setu ata2s v1 system manual

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SETU ATA2S

System Manual

MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.

Corporate Office: Factory:

394-GIDC, Makarpura, 39-GIDC, Waghodia-391 760

Vadodara-390 010, India Dist. Vadodara, India

Tel. : +91 265 2630555 Tel. : +91 2668 263172/73

Fax : +91 265 2636598 Fax : +91 2668 262631

E-mail: [email protected] E-mail: [email protected]

URL: www.MatrixTeleSol.com

Version 1

March 2007



Documentation Information

This is a general documentation and it covers many models with

different specifications. A particular product may not support all the

features and facilities described in the documentation.

Matrix Telecom reserves the right to revise information in this

publication for any reason without prior notice. Information in this

documentation may change from time to time. Matrix Telecom makes

no warranties with respect to this documentation and disclaims any

implied warranties. While every precaution has been taken in

preparation of this system manual, Matrix Telecom assumes no

responsibility for errors or omissions. Neither is any liability assumedfor damages resulting from the use of the information contained

herein.

Matrix Telecom reserves the right without prior notice to make

changes in design or components of the equipment as engineering

and manufacturing may warrant.

Neither Matrix Telecom nor its affiliates shall be liable to the purchaser

of this product or third parties for damages, losses, costs or expensesincurred by purchaser or third parties as a result of: accident, misuse

or abuse of this product or unauthorized modifications, repairs or

alterations to this product or failure to strictly comply with Matrix

Telecom’s operating and maintenance instructions.

All rights reserved. No part of this system manual may be copied or

reproduced in any form or by any means without the prior written

consent of Matrix Telecom.



Contents

Section 1: Introduction ................................................................. 5 Welcome ........................................................................................ 7

Packing List .................................................................................... 9

Warranty Statement ..................................................................... 10

Protecting the System .................................................................. 13

Introducing the System ................................................................ 15

Getting Started ............................................................................. 18

Section 2: Features and Facilities ............................................ 27 Auto Configuration ....................................................................... 29

Dial Plan ....................................................................................... 32 FXS Port ....................................................................................... 34

Jeeves .......................................................................................... 44

Password...................................................................................... 54

Peer-to-Peer Calling .................................................................... 55

Phone Book.................................................................................. 58

Resetting the ATA......................................................................... 59

SIP Accounts ................................................................................ 60

Software Upgrade ........................................................................ 65

Status ........................................................................................... 66 Supplementary Services .............................................................. 70

System Name............................................................................... 81

Time Settings ............................................................................... 82

VoIP Basics .................................................................................. 83

WAN Explanation ......................................................................... 89

WAN Port ..................................................................................... 93

Section 3: Appendices ............................................................... 95 Appendix A: Troubleshooting ....................................................... 97

Appendix B: Applications ............................................................. 99

Appendix C: Product Specifications .......................................... 101

Appendix D: Glossary ................................................................ 104

Appendix E: Features at a Glance .............................................. 110

Appendix F: System Commands ................................................ 111



Index ...........................................................................................112

Notes ..........................................................................................115

Programming Register .............................................................117



Section 1: Introduction



Matrix

SETU ATA2S System Manual 7

Welcome

Welcome to the world of telecom solutions from Matrix and thanks for

purchasing a Matrix product.

We want you to get the maximum performance from our product. If

you run into technical difficulties, we are here to help. But please

consult this system manual first.

If you still can’t find the answer, gather all the information or questions

that apply to your problem and with the product close to you, call your

dealer. Matrix dealers are trained and ready to give you the support

you need to get the most from your Matrix product. In fact, mostproblems reported are minor and can be easily solved over the

phone.

In addition, technical consultation is available from Matrix engineers

every business day. We are always ready to give advice on

application requirements or specific information on installation and

operation of our products.

The system manual is divided in following sections:Section 1: Introduction

Section 2: Features and Facilities

Section 3: Appendices

We suggest the first time users to read this system manual in the

following sequence and then remaining chapters:

• Section 1

• Section 2 (In hierarchy given below)

• Getting Started 18• Jeeves 44

• Resetting the ATA 59

• Peer-to-Peer Calling 55

• Supplementary Services 70

• Status 66

• Time Settings 82

• FXS Port 34



Matrix

SETU ATA2S System Manual8

• WAN Port 93

• VoIP Basics 83

• SIP Accounts 60

• Dial Plan 32

• Phone Book 58• Software Upgrade 65

• Auto Configuration 29

• System Name 81

VoIP ATA ‘Phone Adaptor’ and ‘SETU ATA2S’ are used as

synonymous in this manual.



Matrix


Packing List

The ideal sales package for SETU ATA2S is as mentioned below:

Sr. Accessories Qty.01 SETU ATA2S 1

02 System Manual 1

03 Adaptor 12V, 1.25Amp. 1

04 Mounting Screw 30/7 2

05 Screw Grip 2

06 Warranty Card set 1

07 Support Card 1

08 RJ11 Cable 2

09 RJ45 Cable 1

10 Mounting Template 1

11 SETU ATA2S CD 1

• Please make sure that these components are present.

• In case of short supply or damage detection, contact the sourcefrom where you have purchased the system.

=X=X=



Matrix


Warranty Statement

Matrix Telecom Pvt. Ltd. (Matrix) warrants to its consumer purchaser

any of its products to be free of defects in material, workmanship andperformance for a period of 15 months from date of manufacturing or

12 months from the date of installation which ever is earlier.

During this warranty period, Matrix will at its option, repair or replace

the product at no additional charge if the product is found to have

manufacturing defect. Any replacement product or part/s may be

furnished on an exchange basis, which shall be new or like-new,

provided that it has functionality at least equal to that of the product,

being replaced. All replacement parts and products will be the

property of Matrix. Parts repaired or replaced will be under warrantythroughout the remainder of the original warranty period only.

This limited warranty does not apply to:

1. Products that have been subjected to abuse, accident, natural

disaster, misuse, modification, tampering, faulty installation, lack

of reasonable care, repair or service in any way that is not

contemplated in the documentation for the product, or if the model

or serial number has been altered, tampered with, defaced or

removed.

2. Products which have been damaged by lightning storms, water or

power surges or which have been neglected, altered, used for a

purpose other than the one for which they were manufactured,

repaired by customer or any party without Matrix’s written

authorization or used in any manner inconsistent with Matrix’s

instructions.

3. Products received improperly packed or physically damaged.

4. Products damaged due to operation of product outside theproducts’ specifications or use without designated protections.

Warranty valid only if:

• Primary protection on all the ports provided.

• Mains supply is within limit and protected.

• Environment conditions are maintained as per the product

specifications.



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Warranty Card:

• When the product is installed, please return the warranty card with:

• Date, signature and stamp of the customer.

• Date, signature and stamp of the channel partner.

• Matrix assumes that the customer agrees with the warranty termseven when the warranty card is not signed and returned as

suggested.

The Purchaser shall have to bear shipping charges for sending

product to Matrix for testing/rectification. The product shall be shipped

to the Purchaser at no-charge if the material is found to be under

warranty. The Purchaser shall have to either insure the product or

assume liability for loss or damage during transit.

Matrix reserves the right to waive off or make any changes in its

warranty policy without giving any notice.

If Matrix is unable to repair or replace, as applicable, a defective

product which is covered by Matrix warranty, Matrix shall, within a

reasonable time after being notified of the defect, refund the purchase

price of the product provided the consumer/purchaser returns the

product to Matrix.

In no event will Matrix be liable for any damages including lost profits,

lost business, lost savings, downtime or delay, labor, repair or material

cost, injury to person, property or other incidental or consequential

damages arising out of use of or inability to use such product, even if

Matrix has been advised of the possibility of such damages or losses

or for any claim by any other party.

Except for the obligations specifically set forth in this Warranty Policy

Statement, in no event shall Matrix be liable for any direct, indirect,special, incidental or consequential damages whether based on

contract or any other legal theory and where advised of the possibility

of such damages.

Neither Matrix nor any of its distributors, dealers or sub-dealers

makes any other warranty of any kind, whether expressed or implied,

with respect to Matrix products. Matrix and its distributors, dealers or



Matrix


sub-dealers specifically disclaim the implied warranties of

merchantability and fitness for a particular purpose.

This warranty is not transferable and applies only to the original

consumer purchaser of the Product. Warranty shall be void if thewarranty card is not completed and registered with Matrix within 30

days of installation.

All legal course of action subjected to Vadodara (Gujarat, India)

Jurisdiction only.



Matrix


Protecting the System

Installation Precautions:

• Do not installed In direct sunlight and excessive cold or humid

places.• Do not installed at places at where sulfuric gases produced in

areas where there are thermal springs, etc. because it may

damage the equipment or contacts.

• Do not installed at places in which shocks or vibrations are

frequent or strong.

• Do not installed at dusty places or places where water or oil may

come into contact with the system.

• Do not obstruct area around the system (for reasons of maintenance and inspection be especially careful to allow space

for cooling above and at the sides of the system).

Important Safety Instructions:

When using your telephone equipment, basic safety precautions

should always be followed to reduce the risk of fire, electric shock and

injury to persons, including the following:

• Read and understand all instructions.

• Unplug this product from the wall outlet before cleaning. Do notuse liquid cleaners or aerosol cleaners. Use a dry and soft cloth

for cleaning.

• Do not use this product near water, for example, near a bathtub,

wash bowl, kitchen sink, or laundry tub, in a wet basement, or

near a swimming pool.

• Do not open the system in power ON condition.

• Do not place this product on an unstable cart, stand or table. The

product may fall, causing serious damage to the product.

• WAN and Lifeline interfacing cables should not touch theexposed power line cable.

• This product should be operated with proper supply voltage. If you

are not sure about supply voltage, contact authorized dealer. It is

advisable to give proper, stabilized power.

• Do not allow anything to rest on the power cord of product or AC-

DC Adapter. Do not place this product where the cord will be

misused by people walking on it.



Matrix


• Do not overload wall outlets and extension cords as this can

result in the risk of fire or electric shock.

• To reduce the risk of electric shock, do not disassemble this

product. Take it to a qualified serviceman when some service or

repair work is required. Opening or removing covers may exposeyou to dangerous voltages or other risks. Incorrect reassembly

can cause electric shock when the appliance is subsequently

used.

• Unplug this product from the wall outlet and refer servicing to

qualified service personnel under the following conditions:

a) When the power supply cord or plug is damaged or frayed.

b) If liquid has been spilled into the product.

c) If the product has been exposed to rain or water.

d) If the product does not operate normally by following the

operating instructions. Adjust only those controls, which are

covered by the operating instructions because improper

adjustment of other controls may result in damage and will

often require extensive work by a qualified technician to

restore the product to normal operation.

e) If the product has been dropped or the cabinet has been

damaged.

f) If the product exhibits a distinct change in performance.• Do not use the telephone of the product to report a gas leak in

the vicinity of the leak.

=X=X=



Matrix


Introducing the System

• Matrix offers the “Voice over IP-Analog Telephone Adaptor”, VoIP

ATA or Matrix Phone Adaptor (SETU ATA2S) which is a SIP basedVoIP product.

• Sending voice signals over the Internet is called Voice over IP or

VoIP. Session Initiated Protocol, SIP is an internationally

recognized standard for implementing VoIP.

• VoIP-ATA is an Analog Telephone Adapter that allows a user to

make voice call over IP network using a conventional telephone

instrument.

• VoIP-ATA has one IP port, for WAN connection.

• WAN port is used to connect VoIP-ATA to the Internet usingbroadband modem, cable modem, router, etc. The VoIP calls are

made through WAN port.

• VoIP-ATA has two FXS ports to which conventional telephone

instruments (SLTs) can be connected.

• Call routing mechanisms for IC calls from IP to FXS and OG calls

from FXS to IP are included in VoIP-ATA.

• The IC call routing mechanism helps the user to route the calls to

either or both the FXS ports whereas the OG call routing

mechanism helps the user to route the call through desired SIP

Account which is helpful when services from two different ITSPs

have been taken, each having cost benefit in some terms over the

other.

• When a telephone number (A E.164 number or a SIP number

without domain name. i.e. the prefix of the SIP telephone number

e.g. Suppose a user with SIP number SIP: [email protected] is to

be called then the caller should dial 12345) is dialed by the user,

the VoIP-ATA converts this into an IP call using SIP protocol andmakes a call to the callee.

• Using Phone book, the user can also dial complete SIP number by

just dialing feature code and corresponding index number.

• User can also use Auto Configuration feature if he wants to avail

facility of configuration of ATA by ITSP from central location.

• Refer ‘Feature List’ which includes important features supported by

VoIP-ATA.



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Feature List:

• Configuration of the ATA through Web-pages

• Auto Configuration

• Resetting the ATA

• User Access with Password• Synchronization with Time Server

• Supplementary Services with T.38 Fax capability

• WAN Port IP Address programming

• Use of two SIP accounts, simultaneously

• Routing of Dialed Number as per Dial Plan

• Speed dialing using Phone Book feature

• Peer-to-Peer Calling

• Status page display of main parameters

• By default, Static IP Address of ATA2S is : 192.168.001.151.

• Before proceeding further, please ensure that you have an

Internet access and a SIP Account already set up.

SETU ATA2S Photograph:



Matrix


Relevant Topics:

1. Glossary 103

2. VoIP Basics 83

=X=X=



Matrix


Getting Started

Overview

The Matrix ATA allows you to use the analog telephone to make

phone calls over the Internet. The ATA connects to the Internet

through a DSL Modem or Router as shown below:

Internet

Network

FXS2 WANFXS1

This chapter gets the ATA up and running quickly. The details which

we have skipped to make this brief can be found elsewhere in the

manual. It is divided into four sections:

1. Getting to know the phone adapter.

2. Instruction for connecting the phone adapter.

3. Basic steps for configuration.4. Making phone calls.

Getting to know the phone adapter

The Matrix ATA’s ports are located on side panel and LEDs are on

front panel. Both are explained below:

• Connection of Right Side panel ports.

• Left Side panel LEDs.



Matrix


Right Side Panel Ports

Following picture shows how the ports are connected to various

interfaces:

FXS2 FXS1ETHERNET

12VDC 1A(Max)

Telephone Cable Ethernet CablePower Adaptor

(2) (3) (4)(1)Ethernet Switch

No. Port Name Connector Description

1 FXS 1 RJ 11 To connect analog telephone/fax.

2 FXS 2 RJ 11 To connect analog telephone/fax.

3 Power DC JackPower Adaptor, 12Volt 1.25Amp.

DC.

4 EthernetRJ 45 with

two LED’s

This is a WAN Port. Connection to

cable modem/ ADSL modem or

router or LAN switch. Ethernet

activity LED (Green) is glowing.

PC can be connected from

Ethernet switch to access the

ATA.



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Left Panel LEDs

The ATA has 5 LEDs; First for PWR, Second for FXS 1, Third for

FXS2, Fourth for Registration Status of SIP1 and Fifth for SIP2. These

LEDs are used to signify various events.

LED Label LED Indication

PWR Green

FXS1 Red

FXS2 Red

SIP1 Red

SIP2 Red

The reset sequence:• At power ON, ‘PWR’ LED is glowing continuously, whereas all

other LEDs glow for few seconds time which depends on the

timing cycle from boot loader program to init the sequence.

• On successful completion of initialization cycle, each LED is

glowing as per the normal conditions.

Normal conditions:

The LED indications should be as per following table:

LED Status Meaning

ONIC Call/OG Call-Speech and Off-

Hook.

Fast BlinkingError condition during IC/OG

call/Busy Condition.

Slow BlinkingIC Call-Ring Event/OG Call-Number

being dialed.

OFF FXS port On-hook.

ONIC Call/OG Call-Speech and Off-Hook.

Fast BlinkingError condition during IC/OG

call/Busy Condition.

Slow BlinkingIC Call-Ring Event/OG Call-Number

being dialed.

OFF FXS port On-hook.

FXS1

FXS2



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ONSIP1 Active and ATA registered

with the SIP server.

Fast BlinkingSIP1 Active but ATA not registered

with the SIP server.Slow Blinking SIP1 Inactive .

ONSIP2 Active and ATA registered


Fast BlinkingSIP2 Active but ATA not registered


Slow Blinking SIP2 Inactive.

SIP1

SIP2

Where,

OFF : Continuously OFF

ON : Continuously ON

Slow Blinking : 500ms ON, 500ms OFF

Fast Blinking : 200ms ON, 200ms OFF

IC, OG : Incoming, Outgoing

Instruction for connecting the phone adapter

This section describes the instructions on, how to connect the Matrix

ATA to internet network and telephones.

Unpack the box. Get satisfied with the contents and the condition of

the parts. Refer to ‘Packing List’. If parts are OK, proceed with

connections as shown below:

• Mount your ATA in a safe and convenient location where cables

for your network and phone system are accessible. For this,

mechanical drawing can be used as a reference, which is

provided end of the chapter.

• Connect a standard analog telephone instrument to FXS1 andFXS2 ports, using telephone cables provided.

• Insert one end of the Ethernet cable into the WAN port of ATA and

connect the other end of the Ethernet cable to Ethernet Switch

which is connected to router or DSL Modem.

• Insert one end of the Ethernet cable into the Ethernet Switch and

connect the other end of the Ethernet cable to the PC.

• Check the voltage from the power point from where the supply is

to be accessed. It should be between 90-265VAC, 47-63Hz.



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Insert the DC output terminal of power adapter into the ‘Power’

jack of the ATA and connect the 230VAC pins of the power

adapter to a wall outlet for 230VAC.

• When you power ON the ATA, observe that all LED’s are ON for

few seconds and then SIP1, SIP2 LED’s are flashing.

Basic Steps for Configuration

WAN Port:

• Default WAN IP Address for ATA is 192.168.001.151.

ABSOLUTELY NEEDED !

Use WAN IP Information from your LAN Admin. or ISP

• IP Address (must)

• IP Mask (must)• Gateway

• DNS IP Address (must)

• DNS Domain Name

• Note that subnet of WAN and Internet (Router) are same.

• Read the IP of PC, by clicking on Network Neighborhood→

Properties → TCP/IP Connection → Properties for Windows98.

• Choose the unique IP address of the PC such that its subnet issame as subnet of IP address of ATA, provided by ISP.

PLEASE CHECK !

The subnet of your PC IP Address must be

same as of ATA.

• For example, if IP address of ATA is 192.168.5.10, then PC IP

Address port should be programmed as 192.168.5.X. where ‘X’ can

be any number from 1 to 254, except 10. By doing this, subnetremains same.

• Pick up the handset of the analog phone connected to ATA.

• Enter the programming access code #19 followed by password

1234. You get programming tone.

• Enter the WAN IP Address, using command 11-<WAN IP

Address>-#* . For e.g. to enter IP Address 192.168.001.151. Enter

command 11-<192168001151>-#*.

• You get confirmation tone followed by system restarting.



Matrix


• Start Internet Explorer (IE6 with SP2, Service Pack) of PC

connected to the Ethernet Switch. Type in the IP Address

programmed above in the Address field. On the login page, enter

default password ‘1234’.

• Click on SIP Account from the menu provided on left side of theweb page.

Get Information from your ITSP for SIP Account

• SIP ID

• Registrar Server Address

• Registrar Server Port

• Authentication User ID

• Authentication Password

• Outbound Proxy Server Address

• Outbound Proxy Server Port

• Enter above information in SIP Account1 field. If these information

is not available, keep the values as default. For e.g. Registrar

Server Port = 5060.

• Enable SIP Account for SIP1 and also enable Outbound Proxy, if

information is programmed.

• If you have a second internet phone Service Account (SIP2), thenuse above fields for SIP2 port and enable. It allows you to use

second phone (or Fax) connected at FXS2 port.

• Submit page.

• You can have a look at all above parameters and software version

information by clicking on ‘Status’ from the menu.

• ATA is ready for further IP calling service!

• User can also use Auto Configuration feature if he wants to avail

facility of configuration of ATA by ITSP from central location.

• Refer ‘ Appendix A: Troubleshooting’ or default the system and

start again, if you cannot establish connectivity from your ATA.

Making Phone Calls

• After SIP1 and/or SIP2 LED is ON, just dial the SIP ID numerical

number e.g. 4567 which is registered to the same ITSP provider.

• If SIP account is [email protected] then the SIP ID number is

4567 and SIP service domain is ‘Provider.com’.



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• PSTN number can be called directly, to the area which your ITSP

Service Provider supports.

• You can also use feature of ‘Speed Dialing’ and ‘Peer-to-Peer

Calling’ as explained below:

Speed Dialing: The numbers which you need to dial frequently can

be programmed in the ‘Phone Book’ table. Suppose the number you

want to dial is entered at index 05 of the phone book, then go OFF-

Hook and dial #8-05 . The number will be dialed!

Peer-to-Peer Calls: Peer-to-Peer number can be dialed even if ATA

is not registered to any service provider. If Peer-to-Peer is enabled

and FXS port number is programmed, you can call another ATA bydialing its FXS port number. You can also call another ATA or IP-

Phone by dialing its IP Address if Peer-to-Peer is enabled. For e.g.

Just dial 192*168*1*21, to call the IP number 192.168.1.21.

Mechanical dimension of SETU ATA2S:



Matrix


104.00 mm (4.09 Inch)

8 0 . 0

0 m m ( 3 . 1

5 I n c h )

2 7 . 0

0 m m

( 1 . 0 6

I n c h )

S I M A D O A T A 2 S

=X=X=



Section 2: Features and Facilities



Matrix


Auto Configuration

• ATA configuration includes some basic settings like Registrar

Server Address, Authentication user ID, and User Password which

are provided by the Service Provider (ITSP). Also some specialservices like Call Forward are provided by the ITSP. Hence, it is

desirable that the ATA is configurable by the ITSP from the central

location. Matrix ATA supports this by providing Auto Configuration

feature.

• The ITSP will keep the folders of configuration files in Central

Server for different ATA units as per LAN/WAN MAC Address

number of the ATA. When the customer connects the ATA to the

ITSP’s network, the ATA gets automatically configured and thecustomer starts using the services.

How it Works?

• ATA makes use of TFTP server to down load the configuration files.

• User has to just enter the Server Address provided by ITSP and

specify the config. Folder path provided by ITSP, in the Web page

for ‘Auto Configuration”. Then after selecting option for using the

feature at power ON, Click on “submit” the page.

• Three options for using the feature are provided:• User can select that the ATA is not to be configured provided as

default.

• User can select that the ATA is to be configured, whenever it is

switched ON.

• User can select that the ATA is to be configured, when switched

ON next time.

• ATA can get the Address of FTP Server from the DHCP Server also

if FTP Server Address is not programmed in ATA.

• The MAC Address displayed on the ‘Status Page’ is programmed atthe factory and can not be changed or deleted by the user.

• If the FTP server address is not found, the auto-configuration

process is aborted and the default configuration files or the current

ones are used by the ATA.

• If all the required files are not available in the folder, then there will

be incomplete configuration download to the ATA. Also only those

downloaded files whose version will match to the version of ATA will



Matrix


be updated.

• For example, 7 configuration files are necessary and if only 5 files

are present, then these 5 files only will be downloaded by the ATA

and will be updated in ATA which may not meet the required

Configuration of the ATA.• When Auto Configuration is complete, the ‘Status’ page displays

following information:

• In ‘Status’ field suitable message is displayed such as:

“No Tftp Server Specified”

“Auto Configuration Fail”

“Auto Configuration Success”

• In ‘Statistics’ field, suitable message regarding how many files

are down loaded successfully and total number of configuration

files are provided by the system. For example, if X=Number of

files down loaded and Y=Total number of files updated. Then the

message displayed will be as:

“Files Downloaded/Configured = X/Y”

How to program?

Click on ‘Auto Configuration’ from the menu on the ‘Welcome Page’.

Auto Configuration:

Parameter Value/Default DescriptionAt Next

Power ON

Blank Select this option if you want to Auto

Configure your ATA when you Power it

ON next time.

At Power ON Blank Select this option if you want to Auto

Configure your ATA whenever you

Power it ON.

Never Default = Selected. Select this option if you do not want to

Auto Configure your ATA.

TFTP ServerAddress

N.N.N.N,40 characters Max.

N = 001 To 255

Or Domain Name

Default = Blank

Enter here the server addressprovided by ITSP. Do not enter if it is

to be allocated by DHCP server, on

request by ATA.

Config

Folder Path

40 Characters max.

Default = Blank

Provided by ITSP. Enter here the path

of the folder in which the configuration

files are stored. For e.g.

Matrix/SetuATA.



Matrix


Important Points:

• ATA will not restart, if user clicks on ‘Submit” on this page.

• This feature can also be used when ATA is connected to an IP

PBX.

Relevant Topic:

1. Status 66

=X=X=



Matrix


Dial Plan

• Generally Matrix ATA user can assign two SIP Account numbers to

two FXS Ports, one for each. Hence if Call from FXS1 needs to use

SIP2 Account, then it can be done by assigning SIP2 to FXS1 in

the FXS Port parameters screen.

• But the ATA has also facility to allocate SIP account, as per the

number dialed. The database for this mapping is called Dial Plan.

• Dial Plan has in all 10 entries.

• Each entry has a number and corresponding SIP account which

the system will use to make an OG call.

• If the number string dialed by the user matches with any of the

number string in the Dial Plan, the call is made using thecorresponding SIP Account.

• This flexibility is helpful in following case:

Example:

If Numbers are mapped with SIP Account as given below:

Index Number SIP Account

01 1234 SIP1

02 12 SIP2: : :

10

• Then, a person in the Cyber cafe can call to Africa on ’1267’ even

if another person is talking to US on SIP1 Account on ‘1234’

number. The ATA allocates SIP2 account to the person who wants

to call Africa, based on closest best fit criteria. Thus Dial Plan

works as powerful routing tool.• Similarly, if the user dials a number say 1256783, the call should

be made using SIP2 as this number best fits with the number string

in index 02.

• If entries are not matching or the table is blank, the OG call will use

default SIP1 Account.

• Click on ‘Dial Plan’.



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Dial Plan:

Value/Default Description

Index 1 to 10This is the index number at which

entry is done.

Number

16 digit Maximum is

from 0 to 9.

Default = Blank

Enter the number string which the

user needs to dial frequently.

SIP

Account

SIP 1 and SIP 2

Default = SIP1

Enter the SIP Account Number

provided by ITSP.

Relevant Topics:

1. FXS Port 34

2. SIP Accounts 60

=X=X=



Matrix


FXS Port

Introduction:

For efficient interfacing with Analog telephone connected at FXS1 and

FXS2 and for efficient use of many features like Echo cancellation,VAD, etc., some parameters are programmed on FXS port. This

chapter gives detailed procedure of programming.

• The ATA supports only DTMF dialing (Tone Dialing).

• ATA supports four types of Caller Number Presentation viz. None,

DTMF (not in this version), BellCore FSK and V.23 FSK.

• ATA supports Calling Party Control to disconnect the Trunk which

is still busy when remote called party has gone ON-Hook. It is

implemented by: CPC period of 500 Millisecond.• ATA supports programming at Input Gain (-6dB to +6dB), Output

Gain (-6dB to +6dB), Programmable Flash Timer, Forward Error

Correction, Programmable Ring Timer, Programmable Inter-Digit

Timer and fixed First Digit Wait Timer.

• Some parameters are described as below before explaining

programming:

FXS Port Name:

• This name is sent to the callee in the Display Name field in the SIPcall.

• The callee shall be able to identify the FXS port from which the call

is made along with the SIP number provided by the ITSP for the

SIP Account.

• To elaborate: Suppose the ITSP has provided a SIP number

[email protected]. When the call is made by FXS1 or FXS2, the

callee gets the same number. Hence he is unable to identify that

who is the actual caller; FXS1 or FXS2. The Station name feature

resolves this problem by assigning ‘xyz’ name only to FXS1 andcalling from port FXS1.

• FXS Port Number: It is used for Peer-to-Peer calls only.

Call Progress tone:

• Different types of tones are played during specific event of

establishing the call, speech duration and termination of the call.

As per ITU standard these tones are of different specification for

different country. For example, Dial Tone for India consists of



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400Hz modulated by 25Hz, but for USA/CANADA, it is 350+440Hz,

without modulation. Based on specific frequency, modulating

frequency and cadence, tones are categorized as shown below:

• Dial Tone

• Ring Back Tone• Busy Tone

• Error Tone

• Confirmation Tone

• Programming Tone

• CO Call Waiting Tone (CCWT)

• ATA plays the Dial Tone when it is ready to accept the first digit of a

remote address to make an outgoing call. Normally played when

FXS goes OFF-Hook.

• ATA plays the Ring Back Tone when the called device is alerting

the callee.

• ATA plays the Busy Tone when the callee is busy.

• ATA plays ‘Error Tone’ when, there is no registration with proxy

server or user has performed some wrong operation. It remains for

7 secs. But when it is used as Error Tone due to invalid

programming, then it remains for only 3 secs. and it is called

Programming Error Tone.• ATA plays ‘Confirmation Tone’, to indicate that valid command is

received by the system and it has taken necessary action.

• ATA plays ‘Programming Tone’ which, prompts the user to enter

fresh commands during programming. This tone, remains

continuously till user dials digit ‘0’. But ATA plays this Programming

Tone as Feature Tone when user has activated some feature like

‘Call Forward’, Do Not Disturb (DND)’, etc. and it remains for 7

secs.

• ATA plays the CCWT when an incoming call arrives while the user is connected to another party.

The time duration and the tone, when it is matured are as per

following table 1:



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NameTimer

(Sec.)Matures to

Dial Tone 7 Error Tone → Standby

RBT 60 Error Tone → StandbyBusy Tone 30 Error Tone → Standby

Error Tone 7 Error Tone → Standby

Confirmation Tone 3 Dial Tone/Feature Tone

Programming Tone 7 Error Tone → Standby

CCWT 60 Error Tone → Standby

Timer column shows that for how long the tone will continue and lastcolumn ‘Matured to’ shows type of tone after timer is expired. It goes

in stand by after error tone completes. For example, ATA plays error

tone for 7 secs. and then goes in stand by.

Refer Table 2 for tone specification for different countries, at the

end of this chapter. Just select country name from the list of

countries in the Combo Box for FXS1 and FXS2 to program the

Call Progress Tone.

Ring Type:

• The Matrix ATA supports different cadence for the ring when

incoming call lands on FXS1 or FXS2.

• These cadence is standardized for different country. For example,

India supports: 25Hz, 0.4 ON 0.2 OFF, 0.4 ON 2.0 OFF (Sec.).

USA/Canada supports: 25Hz, 2.0 ON 4.0 OFF (Sec.).

• In order to support a specific ring type, user has to just select the

country. The system will select the cadence as per table 3, at the

end of chapter. Default country is India.

First Digit Wait Timer:

• First digit wait timer signifies the time for which the system waits

for the user to dial something once the FXS port goes OFF-Hook.

On expiry of this timer, the system will give error tone to the user.



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Internal Digit-Digit Wait Timer: (Timer of Relevance)

• This is the time during which if user does not dial a digit after

pressing a digit, the system stops collecting further digits and

processes digits dialed so far.

Echo Cancellation:

• Line echo is caused when conversion between 4-wire to 2-wire line

is carried out, by telephony hybrid. And delay exceeds 50ms.

Acoustic echo is caused when our own voice reverberating the

phone-receiver when we talk.

• Echo Cancellation is a technique that allows for isolation and

filtering of unwanted signals caused by echoes.

• The echo canceller-device puts a signal on the return transmission

path, which is equal and opposite of the echo signal.

• Echo cancellers (digital filters) are used depending on the length of

the echo to be compensated, also depends on the distance

between gateway and line hybrid. (Between 8-32ms for SOHO

products)

VAD:

• Voice Activation Detection (VAD), is a software application that

allows a data network carrying voice traffic over the Internet todetect the absence of audio and conserve bandwidth by preventing

the transmission of “silent packets” over the network.

• Most conversations include about 50% silence.

• VAD (also called “silence suppression”) can be enabled to monitor

signals for voice activity so that when silence is detected for a

specified amount of time, the application informs the Packet Voice

Protocol and prevents the encoder output from being transported

across the network.

• Click on “FXS”

• FXS Ports



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Parameter Values/DefaultDescription (Enter for FXS1

and FXS2)

FXS Port

Name

16 char. Max.

Default = Blank

Enter if name is to be sent

with the dialed number.

FXS Port

Number

4 digits max.

(0,9,*)

Default = Blank

Enter here, the extension

number on which, incoming

Peer-to-Peer call, when

dialed on this number will

land. Both phones at FXS1

and FXS2 will ring is this

number if not matching.

Listening

Volume

-6 to +6 dB, in 1-

dB Steps

Default = 0-dB.

Select to modify audio level

being output to the device

attached to the FXS.

Speaking

Volume

-6 to +6 dB, in 1-

dB Steps

Default = 0-dB.

Select to modify audio level

entering voice before it is sent

over the network to the

remote VoIP.

Call Progress

Tones

List of countries

refer table 2.

Default=India

Select the country where the

ATA is installed. Refer table 2

for the frequency and

cadence of various tones

supported.

Ring Type

List of countries

refer table 3.

Default = India

Select the country where

ATA is installed. Refer table-3

for the cadence of variousrings supported

Ring Timer 01-99 seconds

Default = 60

Select the time period to play

the ring.

Voice Volume Control

Tones and Rings



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Enable Default = YesClick here, to enable the

feature.

TypeFSK-BellCoreDefault = FSK-

ITU (V.23)

Select the CLI Type as per thePhones connected at FXS1

and FXS2.

Send Caller

IDDefault = Yes

Click if CID is to be sent

during an OG call. If NO, CID

will be Blocked.

Enable Default = NoWhen called party goes OFF-Hook caller gets 'Polarity

Reversal'.

Enable CPC Default = No

Click if CPC feature required.

When called party goes ON-

Hook, CPC is detected for

500msec.

Flash Timer

100 to 900

Millisecond in

steps of 100 ms.

Default = 600 ms.

Select as per the Flash period

standards of the terminal or

phone.

Enable Default = Yes Click if the feature required

Tail Length

8 msec

16 msec

32 msec

Default = 32

msec.

Select such that delay is

minimum so that Voice quality

is improved.

Calling Party number Presentation

Polarity Reversal

Call Party Control

Caller ID

Flash Timer

Echo Cancellation



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Enable Default = No

FEC feature is to reduce error

due to the Lost Packet. Click

for improved voice quality.

Enable Default = No Click if VAD feature required.

First Digit

Wait Timer 7 sec Fixed for the system.

Inter DigitWait Timer

1-9 secondsDefault = 4

Enter if time period is

required to change for dialing

digits.

Call Waiting Default = YesClick if this feature is required

only for next OG call.

Outgoing

calls

NoneSIP Account 1

SIP Account 2

Dial Plan

Default = SIP1 for

FXS1 and SIP2

for FXS2

Assign SIP Account SIP1,

SIP2 or the Account

programmed in the Dial Plan

that FXS1 and FXS2 will use

for OG-Call.

Enable Yes/NoDefault = No

Select 'Yes' if direct IPCalling feature is required.

Outgoing Calls

Peer-to-Peer

Forward Error Correction

Voice Activity Detection

Number Dialing Timer

Call Waiting

Note 2:

Polarity Reversal on FXS port is as per following table:



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Condition Battery Polarity

Idle Negative

Off-hook Toggle to Positive

Remote Party Answers Toggle to Negative

On-hook Retain to Negative

Table 2:

Freq. Cadence Freq. Cadence Freq. Cadence Freq. Cadence

Hz second Hz second Hz second Hz second

Australia 425*25 cont. 400*250.4on 0.2off

0.4on 2.0off 425

0.375on

0.375off 425 0.375on 0.375off

Argentina 425 cont. 425 1.0on 4.0 off 425 0.3on 0.2off 425 0.3on 0.4off

Brazil 425 cont. 425 1.0on 4.0 off 425 0.25on 0.25off 425 0.25on 0.25 off

China 450 cont 450 1.0on 4.0off 450 0.35 on 0.36off 450 0.7on 0.7off

Egypt 425*50 cont 425*50 2.0on 1.0off 425*50 1.0on 4.0off 450 0.5on 0.5off

France 440 cont 440 1.5on 3.5off 440 0.5on 0.5off 440 0.25on 0.25off

Germany 425 cont 425 1.0on 4.0off 425 0.48on 0.48off 425 0.24on 0.24off

Greece 4250.2on 0.3off

0.7on 0.8off 425 1.0on 4.0off 425 0.3on 0.3off 425 0.15on 0.15off

India 400*25 cont 400*250.4on 0.2off

0.4on 2.0off 400 0.75on 0.75off 400 0.25on 0.25off

Indonesia 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.25on 0.25off

Iran 425 cont 425 1.0on 4.0off 425 0.5on 0.5off 425 0.25on 0.25off

Israel 400 cont 400 1.0on 3.0off 400 0.5on 0.5off 400 0.25on 0.25off

Italy 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.2on 0.2off

Japan 400 cont 400*20 1.0on 2.0off 400 0.5on 0.5off 400 0.25on 0.25off

Kenya 425 cont 4250.67on 3.0off

1.5on 5.0off 425

0.2on 0.6off

0.2on 0.6off 425 0.2on 0.6off

Korea 350+440 cont 440+480 1.0on 2.0off 480+620 0.5on 0.5off 480+620 0.3on 0.2off

Malaysia 425 cont 4250.4on 0.2off


Mexico 425 cont. 425 1.0on 4.0off 425 0.25on 0.25off 425 0.25on 0.25off

New Zealan 400 cont. 400+4500.4on 0.2off


Phillippines 425 cont. 425+480 1.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off

Poland 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.5on 0.5off

Portugal 425 cont. 425 1.0on 5.0off 425 0.5on 0.5off 450 0.33on 1.0off

Russia 425 cont. 425 0.8on 3.2off 425 0.4on 0.4off 425 0.25on 0.25off

Saudi Arabi 425 cont. 425 1.2on 4.6off 425 0.5on 0.5off 425 0.25on 0.25off

Singapore 425 cont. 425*240.4on 0.2off


South Africa 400*33 cont. 400*330.4on 0.2off


Spain 425 cont. 425 1.5on 3.0off 425 0.2on 0.2off 425 0.25on 0.25off

Thailand 400*50 cont. 400 1.0on 4.0off 400 0.5on 0.5off 400 0.3on 0.3off

Turkey 450 cont. 450 2.0on 4.0off 450 0.5on 0.5off 4500.2on 0.2off

0.6on 0.2off

UAE 350+440 cont. 400+4500.4on 0.2off

0.4on 2.0off 400

0.375on

0.375off 400

0.4on 0.35off

0.225on 0.525off

UK 350+440 cont. 400+4500.4on 0.2off

0.4on 2.0off 400

0.375on

0.375off 400

0.4on 0.35off

0.225on 0.525off

USA/Canada 350+440 cont. 440+480 2.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off

Error Tone

Country

Dial Tone Ring Back Tone Busy Tone



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Remarks:

Frequency notations shall be as:

• f1*f2: f1 is modulated by f2.

For All CountriesConfirmation Tone : 400Hz, 0.1 on 0.1 off,

Programming Tone : 400Hz, 0.1 on 0.9 off,

CCWT : 400Hz, 0.3 on 6.0 off

Table 3:

ON 1 OFF 1 ON 2 OFF 2

01 Australia 25 0.4 0.2 0.4 2.002 Brazil 25 1.0 4.0

03 China 25 1.0 3.0

04 Egypt 25 2.0 4.0

05 France 25 1.5 3.5

06 Germany 25 3.5 5.5 0.79 1.1

07 Greece 25 1.0 4.0

08 India 25 0.4 0.2 0.4 2.0

09 Israel 25 2.0 3.010 Italy 25 1.0 4.0

11 Japan 25 1.0 2.0

12 Korea 25 1.0 3.0

13 Malaysia 25 2.0 3.0

14 New Zealand 25 2.0 3.0

15 Poland 25 2.0 3.0

16 Portugal 25 1.0 5.0

17 Russia 25 1.0 3.0

18 Singapore 25 0.4 0.2 0.4 2.0

19 South Africa 25 0.4 0.2 0.4 2.0

20 Spain 25 1.5 3.0

21 Thailand 25 2.0 3.0

22 UAE 25 2.0 3.0

23 UK 25 0.4 0.2 0.4 2.0

24 USA/Canada 25 2.0 4.0

Code CountryFrequency

(Hz)

CADENCE (In Seconds)



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Relevant Topics:

1. Peer-to-Peer Calling 55

2. SIP Accounts 60

3. Supplementary Services 70

4. Dial Plan 32

=X=X=



Matrix


Jeeves

What’s this?

• As, the set up of SOHO and Enterprise is normally equipped withWAN and Router on the Internet line, it is very convenient if the

device has the capability to be programmed via HTML-based

interface. The Matrix ATA has this provision.

• WWW is an abbreviation of Internet’s World Wide Web. Matrix ATA

needs Web browser IE6 with SP2 (Service Pack) patch or Mozilla

Fire Fox installed on the PC. Description of following features

related to web will help in programming the system.

IP Address• An IP address (Internet Protocol address) is a unique number,

similar in concept to a telephone number, used by network devices

(computers, time-servers, FAX machines, some telephones)

attached to a network to refer to each other when sending

information through a WAN or the Internet for example.

• This allows devices passing the information onwards on behalf of

the sender to know where to send it next, and for the device

receiving the information to know that it is the intended destination.

• An example of IP address is 207.142.131.236. Converting a

number address to a more human-readable form called a domain

address is done via the Domain Name System.

• The Internet Protocol (IP) knows each logical host interface by a

number, the so-called IP address. On any given network, this

number must be unique among all the host interfaces that

communicate through this network.

• Users of the Internet are sometimes given a host name in addition

to their numerical IP address by their Internet service provider.• For all programs that utilize the TCP/IP protocol, the sender IP

address and destination IP address are required in order to

establish communication and send data.

Sub Net

• It is a mechanism that is used to split a network into a number of

smaller sub networks. It can be used to reduce traffic on each sub

network by confining traffic to only the sub networks for which it is



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intended, thereby eliminating issues of associated congestion on

other sub networks and reducing congestion in the network as a

whole.

• Makes entire network more manageable.

• Each sub network functions as though it was an independentnetwork, keeping local traffic local, and forwarding traffic to another

sub network only if the address of the data is external to the sub

network. Such decisions are made on the basis of routing-tables

contained within the various routers, with each table comprising an

IP address table.

• Subnet is a portion of the network, which may be a physically

independent network, which shares a network address with other

portions of the network and is distinguished by a subnet number.

Default Gateway

• A default gateway is a node on a computer network that serves as

an access point to another network.

• In enterprises, the gateway is the computer that routes the traffic

from a workstation to the outside network that is serving the Web

pages.

• In homes, the gateway is the ITSP that connects the user to the

internet.• A default gateway is used by a host when an IP packet’s

destination address belongs to someplace outside the local subnet.

Domain Name System (DNS)

• DNS is a system that stores information about host names and

domain names in a type of distributed database on networks, such

as the Internet.

• Of the many types of information that can be stored, most

importantly it provides a physical location (IP address) for eachdomain name, and lists the mail exchange servers accepting e-mail

for each domain.

• The DNS provides a vital service on the Internet as it allows the

transmission of technical information in a user friendly way.

• While computers and network hardware work with IP addresses to

perform tasks such as addressing and routing, humans generally

find it easier to work with hostnames and domain names (such as

www.example.com) in URLs and e-mail addresses.



Matrix


• The DNS therefore mediates between the needs and preferences

of humans and of software.

Dynamic Host Configuration Protocol (DHCP)

• The DHCP is a client-server networking protocol.• A DHCP server provides configuration parameters specific to the

DHCP client host requesting, generally, information required by the

host to participate on the Internet network.

• DHCP also provides a mechanism for allocation of IP addresses to

hosts. The DHCP protocol provides three methods of IP address

allocation:

• Manual Allocation.

• Automatic Allocation.

• Dynamic Allocation.

How to configure ATA?

To start basic VoIP configuration first enter the default WAN IP

address of the ATA, obtained from SE in the ‘Address’ bar of the Web-

browser screen as explained below:

• Launch the Web browser as explained below:

• Double Click on Internet Explorer icon

Or Click on Start Æ Programs Æ Internet Explorer.

• Enter IP Address at Address Bar after confirming from SE. Let us

say IP Address 192.168.1.10 is entered at Address Bar on opening

page of Microsoft Internet Explorer .

• Note that Matrix ATA2S default IP Address is 192.168.001.151.

• ATA displays Welcome page with a field for entering ‘Password’.

Fie ld Name Value/Default Description

Password Up to '4' digits(0-9), 1234 Enter the default password‘1234’ to access the system

• Click Login after entering valid Password. If there is any error in

typing the Password, the ‘Message page’ is displayed for entering

Password again and click on Submit.

• After entering the correct password, login status page is displayed.

This page contains following table (information) at the centre of the

page.



Matrix


MATRIX SETU ATA2S

Welcome

User Address Login expires if idle for

xxx.xxx.xxx.xxx 10 Minutes

xxx.xxx.xxx.xxx : Address of the computer from where the user

has logged in.

• Login Expiry Time: This is fixed (maximum) time for which user

can keep a page without programming any parameter in it. If this

time is exceeded, he cannot submit current page. He has to login

again.

• All the above fields are non-editable. The user can now go to any

of the screens by clicking on the links provided at the left of the

screen.

• Main Menu is displayed on the left side of Welcome page (without

Password field).

• The ‘Main Menu List’ shows all the important parameters that are

required to be programmed, to make the ATA work as expected.

• Click ……from the ‘Main Menu’.

Common screen command buttons: (General Note for all thescreens)

Note that at the end of the every screen, options are provided for

really applying these changes to the ATA. This is conveyed by

‘Submit’ and ‘Default All’:

Button Meaning

Submit Click, to save changes to ATA

Default AllClick, to set all parameter on the

screen to factory default

The ‘Main Menu List’ will be displayed as shown below:

• Password

• System Name

• Time Settings

• FXS Ports

• Supplementary Services

• WAN Port



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• SIP Accounts

• Dial Plan

• Phone Book

• Peer-to-Peer Calling

• Auto Configuration• Software Upgrade

• Status

• Default the ATA (Resetting the ATA)

Screens Summary:

LINK FUNCTIONS

PASSWORD Use this screen to change password.

System Name Use this screen to assign name to ATA.

Time SettingUse this screen to configure system’s time to

synchronise with the time server.

FXS PortUse this screen to configure FXS-Port parameters

like gain, echo cancellation etc.

Supplementary

services

Use this screen to configure call features like call

forward, DND etc.

WAN Port

Use this screen to configure WAN Port

Parameters.

SIP AccountUse this screen to configure SIP settings, coder,

other parameters for SIP1 and SIP2 Ports.

Dial PlanUse this screen for entering number strings to

select SIP Account.

Phone book This is for programming phone book entries.

Auto

Configuration

This is to configure the ATA of specific MAC

Address number by ITSP.

Software

Upgrade

This is to upgrade any file of the software using

FTP feature.

StatusThis is a brief status of important parameters like

WAN address.

Peer-to-Peer

CallingThis is for programming Peer-to-Peer Call entries.

Default the ATA This is for downloading default parameters.



Matrix


In order to make required changes in features, click on any concerned

feature from the list given on left side of the Main Menu.

Example 1:

To use Dial Plan feature for outgoing call.1. After entering suitable IP address in address field the welcome

page look, as below:

2. After Login, the welcome page displays the time after which login

expires as shown below:



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3. Click on ‘SIP Accounts”. Enter SIP ID Numbers and registrar

Server Address provided by ITSP and enable SIP accounts as

shown below:



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4. Click on ‘FXS Port’ and select ‘Dial Plan for FXS1’ and select

‘SIP Account2’ for FXS2. Thus any number dialed from phone at

FXS1 will use the SIP number mentioned in the ‘Dial Plan’ screen

or SIP1 as default. But any number dialed from FXS2 will use

‘SIP Accounts2’ only as shown below:

5. Click on ‘Dial Plan’. Enter the required number string that will be

dialed from FXS1 and use the specific SIP account as shown

below:



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Example 2:

1. Click on ‘SIP Accounts’ and select ‘Route to BOTH’ for incoming

calls field, as shown below. If any IC call comes on SIP1, both

phones will ring.



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Example 3:

• To program Call Progress Tones and Ring Type for country, click

on FXS port and then select the name of country from the list of

the combo box of ‘Call Progress Tones’ and ‘Ring Type’, as

shown below:

Relevant Topics:

1. Password 54

2. Software Upgrade 65

=X=X=



Matrix


Password

• For system security, it is advisable to change the default ‘System

Password’ after configuring the parameters.

• Matrix ATA supports this feature. For this, old Password, New

Password and Retyping of the new-password are required to be

entered.

• The ‘Password’ will not be visible. Only ‘*’ will be seen for every

character of the password. The fields are as shown in the Web

Page:

• Click on ‘Password’


Password4 digits

Max/1234

Type the default password or the

existing password you use to

access the system, 1234.

New

Password4 digits Max Type the new password

Retype to

confirm4 digits Max Type the new password again

By default, Password is Blank.

Only numerals (0 to 9) are allowed up to four digits maximum. This is

because same password should be used to enter SE mode from SLT.

Relevant Topic:

1. Jeeves 44

=X=X=



Matrix


Peer-to-Peer Calling

What’s this?

• ATA can call another user agent (ATA or Softphone), which is in the

same network (figure 2) or which can be in another network, whichis connected by virtual LAN (VLAN) provided by ITSP (Figure 1).

For example, ITSP provides VLAN services to connect two offices

situated in different cities.

• ATA can call to an extension number of some other ATA or soft

phone without going through any proxy. This is known as Peer-

to-Peer Calling. For this, the extension number (FXS Port number),

which is to be called, is programmed in the number field and IP

address of the called ATA is programmed in ‘IP Address’ field of thePeer-to-Peer Call Table.

• Each entry has two parameter viz. ‘Number’ and ‘IP Address’.

• For example, if IP address of ATA which is to be called, is

192.168.1.12 and extension number 3005, then at Index 01, 3005

is programmed in ‘Number’ field and 192.168.1.12 is entered in

‘IP Address’ field and screen is ‘Submitted’. For Peer-to-Peer OG

Call, user has to follow following steps:

• Enable Peer-to-Peer in FXS Port screen.

• Dial 3005 .• For incoming Peer-to-Peer call (IC Call) , the FXS port number on

which call is to be received, is programmed in ‘FXS Port’ number

field on the screen ‘FXS Port’. For example, if 4001 is programmed

in ‘FXS Port’ screen in FXS1 column then any IC call on 4001 will

land on FXS1 port. But, if incoming call doesn’t have perfect match

and only IP Address of Matrix ATA is called, then call will land on

both FXS1 and FXS2.

• ATA also supports to call the ATA by an IP number. After lifting the

handset, user dials digits with ‘*’ in place of ‘.’. This is called ‘DirectIP Calling’ feature.

• This feature is very much useful in case, Party’s IP Address only is

known and not ‘SIP-ID’.

• For example, a PC phone, with its IP Address 192.167.1.21, as

shown in figure 1 can be called by just dialing 192*167*1*21, after

lifting the handset.



Matrix


Figure 1:

SETU ATA2S

SETU

ATA2S

SETU ATA2S

SETU ATA2S

Internet

192.168.1.10

192.168.1.11

192.168.1.12

192.168.1.13

2001

2002

2004

2005

192.168.1.15

3001

3002

3005

3006

192.167.1.20

192.167.1.21

192.167.1.23

192.167.1.22

Router Router

VLAN

Services

Mumbai Office Delhi Office

192.167.1.24

PC Phone

PC PhonePC Phone

PC Phone

Figure 2:

SETU

ATA2S

SETU

ATA2S

SETU

ATA2S

SETU

ATA2S

192.168.1.10

192.168.1.11

192.168.1.12

192.168.1.13

2001

2002

2004

2005

192.168.1.15

3001

3002

3005

3006

192.167.1.20

192.167.1.21

192.167.1.23

192.167.1.22

Router Router

First Floor Second Floor

192.167.1.24

PC Phone

PC PhonePC Phone

PC Phone



Matrix


Example of Peer-to-Peer table programming:

Index Number IP Address

01 2001 192.168.1.198

02 2002 192.168.1.198

03 1234 192.168.1.200

: : :

64 2010 192.168.1.215

As shown in the programming table, number 2001 is dialed for ATA

bearing IP Address as 192.168.1.198, to make Peer-to-Peer Call.

• Click on Peer-to-Peer, on the welcome page.

Peer-to-Peer:

Parameter Value/Default Description

Index 01 to 64Number at which entry is

made.

Number

Maximum 4 digits

(0-9,*)

Default = Blank

Enter the extension

number of the user agent

to be called.

IP Address

Maximum 15 digits

(0-9, '.')

Default = Blank

Enter IP Address of the

user agent to be called.

Relevant Topic:

1. FXS Port 34

=X=X=



Matrix


Phone Book

• It is the Database for storing Full URI SIP number, name and SIP

account number.

• Maximum 99 numbers can be stored in a phone book.

• Each entry has three parameters viz. the SIP number, name and

SIP Account number.

• The number is dialed out using the SIP Account assigned to it.

• The Access Code for dialing Phone book numbers is #8-Index

Number . The user shall go off-hook and dial #8 followed by the

Phone book index number (from 01 to 99). e.g. #8-64, to dial the

number, stored at Index ‘64’. If number is not programmed at

index 64, user will get ‘Error Tone’.

• Click on ‘Phone Book’.


Index 01 to 99This is the index number at

which an entry is done.

Number

SIP number

maximum 16

characters.Default = Blank

Enter the numberof person to be

called. e.g. 12345678.

Name

Maximum 16

characters Default

= Blank

Enter name of person, whose

SIP number is stored in the

phone book.

SIP

AccountSIP 1 and SIP 2

Enter SIP Account number on

which the OG call will be made,

when dialing the specified

number.

Relevant Topic:

1. SIP Accounts 60

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Matrix


Resetting the ATA

What is this?

• ATA has the provision for changing the password. In case, user has

forgotten the Password or he may need to default/reset the system,

‘System Default’ feature is also provided.

• The system will loose all previous configuration, using this feature.

• ATA supports following methods to default:

• Default only SE Password through Access Code.

• Default System (Including SE Password) through Web Jeeves or

SE command.

How it works?• For resetting through web page, follow following steps:

• Click on ‘Default the ATA’.

• User will find the alert message windows as ‘This will assign

default values to all the parameters of the ATA’.

• Click on ‘OK’ or ‘Cancel’. If clicked on ‘OK’, default values are

assigned to the parameters. If clicked on ‘Cancel’, all the

parameters remain same.

• Use following access code to set only the login password (SE

Password) to default value:#*** .

It is not required to enter programming mode.

• Use following SE command to default the system:

21-Reverse SE Password-#*.

Relevant Topic:

1. Getting Started 18

=X=X=



Matrix


SIP Accounts

This chapter describes how to configure basic SIP settings and

Advanced VoIP and RTP parameters. Before starting it will be useful

to refer some VoIP terms in the “VoIP Basics” chapter and Glossary.Click on ‘SIP Accounts’.

Performance Parameters for SIP Accounts:

• Enter all values for SIP1 and SIP2.

• Parameters in SIP settings column are explained below:

Registrar Settings:

• Enable SIP Account: Click on enable SIP1, SIP2 or bothaccounts. If SIP account is disabled, it will not be used to make OG

calls even if it gets selected by the OG call routing logic.

By default, SIP Account is Disable.

• SIP ID: Enter here, the user part of the full SIP URI. This can be a

number or text.

Field size: 24 characters maximum. Allow all ASCII characters.

e.g. SIP URL provided by the ITSP is

[email protected], enter 12345 in this filed.By default, SIP ID is Blank.

• Registrar Server Address: Enter the SIP registrar server’s

address. It can be IP address also and it’s provided by ITSP.

Field size: 40 characters maximum. Allowed all ASCII chars.

By default, Registrar Server Address is Blank.

• Registrar Server Port: Enter here, the listening port for SIP. SE

may change this if ITSP has provided the number, not same asdefault.

Valid range is from 1025 to 65535.

By default, Registrar Server Port is 5060.

• Re-Registration Timer: As a part of normal process, the register

server deletes an entry of its client from its database on expiry of a

fixed timer. This timer is set by the register server.

Hence, in order to be registered always, the ATA should send a



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registration request before this timer expires. Enter the value of this

timer here. It signifies the time after which the ATA should send

registration request again to be registered.


By default, Re-Registration Timer is 3600 secs.

• Registration Retry Timer: Enter the period between retries for

registration. On failure of registration, the ATA sends the

registration request. If the registration attempts fails, the ATA

should send the registration request on expiry of this timer. The

ATA should keep sending the registration request till it gets

registered and once it gets registered the ATA will start

Re-Registration timer.Valid range is from 00001 to 65535.

By default, Registration Retry Timer is 10 seconds.

Authentication:

• User-ID: Enter the user name for registering the SIP account with

the SIP register server. This field is relevant when SIP user ID and

Authentication User ID are not same. It is provided by ITSP.

Valid range is maximum 40 characters.

By default, Authentication User ID is Blank.

• Password: Enter the password associated with the user name

above.

Valid range is maximum 16 characters.

By default, Authentication Password is Blank.

Vocoders Preference:

• Type : Two options for selecting Vocoder are provided:

• Single• List

By default, Vocoders Type is List.

• Single: If single is selected, preferred Vocoder only will be offered.

Valid Range is G.711 a-law, G.711 u-law, G.723 and G.729.

• List : If clicked here, system will negotiate with remote phone from

the available Vocoders when call is established. If list is selected,



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list of supported Vocoders will be offered with selected Vocoder as

preferred Vocoder.

• Vocoders : ATA supports 4-types of Vocoders:

• G.711 a-law.• G.711 u-law.

• G.729.

• G.723.

By default, Preferred Vocoder is G.729.

DTMF Option:

• Select the option to decide, how the DTMF digits will be sent over

IP, When digit is pressed. ‘Inband’, which means DTMF iscombined in audio signal. Select outband, if digits are to be sent

via RTP, using RFC 2833.

Valid range is Inband, Outband.

By default, DTMF Option is Outband.

Outbound Proxy:

• Enable: Click to enable, if the ITSP service provider has a SIP

outbound server to handle voice calls.

By default, Enable Outbound Proxy is Blank.

• Server Address: Enter to specify the outbound proxy servers

address. Allow IP address also.

Valid range N.N.N.N, Where N=1.0.0.1 to 255.255.255.254,

Maximum 48 characters.

By default, Outbound Proxy Server Address is Blank.

• Server Port: Enter to specify the outbound proxy server’s

listening port for SIP. SE may change this field if the ITSP providesan outbound proxy sever port number other than the default. This

may be same as SIP server’s port address.

Valid Range is from 1025 to 65535.

By default, Outbound Proxy Server Port is 5060.

STUN:

• Enable: Click, if there is a NAT router between the ATA and the

ITSP’s SIP server.



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By default, Enable STUN is Blank.

• Server Address: Enter the STUN server’s address. Allow IP

address also. Allow extended ASCII characters.

Valid range is N.N.N.N, Where N=0 to 255 or maximum 40characters.

By default, STUN Server Address is Blank.

• Server Port: enter the STUN server’s listening port for SIP. SE

may change this field if the ITSP provides a STUN Server port

number other than the default.


By default, STUN Server Port is 3478.

NAT Keep Alive:

• Enable: Click Yes to enable. The ATA’s SIP messages will use NAT

router’s public IP address and SIP port number. The NAT should

be also be configured with this port number to forward the traffic

with this port number.

NAT eliminates the need for STUN.

By default, disable.

• Interval: Assign the time after which the ATA should send SIP

notify messages to the SIP server.

By default, NAT keep Alive interval is 120 seconds.

Incoming Call-Routing:

• Incoming Calls: Select the port on which you want the incoming

calls on SIP Account 1 to be routed and to land.

Similarly specify the port for SIP Account 2 calls.

Valid range is FXS1, FXS2 or Both.By default, FXS1 for SIP1 and FXS2 for SIP2.

QOS (Layer 3):

• SIP QOS: This field specifies the QOS type viz. TOS

(Precedence, also called Priority), or DiffServe for voice traffic.

The ATA sends all the voice packets with this QOS setting.

Precedence uses 3 bits and DiffServe uses 6-bits. Valid range



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for Precedence is from 0 to 7 and for DiffServe is from 00 to 63.

This parameter is same for SIP1 and SIP2.

By default, Precedence is 5 and DiffServe is 26.

• RTP QOS: this field specifies the QOS type viz. TOS(Precedence, also called Priority) Or DiffServe for voice traffic.

The ATA sends all the voice packets with this QOS setting.

Precedence uses 3 bits and DiffServe uses 6-bits.

Valid range for Precedence is from 0 to 7 and for DiffServe is from

00 to 63. This parameter is same for SIP1 and SIP2.

By default, Precedence is 5 and DiffServe is 46.

Relevant Topics:

1. VoIP Basics 83

2. Glossary 104

3. FXS Port 34

4. Time Settings 82

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Matrix


Software Upgrade

What’s this?

ATA support an embedded FTP server which can be used for:

• Uploading/Downloading System files

• File Transfer Protocol (FTP), a standard Internet Protocol, is the

simplest way to exchange files between computers on the Internet.

Like the Hypertext Transfer Protocol (HTTP), which transfers

displayable Web pages and related files, and the Simple Mail

Transfer Protocol (SMTP), which transfers e-mail, FTP is an

application protocol that uses the Internet’s TCP/IP protocols. FTP

is commonly used to download program and other files to your

computer from other servers.• Pop-up should be enabled from; TOOLS-Popup Blocker on the

Internet Explorer.

Procedure to upgrade file:

• To upgrade any ‘file’ of the ATA software, click on ‘S/W Upgrade’.

Four file folders are displayed: appli, config, driver, httpd.

• Modified file from User’s Page is copied in the suitable file folder,

config, driver or httpd. Old file is automatically replaced.

• To upgrade an ‘appli’ file, first old one must be deleted and thennew file is copied.

• Finally ATA has to be restarted for applying new file application.

• Mozilla-Client will not work for upgrading the Software.

Relevant Topics:

1. Password 54

2. Jeeves 44

=X=X=



Matrix


Status

What’s this?

• ATA user may require to view remotely for WAN Addresses and

registration status. Matrix ATA has this facility. The registration

time is provided by the server. User can quickly have a glance for

WAN address and can change it from respective page, if required.

• It also provides ‘Registration Status’ which is useful to know

whether ‘SIP’ is registered or not. For example, if the Ethernet

cable for Internet gets disconnected than after some time,

‘Registration Status’ is displayed on the ‘Status’ screen (if updated)

and user can take corrective action, just by connecting the cable

instead of restarting the system.• The MAC Address used for Auto Configuration is programmable

only at factory. It cannot be changed by user.

• Click on ‘STATUS’ page. Following basic parameters are displayed:

STATUS

Parameters Display Description

System

Name

Matrix SETU

ATA2SThis displays the name of system

Version/

RevisionVxRy

This displays the current version 'x'

and Reviosn 'y' of software

IP Address

192.168.50.96 or

default

192.168.001.151

This displays the programmed IP

Address

Subnet Mask 255.255.255.0This displays the programmed subnet

mask

MAC

Address

Unique Address

Number or Default

00.50.C2.55.B0.18

This displays the MAC Address

programmed at factory. Default

address will be dis]played only if it is

not programmed at factory. It will not

be defaulted, if user will try to default

ATA.

WAN Port:



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Registration

Status

Not Registered/

Registered

When 'Register' button is

presses the ATA sends

registration request to theserver and the 'Status' is

displayed.

Registration

Time

This time is provided

server during registration

Registration

Retry Count 0,1,2……

This shows how many

times after registration

failure, ATA has sent

registration request.

Reg Last Failed

Reason

This displays the reason

for last time failure of

registration. For example,

Message sent failure.

Response timeout. Failed

response 486, etc.

NAT Type

Blank display,

Unknown, full cone,Restricted, Port

Restricted,

Symmetric,

Symmetric Behind

Firewall, Blocked

NAT is configured for

option as shown.(1) If STUN is not enabled,

it is Blank (No Display) (2)

if none of the type of NAT

is detected by ATA, then

'Unknown' is displayed.

Public IP N.N.N.N.

N = 001 to 255

When STUN is enabled

'Public IP' is displayed

here.

SIP Account1:



Matrix


Registration

Status

Not Registered/

Registered

When 'Register' button is

presses the ATA sends

registration request to theserver and the 'Status' is

displayed.

Registration

Time

This time is provided by

server during registration

Registration

Retry Count 0,1,2……

This shows how many

times after registration

failure, ATA has sent

registration request.

Reg Last Failed

Reason

This displays the reason

for last time failure of

registration. For example,

Message sent failure.

Response timeout. Failed

response 486, etc.

NAT Type

Blank display,Unknown, full cone,

Restricted, Port

Restricted,

Symmetric,

Symmetric Behind

Firewall, Blocked

NAT is configured for option as shown.

(1) If STUN is not enabled,

it is Blank (No Display) (2)

if none of the type of NAT

is detected by ATA, then

'Unknown' is displayed.

Public IP N.N.N.N.N = 001 to 255

When STUN is enabled

'Public IP' is displayedhere.

SIP Account2:



Matrix


Status Blank

Message displays after

Auto Configuration is

applied during Power ON:for e.g. 'No tftp Server

Specified', 'Auto

Configuration Fail', 'Auto

Configuration Success'.

Statistics Blank

Message displays after

Auto Configuration is

applied: for e.g. 'Files

Downloded/Configured=X/Y'.

Auto Configuration

Relevant Topics:

1. WAN Port 93

2. SIP Accounts 60

3. System Name 81

4. Auto Configuration 29

=X=X=



Matrix


Supplementary Services

This chapter contains following sections:

• Introduction• Programming Type

• Programming of Web Pages

• Programming using Phones

• Important Points

Introduction:

ATA provides following call features based on supplementary services

provided by the ITSP:

• Making Second Call• Call Hold

• Call Waiting

• Call Toggle

• Call Transfer-Blind

• Call Transfer-Attended

• 3-Party Conference

• Call Forward-Unconditional

• Call Forward-On Busy

• Call Forward-On No Reply

• DND

• Caller ID

These supplementary services should be supported by the service

provider.

Making Second Call

• Telephone user can make second call while talking to first caller

by keeping the first caller on hold. The user should keep the firstcaller on hold and then dial the destination number to which he

wants to talk. On completing the talk, ask the called party to

hang-up. When the called party hangs up, the user gets

connected to the first caller. Also, while in speech with second

call, he can talk to first user by using call toggle access code

(except type 3 users). The user can also have conference

between three parties by dialing conference access code.



Matrix


Call Hold

• Call Hold allows one party to maintain an idle (non-talking)

connection with another party while receiving another call (Call

Waiting) or while initiating another call (Call Transfer), or while

performing some other call management functions.

Call Waiting

• Call Waiting (CW) notifies an engaged caller of an incoming call

and allows them to receive a call from a third party while the party

with whom they have been speaking is put on hold.

Call Transfer (Blind):

• Call Transfer allows one party to reconnect the party with whom,

they have been speaking; to a third party. The first party is

disconnected when the third party becomes connected. Call

transfer without consulting to the party on whom call is to be

transferred is “Blind Transfer”.

Call Transfer-Attended

• Telephone user dials Call Hold access code followed by the

Transfer target’s number to transfer the caller (Transferee) to the

Transfer target. In case of, Call transfer with consulting or CallTransfer-attended the user gets ring back tone followed by

speech or Busy Tone. He may talk to transfer target and hang up.

Call Toggle

• To attend to the second call, phone user has to use this feature

by dialing ‘Call Toggle Access Code’.

• When the user dials the Call Toggle access code when a party is

held, user is connected to the waiting caller. If no party is held,

then user gets error tone.• The user is connected to the current party when he goes

ON-Hook, gets ring and again goes OFF-Hook.

• Type 3 user cannot use this feature.

3-Party Conference

• Telephone user dials Conference access code to have

Conference between the User and the two callers.

• When the user dials the Conference access code (with one held



Matrix


party and one current party):

• Conference is held between the user and the two callers.

While during the Conference if the user dials the access code

to Toggle (Not applicable for Type 3 users).

• User is connected to one of the callers.• Other caller gets music.

Programming Type:

Following is the method of programming the features by phone as

well as Jeeves:

• The following table shows the feature codes for various features to

be used in three different types of feature code sets.

• Type 1 is a set of access codes used by Matrix-ATA.

• Type 2 is made considering the European access codes.

• Type 3 is made considering the American access codes.

Type 1 (Matrix

System)

Type 2

(European

System)

Type 3

(American

System)

Hold Flash Flash Flash

Hold Retrieve

Flash (within

inter-digit timer)

/Remind Ring

Flash (within

inter-digit timer)

/Remind Ring

Flash (within

inter-digit timer)

/Remind Ring

Reject Waiting

callFlash-2 Flash-0 --

Hold current

call & answer waiting call

Flash-1 Flash-2 Flash

Disconnect

current call and

answer waiting

call

Flash-3 Flash-1 --

Call Hold

Call Waiting



Matrix


Blind Call

Transfer

Flash-*1-

Number-ON

Hook

Flash-*98-

Number-ON

Hook

Flash-#90-

Number-ON

Hook

Attended Call

Transfer

Flash-Number-

Speech-OnHook

Flash-Number-

Speech-OnHook

Flash-Number-

Speech-OnHook

Conduct

ConferenceFlash-0 Flash-3 Flash

Conference to

ToggleFlash-1 Flash-2 --

Call Toggle Flash-1 Flash-2 --

Call Toggle

Call Transfer

Conference

Programming of Web Pages:

Programming through Jeeves is as given below:

• Click on “Supplementary Services”

• Type Selection:

Description

Type1 Type2 Type3

Feature

Code

Type

Value/Default

Default = Type 1

Select Type from Type1, Type2

and Type3 as per the country in

which ATA is installed. Select

Type1 for Matrix System, Type2

for European System and Type3

for American System.



Matrix


• Call Forward and DND:

Value/DefaultDescription (Programm for

SIP1 and SIP2)

Call Forward-

Unconditional

16 digits max

(Digits 0 to 9

and *)

(Default: Blank)

Note 1

Enter destination number to

set “call forward”-for call

coming on SIP1. Set another

number for call coming on

SIP2.

Call Forward-

When Busy

16 digits max

(Digits 0 to 9

and *)

(Default: Blank)Note 1

Enter destination numbers to


coming on SIP1 which is

busy on speech. Set another

number for call coming on

SIP2 (busy).

Call Forward-

When No

Reply

16 digits max

(Digits 0 to 9

and *)

(Default: Blank)

Note 1

Enter destination number to


coming on SIP1, in case this

call is not attended. Set

another number for SIP2.

Call Forward

Timer

01-99 sec (45)

Note 2

Set the timer for which ‘No

Reply’ condition remains.

After this time, system will

initiate call forward.

DND (Do Not

Disturb)

Set/Cancel

(Cancel)

This is ‘Do Not Disturb’

feature. Enable if you do not

want to receive any calls.

Note 1: The number field shall remain grey till the feature isenabled by checking it in the check box.

Note 2: This field will be activated only if Call Forward-When No

Reply is selected.

Note 3: Call Forward and DND applies to SIP account and not

‘FXS’.



Matrix


Procedure of Calling:

Call Forward

• Call Forwarding (Always)

Incoming calls may always be forwarded to another designated

party, and call forwarding founder can hear the notified rings.

Example:

1 A calls B. NO TALK.

2

B responds: it is

unavailable. So it returns a

forwarding destination C.

User at A is

unaware of this.

3

A is automatically forwarded

to C. Talk between A, C.

• Call Forwarding (Busy):

Incoming calls can be forwarded to another designated party if the

phone of the called party is busy.

1 A calls B. NO TALK.

2 B responds, (Busy state).

B is in busy state and

the new forwarding

state is C.

3 A is automatically

forwarded to C.Talk between A, C.

• Call Forwarding (No-Answer):

Incoming calls can be forwarded to another designated party when

there is no response, from the original called party.

1 A calls B.B is ringing, A can hear

ring back tone.

2

B responds to A, after 45

secs. (programmable) with

no one picking up the

phone.

No answer and A needs

to contact C.

3 A is automatically

forwarded to C.Talk between A, C.



Matrix


3-Party Call Conference (For Type 1):

• To make 3 party call conference.

1 Speech on.

2 Dial Flash.Feature tone (Second dial tone

B on hold).

3 Dial Station-#.Ring Back Tone on B. C is

ringing.

4 A presses Flash-0. A and B and C are talking.

5 A presses Flash-1.For conference toggle only A

and B are takling.

• Event 1: When A and B are in communication, A puts B on hold

then dials C’s number plus the pound key #.• Event 2: After A and C are in communication, press Flash-0 button

again to initiate the 3 party call conference.

• Event 3: During conference if A goes ON-Hook, B gets transferred

to C.

• Event 4: Press Flash-1 to separate activated 3 party call

conference into two individual connections (One is online the other

is on hold).

Call Hold (For Type 1):

• Caller can use hook flash key to hold the call.

1 A calls B Talk is ON.

2 A presses Flash. A, hears second dial tone, B

hears nothing. B is on Hold.

3 A can press again

Flash.Talk between A, B.

Programming using phone:

• Call Forward and DND:

Use following commands:



Matrix


Feature Command Description

Set Call Forward-Unconditional

51-SIP Account- Destination Number-#*

Enter SIP Account

number 1 or 2 for ‘SIP

Account’. After issuingthe command user

gets Confirmation

tone.

Cancel Call

Forward-

Unconditional

52-SIP Account-#* -Do-

Set Call Forward-

When Busy

53-SIP Account-

Destination Number-#*

-Do-

Cancel Call

Forward-When

Busy


Set Call Forward-

When No Reply

55-SIP Account-

Destination Number-#* -Do-

Cancel Call

Forward-When No

Reply


Set Call Forward

Ring Timer

57-SIP Account-Timer

(Time in seconds is from

00-99)

Enter value in

seconds for XX, for

how long Ring should

continue. Default=60

secs.

Set DND 61-SIP Account-#*

Use command if DND

feature required on a

SIP Account

Cancel DND 62-SIP Account-#*

Use command if DND

feature is not required

on a SIP Account. 1

for SIP1 and 2 for

SIP2.



Matrix


Call Waiting

• Use following commands to enable/disable Call Waiting beeps on

FXS:

Enable Call Waiting 72-FXS-#*

Disable Call Waiting 73-FXS-#*

Where,

FXS can be FXS1 or FXS 2 and ‘1’ for FXS1 and ‘2’ for FXS2.

Note:

Matrix ATA supports display of Caller ID of the third party’s number

when he is trying to call to a Matrix ATA, which is busy with speech.

This feature requires that the phone connected to ATA, which

supports Call Waiting, should be programmed for offline CLIP.

Caller ID

• Caller ID of the user who is making call can be blocked or sent

using Jeeves as well as Phone.

Using Jeeves: Refer the page FXS settings for programming

Caller ID through Jeeves.

Using Phone: Use following commands to send/block Caller ID:

Send Caller ID 66-FXS-#*

FXS can be1 or 2 as per FXS1

or FXS2 on which this feature is

required.

Block Caller ID 67-FXS-#*

FXS can be1 or 2 as per FXS1

or FXS2 on which this feature is

required.

Cancel all the features

• Many times the user does not get calls and he gets confused

whether the calls are not coming due to network problem or the

setting of some feature on the ATA. In such case, it is convenient

to use some command for resetting any feature setting.

Use the following command to cancel all the features:

99-#*



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• This command cancels all the features set for the SIP account or

the Phone (FXS).

Following features get reset:

• Call Forward-All types get canceled

• DND gets canceled.

Important Points:

• When one SIP account is routed to both FXS ports, for incoming

call:

• If one port is IDLE and other in speech, only port which is IDLE

rings, even when other port has CW enabled.

• If both ports are in speech with CW enable, then both will receive

CW tone for IC call. In case any of the FXS ports rejects the

waiting call, other port doesn’t get CW tone.

• ‘Call Forward-always’ has priority over DND.

Relevant Topics:

1. Time Settings 82

2. FXS Port 34

3. SIP Accounts 60

=X=X=



Matrix


System Name

• The System Name is for identification purpose.

• This screen allows user to change the system name. This feature

helps the user identify the system. This is required when multiple

ATA systems are connected in the same network. Now when

accessing the ATA for configuring it, the name programmed in this

field should appear as the system name.

This shall help SE to confirm that he is programming the correct

system.

• Click on ‘System Name’.


System Name

24 char. Max. may be

ASCII. Default = Matrix

SETU ATA2S

Enter the name to

identify ATA

Relevant Topic:

1. FXS Port 34

=X=X=



Matrix


Time Settings

• Matrix ATA can be configured for correct time based on the Simple

Network Time Protocol (SNTP).

• SNTP is a protocol built on top of TCP that ensures accurate local

time-keeping with reference to radio and atomic clocks located on

the Internet.

• The ATA gets synchronized with the specified NTP server (IP

Address), which is a global server and gets the date and global

time. Based on the time zone, settings, Matrix ATA shows correct

time for that country. For example, if time shown by

ntp1.cs.wisc.edu server for India (Culcutta, Chennai, Mumbai, New

Delhi) is 10:44:50AM, then current time shown for Kabul is09:44:50AM. Because for India GMT is ‘GMT +5.30’ but for Kabul it

is ‘GMT +4.30’ (1 hour behind India). The date is displayed as

17May2006.

• Click on ‘Time Settings’

Field Name Value/Default Description

Current Date DD-Month-YYYY ATA displays current

system Date

Current Time HH:MM:SS ATA displays current

system Time

Time Server

Address

ntpl.cd.wise.edu

time.windows.com

time.nist.gov

Select the suitable

Time server

Time Zone

List of places as per

windows software of

PC. Default =

Calcutta………

Select the place name,

based on country where

ATA is installed

Relevant Topics:

1. FXS Port 34

2. Supplementary Services 70

=X=X=



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VoIP Basics

VoIP is the sending of voice signal over Internet Protocol. Calling on

VoIP allows user to make phone calls and send faxes over the

internet at a negligible cost compared to the normal circuit switchedTelephone network. The main components of VoIP are:

• SIP

• Network Servers

• RTP

• NAT and STUN

• Types of NAT

• STUN

• Outbound Proxy• SIP ALG

• SIP Call Flow

• Important Voice Parameters: Jitter Buffer, Voice Compression and

QoS.

SIP:

• The signaling protocol which is used to establish an IP call is called

SIP-Session Initiation Protocol. It is an application layer protocol

that handles the setting up, altering and tearing down of voice andmultimedia sessions over the internet. Refer RFC 3261 for more

information.

• SIP handles telephone calls and can interfere with PSTN network.

• A SIP account uses complete identity called, URI (Uniform

Resource Identifier) or SIP Address, like e-mail address and is

given by: SIP-Number@SIP service Domain.

• SIP number can use letters or numbers like vrajeshp@ITSP

Provider.com or 73121@ITSP Provider.com. A SIP service domain

is the domain name in the SIP URL. For example, if SIP address is73121@VoIP Provider.com, then ‘VoIP Provider.com’ is the SIP

service domain.

• In general, SIP follows the Client-Server Architecture. To support

this, there are two main entities-user agent and network servers.

The peers in a session are called User Agents (UAS). A user agent

can function in one of the following roles:

• User agent client (UAC)-A client application that initiates the SIP

request.



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• User agent server (UAS)-A server application that contacts the

user when a SIP request is received and that returns a response

subject to user’s input.

Network Servers:There are three types of network servers in a SIP network: proxy

server, redirect server and registrar server.

Proxy Server: A SIP proxy receives a request, makes a determination

about the next server to send it to, and forwards the request, possibly

after modifying some of the header fields. As such, SIP requests can

traverse many servers on their way from UAC to UAS. Responses to

a request always travel along the same set of servers the requestfollowed, but in reverse order.

Redirect Server: The redirect server does not forward requests to the

next server. Instead it sends a redirect response back to the client

containing the address of the next server to contact.

SIP Registrar: SIP Registrar is an entity where SIP users can get

themselves registered. Registrar imparts mobility to the SIP users. A

SIP user can register himself with a registrar. If the user changes hislocation, he has to register again with the registrar stating his latest

contact information. Whenever a call is to be delivered to that user,

Registrar can provide the information about the location where the

user was active recently.

RTP:

After a VoIP call is established using SIP, the RTP (Real Time

Protocol) is used to handle voice data transfer in the packets form.

Refer RFC 1889 for more details.

NAT and STUN:

NAT (Network Address Translation-RFC 1631), is the translation of IP

address of a host in a packet. For example, the source address of an

outgoing packet, used within one network is changed to a different IP

address known within another network.



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For using this feature, two terms are explained as below:

a. NAT

b. STUN

• One network is designated the inside network and the other is the

outside. Typically, a company maps its local inside networkaddresses to one or more global outside IP addresses and unmaps

the global IP addresses on incoming packets back into local IP

addresses. This helps ensure security since each outgoing or

incoming request must go through a translation process that also

offers the opportunity to qualify or authenticate the request or

match it to a previous request.

• NAT also conserves on the number of global IP addresses that a

company needs and it lets the company use a single IP address inits communication with the world.

Types of NAT:

Four types of NAT’s can be used:

• Full Cone NAT-This type of NAT maps all requests from the same

internal IP address and port to the same external IP address and

port. Any external host can send a packet to the internal host only if

the internal host had previously sent a packet through the NAT.

• Restricted Cone NAT-This type of NAT maps all requests from thesame internal IP address and port to the same external IP address

and port. An external host (with IP address X) can send a packet to

the internal host only if the internal host had previously sent a

packet to IP address X.

• Port Restricted Cone NAT-This type of NAT maps all requests

from the same internal IP address and port to the same external IP

address and port. An external host (with IP address X and source

port P) can send a packet to the internal host only if the internal

host had previously sent a packet to IP address X and port P.• Symmetric NAT-In a symmetric NAT, all requests from the same

internal IP address and port to a specific destination IP address

and port are mapped to the same external IP address and port. If

the same host sends a packet with the same source address and

port, but to a different destination, a different mapping is used.

Furthermore, only the external host that receives a packet can

send a UDP packet back to the internal host.



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STUN: (RFC 3489)

Note:

STUN is used if there is a NAT router between ATA and SIP Server

(from ITSP).

• STUN stands for Simple Traversal of UDP over NAT. It is aprotocol, which enables an IP phone to detect the presence and

type of NAT behind which the phone is placed. An IP phone that

supports STUN can intelligently modify the private IP address and

port in its SIP/SDP message by using the NAT mapped public IP

address and port through a series of STUN queries against a

STUN server located on the public Internet.

• This will allow SIP signaling and RTP media to successfully

traverse a NAT without requiring any configuration changes on theNAT. STUN presents a working solution for most NATs that are not

symmetric NAT, e.g., most of the SOHO routers have

non-symmetric NAT and in this case, it is OK to use STUN.

However, STUN does NOT work with symmetric NAT and if your

routers have built-in symmetric NAT, do not use STUN.

• A STUN server can help facilitate traversing through most NATs,

except for symmetric NATs.

Outbound ProxyVoIP service provider can host a SIP outbound proxy server to handle

all of the ATA’s VoIP traffic. User has to enable this feature if VoIP

service provider has a SIP outbound server to handle voice calls. This

allows the ATA to work with any type of NAT router and eliminates the

need of STUN.

SIP ALG

Some NAT routers includes the gateway which allows SIP calls to

pass through NAT by checking and translating IP addressesembedded in the data stream to Public IP addresses. This type of

NAT is known as ‘SIP ALG’, SIP Application Layer Gateways. If ATA is

behind ALG, it eliminates need to use STUN or Outbound Proxy for

NAT traversal.

SIP Call flow:

A simple example of how SIP requests and responses are sent will

help understanding SIP and RTP.



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When A calls B, the steps for the set up and tear down of a SIP call is

explained as below:

A B

1. INVITE2. Ringing

3. OK

4. ACK

5. Voice Data-RTP

6. BYE

7. OK

1. A sends SIP INVITE request to B. This message is an invitation to

B to participate in a SIP telephone call.

2. B sends a response that it is ringing.

3. A answers the call. B responds it by ‘OK’.

4. B also answers the call and A sends acknowledgment by sending

‘ACK’ message.

5. Now, voice session starts between A and B in form of RTP (Real

Time Protocol).

6. After the talk, A hangs up and sends a BYE request.

7. Finally, B sends the reply by sending ‘OK’ response and the call is

terminated.

Important Voice Parameters:

Jitter Buffer:

• The clarity of voice heard from the called party is also distorted due

to non-uniform delay introduced by the Telephony network known

as ‘Jitter’.• In order to reduce the effect of ‘jitter’, some sort of ‘filters’ are used

in the voice-channel called ‘Jitter Buffer’. The ATA has built-in

adaptive buffer that helps to smooth out variations in delay (jitter)

for voice traffic. This will ensure good voice quality. It is not

programmable.

Voice Compression:

• Voice compression is a technique to reduce the band width of voice

channel, using suitable codec or Coder-Decoder. The codec



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converts voice signals from analog form to digital signals and vice

versa. It uses the compression alongwith, CELP.

• Compression-decompression is a voice compression-

decompression algorithm that defines the rate of speech

compression, quality of decompressed speech and processingpower requirements. It is called companding.

• Codec of suitable bandwidth is selected during the speech

reception or transmission, to minimize the digital voice channel

bandwidth.

QoS:

Quality of Service, QoS is the capability of a network to provide better

service to selected network traffic over various technologies, includingEthernet and 802.1 networks, and IP-routed networks. ATA supports

Type of Service tagging and Differentiated Services tagging as

mentioned below: (This allows ATA to tag voice frames so they can be

prioritized over the network).

1. Type of Service (ToS)

2. DiffServe

ToS:

• Network traffic can be classified by setting the ToS values at thedata source (at the ATA) so that a server can decide best method

of delivery, such as the least cost, fastest route etc.

• This parameter allows you to configure Type of Service (ToS) bits

by specifying the precedence and delay of audio and signaling IP

packets.

DiffServe:

DiffServe is differentiated Services, which defines a Class of Service

model that makes packets so that they receive per hope treatment atDiffServe-compliant network services along the route, based on the

application types and traffic flow. Packets are marked with DiffServe

Code Point (DSCP) indicating the level of service desired.

Relevant Topic:

1. Glossary 104

=X=X=



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WAN Explanation

What is this?

• This chapter explains programming aspects of WAN ports, in detail.

If you have already started programming using chapter ‘Getting

Started’, do not use this information. ATA has facility to program

WAN port by normal phone as well as the computer.

• For this, “DHCP server used in the LAN’ is also discussed. For

more details on IP and DHCP refer Glossary and Jeeves topics.

How to program?

Configuring the ATA

• ATA has a WAN port whose IP address is 192.168.001.151

(default).

• The WAN port of ATA can be connected to one of the ports of the

LAN Switch/Hub/Router (in Corporate Offices) OR it can be

connected to the Ethernet port of the Broadband modem (in

SOHO).

• When connected to LAN of the organization, the IP address of the

WAN port should be changed according to the LAN addressing

scheme. The LAN might be using static addresses or dynamic

addresses. If static addresses are used, the SE should take thestatic address of the WAN port from the LAN administrator and

program it as the IP address for the WAN port of ATA.

• If DHCP server is used in the LAN, then DHCP client should be

enabled in the product using the embedded Web server. Doing so,

the DHCP server in the LAN shall assign an IP address

dynamically to WAN port of the ATA whenever ATA is restarted.

• The IP address of the WAN port can be configured using a

computer as well as conventional instrument. (Please note that,

only the IP address of WAN port can be programmed fromconventional Phone and no other parameter/feature of the WAN

port can be programmed from conventional Phone.)

• Also WAN IP address is requested from server if PPPoE server is

enabled and User Name and Password are entered.

• Programming are explained with two options as mentioned below:

• Configuring IP Address of WAN port of ATA using computer

• Configuring IP Address of WAN port of ATA using conventional

phone instrument.



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Configuring the IP address of WAN port of ATA using Computer

• Make the connections as shown in the figure given below:

WAN PortFXS1

FXS2

SETU ATA2S

Computer

• Get the IP address for the WAN port of ATA from the LAN

administrator or ITSP.• Open Web browser (Internet Explorer or Netscape) on the

computer connected to the WAN port of the ATA.

• Enter the default WAN port address (192.168.001.151) in the URL

field and press Enter.

• Embedded Web server takes the System Engineer (SE) to the

Home page.

• The SE has to Log in into the ATA using password (default

password = 1234).

• The SE will have to go to ‘WAN port settings’ link and change theIP address of the WAN port from default to that given by the LAN

administrator. The SE may change the Subnet Mask, Gateway

address and DNS address if required.

• Once the new setting take effect the SE will have to re-enter the

web server with new URL.

Note: Ensure that the subnet of WAN and PC is same.



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Configuring the IP address of WAN port of ATA using

Conventional Phone Instrument

• Make the connections as shown in the figure given below:

WAN PortFXS1

FXS2

SETU ATA2S

• Get the IP address for the WAN port of ATA from the LAN

administrator or ITSP.• Pick up the Handset of the Conventional Phone.

• You get error tone, since you are not registered.

• If you are registered, you shall get dial tone. (This is less likely

when configuring the ATA for the first time)

• Enter the Programming Access Code ‘#19’ followed by password

1234.

• You get programming tone.

• Enter the WAN IP address using following command:

11-WAN IP Address-#* Where,

WAN IP Address is of 12 digits. Each octet is of three digits. If only

single digit is used then it should be converted to three digits and

then entered.

For e.g. To enter IP address 192.168.9.10, enter command

11-192168009010-#*.

• You get confirmation tone for 5 secs. followed by programming

tone.



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WAN Port

Matrix ATA is receiving and transmitting data to the Internet Network

through Wide Area Network (WAN) port.

• There are three ways to obtain WAN IP-address as mentioned

below. Depending upon type of Internet connection, one of this is

used. e.g. if Internet connection is through xDSL (Digital Subscriber

Loop), PPPoE is selected.

• By manually entering a static IP address.

• By configuring the ATA in DHCP mode.

• By configuring the ATA in PPPoE mode.

• Point-to-Point Protocol over Ethernet (PPPoE) defined in RFC

2516, is a means of connecting one premise to Internet ServiceProvider.

• PPPoE is a method of encapsulating data for transmission to a far

point.

• Its advantage is that, it eliminates need for the ITSP to manage the

allocation of IP address.

• If DHCP or PPPoE is enabled, IP address is allocated dynamically

by the respective server and static IP address need not be entered.

• Matrix provides unique MAC address for each ATA, which is

programmed at factory. User cannot change it. It can be in therange of 00.50.C2.55.B0.18 to 00.50.C2.55.B0.1F (HEX Format).

• The registered series of MAC address can have a maximum of 8

different MAC addresses. Hence maximum of 8-ATA units can be

connected in a single LAN.

• The MAC address is used by ITSP to maintain the files of individual

ATA in the TFTP Server.

• Click on ‘WAN’.

WAN Port:

Parameters Value/Default Description

Enable DHCP Default = DisableClick, if DHCP is supported

by the ITSP

Enable PPPoE Default = DisableClick, if PPPoE is

supported by the ITSP



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User Name

Maximum 16

characters. Allow

ASCII characters

Enter Name e.g. Matrix

Password Maximum 16characters. Allow

ASCII characters

Enter Password

Enable Static IP

AddressDefault = Enabled

Selete if static address

available from ITSP

IP Address

N.N.N.N

Default =

192.168.001.151

Ener IP address provided

by SE or ITSP while

Network installation at your

site. Do not enter if DHCP

is enabled.

IP Mask

N.N.N.N.

Default =

255.255.255.0.

Not required to change if

other IP addresses are with

same subnet

GatewayN.N.N.N

Default = BlankProvided by ITSP

IP Address N.N.N.N = 1.0.0.1 To255.255.255.254.

Default = Blank.

Provided by ITSP

Domain Name

Blank (40 char.).

Allow Extended ASCII

set.

Provided by ITSP

DNS:

Relevant Topics:

1. Glossary 104

2. VoIP Basics 833. Supplementary Services 70

=X=X=



Section 3: Appendices



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Appendix A: Troubleshooting

This section describes trouble shooting procedures for the Matrix ATA.

Some most common Problems and Corrective Action are described

below:

Symptom Probable Corrective Action

The Power LED does notlight up

Go through this checklist :

• Unplug the power adapter from the Phone Adapter.Wait for five seconds. Thenplug the power adapter intothe Phone Adapter again.

•

Make sure the power adapter you are using is the oneincluded with the ATA.

• Make sure that power sourceis turned ON.

After Power ON, SIP1, SIP2LEDs are fast blinking

Check that SIP Account isenabled and Registration withSIP Server is successful

After Power ON, SIP1, SIP2

LEDs are slow blinking

Check that on SIP Account page,

SIP1 and SIP2 are not disabledJeeves:

Matrix ATA cannot beaccessed from the ‘WANPort’ using Jeeves

• Make sure that computer’s IPaddress is in the samesubnet as ATA’s IP address.

• Ping the ATA. Click StartÆ Programs Æ AccessoriesÆ Command Prompt. Type‘ping’ in the command

prompt followed by ATA’s IPaddress and ‘Enter’. ATAshould respond. Otherwisemake sure that computer’sEthernet adaptor is ok.

• Make sure that ‘Password’ iscorrectly entered. Otherwisedefault the system.



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Calling via PBX:

Not only SOHO applications, but Enterprises also have wide

applications of Matrix-ATA. This is by way of providing facility to call

PSTN and IP network through the PBX. Refer following figures:

PABX

Internet

MatrixATA-1

MatrixATA-2

LAN

Telephone

Telephone

TelephoneTelephone

Router

Router

LAN

Telephone Telephone

MatrixATA-3

SIP Proxy

PSTN

FXO1 FXO2

=X=X=



Matrix


Appendix C: Product Specifications

Configuration and Capacity:

Application No. of Ports

Connector Type

Maximum FXS

Port

For connecting Analog

Phone or Fax Machine2 RJ 11

WAN Port

(Ethernet)

To connect Ethernet

cable router provided by

the service provider

1RJ 45, 10/100

Base T

DC Jack

To connect the power

adapter for DC supplyvoltage

1 --

VoIP and Networking Protocol:

• MAC Address (IEEE802.3)

• IPv4-Internet Protocol Version 4 (RFC791) Upgradable to V6

(RFC1883)

• ARP-Address Resolution Protocol

• DNS-A record (RFC1706), SRV Record (RFC 2782)

• DHCP Client-(RFC 2131)• ICMP-Internet Control Message Protocol (RFC729)

• TCP-Transmission Control Protocol (RFC793)

• UDP-User Datagram Protocol (RFC768)

• RTP-Real Time Protocol (RFC1889) (RFC1890)

• RTCP-Real Time Control Protocol (RFC1889)

• DiffServe (RFC2475), Type Of Service TOS (RFC791/1349)

• SNTP-Simple Network Time Protocol (RFC 2030)

Voice Functionalities:

• SIP-Version 2

• Two SIP Accounts can be used simultaneously

• DTMF: Inband and Outband (RFC2833)

• Call Progress Tone Generation: Select as per the country

• QoS: RTP QoS and SIP QoS using DiffServ and ToS

(Precedence)

• Full Duplex Audio



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• Echo Cancellation: G.168, with programmable tail length of 8, 16

and 32 msecs.

• Forward error correction (FEC)

• Voice Activity Detection (VAD)

• Fax using T.38 and Pass Through• Speech Volume Setting-Transmit and Receive

• Flash Timer Programming

Voice Codecs: G.711 (a-Law and u-law), G.729, G.723

Telephony features:

• Voice Calls using SIP proxy and Voice calls without using SIP

proxy (Peer-to-Peer Calling)

• Call Waiting and Cancel Call Waiting

• Caller ID Blocking

• Call Forwarding

• 3-Party Conference

• Call Transfer

• Caller ID: Generation and Display

• Calling Party Control (CPC)

• Speed Dialing with Phone Book feature

Time Settings: Synchronizing with specific Time Server

Firewall Features;

• Outbound Proxy

• STUN

• NAT Keep Alive

Provisioning, Administration and Maintenance:

• Auto Configuration• Programming Using Web Page

• Software Upgrade using FTP

• Phone Programming of some parameters like:

• WAN IP Address

• DND



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speech quality similar to 32-kbps ADPCM. It is described in the ITU-T

standard in its G-series recommendations.

HTTP

Hyper Text Transport Protocol-The communications protocol used toconnect to servers on the World Wide Web.

IP

It is the ‘Internet Protocol’, Network layer protocol in the TCP/IP stack

offering a connectionless internetwork-service. IP provides features

for addressing, type-of-service specification, fragmentation and

reassembly and security. It is defined in RFC 791.

MAC

Media Access Control Address - The unique address that a

manufacturer assigns to each networking device.

• MAC is Media Access Control address in the form of 48-bit

number, which is unique to the LAN NIC (network interface card).

• It is programmed into the card at the time of manufacture. IEEE

registration authority administers MAC address scheme for all

LANs which conform to IEEE, 802stds. Including both Ethernet

and token ring.• Consists of two parts: 24-bit company id (manufacturer ID0 and

24-bit Extension id (board id)

• Destination and source MAC names are contained in the header

of the LAN packet and are used by various devices like hubs,

bridges.

• A VoIP Service Provider will typically have its subscribers register

with the Service Provider’s VoIP service server before starting a

subscriber’s service. Often, an ITSP will require the registration of

the MAC addresses of any devices directly connected to their network.

NAT

It is the Network Address Translation. Mechanism for reducing the

need for globally unique IP addresses. NAT allows an organization

with addresses that are not globally unique to connect to the internet

by translating those addresses into globally routable address spaces,

also known as Network Address Translator.



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NAT-Traversal

It is a method of enabling specialized applications, such as Internet

phone calls, video and audio, to travel between your local network

and the Internet. STUN is a specific type of NAT traversal.

NAT Server

It is the Network Address Translation, an Internet standard that

enables a local-area network (LAN) to use one set of IP addresses for

internal traffic and a second set of addresses for external traffic.

NTP

It is the Network Time Protocol. Protocol built on top of TCP that

assures accurate local time-keeping with reference to radio and

atomic clocks located on the Internet. This protocol is capable of

synchronizing distributed clocks within milliseconds over long time

periods.

Ping

Packet Internet Groper-An Internet utility used to determine whether a

particular IP address is online.

Proxy Server It is an intermediary program that acts as both a server and a client

for the purpose of making requests on behalf of other clients.

Requests are serviced internally or by passing them on, possibly after

translation, to other servers. A proxy interprets, and if necessary,

rewrites a request message before forwarding it.

QoS

It is the ‘Quality of Service’, the capability of a network to provide

better service to selected network traffic over various technologies,including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet

and 802.1 networks, SONET, and IP-routed networks that may use

any or all of these underlying technologies. The primary goal of QoS

is to provide priority including dedicated bandwidth, controlled jitter

and latency (required by some real-time and interactive traffic), and

improved loss characteristics.



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VAD

It is the Voice activity detection, when enabled on a voice port or a

dial peer, silence is not transmitted over the network, only audible

speech. When VAD is enabled, the sound quality is slightly degraded

but the connection monopolizes much less bandwidth.

VoIP

It is the Voice over IP, the capability to carry normal telephony-style

voice over an IP-based Internet with POTS-like functionality, reliability,

and voice quality. VoIP enables a router to carry voice traffic (for

example, telephone calls and faxes) over an IP network. In VoIP, the

DSP segments the voice signal into frames, which then are coupled in

groups of two and stored in voice packets.

=X=X=



Matrix


Appendix E: Features at a Glance

Description Feature Code

To enter SE Mode. #19-SE Password To exit SE Mode. 0

To dial a phone book number. #8-Index Number

To default SE Password #***

=X=X=



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Appendix F: System Commands

Meaning Command

To program the WAN IP Address 11-WAN IP Address-#*

To default ATA 21-Reverse SE Password-#*

Set Call Forward-Unconditional51-SIP Account-Destination

Number-#*

Cancel Call Forward-Unconditional 52-SIP Account-#*

Set Call Forward-When Busy53-SIP Account-Destination

Number-#*

Cancel Call Forward-When Busy 54-SIP Account-#*

Set Call Forward-When No Reply55-SIP Account-Destination

Number-#*

Cancel Call Forward-When No Reply 56-SIP Account-#*

Set Call Forward Ring Timer 57-SIP Account-Timer

Set DND 61-SIP Account-#*

Cancel DND 62-SIP Account-#*

Send Caller ID 66-FXS-#*

Block Caller ID 67-FXS-#*

Enable Call Waiting beeps 72-FXS-#*

Disable Call Waiting beeps 73-FXS-#*

Cancel all features 99-#*

=X=X=



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Index

A

Applications-Refer Appendix B: Applications 99

Authentication-Refer SIP Accounts 60 Auto Configuration 29

B

C

Call Progress Tones-Refer FXS Port 34

Caller ID-Refer FXS Port 34

Call Party Control-Refer FXS Port 34Calling Party Number Presentation-Refer FXS Port 34

Call-Routing-Refer SIP Accounts 60

Call Waiting-Refer Supplementary Services 70

Call Hold-Refer Supplementary Services 70

Call Toggle-Refer Supplementary Services 70

Call Transfer-Refer Supplementary Services 70

Call Forward-Refer Supplementary Services 70

Conference-Refer Supplementary Services 70

D

Dial Plan 32

Default-Refer Resetting the ATA 59

Do Not Disturb-Refer Supplementary Services 70

DTMF Option-Refer SIP Accounts 60

Direct IP Calling-Refer Peer-to-Peer Calling 55

E

Echo Cancellation-Refer FXS Port 34

F

Flash Timer-Refer FXS Port 34

Forward Error Correction-Refer FXS Port 34

FXS Port Name-Refer FXS Port 34

FXS Port 34



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G

H

IIncoming Call-Refer FXS Port 34

J

Jeeves 44

K

L

M

MAC Address-Refer WAN Port 93

N

Number Dialing Timer-Refer FXS Port 34

New Password-Refer Password 54

NAT Keep Alive-Refer SIP Accounts 60

O

Outgoing Calls-Refer FXS Port 34

Outbound Proxy-Refer SIP Accounts 60

P

Password 54

Peer-to-Peer Calling 55

Phone Book 58

Polarity Reversal-Refer FXS Port 34Precedence-Refer SIP Accounts 60

Q

QoS-Refer SIP Accounts 60

R

Ring Types-Refer FXS Port 34

Registrar Server-Refer SIP Accounts 60



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Re-Registration Timer-Refer SIP Accounts 60

Resetting the ATA 59

S

SIP Accounts 60SIP ID-Refer SIP Accounts 60

SIP Protocol-Refer Introducing the System 15

Software Upgrade 65

Status 66

STUN-Refer SIP Accounts 60

Supplementary Services 70

System Name 81

T

Tones and Rings-Refer FXS Port 34

Time Settings 82

ToS-Refer SIP Accounts 60

U

Upgrade-Refer Software Upgrade 65

VVAD-Refer Glossary 104

VoIP Basics 83

Volume Control-Refer FXS Port 34

Vocoder-Refer SIP Accounts 60

W

WAN Explanation 89

WAN Port 93

X

Y

Z

=X=X=



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Notes



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R e g i s t e

r o f C h a

n g e s

S.N. Date Major Programming Changes made

Programming Register



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f C h a

n g e s

Programming Register

S.N. Date Major Programming Changes made

setu ata2s v1 system manual

Documents