Setu Ata2ll System Manual

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SETU ATA2LL

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 2 August 2006



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 anyliability assumed for 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, costsor expenses incurred 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 ................................................................ 5Welcome ...................................................................................... 7

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

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

Introducing the System .............................................................. 13

Getting Started ........................................................................... 16

Section 2: Features and Facilities ............................................. 25Dial Plan ..................................................................................... 27

FXS Port .................................................................................... 29

LAN/WAN Explanation ............................................................... 39Jeeves ........................................................................................ 46

LAN Port ..................................................................................... 56

Life Line ...................................................................................... 58

Password .................................................................................... 60

Peer-to-Peer Calling ................................................................... 61

Phone Book ............................................................................... 64

Resetting the ATA ...................................................................... 65

SIP Accounts ............................................................................. 66

Software Upgrade ....................................................................... 71Status ......................................................................................... 72

Supplementary Services ............................................................ 75

System Name ............................................................................ 86

Time Settings ............................................................................. 87

VoIP Basics ............................................................................... 88

WAN Port ................................................................................... 95

Section 3: Appendices ............................................................... 97 Appendix A: Troubleshooting ..................................................... 99

Appendix B: Applications ......................................................... 101

Appendix C: Product Specifications ........................................ 103

Appendix D: Glossary .............................................................. 106

Appendix E: Feature at a Glance ............................................. 112

Appendix F: System Commands .............................................. 113



Index ......................................................................................... 114

Notes .........................................................................................117

Programming Register ...........................................................119



Section 1: Introduction





Matrix

SETU ATA2LL 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, most problems 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 16• Jeeves 46

• Life Line 58

• Peer-to-Peer Calling 61

• Supplementary Services 75

• Resetting the ATA 65

• Status 72

• Time Settings 87



Matrix

SETU ATA2LL System Manual8

• FXS Port 29

• LAN Port 56

• WAN Port 95

• VoIP Basics 88

• SIP Accounts 66• Dial Plan 27

• Phone Book 64

• Software Upgrade 71

• System Name 86

• Troubleshooting 99



Matrix


Packing List

The ideal sales package for SETU ATA2LL is as mentioned below:

Sr. Accessories Qty.01 SETU ATA2LL 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 3

09 RJ45 Cable 2

10 Mounting Template 1

11 SETU ATA2LL CD 1

• Please make sure that these components are present.

• In case of short supply or damage detection, contact the sourcefor 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 and performance 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 theproperty of Matrix. Parts repaired or replaced will be under warranty

throughout 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 the

products’ 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



Matrix


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

terms even 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 WarrantyPolicy 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



Matrix


implied, with respect to Matrix products. Matrix and its distributors,

dealers or 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 originalconsumer purchaser of the Product. Warranty shall be void if the

warranty 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


Introducing the System

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

ATA or Matrix Phone Adaptor (SETU ATA2LL) which is a SIPbased VoIP 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 two IP ports, one for WAN connection and other

for LAN connection.• WAN port is used to connect VoIP-ATA to the Internet using

broadband modem, cable modem, router, etc. The VoIP calls are

made through WAN port.

• LAN port is used to connect a Computer to VoIP-ATA. Doing so,

the user can surf the Net as well as make VoIP calls from the

telephone instrument simultaneously.

• VoIP-ATA has a Lifeline port to which a Trunk line from PSTN can

be connected. However, the Lifeline port is not a true FXO 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 shall help the user to route the calls

to either or both the FXS ports whereas the OG call routing

mechanism shall help the user to route the call through desired SIP

Account which shall be helpful when services from two differentITSPs 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 and

makes a call to the callee.



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

by just dialing feature code and corresponding index number.

• VoIP-ATA uses SIP protocol (RFC 3261) to make Voice calls

over IP.

• User can make a voice call on PSTN also, but using an accesscode (Prefix).

• Emergency calls can be placed on the Lifeline port. (However,

the caller has to dial the Emergency number twice).

• Refer ‘Feature List’ which includes important features supported

by VoIP-ATA.

Feature List:

• Configuration of the ATA through Web-pages

• Reset the ATA

• User Access with Password

• Synchronization with Time Server

• LAN Port IP Address programming

• 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

• Emergency Number Dialing• Speed dialing using Phone Book feature

• Peer-to-Peer Calling

• Status page display of main parameters

Before proceeding further, please ensure that you have an

Internet access and a SIP Account already set up.



Matrix


SETU ATA2LL Photograph:

Relevant Topics:

1. Glossary 1062. VoIP Basics 88

=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:

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 side panel ports

• Front panel LEDs

Side Panel Ports

Following picture shows how the ports are connected to various

interfaces:



Matrix


No. Port Name Connector Description

1 Power DC JackPower Adaptor, 12Volt 1.25Amp.

DC.

2 WANRJ 45 with

two LED’s

Connection to cable modem/

ADSL modem. Ethernet activity

LED (Green) is glowing.

3 LANRJ 45 with

two LED’s

LAN cable connection for co-

located PC.

4 FXS 1 RJ 11To connect analog

telephone/fax.

5 FXS 2 RJ 11To connect analog

telephone/fax.

6Life Line

FXORJ 11

For PSTN bypass/Lifeline

Feature.





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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

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

with the SIP server

Fast BlinkingSIP2 Active but ATA not registered

with the SIP server

Slow Blinking SIP2 Inactive

FXS2

SIP1

SIP2

Where,

OFF : Continuously OFF

ON : Continuously ON

Slow Blink ing : 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 elsewhere in the manual.

• Connect a standard analog telephone instrument to FXS1 and

FXS2 ports, using telephone cables provided.



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• Insert a standard telephone cable into the Life-Line port of ATA

and connect the other end of the telephone cable to a wall jack

for the PSTN line.

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

connect the other end of the Ethernet cable to an uplink port (arouter or a DSL modem), which is used to access internet.

• Insert one end of the Ethernet cable into the LAN port of ATA 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.

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

LAN Port:

• Consult your LAN Administrator to allocate the IP Address for ATA

or read the IP of PC, by clicking on Network Neighborhood →

Properties → TCP/IP Connection → Properties for Windows98.

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

PLEASE CHECK !

The subnet of your ATA IP Address must be

same as of PC to be connected at LAN port

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

should be programmed as 192.168.5.X. where ‘X’ can be any

number from 1 to 254, except 10. By doing this, subnet remains

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 LAN IP Address, using command 12-<LAN IP

Address>-#*. For e.g. to enter IP Address 192.168.1.120. Enter

command 12-<192168001120>-#*.

• You get confirmation tone followed by system restarting.



Matrix


• Start Internet Explorer (IE6 with SP2, Service Pack) connected to

the LAN port of the ATA. Type in the IP Address programmed

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

password ‘1234’.

WAN Port:

• Click on WAN Port from menu on left side of the page.

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 Nmae

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

different from subnet of LAN IP Address.

• For example, if LAN IP Address is 192.168.A.120, then WAN IP

Address can be 192.168.B.C, where B can be any number from 1

to 254, except A and C can be any number from 1 to 254.

• Now, enter above information in the respective field. If informationof some parameter is not provided, keep the field value as default

or unchanged.

• Click on ‘Submit’. The system starts rebooting.

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

web 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



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• 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), then

use above fields for SIP2 port and enable. It allows you to use

second phone (or Fax) connected at FXS2 port.

• Submit page. If registration is successful, SIP LED will glow

steady.

Note:

The computer need not be dedicated tothe ATA, after installation is complete.

• 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!

• Refer ‘Troubleshooting’ in the Appendix section or default the

system with the push button switch 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’.

• PSTN calls can be made even if ATA is not registered to ITSP

provider (you get ‘error tone’ in this case).

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

Service Provider supports.• Dial ‘##’ to get PSTN dial tone. If the ATA is powered OFF, you

can access the PSTN directly. Dial PSTN number as usual e.g.

02652630555.

• If Emergency Number is programmed, dialing it gives you the

PSTN dial tone. Then dial the same number again.

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

Calling’ as explained below:



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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 (you get ‘error tone’ in this

case). If Peer-to-Peer is enabled and FXS port number is

programmed, you can call another ATA by dialing 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 ATA:

1 9 7 . 0

0 m m ( 7 . 7

6 I n c h )

=X=X=





Section 2: Features and Facilities





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’.



Matrix


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 29

2. SIP Accounts 66

=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 Echocancellation, 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 theSIP call.

• 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





Matrix


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 andlast column ‘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 thatallows a data network carrying voice traffic over the Internet to

detect 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/DefaultDescript ion (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

Lisr of countries

refer table 3.

Default = India

Select the country where ATA

is installed. Refer table-3 for

the cadence of various ringssupported

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





Matrix


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

Argenti na 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*380 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 Zealand 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 Arabia 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

Country

Dial Tone Ring Back Tone Busy Tone Error Tone



Matrix


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)



Matrix


Relevant Topics:

1. Peer-to-Peer Calling 61

2. SIP Accounts 66

3. Supplementary Services 75

4. Dial Plan 27

=X=X=



Matrix


LAN/WAN Explanation

What is this?

• This chapter explains programming aspects of LAN and WANports, in detail. If you have already started programming using

chapter ‘Getting Started’, do not use this information. ATA has

facility to program both, LAN and WAN ports 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?

• ATA has a WAN port whose IP address is 192.168.10.10 (Bydefault) and a LAN port with IP address 192.168.1.10 (By

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 the static 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 server should

be enabled in the product using the embedded Web server.

Doing so, the DHCP server in the LAN switch shall assign an IP

address dynamically to WAN port of the ATA whenever ATA isrestarted. If required, the IP address assigned dynamically to the

WAN port of ATA by the DHCP server can be known by logging in

the ATA from the LAN port.

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

computer as well as conventional instrument.

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

is enabled and user name and password are entered.

• Programming methods are explained with 4 options as mentioned



Matrix


below:

• Configuring the IP Address of WAN port of ATA, using computer.

• Configuring the IP Address of WAN port using conventional

phone instrument.

• Configuring the IP Address of LAN port of ATA using computer.• Configuring the IP Address of LAN port using conventional

phone instrument.

Configuring the IP address of WAN port of ATA using

Computer

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

WAN PortFXS1

FXS2

LAN PortLifelinePort

SETU ATA2LL

Computer

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

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

computer connected to the LAN port of the ATA.

• Enter the default LAN port address (192.168.1.10) in the URL

field and press Enter. (Note: PC and LAN port should be on same

subnet: 192.168.1.XXX)

• Embedded Web server takes the SE to the Home page.

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

password = 1234).



Matrix


• The SE will have to go to ‘WAN port settings’ link and change the

IP 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.

Note: WAN port and LAN port must be on different subnets.

Configuring the IP address of WAN port of ATA using

Conventional Phone Instrument


WAN PortFXS1

FXS2


SETU ATA2LL

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

administrator.

• 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-#*





Matrix


WAN PortFXS1

FXS2


SETU ATA2LL

Computer

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

administrator.

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

computer connected to the WAN port of the ATA.

• Enter the default WAN port address (192.168.10.10) (if not

changed) in the URL field and press Enter (Note: PC and LAN

port should be on same subnet).

• Embedded Web server takes the SE to the Home page.

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

password = 1234).

• The SE should go to ‘LAN port settings’ link and change the IP

address of the LAN port from default to that given by the LAN

administrator. The SE may change the Subnet Mask if required.

• The SE should note the changed IP address of the LAN port of the ATA so that he can access the Embedded Web Server using

this IP address. (This shall be useful if dynamic addresses are

assigned to the WAN port of the ATA by the DHCP server)

Note: WAN port and LAN port must be on different subnets.

Configuring the IP address of LAN port of ATA using

Conventional Phone Instrument



Matrix



WAN Port FXS1

FXS2

LAN Port

Lifeline

Port

SETU ATA2LL

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

administrator.

• Pick up the Handset of the Conventional Phone.

• You get dial tone.

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

1234.

• You get Programming tone.

• Enter the LAN IP address using following command:

12-LAN IP Address-#*

Where,

LAN 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

12-192168009010-#*.

• You get confirmation tone for 3 seconds followed by programming

tone.

• Enter ‘0’ to get out of programming mode.

• The SE can dial any programming command during confirmation

tone.



Matrix


Important Point:

• When SIP is not registered or Ethernet cable is not connected,

error tone is coming on both FXS ports, but programming by

phone is still possible.

Relevant Topics:

1. Jeeves 46

2. Glossary 106


4. LAN Port 56

5. WAN Port 95

=X=X=





Matrix


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 intended, thereby eliminating issues of associatedcongestion 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 independent

network, 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.

• The default gateway address is usually an interface belonging tothe LAN’s border router.

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 each



Matrix


domain 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 addressesto 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.

• 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 LAN 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 192.168.5.1 is entered at Address Bar on opening

page of Microsoft Internet Explorer .

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









Matrix


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

expires as shown below:

3. Click on ‘SIP Accounts”. Enter SIP ID Numbers 978 and 651

provided by ITSP and enable SIP accounts as shown below:



Matrix


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 be ‘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:



Matrix


Example 2:

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

calls field, as shown below:



Matrix


Example 3:

• To program Ring Type for country, click on FXS port and then

select the name of country from the list of the combo box of the

‘Ring Type’, as shown below:

Relevant Topics:

1. Password 602. Software Upgrade 71

=X=X=



Matrix


LAN Port

The Matrix ATA has two Ethernet ports:

• LAN

• WAN

This chapter describes, LAN Parameters.

• When trying to program the ATA from LAN port, the SE should

ensure that the computer which he is using to configure the ATA

is in the same subnet as ATA. However, this shall be required in

very few cases. The IP of the computer can be changed by going

to the properties of TCP/IP connection. This can be reached

using the path:Network Neighborhood J Properties J TCP/IP connection J

Properties (for Windows-98).

• Normally the MAC address will not be changed. It needs to be

changed only in one case i.e. when more than one ATA are to be

connected to the same LAN.

• Refer Glossary for more details on MAC and Web Jeeves for IP

Address and Subnet Mask.

• Click on ‘LAN’ to program these parameters:





Matrix


Life Line

Introduction:

• This is one of the Ports of the Matrix ATA for interfacing with

PSTN, called the ‘Life Line Port’. It is not a true FXO port.

• The basic purpose of this port is to provide connectivity to the

PTSN in following cases:

• When a number from a list of numbers is dialed. This list can

have maximum of 5 numbers. Generally, Emergency numbers

can be stored by the user in this list. By default, this list is blank.

• When the user wishes to dial a number using PSTN. The user

prefers this option when he is unable to achieve toll quality

speech through IP due to some reasons.• The user has to dial access code ‘##’ to access the Life Line.

The access code is fixed and not programmable.

• It is required to dial emergency number twice to make a call from

PSTN port.

How it works:

• On receipt of the number from the user dialed from FXS port, the

system compares it with the numbers in the number list and if a

perfect match is found it connects the user to the PSTN. Theuser will listen to the PSTN dial tone if number match is found in

emergency number list and dial plan.

• The user will dial the number again to actually dial it on the PSTN.

• Click on ‘Lifeline’

Emergency Numbers Table


Index 1 to 5This is the index number atwhich an emergency number

is stored.

Number

4 digit Maximum is

from 0 to 9.

Default = Blank

Enter emergency number

likeHospital, Fire, etc.





Matrix


Password

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

Password’ after configuring the parameters. 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’


Password 4 digitsMax/1234

Type the default password or the

existing password you use toaccess 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. Thisis because same password should be used to enter SE mode from

SLT.

Relevant Topic:

1. Jeeves 46

=X=X=



Matrix


Peer-to-Peer Calling

What’s this?

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

the same LAN (figure 2) or which can be in another network,which is connected by virtual LAN 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 the Peer-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, 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 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‘Direct IP 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 ATA2LL

SETU

ATA2LL

SETU ATA2LL

SETU ATA2LL

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

Baroda Office Mumbai Office

192.167.1.24

PC Phone

PC PhonePC Phone

PC Phone

Figure 2:

SETU

ATA2LL

SETU

ATA2LL

SETU

ATA2LL

SETU

ATA2LL

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


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 or

as per the Dial Plan.

• 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’.

• 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 66

=X=X=



Matrix


Resetting the ATA

What is this?

• ATA has the provision for changing the password. Hence if you

have forgotten the Password or you may need to default/reset the

system, ‘System Default’ feature is required.

• The system will loose all previous configuration.

How it works?

• Keep pressed the ‘reset’ button, till you see the LED sequence

for reset cycle. (Approx. time for pressing is 3 secs.)

Relevant Topic:1. Getting Started 16

=X=X=









Matrix


STUN:

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

ITSP’s SIP server.

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 40

characters.

By default, STUN Server Address is Blank.

• Server Por t: enter the STUN server’s listening port for SIP. SE

may change this field if the ITSP provides a STUN Server portnumber other than the default.

Valid range is from 1025 to 65535.

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 Cal ls : Select the port on which you want theincoming 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.







Matrix


Status

What’s this?

• ATA user may require to view remotely for LAN, 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 LAN and 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 connectingthe cable instead of restarting the system.

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

displayed:

STATUS

Parameters Display Description

System NameMatrix SETU

ATA2LL

Display the name of

systemVersion/

RevisionVxRy

Display the current version

'x' and Reviosn 'y' of

software

IP Address 192.168.50.96Display the programmed

IP Address

Subnet Mask 255.255.255.0display the programmed

subnet mask

MAC Address 00.50.C2.55.B0.18Display the programmed

MAC Address

IP Address 192.168.1.98

Subnet Mask 255.255.255.0

MAC Address 00.50.C2.55.B0.12

WAN Port:

LAN Port:





Matrix


Public IPN.N.N.N.

N = 001 to 255

When STUN is enabled

'Pulblic IP' is displayed

here.

Relevant Topics:

1. LAN Port 56

2. WAN Port 95

3. SIP Accounts 66

4. System Name 86

=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 thefirst caller 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 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 Call Transfer-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 isheld, 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.





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


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


Call Transfer (for Type 1):

• Call Transfer without consultation (Blind transfer):

Assuming that call party A and B are in conversation. A wants to

Blind Transfer B to C: To transfer incoming call to another phone:

1 Speech ON. A and B under talk.

2 Dial Flash on SLT-A. Feature tone

3 Dial * 1-Station-#.Ring back tone on B

and C is ringing.

4 Note1. B and C talking.

Note 1 is:

(a) A gets confirmation tone followed by Dial tone if transfer issuccessful.

(b) A gets busy tone for 2 seconds followed by speech, if

transfer target is busy.

(c) A gets continuous busy tone, if no response comes from

transfer target.

(d) A gets error tone, if transfer isn’t successful due to either

timeout or invalid codes have been entered.

By default, Matrix Flash key Command

• Call transfer with consultation (Attended Transfer)

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

then dials Number of C plus the pound key #.

Event 2: After A and C are in communication, A goes ON-Hook

and C gets connected to B.

Call Waiting (For Type 1):

1C calls B, when A and B are

talking.

B hears waiting tone, C can hear

ring back tone.

2

B press “Flash 2” or

B press “Flash 3” or just

hang-up

B press “Flash 1”

Reject the second call.

Disconnect first call and answer

second call.

A is on hold and B can answer

to 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


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 ATA2LL.

Enter the name to

identify ATA

Relevant Topic:

1. FXS Port 29

=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 Kabulis 09:44:50AM. 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 DateCurrent 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 places

name, based on

country where ATA isinstalled

Relevant Topics:

1. FXS Port 29


=X=X=





Matrix


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

SIP request.

• 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. Assuch, 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 request followed, 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 getthemselves registered. Registrar imparts mobility to the SIP users.

A SIP user can register himself with a registrar. If the user changes

his location, 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 TimeProtocol) 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.



Matrix


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 addressin its 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

the same 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 theinternal 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





Matrix


SIP Call flow:

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

help understanding SIP and RTP.

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

is explained as below: A B

1. INVITE

2. 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.



Matrix


Voice Compression:

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

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

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 processing

power 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,

including Ethernet 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 the

data 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

at DiffServe-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.





Matrix


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.

• The IP address (Refer Glossary)• Click on ‘WAN’

WAN Port:

Parameters Value/Default Description

Enable DHCP Default = DisableClick, if DHCP is supported

by the ITSP

Enable PPPoE Default = Disable

Click, if PPPoE is

supported by the ITSP

User Name

Maximum 16

characters. Allow

ASCII characters

Enter Name e.g. Matrix

Password

Maximum 16

characters. Allow

ASCII characters

Enter Password



Matrix


Enable Static IP

AddressDefault = Enabled

Selete if static address

available from ITSP

IP Address

N.N.N.N

Default =

192.168.10.10

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

MAC AddressDefault =

00.50.C2.55.B0.18

Matrix registered address.

Change, if multiple ATA are

used in the network.

IP Address

N.N.N.N = 1.0.0.1 To

255.255.255.254.

Default = Blank.

Provided by ITSP

Domain Name

Blank (40 char.).

Allow Extended

ASCII set.

Provided by ITSP

DNS:

Relevant Topics:

1. Glossary 106

2. VoIP Basics 88


=X=X=







Matrix


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 ‘LANPort’ using Jeeves

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

• Ping the ATA. Click StartÆ Programs Æ Accessories Æ 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 with‘Reset’ button



Matrix


Password was changed butnow it isforgotten

No other way, except to restorethe default configuration file byresetting ATA.

Cannot change the

Password

Ensure that ‘retyping of new

Password’ is correct.Telephone port :

No dial tone, Error tone isheard.

• Error tone may be due to‘SIP Account not registered’.Check Phone portconfiguration.

• Ensure that it is the samephone port which is assignedthe SIP Account number

from the ISP.• Make sure the telephone is

plugged in appropriate port,Phone1 or Phone2.

• Verify the SIP ID number,entered is correct andactive/enabled.

Only Error Tone is heardwhen user of any phoneconnected at FXS1 or FXS2, goes OFF-Hook

Check that, on SIP Accountpage, SIP1 and SIP2 are enabledand on FXS port page for OGCalls, SIP1 assigned to FXS1and SIP2 is assigned to FXS2

=X=X=



Matrix


Appendix B: Applications

Typical deployment of “SETU ATA2LL” is explained for following

applications:

1. Making normal VoIP calls

2. Making calls via PBX and connecting multiple ATA

Normal VoIP calls:

• Today the Internet Connection from the service provider is

available at most of the places. Hence, Matrix ATA can be used

widely in a home or small offices, for making and receiving VoIP

calls through the ITSP.

Service Provider Network Operations

PSTNVoIP

Gateway

IP Network

Telephone or Fax Machine

IP Network Access(Modem or Router)

SETU ATA2LL

Telephone or Fax Machine

SETU ATA2LL

IP Network Access(Modem or Router)

• The above figure shows how the IP call is routed to the

destination. The analog phone and the ATA convert the call into

VoIP and sends to the Internet and ITSP’s SIP server through the

Modem or router. The server forwards it to the IP phones through

the Internet or to analog phones through the VoIP Gateway and

PSTN.





Matrix


Appendix C: Product Specifications

Configuration and Capacity:

Application No. of Parts

Connector Type

Maximum FXS

Port

For connecting Analog

Phone or Fax Machine2 RJ 11

PSTN Life Line

PortFor accessing PSTN line 1 RJ 11

LAN PortTo connect PC for

programming ATA1

RJ 45, 10/100

Base T

WAN Port

To connect Ethernet

cable router provides by

the service provider

1RJ 45, 10/100

Base T

Push Button

Switch

To default the system by

pressing a switch1 --

DC Jack

To connect the power

adapter for DC supply

voltage

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)



Matrix


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

• Full Duplex Audio

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

and 32msecs.

• 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

PSTN Life Line Support:

• Send Emergency Call to PSTN• Make PSTN Call by Dialing Prefix Number

• PSTN Relay when Device Power Down

• Receiving PSTN Phone calls

Time Settings: Synchronizing with specific Time Server



Matrix


Firewall Features;

• Outbound Proxy

• STUN

• NAT Keep Alive

Provisioning, Administration and Maintenance:

• Programming Using Web Page

• Software Upgrade using FTP

• Phone Programming of some parameters like:

• LAN IP Address

• WAN IP Address

• DND

LED Indication:

1 LED for Power

1 LED for each FXS Port

1 LED for each SIP Account

Security : Password Protected Administrator

Dimension (W x H x D) : 130x197x45mm (5.12” x 7.76” x 1.77”)

Power Supply:

External Adaptor : [email protected].

Power Consumption : 12W

Environmental:

Operating Temperature : -10°C to 50°C (140° to 122°F)

Operating Humidity : 5-95% RH, Non-Condensing

Storage Temperature : -40°C to 85°C (-40°F to 185°F)

Storage Humidity : 0-95% RH, Non-Condensing

=X=X=



Matrix


Appendix D: Glossary

10BaseT

10-Mbps base band Ethernet specification using two pairs of twisted-pair cabling (Categories 3, 4, or 5): one pair for transmitting

data and the other for receiving data. 10BASET, which is part of the

IEEE 802.3 specification, has a distance limit of approximately 328

feet (100 meters) per segment.

A-l aw

It is an ITU-T companding standard, used in the conversion between

analog and digital signals in PCM systems. A-law is used primarily

in European telephone networks and is similar to the North American μ-law standard. See also companding and μ-law.

CELP

It is the Code Excited Linear Prediction compression. It is the

compression algorithm used in low bit-rate voice encoding. Used in

ITU-T Recommendations G.728, G.729, G.723.

Codec

It is ‘Coder decoder’. In Voice over IP, Voice over Frame Relay, and

Voice over ATM, a DSP software algorithm used to compress/

decompress speech or audio signals.

Companding

Contraction derived from the opposite processes of compression

and expansion. Part of the PCM process whereby analog signal

values are rounded logically to discrete scale-step values on a

nonlinear scale. The decimal step number then is coded in itsbinary equivalent prior to transmission. The process is reversed at

the receiving terminal using the same nonlinear scale. Compare with

compression and expansion. See also a-law and μ-law.

Compression

The running of a data set through an algorithm that reduces the

space required to store or the bandwidth required to transmit the

data set. Compare with companding and expansion.



Matrix


DHCP

It is the Dynamic Host Configuration Protocol. It provides a

mechanism for allocating IP addresses dynamically so that

addresses can be reused when hosts no longer need them.

DNS

It is the Domain Name System. System used on the Internet for

translating names of network nodes into addresses.

Firewall

Router or access server or several routers or access servers,

designated as a buffer between any connected public networks and

a private network. A firewall router uses access lists and other

methods to ensure the security of the private network.

FTP

File Transfer Protocol - A protocol used to transfer files over a TCP/

IP network.

FXO

It is Foreign Exchange Office. An FXO interface connects to the

public switched telephone network (PSTN) central office and is theinterface offered on a standard telephone. Matrix-ATA’s FXO

interface is an RJ-11 connector that allows an analog connection at

the PSTN central office or to a station interface on a PBX.

FXS

It is Foreign Exchange Station. An FXS interface connects directly

to a standard telephone and supplies ring, voltage, and dial tone.

Matrix-ATA’s FXS interface is an RJ-11 connector that allows

connections to basic telephone service equipment and PBXs.

G.711

Describes the 64-kbps PCM voice coding technique. In G.711,

encoded voice is already in the correct format for digital voice

delivery in the PSTN or through PBXs. Described in the ITU-T

standard in its G-series recommendations.



Matrix


G.723

It is an ITU standard for speech codecs that uses the ADPCM

method and provides toll quality audio of 20 and 40 Kbps.

G.729Describes CELP compression where voice is coded into 8-kbps

streams. There are two variations of this standard (G.729 and G.729

Annex A) that differ mainly in computational complexity; both

provide 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 to

connect 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.

MACMedia 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 and24-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



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network.

NAT

It is the Network Address Translation. Mechanism for reducing the

need for globally unique IP addresses. NAT allows an organizationwith 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.

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 andatomic 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





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UDP

User Datagram Protocol-A network protocol for transmitting data

that does not require acknowledgment from the recipient of the data

that is sent.

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=





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Appendix F: System Commands

Meaning Command

To program the WAN IP Address 11-WAN IP Address-#*

To program the LAN IP Address 12-LAN IP Address-#*

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-#*

Block Caller ID (per call) 65-FXS-#*

Send Caller ID (per call) 66-FXS-#*

Send Caller ID (all subsequent calls) 67-FXS-#*

Disable Call Waiting beeps (per call) 71-FXS-#*

Enable Call Waiting beeps (per call) 72-FXS-#*

Disable Call Waiting beeps (all

subsequent call)73-FXS-#*

Cancel all features 99-#*

=X=X=



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Index

A

Applications 101

Authentication 66

B

C

Call Progress Tones 33

Caller ID 34

Call Party Control 34

Calling Party Number Presentation 34Call-Routing 69

Call Waiting 75

Call Hold 75

Call Toggle 75

Call Transfer 75

Call Forward 75

Conference 75

DDial Plan 27

Default 65

Do Not Disturb 75

DTMF Option 68

E

Echo Cancellation 31

Emergency Number 58

F

Flash Timer 29

Forward Error Correction 29

FXS Port Name 29

G

H



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I

Incoming Call 36

J

Jeeves 42

K

L

LAN IP Address 52

Lifeline 58

M

MAC Address 56

N

Number Dialing Timer 35

New Password 60

NAT Keep Alive 69

O

Outgoing Calls 35Outbound Proxy 68

P

Polarity Reversal 34

Peer-to-Peer 35

Q

QoS 69

R

Ring Types 33

Registrar Server 66

Re-Registration Timer 66

S

SIP Account 66

SIP ID 66



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Status 72

SIP Protocol 13

STUN 69

TTones and Rings 33

Time Settings 87

U

Upgrade 71

V

VAD 111

Volume Control 33

Vocoder 67

W

WAN Port 95

X

Y

Z

=X=X=



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Notes



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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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fC h a

n g e s

Programming Register

S.N. Date Major Programming Changes made

Documents

Setu Ata2ll System Manual