTP client on a remote machine.2 Connect to the FTP service running on the Sphericall Manager.

From the Windows taskbar on the remote machine:a. Click Start\Run.b. Type cmd in the Open field.c. Type ftp <machine_name> where machine_name is the name of the Sphericall

Manager running the FTP service.d. Type the appropriate username and password for the FTP service account

(PlcmSpIp).

If the FTP service has been configured appropriately, a successful login message appears along with the ftp> command prompt.

3 Type ls to view a listing of the remote directory.4 Type quit to quit the cmd.exe application.5 Type exit to exit cmd.exe application.

S N T P S E R V E R C O N F I G U R A T I O NFor the most current information on SNTP server installations, see the Microsoft web site or the product documentation for your SNTP service.SNTP is a default service that runs on Window 2003 Active Directory server.


. . .

. .F T P, S N T P & D H C P N O T E SDHCP Configuration

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . D H C P C O N F I G U R A T I O N

For the most current information on DHCP server installations, see the Microsoft web site or the product documentation for your DHCP service.IP phones require the following scope options:

Table A.1 Scope Options

C O N F I G U R E D H C PIf IP addresses are to be dynamically assigned throughout the network (and, later, throughout the Sphericall system), a DHCP must be running on at least one server per subnet in all subnets to be configured for DHCP. Refer to the System Requirements documentation for more information concerning DHCP requirements on the network.

To insta l l DHCP service on a Microsoft 2003 ServerFrom the Windows taskbar:

1 Click Start\Settings\Control Panel.2 Double-click Add or Remove Programs.3 Click Add or Remove Windows Components to initiate the Windows Components

Wizard.

From the Windows Component Wizard window, In the Components area:

4 Select the Networking Services check box.5 Click Details...

In the Subcomponents of Networking Services area:6 Select the Dynamic Host Configuration Protocol (DHCP) check box.7 Click OK.

DHCP service has now been installed for the network.

To create a scope wi th a range of IP addresses in Windows 2003:From the Windows taskbar:

1 Click Start\Programs|Administrative Tools\DHCP.2 Highlight to select [server name.domain name].

From the DHCP Server - New Scope window:

Number Action Notes

002 Time Offset See end of DHCP section for table

003 Router

004 Time Server

066 Boot Server Host Name IP Address of FTP Server


F T P, S N T P & D H C P N O T E SDHCP Configuration

3 Click Action\New Scope.4 Click Next to the Welcome Wizard.

Figure A.2 DHCP - Scope Name

5 Enter the Name of the scope.6 Enter the Description of the scope.7 Click Next.


. . .


Figure A.3 DHCP - IP Address Range

8 Enter the Start IP address in the open field.9 Enter the End IP address in the open field.

10 Enter the Subnet mask in the open field.11 Click Next.

From the DHCP - Add Exclusions window:12 Enter Start IP address of addresses you want to exclude.13 Enter End IP address of addresses you want to exclude.14 Click Next.

From the DHCP - Lease Duration window:15 Enter the Lease Duration Days.16 Enter the Lease Duration Hours.17 Enter the Lease Duration Minutes.18 Click Next.

From the DHCP - Configure DHCP Options window:19 Select Yes to configure these options now.20 Click Next.

From the DHCP - Router (Default Gateway) window:21 Enter any Routers or Default Gateways.22 Click Add to add them to the lower window.23 Repeat for all Router IP addresses and Gateway IP addresses.24 Click Next.

From the DHCP - Domain Name and DNS Servers window:25 Enter the Domain Name and DNS Servers.



26 Click Next.

From the DHCP - WINS Servers window:27 Enter any WINS Server information.28 Click Next.

From the DHCP - Activate Scope window:29 Click Yes to activate this score now.30 Click Next.31 Click Finish.32 Click on Action\Authorize.

This action will change Red indication to Green when authorization is established.33 Click Finish.

DHCP - Scope Range is complete.

To conf igure the Microsoft Windows 2003 Server DHCP scope for IP Phones onlyFrom the Windows taskbar:

1 Click Start\Programs\Administrative Tools\DHCP.2 Highlight the appropriate scope in the DHCP Tree.

Figure A.4 DHCP window

3 Double-click Scope Options in the Contents of Scope pane.4 Click Action\Configure Options.


. . .


Figure A.5 Scope Options window

5 Add 003 option for Router.6 Enter IP Address of router.7 Select the 066 Boot Server Host Name check box.

Adding the 066 Boot Server Host Name option will allow the DHCP server to pass the FTP server address to the IP phone.

8 Type the IP address of the FTP server in the String Value field.9 Click Apply.

10 Select the 004 Time Server check box.

Adding the 004 Time Server option will allow the DHCP server to pass the SNTP server address to the IP phone.




11 Type the name of the SNTP server in the Server name field.12 Type the IP address of the SNTP server in the IP address field.13 Click Apply.14 Select the 002 Time Offset check box.

Adding the 002 Time Offset option will allow the DHCP server to pass the GMT offset to the phone.


15 Type the appropriate unsigned integer as the time offset value in the Long field.

The time offset value is the number of seconds difference from Greenwich Mean Time (GMT), a signed integer. To calculate the time offset value:


. . .


• Calculate the number of seconds of offset for the location of your organization’s Sphericall system.• For example, the GMT offset for Chicago, Illinois (Stardard Time), is -6 hours.

-6 hours * 60 minutes/hour * 60 seconds/minute = -21600 seconds

Note: If the value is negative, enter the (-) sign before converting the offset to a hexadecimal value.

• Convert the number of seconds of offset to a hexadecimal value.• If using a scientific calculator, make certain that the Dword value is selected

when converting the hexadecimal value back to a decimal value.• -21600 as a hexadecimal value = FFFFABA0

• Convert the number of seconds of offset back to a decimal value. The resulting value, in seconds, is the unsigned integer.• FFFFABA0 as a decimal value = 4294945696

Note: This calculation is based upon Standard Time. The IP phone automatically calculates Daylight Savings Time.

T I M E O F F S E T V A L U E SRefer to the Time Offset table or www.greenwichmeantime.com for the appropriate unsigned integer.

Table A.2 US Standard Time Offsets (GMT)


The scope options appear in the Contents of Scope pane.

Time Zone Major Cities GMT Offset Unsigned Integer

Pacific Seattle, WASan Francisco, CALos Angeles, CASan Diego, CA

-8 4294938496

Mountain Calgary, CanadaSalt Lake City, UTAlbuquerque, NMBoise, ID

-7 4294942096

Central Chicago, ILNew Orleans, LAHouston, TXMexico City, Mexico

-6 4294945696

Eastern New York, NYToronto, CanadaWashington D.C.Miami, FL

-5 4294949296



Figure A.8 DHCP window

18 Exit DHCP Manager.


. . . .

. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .INDEX

Numerics480i

time server 2-26time stamp 2-26

480i IP address assignment 2-26480i IP Phone 2-25

AAastra

SIP 9112i 2-7, 2-22SIP 9133i 2-7, 2-22

Aastra 480i IP Phone InstallationStatic IP Addressing 2-25, 2-26

Aastra 480i login and password 2-7, 2-22

Aastra 9133i SIP Phone 3-89Aastra password 2-23Aastra SIP Phone 3-52account on the domain 2-8, A-297A-law 18-289AudioCodes access and setup 14-203,

14-212AudioCodes MP11X overview 14-203Authorization Header 3-99, 17-277, 18-

296Auto Attendant Group Number 9-144

BBogen Transformer 7-124Boot Screen Messages 2-15Boot Sequence Errors 2-15BranchHub Manual 1-3Busy Lamp Field 2-29Busy Lamp Field States 2-29Busy-Lamp-Field 2-28

Ccable 2-25Call loggers 9-147Call logging 7-128, 8-136, 10-160Call progress tones 9-149

amplitude levels 9-150busy 9-149disconnect 9-149external dial tone 9-149flashtime 9-150internal dial tone 9-149reorder 9-149ringback 9-149

transmission and noise measuring 9-150

Call records 10-169caller ID 3-99, 17-276, 18-295Caller ID name 18-288CallNOW implemetation 1-2CallStateConstants 16-227Carrier traffic studies 9-148CBeyond 17-252Changing the IP Phone User Account 2-

8, 2-23Changing the SNMP write

community 12-186COHub Manual 1-3COM API

Getting Started 16-228Com API 16-228Common message coding 10-170

blocked message waiting 10-171direct call message 10-170forward all calls 10-170forward on no answer 10-170forward on unknown reason 10-171invalid message waiting 10-171message waiting indicator

messages 10-171configure a voice mail station 10-159configure the VM and AA 10-162configure the voice mail line setting 10-

161Constants 16-227

DDaily Management 1-2DHCP 2-25, A-301

configure service A-304install service A-301

DHCP Configuration A-300, A-301DHCP scope for IP Phones A-304Diagnostic Star Codes 3-98DID maps

SIP 3-99, 17-276, 18-295DNS domain name 17-243DNS Test 3-84DNS test 17-244Drop loop current 7-128, 8-136Dynamic Host Configuration Protocol

(DHCP) A-301

EEmergency Service Installations 1-2Erlang model 9-146

Blocking 9-147Busy hour call multiplier 9-147Busy hour traffic 9-146Estimating busy hour traffic 9-148IVR port 9-147Recall factor 9-146

Error messages 2-15Eutectics IPP520 5-109Eutectics IPP520 Phone

configuration 5-109functionality 5-110installation 5-109overview 5-109

Expansion slot types 9-142Export Phone Distribution Map 3-86Extensions

assigning to a music-on-hold station 7-128

assigning to stations 10-161configuring for voice mail 10-158

FFax Relay 18-289File Upgrade 2-17Flow control settings 10-154FP *96 9-150, 10-161FTP Preparation 2-9FTP server 2-13, A-297

configuring A-297installation A-298moving the service A-297upgrades A-297

FTP server address 2-6, 2-20FTP Server Configuration Topics A-297FTP Service A-298FTP service functionality A-300

GG711

caveat for Global Crossing 17-241Generic SIP Trunk 17-244Getting Started

Sphericall COM Type Libraries 16-228

Global Crossing 17-241Grandstream GPX2000 SIP phone 3-66

Integrate Partner Technologies Index I-1

I N D E X

Grandstream GXP-2000speed dial configuration 3-63, 3-71time zone configuration 3-63, 3-71upgrades 3-64, 3-72

Grandstream GXP-2000 Phone 3-58, 3-66

Group Policy for Windows Installer login requirements 2-6, 2-21

HHangup detection 9-150Hard disk storage 9-142Hardware Manuals

BranchHub 1-3COHub 1-3MeetingHub 1-3MG Command Line Reference 1-3PhoneHub 1-3

Help for programming ideas 16-227Help with the Visual Basic 16-227hostname 3-99, 17-276, 18-295hunt order

auto attendant extension 10-162voice mail extension 10-162

IInstall & Configure the Sphere

System 1-2Install Sphericall Voice Mail 1-2Integrate Partner Technologies 1-2Integration notes 9-149IP address 2-6, 2-20, 3-99, 17-276, 18-

295IP Addresses

see DHCP scope for IP Phones A-304

IP Phonepreviously on network 2-9

IP phone initialization 2-9IP Phone Installation

Dynamic IP Addressing 2-12IP Phone Upgrades 2-19, 2-39IVR port 9-147

LLink Wireless Telephone 15-223Load Balancing for Sayson IP Phone 2-

11, 2-24local user account 2-23Logins & Permissions

Group Policy for Windows Installer 2-6, 2-21

Polycom IP phone 2-7, 2-22

MMAC address 17-246Manage, Monitor & Support

Sphericall 1-2Media servers 10-164

media serverscreating 10-164

MeetingHub Manual 1-3Memory 9-142Message Server

voice port requirements 9-146Message Storage Retrieval Identifier 9-

144Message stores

MSRLineID 10-165MG CLI Reference Manual 1-3MGC-to-IP phone connections 2-12, 2-

24MGC-to-MG Connection Control

configuring 2-12, 2-24MGCP phones 2-12, 2-24

MIBsinstallation on the Sphericall

Manager 12-189installing 12-189sphere-reg.mib 12-189sphere-tc.mib 12-189table of channel description

objects 12-190table of e2prom objects 12-190table of system objects 12-190table of telephony board description

objects 12-190table of version objects 12-190table of voltage/temperature

objects 12-191Microsoft Windows Server 2-6, 2-20MLPP 1-2MOH and Upgrades 7-130MOH behavior during MGC timeout 7-

130MoH for fallback 7-125MoH Options 7-125MOH Upgrade path 7-129, 7-130Moves, Adds & Changes to the

System 1-2MSRLineID 9-144, 10-165Mu-law 18-289Music-on-Hold 7-123

Bogen Transformer 7-124device sources 7-124interface device 7-124moh1 7-128music source 7-123re-broadcasting rights 7-124

MWI light flashes 10-168

NNetwork

DHCP A-301Normal Boot Screens 2-15normal messages during boot

sequence 2-15Numbering plan

PBX properties 2-11

OOID index 12-191OIDs 12-186Outbound Proxy 3-99, 17-276, 18-295

PPaging lines 8-133

adding a line 8-134configuration 8-134installation and integration 8-136profile 8-136

Paging systeminstallation test 8-138recommended products 8-133zone configurations 8-136

password 2-8, A-297PBX properties

configuring 2-11PBX statistics 9-147Phone failover 2-19Phone Setup 2-25PhoneHub Manual 1-3phones 5-109

Aastra 480i 2-20PING 2-16Plan & Prepare the Sphere System 1-2Planning 2-20Plantronics 5-112

CS50-USB 5-112platform hardware requirements 9-142Polycom IP phone account

permissions 2-7, 2-22Polycom IP501

Polycom IP601Polycom IP650

Polycom IP4000 3-

74Polycom SIP Phone 3-74Preparing 2-21PreviousState properties 16-227Primary addresses

group extensions and voice mail extensions 7-129

Processor 9-142

QQuick Reference Guides 1-2quick start guide 2-12Quintum 18-279

Caller ID on the station ports 18-288Hook Flash 18-288

Rregister SIP 3-98, 17-276, 18-295Registration Type 17-238Release Notes & Upgrade

I-2 Index Integrate Partner Technologies

. . .

. .I N D E X

Procedures 1-2report

phone distribution map 3-86Reports, Statistics and Tools 1-2Reset 2-33Restarting the Ip Phone 2-16Restarting the phone 2-16RS-232 serial connection 9-139, 9-141

SSample Applications for Sphericall Type

Libraries 16-228SCCall 16-227SCCalls 16-227scope

DHCP Scope Range A-304Scope Options A-301SCPhone 16-227Screen Messages 2-15Service Provider 3-99, 17-276, 18-295Service Provider Domain 3-99, 17-246,

17-276, 18-295Silence Suppression 18-287SIP 3-45, 3-98, 17-234, 17-254, 17-276,

18-295Account 17-241, 17-246Account field 3-98, 17-276, 18-295Authentication 3-42Authorization 17-248Call center 17-239Challenge authentication 17-249Contact Domain 17-243credentials 3-99, 17-277, 18-296Description of trunk 17-241DNS Name 3-43Inbound Registration 17-239INVITE 17-249IP Address 3-43Outbound Proxy 3-43, 17-242Outbound Registration 17-239Planning 17-240Port 17-246Port information 17-242Primary MGC 17-243Proxy 3-43REGISTER 17-249REGISTER requests 17-243Registrar 3-42Registration Type 17-246Registration type 17-242Requirement 3-45, 17-234, 17-254Respond authentication 17-249Secondary MGC 17-243Service Provider 17-239service provider 17-246Service Provider Domain 17-242SIP Domain 3-43SIP Phone MAC Address 3-42Softtrunk Total Capacity 17-247, 17-

272softtrunks 17-238System Initialization Settings 17-242Tie line solution 17-253Tie-lines 17-239Unknown authentication 17-249User Agents 3-45, 17-234, 17-254User Agents Upgrade 3-48, 17-237User Name/User ID 3-42Userinfo 3-98, 17-276, 18-295

SIP Configuration Note 3-42SIP connection overview 3-98, 17-276,

18-295SIP Failover 3-49SIP phone

restart 3-65, 3-73upgrade 3-65, 3-73

SIP phonesrestart 3-57

SIP phones upgradefirmware 3-57

SIP Registration 17-238SIP Server 3-42SIP Terminal location logic 3-98, 17-276,

18-295SIP tie line to Call center 17-266SIP trunk configuration

Global Crossing Requirements for SIP 17-241

SIP Trunk MAC Address 17-244SIP Trunking Configuration 17-252SIP Trunking Tie Line 17-253SIP Trunking to SIP Service

Provider 17-239SIP trunks

DID maps 3-99, 17-276, 18-295SMDI 9-139, 10-157, 11-173

access hangup detection 9-150Auto Attendant Group Number 9-144cables 9-141cabling requirements 9-141Call Center 9-139call processing 9-144call records 10-169common message coding 10-170enabling SMDI process 10-156Fax Center 9-139hardware requirements 9-140integration notes 9-149Message Storage Retrieval

Identifier 9-144messages from the voice mail

system 9-140messages to the voice mail system 9-

140MSRLineID 9-144overview 9-139requirements 9-140RS-232 compliant serial connector 9-

141

RS-232 serial connection 9-139setting flow control settings 10-154setup and operation 9-143supported features 9-140system planning 9-146transfer initiation and release 9-150Voice Messaging 9-139voice messaging platform

requirements 9-141SMDI integration 9-139SMDI platform hardware

requirements 9-142SMDI process 10-156SNMP

agent 12-186definition 12-185manager 12-185manager and agent setup 12-187monitoring process 12-187OID index 12-191OIDs 12-186overview 12-185requirements 12-186Sphericall MIBs 12-189Sphericall traps 12-188trap function 12-187trap information 12-188traps 12-186

SNMP access 12-186SNMP Agent 12-186SNTP server 2-13SNTP server address 2-6, 2-20SNTP Server Configuration A-300SNTP time offset 2-13softtrunk 17-238SoundPoint MGCP Phone Installation

Static IP Addressing 2-13Spanish 1-2Spectralink 15-223Sphere

preparing the system 10-154Sphere Document Index 1-2Sphere Star Codes 1-2Sphere System Requirements 1-2, 2-6,

2-20Sphericall COM API 16-228Sphericall Desktop Users Manual 1-2Sphericall Manager

Primary vs Secondary 10-156Sphericall Manager Commissioning

wizardchoosing a Pirmary or Secondary

Sphericall Manager 10-156choosing voice mail 10-157

Sphericall Manager Configuration utilitySMDI 10-157

Sphericall Phone Type Library 16-227Star Codes

administrative 3-97static IP address 2-13, 2-26

Integrate Partner Technologies Index I-3

I N D E X

Static IP AddressingAastra 480i phones 2-25, 2-26Polycom SoundPoint phones 2-13

Station settingscall logging 7-128, 8-136, 10-160drop loop current 7-128, 8-136extension assignment 10-161properties 8-135, 10-160stutter dial tone 7-128, 8-136

Stutter dial tone 7-128, 8-136Support

Diagnostic Star Codes 3-98Symptoms 11-181Sync IP Phone Files 2-16

TT.38 Fax Relay 18-289test coverage to voice mail and to test

call disconnect 10-168test direct calls into voice mail by the

operator 10-167test direct subscriber access of

mailboxes 10-167test the setting and cancelling of

MWI 10-167Third-party permissions required 2-6, 2-

21Time offset values A-307To change the SNMP access write

community name 12-186Traps 12-186

critical information 12-188function 12-187Sphericall traps 12-188

Troubleshooting 2-30error messages 2-15phone failover 2-19restarting the phone 2-16

troubleshootingSIP 3-98, 17-276, 18-295SIP stations 3-99, 17-276, 18-295SIP trunks 3-99, 17-276, 18-295

Troubleshooting informationHyperTerminal monitoring 11-173,

11-177Trunk

Inward Routing 17-249

UUpgrades 2-19, 2-39user account 2-8, 2-23, A-297User Agent 3-99, 17-276, 18-295user agent 3-46, 17-235User Agents 18-286UTStarcom 2-8, 2-22UTStarcom F1000

create a DNS Record 3-49, 3-50firmware updates 3-94Sphericall Voice Mail settings 3-94

user settings 3-93web configuration 3-91WiFi and network settings 3-90wireless access point settings 3-93

UTStarcom F1000Goverview 3-89

UTStarcom F3000overview 3-89

Vverify Monitor privileges for the SMDI

instance 10-163view all the SNMP community

names 12-187VLAN use for IP phone 2-9Voice 11-180, 11-181voice mail extensions

configuring 10-158voice message deletion 10-168voice message storage 9-142Voice messaging

configuring 10-164media servers 10-164Message Waiting Indicators 10-169platform requirements 9-141restarts and refreshes 10-169system testing 10-166

voice part requirements 9-146volume 18-288volume settings 5-108

WWeb Client 2-26

XXML file 2-23

I-4 Index Integrate Partner Technologies

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