Setu Ata1s v1 System Manual

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SETU ATA1S

System Manual





SETU ATA1SSIP-FXS Gateway

System Manual



Documentation Disclaimer

Matrix Comsec reserves the right to make changes in the design or components of the product as engineering and

manufacturing may warrant. Specifications are subject to change without notice.

This is a general documentation for all models of the product. The product may not support all the features andfacilities described in the documentation.

Information in this documentation may change from time to time. Matrix Comsec reserves the right to revise

information in this publication for any reason without prior notice. Matrix Comsec makes no warranties with respect

to this documentation and disclaims any implied warranties. While every precaution has been taken in the

preparation of this system manual, Matrix Comsec assumes no responsibility for errors or omissions. Neither is any

liability assumed for damages resulting from the use of the information contained herein.

Neither Matrix Comsec nor its affiliates shall be liable to the purchaser of this product or third parties for damages,

losses, costs or expenses incurred by the 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

Comsec's operating and maintenance instructions.

Copyright

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 Comsec.

Version 1

Release date: July 30 , 2010



Table of Contents i

Contents

Introduction..................................................................................................................................................... 1

Welcome! ............................................................................................................................................................ 1

About this System Manual .................................................................................................................................. 1

How to Read this System Manual ....................................................................................................................... 2

Knowing SETU ATA1S.................................................................................................................................... 5

The Overview ...................................................................................................................................................... 5

Programming Options ......................................................................................................................................... 8

Applications of SETU ATA1S ............................................................................................................................ 16

Getting Started.............................................................................................................................................. 19

Protecting SETU ATA1S ................................................................................................................................... 19

Installing SETU ATA1S ..................................................................................................................................... 21

Checking the Status .......................................................................................................................................... 35

Using SETU ATA1S....................................................................................................................................... 39

Making Calls ...................................................................................................................................................... 39

Receiving Calls ................................................................................................................................................. 41

Managing Calls ................................................................................................................................................. 42

Call Features ..................................................................................................................................................... 46

Special Features ............................................................................................................................................... 50

Configuring SETU ATA1S............................................................................................................................. 57

Admin Password ............................................................................................................................................... 57

Auto Configuration ............................................................................................................................................ 59

Automatic Number Translation .......................................................................................................................... 62

Date and Time Settings ..................................................................................................................................... 64

Daylight Savings Time Adjustments .................................................................................................................. 66

Debug ................................................................................................................................................................ 71

Dialed Number Table ........................................................................................................................................ 73

Digest Authentication ........................................................................................................................................ 77

FXS Port ............................................................................................................................................................ 79

LAN Port ............................................................................................................................................................ 85

PCAP Trace ...................................................................................................................................................... 87

Peer-to-Peer Call Table .................................................................................................................................... 90

Restart ATA1S .................................................................................................................................................. 92

SIP Trunks ........................................................................................................................................................ 93

Speed Dialing .................................................................................................................................................. 104

Static Routing Table ........................................................................................................................................ 106



ii Table of Contents

Supplementary Services ................................................................................................................................. 107

Supplementary Services of Service Provider .................................................................................................. 109

System Parameters ......................................................................................................................................... 110

User Password ................................................................................................................................................ 117

WAN Port ........................................................................................................................................................ 119

Software Upgrade ....................................................................................................................................... 127

Reinstate the Default Settings ................................................................................................................... 129

Appendix ..................................................................................................................................................... 131

Frequently Ask Questions (FAQs) .................................................................................................................. 131

Acronyms ........................................................................................................................................................ 134

Features at a Glance ....................................................................................................................................... 136

Product Specifications ..................................................................................................................................... 137

System Commands ......................................................................................................................................... 140

Warranty Statement ........................................................................................................................................ 141

Open Source Licensing Terms and Conditions ............................................................................................... 143

Index ............................................................................................................................................................ 149



Matrix SETU ATA1S System Manual 1

CHAPTER 1 Introduction

Welcome!

Welcome to the world of Telecom Solutions. Thank You for choosing SETU ATA1S (VoIP Analog Telephone

Adaptor). This product is designed to give you the highest performance, combined with real ease of use. We hopeyou will make optimum use of this intelligent, intuitive, feature-packed VoIP Phone Adaptor. Please read this

document carefully before installing SETU ATA1S.

About this System Manual

SETU ATA1S System Manual contains detailed information and instructions for installing, configuring and using the

system. This System Manual also contains information on protecting and maintaining the system and steps to be

followed to default the system in case you are unable to configure it properly. We have made the best efforts to let

you know your SETU ATA1S so that it becomes a fun for you to use it.

Intended Audience

This System Manual is aimed at:

Network and System Engineers, who will install, maintain and support SETU ATA1S. It is assumed that they

have some experience in installing VoIP Phone Adaptor and are familiar with VoIP technology.

End Users, the persons/organizations who will actually use SETU ATA1S. They include residential users,

personnel of small and medium businesses, large enterprises, other commercial and public organizations/

institutions.

It is assumed that the End Users have some previous experience in operating VoIP Phone Adaptor. End Users are

not expected to configure the system or program its features. However, it is anticipated that some of them may

have to or want to configure the phone adaptor and program the features. Therefore, this document provides

instructions on installation and configuration of the Phone Adaptor in as lucid manner as possible.

Organization of this document

This System Manual contains following topics:

Introduction: Gives information about this system manual.

Knowing SETU ATA1S: Helps you to know your SETU ATA1S.



2 Matrix SETU ATA1S System Manual

Getting Started: Contains information on how to install SETU ATA1S.

Using SETU ATA1S: Gives instructions on using your SETU ATA1S.

Configuring SETU ATA1S: Contains instructions for configuring the parameters of SETU ATA1S.

Software Upgrade: Gives you information on how to upgrade the software and configuration files of SETU

ATA1S.

Reinstate the Default Settings: Helps you to default the system when you are stuck somewhere or when

you are unable to configure your system properly.

Appendices: Contain information such as Frequently Asked Questions, Acronyms, System commands etc.

How to Read this System Manual

This System Manual is prepared in such a way that you will find all the information quickly and easily. The table of contents given at the beginning of this System Manual shows all the chapters along with the main topics covered in

that chapter. Page numbers are given opposite to every topic which helps you to refer that particular topic easily.

Also index at the back of this System Manual contains a keyword reference with page numbers which will help you

to find the particular information easily.

Conventions used in this System Manual

Notes: It indicates something that requires your special attention or to remind you of something you need

to do when you are using SETU ATA1S.

• Cross-references: It is given in blue shaded text in the soft copy. You will be able to navigate easilythrough the System Manual with the help of Cross-references.

• Instructions: Instructions to be followed are given in the arrow format as shown below:

For Windows XP, Click on, Control panel, Select Network Connections Right click on 'Local

Area Connection' Properties TCP/IP Connection Properties.

Terminology used in this System Manual

In this manual words 'ATA', 'ATA1S', 'SETU ATA1S', 'System' and 'gateway' are used synonymously. Similarly

'user' and 'you' have same meaning and are referred to the user of SETU ATA1S. Also 'Caller' and 'Calling party'

are same and 'Callee' and 'Called party' are used synonymously.

Some of the terms used in the manual are defined below:

• Admin: A person who installs, configures and maintains SETU ATA1S. Admin can also be the user. Admin

has full accessibility of the system.

• User: A person who uses SETU ATA1S. User has limited access to the system

• Caller/Calling party: A person who makes calls.

• Callee/Called party: A person who receives calls.




• Remote user/Remote party/Remote end: The person with whom the user interacts.

• Transferor: The user who transfers the call.

• Transferee: The remote user whose call is to be transferred (first party) to another remote user (second

party).

• Transfer Target: The remote user to whom the call is to be transferred (second party).

• LAN administrator: A person who maintains the LAN network of an organization.

Using this manual, you will be able to install, operate and make optimum use of SETU ATA1S. In case you face any

technical difficulties, you are advised to contact our dealers for help. All Matrix dealers are properly trained and

ready to give you support whenever needed. In addition to this, technical consultation is also available from Matrix

engineers every business day. For technical consultation from our engineers, you are requested to dial our support

numbers.







CHAPTER 2 Knowing SETU ATA1S

The Overview

ATA1S is a VoIP Analog Telephone Adapter used as a SIP-FXS Gateway. It allows you to make and receive voice

calls over IP Network using conventional telephone instrument.

ATA1S uses SIP (Session Initiation Protocol) protocol to make voice calls over the IP network. ATA1S converts the

voice traffic into IP packets for transmission over the internet. When a telephone number is dialed by you, ATA1S

converts it into an IP call using the Session Initiation Protocol (SIP) and initiates a call, thereby making VoIP calls

as easy as normal telephone call. You have flexibility to make a call by selecting any SIP trunk.

The Incoming call on a SIP trunk can be routed to FXS or to a fixed destination number. The Out going call initiated

from FXS port can be routed from the SIP trunk based on the number dialed.

ATA1S can be used in Residence, SOHO, Corporate Offices, Industries, Workshops, etc. ATA1S can be used to

make voice calls to remotely located branch offices of an enterprise using Internet network.

ATA1S also offers flexibility to make VoIP calls over the Internet and browse the Internet simultaneously using

inbuilt router.

ATA1S has 1 WAN port, 1 LAN port, 1 FXS port, a Power Socket and 5 LEDs.

F X S

RJ11

Computer

P C / LA N

Router

WA N

1 2 V D C 1A ( M

a x )




WAN Port

The WAN port labeled as WAN is used to connect the ATA1S to Internet Network or to the LAN Network. Refer

topic “Applications of SETU ATA1S” for more details.

LAN Port

The LAN port labeled as LAN is used to connect a PC to the ATA1S. LAN port can also be connected to LAN switch

in certain applications. Refer topic “Applications of SETU ATA1S” for more details.

FXS Port

The FXS port labeled as FXS is used to connect telephone instrument to the ATA1S.

Power Socket

A Power Socket labeled as 12V DC is used to power the system using 12VDC, 1.00A power adaptor.

LEDs

The LEDs on the module are as shown in the figure below.

1. ATA1S has five LEDs namely FXS, SIP1, SIP2, SIP3 and POWER. LEDs indicate the status of the ports,

various events occurring on the ports and also the error conditions.

2. The reset sequence: At Power ON, POWER LED will glow Green, continuously whereas all other LEDs

(FXS, SIP1, SIP2 and SIP3) will glow Red for 3 times (250 ms On-time and 250 ms Off-time).

3. On successful completion of initialization cycle, each LED will glow as per the call conditions as described

below:

LED Status Meaning

FXS Continuously ON Indicates that FXS port is either OFF-hook or in speech during IC

Call/OG Call.

Blinks fast 200ms ON,200ms OFF

Indicates that FXS port is in Error state during IC/OG call.

Blinks slow 500ms ON,

500ms OFF

Indicates that FXS port is Ringing for IC Call or Number is being

dialed for OG call (After end of dialing).

Continuously OFF Indicates that FXS port is in ON-Hook.

SIP1 Continuously ON Indicates that SIP1 is enabled and is registered with the SIP server

or 'Active' status of SIP trunk.

Blinks fast 200ms ON,

200ms OFF

Indicates that SIP1 is enabled but not registered with the SIP

server (SIP is configured as proxy).


500ms OFF


server (SIP is configured for P2P calling).

Continuously OFF Indicates that SIP1 is disabled.

SIP3

SIP2

SIP1

FXS

POWER




Remarks:

If you are using SIP Trunks, for proxy calling, registration status for SIP Trunk can only be known from the 'Status

page' of Web Jeeves or issuing command from SLT. Refer appendices “System Commands” for display of status

using command.




200ms OFF



Blinks slow 500ms ON,500ms OFF

Indicates that SIP2 is enabled but not registered with the SIPserver (SIP is configured for P2P calling).





200ms OFF




500ms OFF


server (SIP is configured for P2P calling).


LED Status Meaning




Programming Options

• SETU ATA1S gives flexibility to configure its parameters either (A) Configuring using Conventional Phone

Instrument or (B) Configuring using Web Jeeves. ATA1S supports embedded Web server (http based) so

that the parameters can be configured using GUI based special software called Web Jeeves.

• You will be able to configure only limited Parameters and features through phone. Please refer appendices

“System Commands”.

• Using computer, you can configure all the parameters of ATA1S. To configure the parameters using

computer, configure the IP address and subnet mask of LAN/WAN port of ATA1S using phone and open

the Web Jeeves.

Configuring using Phone

You can configure the IP Address and Subnet Mask of LAN/WAN port and Connection Type of ATA1S using

Conventional Phone Instrument as explained below:

1. Pick up the handset of Analog Phone connected to ATA1S.

2. Dial the programming access code #19 followed by the Admin Password (default is 1234). OR dial the

access code #18 followed by User Password (default is 1234). You will get programming tone.

3. To change the connection type of the WAN port, viz. Static, DHCP or PPPoE, enter command:

15<Code>#* . To enable/disable 'L2 VLAN/CoS' on WAN port, enter command: 16<Code>#* . Where,

code is 0 for disable and 1 for enable.

4. Enter LAN IP address using the command: 12<LAN IP Address>#* . For example to enter IP address

192.168.1.120, enter the command 12<192168001120>#*.

5. Similarly enter Subnet mask of LAN port using command: 14-LAN Subnet Mask-#* .

6. Enter the WAN IP address using the command: 11-WAN IP Address-#* and enter WAN subnet mask

using the command: 13-WAN Subnet Mask-#* .

7. You can know the status of various network parameters on the display of your SLT. You can know the

connection type viz. DHCP, PPPoE or Static, WAN IP address and subnet mask, LAN IP address and

subnet mask, Gateway address and DNS address.

8. Entering programming mode using command #18-1234 or #19-1234 and then enter the command 37-#* to

know the current connection type of ATA1S. ATA1S will display the connection type on SLT display. If theconnection type is programmed as DHCP, you will get display for DHCP. Similarly you will get display as

PPPoE or Static as per type of connection.

9. Similarly, enter the command 31-#* to display the WAN IP address on SLT.

10. Enter the command 32-#* to display the LAN IP address on SLT.

11. Enter the command 33-#* to display the WAN subnet mask on SLT.

12. Enter the command 34-#* to display the LAN subnet mask on SLT.

13. Enter the command 35-#* to display the Gateway address on SLT.




14. Enter the command 36-#* to display the DNS address on SLT.

15. Enter the command 38-SIP Trunk-#* to display the registration status of SIP trunk on SLT. For e.g.

The display will be '001' for 'Disable' status, when the SIP trunk is disabled.

The display will be '002' for 'Active' status, when SIP trunk is registered (for Proxy) or configured for Peer-

to-Peer.

The display will be '003' for 'Registering' status, when SIP trunk is in process of registration.

The display will be '004' for 'Failed' status, when any error condition occurs and no call can be made using

this SIP trunk.

16. Whenever you issue the command to display any parameter on the display of SLT, you need to go on-hook

after issuing the command. You will get 'ring' and ATA1S will display the desired parameters for which

command is issued on SLT. You will again reach the programming mode on going off-hook.

17. When you issue valid command in programming mode, ATA1S plays confirmation tone for 3 seconds

followed by programming tone. To exit programming mode, enter '0'. You get dial tone if network

parameters are not changed or else the ATA1S will restart without dial tone/confirmation tone.

When ATA1S gets IP from DHCP/PPPoE server, you can know these values by displaying it on FXS port

also. Refer appendices “System Commands” for other commands to display other parameters for LAN and

WAN port.

This is how LAN port and WAN port of ATA1S can be configured using Conventional Phone.

Configuring using Computer

To configure the IP Address and Subnet Mask of LAN/WAN port of ATA1S using Web Jeeves follow the steps

given below:

Changing the IP address of LAN/WAN port through LAN port:

1. Connect a computer to the LAN port of ATA1S.

2. Know the IP Address and Subnet Mask of the computer by following the steps given below:

For Windows XP, Go to Control panel, Select Network Connections Right click on 'Local Area

Connection' Properties TCP/IP Connection Properties.

If you are using Operating System other than Windows XP, contact your LAN administrator to determine

the IP address of the computer.

WAN Port FXS

SETU ATA1S

LAN Port






5. Click on 'Admin' and Login into the Home page using the Admin password (default 1234). Alternatively, to

login as User, click on 'User' and enter the user password (default '1234'). You will get Welcome page as

shown in web pages (a), (b) and (c) below:

a. Login as Admin:

b. Login as User:




c. Welcome Page after successful login by Admin:




d. Welcome Page after successful login by User:

Now, 'Admin' or 'User' can click on any link from the Menu provided on left side on the Welcome Page to

start further programming of the web pages.

6. Login into the Home page using the Admin password (default 1234).

7. Go to 'WAN port Parameter' link and change the required parameters there. If the IP Address is changed,

system will restart and you will have to login again!

8.If required, change the parameters of 'LAN port' also.

9. Please note that LAN port and WAN port of ATA1S must be in the different subnet else routing functionality

of ATA1S shall get affected.

10. Click on 'Submit'. The system shall reboot and user will be required to log in again using the new IP

Address.

11. Once the necessary changes are made it is advisable to restore the earlier settings of the computer while

removing the computer.




Changing the IP address of LAN/WAN port through WAN port:

1. Connect a computer to the WAN port of ATA1S.

2. Know the IP Address and Subnet Mask of the computer by following the steps given below:


Connection' Properties TCP/IP Connection Properties.

If you are using Operating System other than Windows XP, contact your LAN administrator to determine


3. Change IP address and Subnet Mask of the computer such that the WAN port of ATA1S and computer

connected to WAN port are in the same subnet.

If you cannot change the IP address and subnet mask of your computer, to change IP address and subnetmask of WAN port of ATA1S, using phone commands.

4. Open Web Jeeves of ATA1S by entering the default WAN port address in the URL field and press 'Enter'.

The default IP address of WAN port of ATA1S is 192.168.001.191 and default subnet mask is

255.255.255.0.

5. Login into the Home page using Admin password (default 1234).

6. Go to 'LAN port' link and make the required changes (after changes are done on this page, the system will

restart and you will have to login again).

7. If required, change the parameters of 'WAN port' also.

8. Please note that LAN port and WAN port of ATA1S must be in the different subnet else routing functionality

of ATA1S shall get affected.

9. Click on 'Submit'. The system shall reboot and user will be required to log in again using the new IP

Address.

You will get display of current Software Version/Revision number and Kernel Date on the Status page.

10. Once the necessary changes are made it is advisable to restore the earlier settings of the computer.

WAN Port FXS

SETU ATA1S

LAN Port




Please refer “Configuring SETU ATA1S” for configuration of other parameters such as SIP Trunks, FXS

port etc.

Admin Password and User Password

Admin is a person who installs, configures and maintains the SETU ATA1S. Admin can also be the User.User is a person who uses the SETU ATA1S.

You can login into the Web Jeeves either using Admin Password or User Password.

Admin Password

Login using Admin Password will allow you to access all the parameters of ATA1S. Default Admin Password is

1234.

User Password

Login using User Password will allow you to access only the limited parameters of ATA1S. Default User Password

is 1234.

User and Admin both can access the pages of Web Jeeves as mentioned in topic 'Reinstate the Default Settings'

(refer chapter “Reinstate the Default Settings”). All remaining pages are accessible only by Admin.




Applications of SETU ATA1S

ATA1S finds its application in following scenarios:

1. Residential Application - Home users use ATA1S to make low-cost calls to their relatives staying abroad.

2. SOHO Application - Small Office owners use ATA1S to make low-cost calls to their overseas customers,

employees, suppliers, etc.

FXS

LAN

ATA1S

IP Network

Proxy

Broadband

Modem/Router

FXS

LAN

PBXWAN

Proxy

IP Network

BroadbandModem/Router

ATA1S




3. Business Application - Corporate Offices use ATA1S to make low-cost calls to their overseas customers,

employees, suppliers, etc.

4. Peer-to-Peer Calls Application - Corporate Offices also use ATA1S to communicate between various

branch offices free of cost.

Hope, now you know your SETU ATA1S!

FXS1

LAN

WAN

IP Network

Switch

Proxy

Broadband

Modem/Router

ATA1S

PBX

FXS

LAN

WAN


IP Network

FXS

LAN

WAN

FXS

LAN

WAN

FXS

LAN

WAN

ATA1SATA1S

ATA1S ATA1S

Location D

Location C

Location A

Location B

Broadband

Modem/Router







CHAPTER 3 Getting Started

Protecting SETU ATA1S

You should take following measures to protect your SETU ATA1S.

Installation Precautions:

• Do not install the system in direct sunlight and the place where there is excessive cold or humid

atmosphere.

• Do not install the system where sulfuric gases are produced and in areas where there are thermal springs

because it may damage the equipment.

• Do not install the system at the places where shocks or vibrations are frequent or strong.

• Do not install the system at dusty places or places where it may come in direct contact with oil or water.

• The system should be installed at the place where enough open space is there for ventilation.

Safety Instructions:

You should always take basic safety precautions to reduce the risk of fire, electric shock and injury to the system as

well as the person using it. Follow the steps given below:

• Read and understand all the instructions given in the manual properly.

• Unplug the product from the wall outlet before cleaning and do not use liquid cleaners. Use only dry and

soft cloth.

• Do not use the product near water i.e. near a bath-tub, kitchen sink or near a swimming pool.

• Do not open the system in power ON condition.

• Do not place the product at the place from where it can fall and serious damage may be caused to the

product.

• Interfacing cables should not touch the exposed power line cable.

• The product should be operated with proper power voltage supply. It is advisable to give stabilized power

to the product.• Do not place the product where power cord can be misused.

• 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 or damage to the system, take the product to the qualified serviceman

when some repair work or servicing is required. Removing covers or opening the system or incorrect

reassembly may cause electric shock when used subsequently.

Unplug the system from the wall outlet and contact the qualified service personnel under the following conditions:

• When the power supply cord or plug is damaged or frayed.

• If liquid has been spilled into the product.

• If the product has been exposed to rain or water.

• If the product does not operate normally by following the operating instructions. Adjust only those controlswhich 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.

• If the product has been dropped or the cabinet has been damaged.

• If the product exhibits a distinct change in the performance.




Installing SETU ATA1S

Before installing SETU ATA1S, verify the package contents. The Sales Kit of SETU ATA1S contains:

Make sure that all the above mentioned components are present when you open the Sales Kit of SETU ATA1S. Incase any of the part is missing or damaged, contact the vendor/ITSP/system administrator from whom you have

purchased the system.

Mounting the ATA1S on wall

• Select a suitable place on the wall for mounting the ATA1S.

• Drill a hole of appropriate size.

• Insert the nail grip in the hole.

• Insert the screw and tighten it leaving the screw head a few millimeters protruded of the wall.

• Check the strength of the nail.

• Hang the ATA1S on the wall.

ATA1S can be used for various applications as shown in the topic "Applications of SETU ATA1S". ATA1S can be

installed in different scenarios as shown below.

Residential Application

A typical home user has an Internet connection for surfing the net. He can purchase an ATA1S to make Voice calls

over the Internet.

The figure given below shows a typical connection made by the home user to surf the Internet before purchasing

ATA1S.

Two Screws M 7/30

Two Screw Grips

Warranty Card Set

SETU ATA1S Adaptor 12V, 1 Amp(Country Specific)

CD containing System Manual,Quick Start and User Card

Quick Start and User Card Ethernet Cable (RJ45)

IP


Computer




How to connect?

After purchasing the ATA1S, the home user should connect it as shown in the figure below:

1. Connect the WAN port of the ATA1S to the Ethernet (LAN) Port of the broadband modem/router using a

straight Ethernet cable.

2. Connect telephone instruments to the FXS port using RJ11 cable.

3. Connect a computer to the LAN port of ATA1S using straight Ethernet cable.

4. Power the ATA1S by connecting the Power Adaptor to the Power jack.

5. At Power ON, the LEDs will glow as per the Table shown under the sub-topic LEDs in 'Knowing your SETU

ATA1S'.

How to configure?

1. To access Jeeves using the computer connected to LAN port of ATA, ensure that the LAN port of ATA1S

and the Computer are in the same subnet. The default IP address of the LAN port of ATA1S is

192.168.002.16 and the subnet mask is 255.255.255.0.

2. Change the IP address of the computer to 192.168.002.x where, x can be any number from 001 to 254

(except 016) and subnet mask of the computer to 255.255.255.0.

3. To know IP address of your computer (to change it if required), follow the steps below:


Connection' Properties Internet Protocol (TCP/IP) Properties.

If you are using Operating System other than Windows XP, contact your LAN Administrator to determine


4. If it is not possible to change the IP address of the computer, change the IP address of the LAN port of

ATA1S using telephone instrument. Please refer topic “Programming Options”.

5. Open Web Jeeves of ATA1S by entering the LAN IP address of the ATA1S in the URL field and press

'Enter'.

6. Login as 'Admin' into the Jeeves of ATA1S using the Admin password (default 1234). You will reach the

Home page.

7. Click on 'WAN Port' link and program parameters such as IP address, Subnet mask, Gateway address,

DNS address etc. ISPs generally use PPPoE or DHCP protocols to provide internet services. Enable

relevant option and configure its parameters. If your ISP provides Static IP address, then select "Enable

Static IP" option and configure the relevant parameters.

IP WAN Port FXS

SETU ATA1S

LAN Port

Broadband

Modem/Router




For more details, please refer topic “WAN Port” in Configuring SETU ATA1S.

8. Click on the link 'SIP Trunks' and enter the values of parameters such as Registrar Settings,

Authentication, and Outbound Proxy etc. in the respective fields as provided by your ITSP.

For more details on programming SIP Trunk parameters, please refer topic “SIP Trunks” in Configuring

SETU ATA1S.

9. Ensure that the SIP trunk gets registered to the SIP server by checking the 'Status' page in Jeeves and

respective SIP LED on the product.

10. The set up is ready to make and receive VoIP calls. All incoming calls will land on your phone connected to

the FXS port and the calls will be made from SIP trunk SIP1.

It is recommended to configure your broadband modem/router in 'Bridge Mode'. If doing so, configure the

IP address, Subnet Mask, Gateway Address and DNS address provided by your ISP in WAN port of

ATA1S.

To surf the internet using computer connected to LAN port of ATA, please program the IP address of the

LAN port of ATA1S as the 'Gateway Address' of the computer. Also, program the DNS address in the

computer as provided by the ISP.

SOHO Application

SOHO user purchases ATA1S to make calls to their customers, employees, suppliers etc staying abroad with

minimum cost. While making overseas calls they can also browse the Net simultaneously.

SOHO users generally have a typical set up in their offices as shown in the figure below:

PSTN

PBX

FXO

FXS1

FXS2

FXS3

FXSn

IP

Broadband

Modem/Router




How to connect?

After purchasing the ATA1S, the SOHO users should connect ATA1S as shown in the figure below:

Figure 2:

1. Connect the WAN port of the ATA1S to the Ethernet (LAN) Port of the broadband Modem/Router using a

straight Ethernet cable.

2. Connect LAN port of ATA1S to the Computers via Switch/Hub.

3. Connect FXS port of ATA1S to the FXO port of PBX using RJ11 cable.


5. At Power ON, the LEDs will glow as per the sub-topic LEDs in 'Knowing your SETU ATA1S'.

How to Configure?

1. To access Jeeves using computer/s connected to LAN port of ATA, ensue that LAN port of ATA1S and the

computers in the LAN should be in the same subnet. Since the computers were already in a LAN network

before connecting ATA1S, it would be inconvenient to change the IP address and subnet of all the

computers in the network. Hence, it would be better to change the IP address and subnet mask of the LAN

port of ATA1S. This can be done using telephone instrument.

Please refer topic “Programming Options”.

2. Please note that the default IP address of the LAN port of ATA1S is 192.168.002.16 and the subnet mask

is 255.255.255.0.

3. If net surfing is also to be done using the computers, please program the IP address of the LAN port of

ATA1S as the 'Gateway Address' of the computer. Also, program the DNS address in the computer as

provided by the ISP.


'Enter'.

IP

WAN

LAN

ATA1S

FXS FXOn

PBX

PSTN FXO2FXO1

FXO3

FXS1

FXS2

FXS3

FXSn

Broadband

Modem/Router




5. Login as 'Admin' into the Jeeves of ATA1S using the Admin Password (default 1234). You will reach the

Home Page.

6. Click on 'WAN Port' link and program parameters such as IP address, Subnet mask, Gateway address,

DNS address etc. ISPs generally use PPPoE or DHCP protocols to provide internet services. Enable

relevant option and configure its parameters. If your ISP provides Static IP address, then select "Enable

Static IP" option and configure the relevant parameters.



Authentication, Outbound Proxy, etc. in the respective fields as provided by your ITSP.


SETU ATA1S.

8. Ensure that the SIP trunk gets registered to the SIP server by checking the 'Status' page in the Jeeves and


9. The set up is ready to make and receive VoIP calls. All incoming calls will land on your phone connected to

the FXS port and the calls will be made from SIP trunk SIP1.

It is recommended to configure your broadband modem/router in 'Bridge Mode'. If doing so, configure the

IP address, Subnet Mask, Gateway Address and DNS address provided by your ISP in WAN port of

ATA1S.

Business Application (In corporate Office)

Corporate Offices use ATA1S to make low-cost calls to their overseas customers, employees, suppliers, etc.

In Offices, set-up before purchasing ATA1S is generally as shown in the figure below:

PSTN

PBX

FXO

FXS1

FXS2

FXS3

FXSn

IP





How to connect?

After purchasing the ATA1S, Corporate users should connect ATA1S as shown in the figure below:

Figure 3:

1. Connect WAN port of ATA1S to the LAN Switch/Network of the Enterprise.

2. Connect FXS port of ATA1S to the FXO port of PBX using RJ11 cable.


4. At power ON, the LEDs will glow as per the sub-topic LEDs in 'Knowing your SETU ATA1S'.

How to Configure?

1. To access Jeeves using computer connected to LAN switch, ensure that WAN port of ATA1S and the

computers in the LAN should be in the same subnet. Since the computers were already in a LAN network

before connecting ATA1S, it would be inconvenient to change the IP address of all the computers in the

network. Hence, it would be better to change the IP address and subnet mask of the WAN port of ATA1S.

This can be done using telephone instrument.


2. Please note that the default IP address of the WAN port of ATA1S is 192.168.001.191 and the subnet

mask is 255.255.255.0.

3. Open Web Jeeves of ATA1S by entering the WAN IP address of the ATA1S in the URL field and press

Enter.


Home Page.

5. Click on 'WAN Port' link and program parameters such as IP address, Subnet mask, Gateway address,DNS address etc.

IP

PSTN

WAN

ATA1S

FXS

FXOn

PBX

FXO1

FXOn-1

FXS1

FXS2

FXS3

FXSn

LAN

Broadband

Modem/Router




Generally LAN administrators use DHCP or Static IP address with their network. Enable relevant option

and configure its parameters.

Program the Broadband modem's or Router's LAN port IP address as Gateway Address in the WAN port

parameter page.



Authentication, Outbound Proxy, etc. in the respective fields as provided by your ITSP.

For more details on programming SIP Trunk Parameters, please refer topic “WAN Port” in Configuring

SETU ATA1S.

7. Ensure that the SIP trunk gets registered to the SIP server by checking the 'Status' page in the Jeeves and


8. The set up is ready to make and receive VoIP calls. By default, all incoming calls will land on your phoneconnected to the FXS port and the calls will be made from SIP trunk SIP1.

In this application LAN port of ATA1S can also be connected to a computer or to one of the port of another

LAN Network.

Peer-to-Peer Call Application

Corporate Offices can use ATA1S to communicate between their branch offices free of cost.

Residential Users can use ATA1S to communicate free of cost with their near and dear ones staying geographically

away from each other.

How to connect?

Corporate Users - Scenario1:

Suppose an organization has one branch office located overseas. The organization wants to establish a

mechanism whereby the calls between the branch office and head office become free. The organization can make

following set-up and configure the ATA1S as explained below.

Figure 4: Corporate Users-Scenario1.

IP Network

Broadband

Modem/Router

Broadband

Modem/Router

WAN

ATA1S

FXO1 FXS

FXO2

FXOn

FXS1

FXS2

FXS3

FXSn PSTN

WAN

ATA1S

LAN

Head Office Branch Office

FXS

PBX FXO




At Head office:

• Connect the WAN port of ATA1S to one of the ports of the LAN switch of the organization as shown in the

figure above.

• Connect the FXS port of the ATA1S to the FXO port of the PBX used by the organization.

• Power the ATA1S by connecting the Power Adaptor to the Power jack.

At Branch office:

• Connect the WAN port of ATA1S directly to the Ethernet port of the Broadband modem/ router.

• Connect the LAN port of ATA1S to the computer.

• Connect the telephone instrument to the FXS port of the ATA1S.


• Corporate Users - Scenario2: (refer fig. 5 - when ATA1S is connected behind the Broadband router NAT

Type 'other than symmetric' at both locations.)

Figure 5: Corporate Users - Scenario 2

At each location:


figure.



Corporate Users - Scenario3: (refer fig.5 - when ATA1S is connected behind theBroadband router with NAT type 'Symmetric' at both the locations.)

At each location:


figure. The ATA1S is located behind the NAT.



Residential Users:

Suppose a family wants to talk frequently to their relatives staying overseas. They can use Peer-to-Peer application

of ATA1S. They need to make following set-up and configure the ATA1S as explained below.

IP Network Broadband

Modem/Router

Broadband

Modem/Router

WAN

ATA1S

FXO1 FXS

FXO2

FXOn

FXS1

FXSn

PSTN PSTN

ATA1S

FXSWAN FXO1

FXO2

FXOn

FXS1

FXSn

PBX PBX

192.168.1.1 192.168.1.2

199.57.74.92102.98.54.69

Location-A Location-B




Fig. 6: Residential Users:

At each location:

• Connect the WAN port of ATA1S directly to the Ethernet port of the Broadband modem/ router.

• Connect the LAN port of ATA1S to the computer.

• Connect the telephone instrument to the FXS port of the ATA1S.


How to configure?

Corporate Users - Scenario1:

Following steps should be done at the head office:

Configure the Broadband Router as follows:

• Forward SIP listening port (e.g., 5060) in Router to IP address of WAN port of ATA1S (for allowing

Incoming Call)

Configure the ATA1S as follows:

1. To access Jeeves using computers connected to LAN switch, ensure that the WAN port of ATA1S and the

computers in the LAN are in the same subnet. The default WAN IP address of ATAT211 is


2. Change the IP address and the subnet mask of the WAN port of ATA1S using telephone instrument.


3. Open Web Jeeves of ATA1S by entering the WAN IP address of the ATA1S in the URL field and press'Enter'.

4. Login as 'Admin' into the Jeeves of ATA1S using the Admin Password (default 1234). You will reach the

Home page.

5. Click on 'WAN Port' link and configure required parameters in the respective fields as mentioned below:

• Select 'Enable DHCP', if you want IP address, Subnet mask and Gateway address to be assigned by

DHCP server automatically.

• Select 'Enable Static IP', if you want to assign IP address, Subnet Mask and Gateway address

manually. Configure all the fields manually if you have selected this option.




6. On the 'WAN Port' page, also keep the default parameter 'Use SIP port fetched using STUN?' as 'YES' (if

NAT type is 'other than symmetric) and program STUN server's address (the STUN server port is '3478',

by default).


7. Click on 'SIP Trunks'. Configure the parameter 'Source Port IP Address' as 'Use IP Address fetch using

STUN'.

8. On 'SIP Trunk' program 'Enable SIP Trunk' for the SIP Trunk which is to be used. Program the parameter

'SIP ID' for this SIP Trunk as '*'.


SETU ATA1S.

9. Click on 'Peer to Peer Call Table' link and program the number (say, 2001) assigned to the FXS port of the

ATA1S located at the branch office. Also program the IP address of broadband router/modem located at

the branch office. If you are comfortable dialing IP addresses directly after lifting the handset. Peer-to-Peer table need not be programmed.

For more details, please refer topic “Peer-to-Peer Call Table” in Configuring SETU ATA1S.

Following steps should be done at the branch office:

Configure the Broadband Router as follows:

• Configure the Router in Bridge mode.

Configure the ATA1S:

1. Ensure that the LAN port of ATA1S and the computer connected to the LAN port are in the same subnet.

The default LAN IP address of ATAT211 is 192.168.002.16 and the subnet mask is 255.255.255.0.

2. Change the IP address of the computer to 192.168.002.x where, x can be any number from 001 to 254

(except 016). Also, change the subnet mask of the computer to 255.255.255.0.


3. If it is not possible to change the IP address of the computer then change the IP address and the subnet

mask of the LAN port of ATA1S using telephone instrument.


4. If net surfing is also to be done using the computer, please program the IP address of the LAN port of

ATA1S as the 'Gateway Address' of the computer. Also, program the DNS address in the computer as

provided by the ISP.


Enter.


Home page.





• Select 'Enable DHCP', if you want IP address, Subnet mask and Gateway address to be assigned by


• Select 'Enable Static IP', if you want to assign IP address, Subnet Mask and Gateway address



8. Click on 'SIP Trunks' and on 'SIP Trunks' program 'Enable SIP Trunk' for the SIP Trunk which is to be used

and program the parameter 'SIP Id' for this SIP Trunk as '*'.

Also program the parameter 'Use Symmetric RTP?' as 'Yes' (for NAT type of Router at head office

'Symmetric').


SETU ATA1S.

9. Click on 'Peer to Peer Call Table' link and program following parameter:

a. Number field with the number (say, 3001) assigned to the FXS port of the ATA1S located at the headoffice.

b. 'Destination Address' with the IP address of broadband router/modem located at the head office. If you

are comfortable dialing IP addresses directly after lifting the handset, Peer-to-Peer table need not be

programmed.

For more details, please refer topic “Peer-to-Peer Call Table” in Configuring SETU ATA1S.

Corporate Users - Scenario2: (refer Fig. 5 - When ATA1S is connected behind theBroadband router with NAT type 'Other than Symmetric' at both locations.)

Following steps should be done at both the Locations:Configure the Broadband Router at both locations as follows:

1. Forward SIP listening port (e.g., 5060 for both Routers) in Router to IP address of WAN port of ATA1S

connected behind the respective router.

Configuring ATA1S:

1. To access Jeeves using computers connected to WAN port of ATA, ensure that the WAN port of ATA1S

and the computers in the LAN are in the same subnet. The default WAN IP address of ATA1S is


2. Change the IP address and the subnet mask of the WAN port of ATA1S (if required) using telephone

instrument.


3. Program following parameters (for both locations):

• Open Web Jeeves of ATA1S by entering the WAN IP address of the ATA1S in the URL field and press

Enter.

• Login as 'Admin' into the Jeeves of ATA1S using the Admin password (default 1234). You will reach the

Home page.

• Click on 'WAN Port' link and enter the values of IP address, Subnet mask, Gateway and DNS address

in the respective fields.

• On the 'WAN Port' page, enable STUN by the parameter 'Use SIP port fetched using STUN?' (default

'YES'). Also program the STUN server address (By default STUN server port is '3478' ).




• Click on 'SIP Trunks' and program 'Enable SIP Trunk' for the SIP Trunk which is to be used. Program

the parameter 'SIP Id' for this SIP Trunk as '*' (for configuring it as P2P Trunk).

• On 'SIP Trunks' configure the parameter 'Source Port IP Address' as 'Use IP Address fetched using

STUN'.

4. Program following parameters (for Location-A):

• Click on 'Peer to Peer Call Table'. Program the number (say, 3001) assigned to the FXS port of the

ATA1S located at Location-B.

• Program the IP Address of broadband modem/router located at the Location-B.

• If the users are comfortable dialing IP addresses directly after lifting the handset, peer-to-peer table

need not be programmed.

5. Program following parameters (for Location-B)

• Click on 'Peer to Peer Call Table'. Program the number (say 2001) assigned to the FXS port of the

ATA1S located at Location-A.

• Program the IP address of broadband modem/router located at the Location-A.



Corporate Users - Scenario3: (refer Fig. 5 - When ATA1S is connected behind theBroadband router with NAT type 'Symmetric' at both the locations.)

Following steps should be done at both the Locations:

Configure the Broadband Router at both locations as follows:

1. Forward SIP listening port of client (e.g., 5060 for both Routers) in Router to IP address of WAN port of

ATA1S connected behind the respective router.

2. Forward RTP listening port in Router (to WAN IP address of ATA1S which is connected behind the

respective router).

Configuring ATA1S:

1. To access Jeeves using computers to WAN port of ATA, ensure that the WAN port of ATA1S and the

computers in the LAN are in the same subnet. The default WAN IP address of ATA1S is 192.168.001.191

and the subnet mask is 255.255.255.0.

2. Change the IP address and the subnet mask of the WAN port of ATA1S (if required) using telephone

instrument.


3. Program following parameters (for both Locations):• Open web Jeeves of ATA1S by entering the WAN IP address of the ATA1S in the URL field and press

Enter.

• Login as 'Admin' into the Jeeves of ATA1S using the Admin password (default 1234) you will reach the

Home page.

• Click on 'WAN Port' link and enter the values of IP address, Subnet mask, Gateway and DNS address

in the respective fields.

• On the 'WAN Port' page, also program the parameter 'Use SIP port fetched using STUN?' as 'No' (by

default it is 'YES').

• Click on 'SIP Trunks' and program 'Enable SIP Trunk' for the SIP Trunk which is to be used. Program

the parameter 'SIP Id' for this SIP Trunk as '*' (for configuring it as P2P Trunk).

• On 'SIP Trunks', configure the parameter 'Use Symmetric RTP? as 'Yes' for the SIP Trunk to be used.




4. Program following parameters (for Location-A):


ATA1S located at Location-B.



5. Program following parameters (for Location-B)


ATA1S located at Location-A.

• If the user are comfortable dialing IP Addresses directly after lifting the handset, Peer-to-Peer table

need to be programmed.

Residential Users: (Refer fig. 6)

Following steps should be done at both the locations:

Configure the Broadband Routers at both the locations as follows:

Configure the Routers in Bridge mode.

Configuring ATA1S:


'Enter'.

2. Login as 'Admin' into the Jeeves of ATA1S using the Admin password (default 1234).You will reach the

Home Page.


• Select 'Enable DHCP', if you want IP address, Subnet mask, and Gateway address to be assigned by


• Select 'Enable PPPoE', if your ISP provides internet services using PPPoE. Configure Username,

Password and Service Name provided by your ISP, if you have selected this option.• Select 'Enable Static IP', if you want to assign IP address, Subnet Mask and Gateway address


For more details, please refer topic “WAN Port”.

4. On 'SIP Trunks' program 'Enable SIP Trunk' for the SIP Trunk which is to be used. Program the parameter

'SIP Id' for this SIP Trunk as '*' (for configuring it as P2P Trunk).

5. Program following parameters in ATA1S (for Location-A):


ATA1S located at Location-B, as Destination SIP-ID Number.• Program the IP address of broadband modem/ router located at the Location-B.

• If the users are comfortable dialing IP addresses directly after lifting the handset, Peer-to-Peer table


6. Program following parameters in ATA1S (for Location-B):


ATA1S located at Location-A, as Destination SIP-ID Number.

• Program the IP address of broadband modem/router located at the Location-A.

• If the users are comfortable dialing IP addresses directly after lifting the handset, Peer-to-Peer table





If you are stuck somewhere while configuring you SETU ATA1S, you may default the system as mentioned

in the topic “Reinstate the Default Settings" and start configuration process again.

Configuring ATA1S for Gateway functionality (Call Routing)

FXS to SIP routing:

• You can dial the number from FXS port. ATA will route this call using the SIP1 trunk, in default condition.

• You can also dial the number from FXS port and configure the ATA to route the call to desired SIP trunk;

SIP1/SIP2/SIP3. You can also configure ATA to route the call to the destination route as configured in

Dialed Number Table.

• To achieve this, ATA needs to be configured for the parameter 'OG call route' in the Web page 'FXS Port'

and also configure the dialed number table for desired destination route in Web page 'Dialed Number

Table'.

• Refer topics “FXS Port” and “Dialed Number Table”.

SIP to FXS routing:• By default ATA will route the IC call on SIP trunk, to the FXS port as routing type is 'Route to FXS' (by

default).

• Refer topic “SIP Trunks”.

SIP to Fixed Destination Number:

• The ATA can be configured for the option 'Collect Number and Route to FXS' in the web page 'SIP Trunks'.

Refer topic “SIP Trunks”.

• The caller can also get the dial tone for the IC call on SIP trunk of ATA1S and then dial the desired

destination number of the user of the PBX. (The PBX is interfaced with FXS port of the ATA, and

configured properly for CLI based routing, for routing the call to the destination number).




Checking the Status

SETU ATA1S facilitates you to check whether you have installed it properly or not by displaying the status of all the

important parameters such as LAN Port, WAN Port, SIP Trunks and Auto Configuration on the Status page in the

Web Jeeves and also by indicating port status through LEDs. Refer sub-topic “LEDs” in Knowing SETU ATA1S to

check port status through LEDs.

You can also check Registration status of SIP Trunks on LCD display of the Phone connected to FXS port

by using the command. Refer appendices “System Commands” .

You can get status of Software version and Revision currently used in your ATA1S, from 'Login' page of

the Web Jeeves.

You can check the status of any of these parameters through Jeeves as mentioned below: Open the Web Jeeves

of SETU ATA1S and click on 'Status' link. (Both Admin and User can access this Web page). The Web page will be

as shown below:




Following is a brief description of the parameters displayed on the 'Status' page:

Parameter Description

System Name Displays the Name of System, configured in the 'System Parameter' page.

Version/Revision Displays the version & revision of the software, as VxRy. Where 'x' is the

version number and 'y' is the revision number of the current software.

Kernel Date Displays the date, when the current Kernel in the system was compiled. It will

be displayed in DD-MMM-YYYY formata. e.g. 14-OCT-2009

LAN Port

IP Address Displays the IP address of LAN port currently used.

Subnet Mask Displays the subnet mask currently used.MAC Address Displays the MAC address of LAN port.

WAN Port

IP Address Displays the IP address of WAN port currently used.

Subnet Mask Displays the subnet mask currently used.

Gateway Address Displays the gateway address currently used.

Primary DNS Address Displays the primary domain name server address currently used.

Secondary DNS Address Displays the secondary domain name server address currently used.

DNS Domain Name Displays the DNS domain name currently used.

NTP Address Displays the NTP server address currently used.

MAC Address Displays the MAC address of WAN port currently used.

NAT Type Displays the type of NAT received in STUN response.




Now your SETU ATA1S is ready for use!

Router's Public IP Address. Displays the programmed router's public address.

IP Address fetched using STUN. Displays the IP address fetched using STUN.

SIP port fetched using STUN Displays the port fetched using STUN.

SIP Trunk1

Status Displays the status for registration for the SIP trunk, as Disabled or Registering

or Active or Failedb.

Registration Time Displays the registration time provided by server during registration.

Registration Retry Count Displays the number of attempts after failure, ATA sent registration request.

Reg. Last Failed Reason Displays the messages about the reasons for registration last failed e.g.

'Message send fail', 'Authentication Fail', 'Router's IP address is not

programmed' etc.

SIP Trunk2 Same as for SIP Trunk1.

SIP Trunk3 Same as for SIP Trunk1.

Auto Configuration

Status Displays the status after auto configuration is applied during power ON. e.g.,"TFTP Server not found", "Auto Configured Successfully" etc.

Next Resync Displayed in the form of DD:MM:YY at HH:MM:SS’, where the Date and Timestamp

signifies when the ATA will again resynchronized with the server.

• If Next Resync time is not available or next Auto-Configuration is not

scheduled, this field remains blank.

Last Resync Displayed in the form of DD:MM:YY at HH:MM:SS’, where the Date and Timestamp

signifies when the ATA was last resynchronized with the server.

• If Last Resync time is not available, this field remains blank.

a. The Kernel Date helps Admin to know whether the Kernel is required to be compiled for new version, to support any new featurerelated to Kernel change (e.g. L2 VLAN) . Admin can decide to send the ATA to Factory, in case Kernel is required to be compiled.

b. The status for registration as 'Active' will be displayed for following cases:

When Proxy SIP trunk is enabled, registered and 'Send REGISTER?' flag is set to 'Yes' or

When Proxy SIP trunk is enabled, not registered and 'Send REGISTER?' flag is set to 'No'

or

When P2P trunk is enabled (irrespective of 'Send REGISTER?' flag).

Refer topic “SIP Trunks” for more details of parameter 'Send REGISTER?'

Parameter Description







CHAPTER 4 Using SETU ATA1S

Making Calls

SETU ATA1S is ready for making calls as soon as the initial configuration is done.

You can make the call by different ways using FXS port as mentioned below:

Making calls by dialing a number from FXS

Go OFF-Hook Dial the number Talk

Making calls using ATA1S is as easy as making a normal telephone call provided the entire necessary

configuration is done.

The calls made from FXS are routed as per 'Outgoing Call Route' selected viz. if 'Outgoing Call Route' is selected

as SIP1, calls made from FXS are routed using SIP1. However, the desired SIP trunk, selected for 'Outgoing Call

Route' should be enabled.

The default settings can be changed by making necessary changes in "FXS Port" link in the Web Jeeves. For more

details, refer topic “FXS Port” in Configuring SETU ATA1S.

For dialing number from SIP and configure routing, refer topic “SIP Trunks” in configuring SETU ATA1S.

Making calls by selecting the trunk

For dialing number from SIP1:

Go OFF-Hook Dial #84 Dial the number Talk

For dialing number from SIP2:Go OFF-Hook Dial #85 Dial the number Talk

For dialing number from SIP3:

Go OFF-Hook Dial #86 Dial the number Talk

Using this feature you can make OG calls using specific trunk. You can select the required trunk (SIP1/SIP2/SIP3)

by dialing corresponding Access Code and then making call by dialing the number, irrespective of the port set for

'Outgoing Call Route'.

Making Calls using Speed Dialing

Go OFF-Hook Dial #9-Index number Talk




You can dial the number by dialing access code i.e. #9 from FXS, followed by the Index for that number where it is

configured in the speed dial table.

For more details on configuration of speed dialing, please refer topic “Speed Dialing” in 'Configuring SETU ATA1S'.

Making Peer to Peer Calls

Go OFF-Hook Dial number stored in P2P table Talk

Peer to Peer calls are the calls made between two SIP clients without the intervention of a proxy server.

To make Peer to Peer call, you should configure peer to peer table in the Web Jeeves of ATA1S. Also, you need to

enable the desired 'SIP Trunk', in web page 'SIP trunks', which is to be used as P2P trunk (or keep the registrar

server address for the SIP trunk, as blank).

Peer to Peer application is used by various branch offices of a corporate office and also by the residential user to

communicate with their relatives staying geographically away.

For more details, please refer “Peer-to-Peer Call Application” in 'Installing SETU ATA1S' and “Peer to Peer Calls” in

'Call Features'.

Direct IP Calling: You can make a call from FXS port by dialing 190*1*1*100 for making call to 190.1.1.100. (dot '.'

Should be dialed by using '*').

To avoid delay of Inter Digit Wait timer before the ATA dials the number, you can dial '#' at the end of

dialing digits for the number string while making calls.




Receiving Calls

SETU ATA1S starts receiving calls as soon as it is switched ON after initial configuration.

Ring Go OFF-Hook Talk

Once you have configured your ATA1S properly then receiving calls through ATA is as easy as receiving a normal

telephone call.

If you have configured all the SIP Trunks then the calls made to SIP1 or SIP2 or SIP3 will be placed on FXS as per

the default settings.

SIP: When you receive call on SIP, the number will be further routed to the port as per settings of routing

type. By default, call will be routed to FXS port and you will get 'ring' on the phone connected to the FXS

port. For changing routing type, refer topic “SIP Trunks” in Configuring SETU ATA1S.




Managing Calls

ATA1S gives you the facility of managing the calls by providing various Supplementary call services such as:

• Call Hold

• Making Second Call

• Call Transfer- Blind

• Call Transfer- Attended

• Call Toggle

• 3-Party Conference

• Call Waiting

Call Hold

Call Hold feature allows you to hold the remote party. Maximum two calls can be put on Hold. Ensure that feature is

enabled in Class of Service. Refer “Class of Service” in topic 'FXS Port'.

How it works?

You can hold the call only from FXS Port viz. only the calls made/received from/on FXS port can be held. You canhold all type of calls made/received from/on FXS port.

Call Hold feature will also be used with other features like Making Second Call, Call Waiting, Call Transfer- Blind,

Call Transfer-Attended, Call Toggle and 3-Party Conference.

Remote Held: ATA allows notifying the local user that, remote user has put that local user on hold. It also allows

disconnecting the remote held call on IP.

Making Second Call

This feature helps you to make a second call from FXS, by keeping the first call on 'Hold'. Ensure that 'Call Hold'feature is enabled in Class of Service. Refer “Class of Service” in topic 'FXS Port'.

How it works?

While making second call, the first call which is to be 'held', can be any type of call viz. Incoming and Outgoing

calls received/made from FXS port.

You can make maximum of two simultaneous calls.

A is in speech with B

Hold a call

A dials Flash A gets feature tone. B goes on hold.

Retrieve Held Call

A dials Flash again within feature tone timer (7 seconds)

OR, A can dial 'Flash' within error state.

A in speech with B

A goes on-hook A gets Ring

A goes off-hook A in speech with B


A dials Flash B goes on hold. A gets feature tone.

A dials number of C A in speech with C




Making second call feature is also used with other features such as Call Transfer-Attended, Call Toggle

and 3-Party Conference.

Call Transfer

Call Transfer helps transferor to connect the transferee to the transfer target.

ATA1S will wait for the notification from the remote peer till expiry of 'Transfer Notification Timer'. The 'Transferor'

will be notified about Transfer Failure/ Error or Transfer Successful or Transfer Timeout (Default timer is 60

seconds). Transferor will get confirmation tone if call transfer is successful.

Call Transfer-Blind

Call transfer without consulting the party to whom call is to be transferred is called Blind Transfer. Ensure that

feature is enabled in Class of Service. Refer “Class of Service” in topic 'FXS Port'.

How it works?

Blind Transfer feature is applicable only from FXS port.

To know the call transfer results, 'A' should remain off-hook after dialing C's number. One of the following

results may occur:

• 'A' gets confirmation tone for 3 seconds followed by Dial tone if transfer is successful.

• 'A' gets busy tone for 3 seconds followed by speech with transferee, if transfer target is busy.

• 'A' gets continuous busy tone, if no notification is received from transferee about transfer activity.

Call Transfer-Attended

Call Transfer after consulting the party to whom the call is to be transferred is called Attended Transfer. Ensure that

feature is enabled in Class of Service. Refer “Class of Service” in topic 'FXS Port'.

How it works?

Attended Transfer feature is applicable only from FXS Port.

To know the call transfer results, 'A' should remain off-hook after dialing C's number.

• 'A' gets confirmation tone for 3 seconds followed by Dial tone if transfer is successful. 'A' gets error tone

followed by disconnecting the call, if call transfer is not successful.

Call Toggle

Call Toggle feature allows you to toggle between an active call and a held call without getting disconnected. Ensure

that feature is enabled in Class of Service. Refer “Class of Service” in topic 'FXS Port'.


A dials Flash-#6 B goes on hold. A gets dial tone.

A dials number of C B in speech with C


A dials Flash B goes on hold. A gets feature tone.


A dials Flash-#7 B in speech with C




How it works?

You can perform toggle activity between two IP calls.

Call Toggle feature is applicable only for FXS Port.

3-Party Conference

3-Party Conference enables you to have conference between three parties. During conference all the three parties

can talk to each other. Ensure that feature is enabled in Class of Service. Refer “Class of Service” in topic 'FXS

Port'.

How it works?

You can use Conference feature between two OG Calls or between two IC Calls or between One OG Call and One

IC Call.

When you go on-hook during active Conference, the Conference will be terminated and both the calls will be

disconnected.

Conference feature is applicable only for FXS Port.

3-Party Conference can be converted to Call Toggle by dialing Call Toggle access code. User will be

connected to one of the parties and the other party goes on hold. While initiating the conference, the

ATA1S supports only G.711 (Mu-law) Vocoder.

Call WaitingThis feature notifies you of another incoming call, when you are busy on a call. You will get beeps during speech

when you have waiting call. Ensure that feature is enabled in Class of Service. Refer “Class of Service” in topic

'FXS Port'. For frequency and cadence of the Call Waiting Tone, refer “Call Progress Tone Generation (CPTG)” in

topic 'System Parameters'. You have three options, when you are busy with the current call:

• You can reject the waiting call.

• You can hold the current call and attend the waiting call.

• You can disconnect the current call and attend the waiting call.

If the feature is disabled, you will not get waiting beeps for the waiting call and the call gets rejected.

How it works?

A in speech with B

A dials Flash B goes on hold. A gets feature tone


A dials Flash-#4 A in speech with B. C goes on hold

A dials Flash-#4 again A in speech with C. B goes on hold

A in speech with B

A dials Flash B goes on hold. A gets feature tone


A dials Flash-#2 A in speech with B and C

A dials Flash-#4 A in speech with B. C goes on hold (3-Party Conference converted to Call

Toggle)


C calls A A gets beeps while talking to B




This feature is applicable only for FXS Port.

This feature can be set/cancel through Web Jeeves as well as by issuing commands from FXS port.

Using Web Jeeves

Open the Web Jeeves and click on 'Supplementary Services' link. (Both Admin and User can access this Web

page) The Web page will be shown below:

Call Waiting: Check the box 'Call Waiting' to set/cancel the feature. By default, it is 'set'.

Using Commands

• Enter programming mode by dialing #19-Admin Password or #18-User Password and then issue

following commands from the programming mode:

• To set/cancel Call Waiting, issue command 72-Code-#*

Where,

Code=1 to Set Call Waiting

Code=0 to Cancel Call WaitingBy default it is 'Set'.

Reject waiting call

A dials Flash-#3 Beeps stop. The call with C is rejected. A in speech with B

Hold current call and attend waiting call

A dials Flash-#4 B goes on Hold. A in speech with C

Disconnect current call and attend waiting call

A dials Flash-#5 B gets disconnected. A in speech with C




Call Features

ATA1S supports various call features such as:

• Call Forward

• Caller ID

• Do Not Disturb (DND)

Call Forward

This feature enables you to forward incoming calls to desired number when you are busy or not available at your

place. You can forward all 'incoming calls on SIP trunk' only if the call is placed on your FXS port (routing type is

configured to route the call to FXS).

The ATA1S supports three call forward options:

• Call Forward-Unconditional

Use 'Call Forward-Unconditional' when you want to forward all incoming calls on SIP trunk.

• Call Forward-When Busy

Use 'Call Forward-Busy' when you want to forward the incoming calls on SIP trunk, when you are busy

with other call or off-hook.

• Call Forward-When No Reply

Use 'Call Forward - When No Reply' when you want to forward the incoming calls on SIP trunk only after

specified time. The call will be forwarded to the programmed destination, on expiry of the timer called 'Call

Forward Timer'. You can set different destination numbers for CF-Busy, CF-No Reply and CF-

Unconditional.

This feature can be enabled/disabled through Web Jeeves as well as by issuing commands from FXS port.

Using Web Jeeves

Open the Web Jeeves and click on 'Supplementary Services' link.

(Both Admin and User can access this Webpage). The Web page will be as shown below:




• To enable 'Call Forward' through Web Jeeves, select required Call forward option and enter the number or

IP address on which you want to forward the calls in the field for 'Destination Number'. By default all

options are disabled.

• The number can be of maximum 40 characters including ASCII characters. By default, it is blank.

• Also set the time after which call is to be forwarded to another number in "Call Forward Timer". The range

of Call Forward timer is 01 to 99. By default, it is 45 seconds. This timer is applicable only when Call

Forward- No Reply is selected. Refer topic “Supplementary Services” in Configuring SETU ATA1S.

Using Commands

Enter programming mode by dialing #19-Admin Password (default password 1234) or #18-User Password

(default password 1234) and then issue following commands from the Programming mode:

• To program Call Forward-Unconditional number, issue command 51-Destination Number-#*.

• To set/cancel Call Forward-Unconditional, issue command 52-Code-#* ,

Where,

Code=1 to Set the featureCode=0 to cancel the feature

• To program Call Forward-When Busy number, issue command 53-Destination Number-#* .

• To set/cancel Call Forward-When Busy, issue command 54-Code-#*

Where,

Code=1 to set the feature

Code=0 to cancel the feature

• To program Call Forward-When No Reply, issue command 55-Destination Number-#* .

• To set/cancel Call Forward-When No Reply, issue command 56-Code-#*

Where,

Code=1 to set the feature

Code=0 to cancel the feature

• To set Call Forward Ring Timer , issue command 57-Timer-#*

Where,

Timer is time in seconds from 01-99.

Caller ID

Caller ID helps you to hide your identity from the called party. When the Caller ID is blocked, the outgoing calls will

be sent as Anonymous calls.

Please note that this feature is applicable only for SIP Calls.

Caller ID feature can be enabled/disabled for the SIP Trunk, through Web Jeeves as well as by issuing commands

from FXS port.




Using Web Jeeves

Open the Web Jeeves and click on 'SIP Trunks' link. (Only Admin can access this Webpage). The Web page will be

as shown below:

To enable 'Caller ID' feature through Jeeves, set Send Caller ID = YES for the SIP Trunk.

By default, it is set as 'Yes' for all SIP Trunks.

Using Commands

Enter programming mode by dialing #19-Admin Password (default password 1234) and then issue following

commands from the Programming mode:

• To send/block Caller ID for SIP1, SIP2, SIP3 issue command 66-SIP Trunk-Code-#*

Where,

SIP Trunk is '1' for SIP Trunk1, '2' for SIP Trunk2 and '3' for SIP Trunk3.Code 1 is to Block

Code 0 is to Send the Caller ID.

DND (Do Not Disturb)

DND feature is useful when you do not want to receive any incoming calls. When DND feature is set on FXS port,

all IC calls on FXS port will be rejected by ATA1S, but OG calls can be made as usual.

DND feature is applicable only for IC calls on FXS port.

DND feature can be enabled/disabled through Web Jeeves as well as by issuing commands from FXS port. Ensure

that this feature is allowed from Class of Service in 'FXS Port' web page.




Using Web Jeeves

Open the Web Jeeves and click 'Supplementary Services' link. (Both Admin and User can access this Webpage).

The Web page will be as shown below:

On above page, select 'Set/cancel' to enable/disable DND. Default is 'disable'.

Using Commands

Enter programming mode by dialing #19-Admin Password (default password 1234) or #18-User Password

(default password 1234) and then issue following commands from the Programming mode:

• To set DND, issue command 61-Code-#*

Where,

Code = 1 to set the feature

Code = 0 to cancel the feature

How to allow the feature from COS?

Open the Web Jeeves and click on 'FXS Port' link. (Only Admin can access this Webpage). The Webpage will be

as shown below:

Ensure that Do Not Disturb (DND) is enabled from Class of Service as shown in above web page, to allow DNDfeature.




Special Features

ATA1S supports following special features:

• Peer to Peer Calls

• Multi Stage Dialing through SIP

• Multiple Gateway Application

Peer to Peer Calls

Calls made between two IP clients without the intervention of a Proxy Server are defined as Peer-to-Peer Calls

(P2P calls). P2P call can be made from FXS.

Generally, Peer-to-Peer application is used for making intra-offices calls. Maximum of 500 offices can be interfaced

using ATA1S. Each office should possess ATA1S with a static IP address. The FXS port of ATA can be connected

to the FXO port of the PBX at each branch office as shown in the figure. The Peer-to-Peer table should be

programmed at each location with the 'Number' of FXS port of the PBX and 'Destination Address' of the gateway of all the other locations, to be called.

Broadband Modem,Cable Modem, DSL

Modem, etc.


Modem, etc.

WANFXO1 FXSFXS1

FXS2

FXS3

FXSn

FXS1WAN FXO 1 FXS1

FXS2

FXS3

FXSn

PBX PBX

IP Network

30014001

191.1.3.200 190.1.1.100

Broadband Modem,Cable Modem, DSLModem, etc.


Modem, etc.

WAN

ATA1S

FXO1 FXSFXS1

FXS2

FXS3

FXSn

FXSWAN FXO1 FXS1

FXS2

FXS3

FXSn

PBX PBX

20015001

192.1.3.5 191.1.1.1

LAN

LAN LAN

LAN

Location C

Location D

Location A

Location B

ATA1S ATA1S

ATA1S




How to program?

Open the Web Jeeves and click on the link 'Peer-to-Peer Call Table'. (Only Admin can access this Webpage). The

Webpage will be as shown below:

• This screen shows a Peer-to-Peer table.

• Each entry has following parameters viz. Number, Name, Minimum Digit, Maximum Digit and Destination

Address. Refer topic "Peer-to-Peer Call Table" in Configuring SETU ATA1S for details about these

parameters.

• With reference to the figure shown, following programming should be done in the Peer-to-Peer table of

ATA1S at Location-A.

• When the user 2001 at Location-A dials 3001, a call is placed at IP address 190.1.1.100 since dialed

number gets best matched with the entry programmed at index 002. On receiving a call, the ATA atLocation-B routes this call to the called number 3001 interfaced with ATA.

• Similarly the user 2001 at Location-A can dial 4001 to call the ATA at Location-D or can dial 5001 to call

the ATA at Location-C.

Making P2P call:

• For making 'Peer-to-Peer' call at all the locations, the SIP trunk (SIP1/SIP2/SIP3) should be configured as

P2P trunk. This can be done by entering SIP Id provided by ISP or configuring as '*' for that trunk in the

Web page 'SIP Trunks ' (by default '*' is programmed). You should also enable the desired SIP Trunk.

Receiving P2P call:

• To receive IC call on P2P trunk, at least one SIP trunk should be programmed with '*' or SIP Id .The IC call

will be routed to destination route FXS. Refer topic “SIP Trunks” for call routing options.

Index Number Name Minimum Digit Maximum Digit Destination Address

001 No Match Found 01 24 190.1.1.100

002 3001 01 24 190.1.1.100

003 4001 01 24 191.1.3.200

004 5001 01 24 192.1.3.5




Refer topic “Making Calls” in this chapter, for more information about making P2P call.

Multi Stage Dialing through SIP

There are some applications where user needs to dial a fixed number string before dialing the actual number.The ATA can be configured to take care for dialing the fixed number string every time when the actual number is

dialed through SIP trunk.

For e.g. consider the following scenario: (In this figure, ATA is referred as ATA1S)

• Customer has multiple offices across the globe which he wants to connect over IP network for voice

calling. But existing voice infrastructure which includes PCM/ TDM PBX does not support VoIP

connectivity. To integrate VoIP functionality in existing voice infrastructure, customer installs ATA1S as

shown in the figure with existing PCM/ TDM PBX.

• Customer has installed IPPBX on public IP network to which all ATA1S are registered. This is to take care

of NAT traversal and for easy maintenance.

• Now the voice communication over IP between the offices takes place as follows:

a. If PBX1 extension 101 wants to connect to extension 102 of PBX2, caller first dials 22 (22 is the SIP-ID

assigned to ATA2 where caller wants to reach) with some trunk access code. The PBX1 routes this call

to its FXO port which is connected to the FXS port of ATA1.

b. ATA1 then routes this call to IPPBX which is again routed by IPPBX to ATA2.

c. ATA2 places the incoming call to its FXS port as per incoming call logic.

d. When PBX2 receives the incoming call, Auto-Attendant or DID answers the call on its FXO port and

play dial message/ dial tone. Caller then dials the desired called party number viz. 102. PBX places the

call to its extension viz. 102.

FXS2

FXS1

PBX1

FXS3

FXS4

101

102

103

104

FXO3

FXO2

FXO1

FXS2

FXS1

FXS3

FXS4

101

102

103

104

PBX2

FXO2

FXOn

FXO1

FXS2

FXS1

FXS3

FXS4

101

102

103

104

PBX4

FXS2

FXS1

PBX3

FXS3

FXS4

101

102

103

104

IP PBX

ATA1S 1 - 21

ATA1S 2 - 22

ATA1S 3 - 23

ATA1S 4 - 24

IP IP

IP IP

FXO2

FXO1

PSTN

FXO3 FXS

21

ATA1S

FXO1 FXS

23

FXO4

FXO2

PSTN

PSTN

FXS1

22

FXS

24

ATA1S

PSTN

FXOn ATA1S

FXS

FXOn

ATA1S




• To achieve the above calling methods between office sites, caller needs to dial the number in two stages

each time, to reach the extension of other office. In first stage, caller first dials the SIP-ID of ATA which is

followed by the number of extension to be called in second stage.

• However, in the real scenarios, caller only wants to dial the complete number viz. SIP-ID of ATA and called

extension in one stage instead in two stages.

• To avoid dialing the number in two stages by caller, the feature 'Multi Stage Dialing' is supported by ATA1S

where caller only dials the complete number in one stage and ATA1S converts the dialing method in two

stages.

• When this feature is used, ATA1S (at calling end) will accept the complete number from caller and only dial

the number of remote SIP device which is also ATA1S (at called side). When call gets mature, ATA1S (at

caller side) then sends the remaining number in DTMF to the called device.

• While ATA1S is sending the DTMF (actual destination number which is the extension of called PBX), caller

will hear the routing tone or silence depending on the 'Play Routing Tone' flag programmed in 'System

Parameters' page.

• Customer should program following parameters for this feature: (refer 'How to program?' topic to know

which web page should be configured for these parameters).

a. Program Dialed Number List and Substitute Number List

b. Enable Automatic Number Translation (ANT) on desired SIP Trunkc. DTMF ON Time

d. DTMF OFF Time

e. Play Routing Tone flag

f. Pause Timer

How to Program?

This feature can be configured by programming the web pages as explained below:

Open the Web Jeeves of ATA1S:

• Click on the link 'Automatic number Translation' and program Dialed Number List and Substitute Number

List (Refer topic “Automatic Number Translation” to program Dialed Number List and Substitute Number

List). The ATA1S will consider character '~' as Wait for Answer and character '^' as Pause when they are

programmed in the Substitute number List and will provide suitable delay.




• Click on 'SIP Trunks' link to enable Automatic Number Translation (ANT). (Refer chapter “SIP Trunks” to

enable Automatic Number Translation (ANT))

• Program Pause Timer (if required) (Refer topic “SIP Trunks” to program Pause Timer for SIP)

• Program DTMF ON Time and DTMF OFF Time (Program this only if required. Most of the times you do not

need to change the default setting) (Refer topic “SIP Trunks” to program DTMF ON Time and DTMF OFF

Time for SIP)

• Click on 'System Parameter' link and program 'Play Routing Tone?' flag as per your requirement.

• Routing Tone will be played only on FXS port when 'Play Routing Tone' flag is enabled. When caller is

calling using the SIP trunk, routing tone will not be played (user will get silent) on SIP. (Refer topic

“System Parameters” to enable/ disable playing routing tone)

In the above scenario, if extensions of PBX1 want to reach the extensions of PBX2 by dialing 22 + Extension

Number (of PBX2), program the Dialed Number List-01 and Substitute Number List-02 (at calling side ATA

viz.ATA1) as follows: (Refer web page for “Automatic Number Translation”)

Number List INDEX1 .. INDEX24

01 22

02 22~03




• Ensure that 'Automatic Number Translation (ANT)' is enabled on SIP Trunk from where you want to route

the call.• Keep 'DTMF ON and OFF Time' and 'Play Routing Tone?' parameters to default values if you don't have

any specific requirement.

• Now when any extension user of PBX1 dials '22 + Extension Number (of PBX2)' say 22101 (to reach the

extension 101 of PBX2),

• ATA1 will first dial out the ATA2 number viz. '22 and wait for answer' (22~). Caller will only hear routing

tone. (DID/Auto-Attendant must be enabled on the FXO port of PBX at called side viz. PBX2).

• Once the call has been matured, ATA1 will send the remaining number viz. 101 in RFC 2833 (Outband)

to called device viz. ATA2.

• These DTMF digits are then detected by PBX2 on its FXO port and will place the call to extension 101.

• In the above scenario, if you want some delay after call gets mature and before ATA1S sends the

remaining DTMF digits, program '22~^' in Substitute Number List instead '22~' only.• Similar programming is required to be done in each ATA1S connected at other sites.

Multiple Gateway Application

'Static routing table' helps you routing the call between point to point sites (connected through 'MPLS', Frame Relay

etc) and to public internet at the same time. Refer topic “Static Routing Table” for more details.

Generally, Static routing does not need to configure, when ATA1S is connected behind the NAT router. LAN

interface of NAT router acts as default gateway for ATA1S and all calls initiated from ATA1S get routed from LAN

port of NAT router. But, if you have connected multiple offices through MPLS, Frame Relay etc and want to make/

receive Peer to Peer calls between various offices and Proxy calls to public internet at the same time, you need to

configure the Static Routing Table in ATA1S.

As in the above figure, two offices are connected through MPLS or Frame Relay. User wants to make Peer to Peer

calls between two offices through Frame Relay/ MPLS network. At the same time, user also wants to make proxy

calls through public internet. Refer 'How to program?' topic below, to know about the programming of suchconfiguration.

::

06

Number List INDEX1 .. INDEX24

Public

IP

SIP Proxy

A B

59.162.252.82

192.168.1.0/24 192.168.2.0/24

ATA1S

Frame Relay/MPLS

ATA1S




How to program?

Open the Web Jeeves and click on the link 'Static Routing Table'. (Only Admin can access this Webpage). The


• Total 5 entries can be programmed in Static Routing Table.

• Each entry has three parameters viz. Destination Address, Subnet Mask and Gateway Address.

• Destination Address is the address of final destination where the call is to be made. This can be IP address

of called SIP device or network address where called SIP device resides.

• Subnet Mask is the mask to be applied on destination address.

• Gateway Address is the IP address of LAN interface of your router from where you want to route the callfor specific destination. This address must be in the same subnet where calling ATA1S resides.

By programming this table, ATA1S can route the calls on the basis of destination address. Calls for

specified network can be routed to specific gateway.

• With reference to the above figure, following programming should be done in the Static Routing Table of

ATA1S at location A.

• Only one entry in Static Routing Table needs to be configured when only one gateway is required to becalled.

• Destination Address viz. 192.168.2.0 specifies the network address of location B.

• Subnet Mask specifies the mask to be applied on destination address.

• Gateway Address viz. 192.168.1.1 specifies the LAN address of router which is used to connect the ATA at

location A with ATA at location B.

• IP address of LAN interface of router which connect location A to public internet should be configured as

Default Gateway as in normal scenario. Refer topic “WAN Port” for more details.

• Doing above programming, all calls made by ATA1S to 192.168.2.0/24 will be routed to router which

connects location A to location B. Whereas, all calls made by ATA1S to address other then 192.168.2.0/24

will be routed to default gateway.

• Similar programming is required in ATA1S at location B to enable calling from location B to location A.

Hope, you know how to use SETU ATA1S now!

Index Destination Address Subnet Mask Gateway Address

1 192.168.2.0 255.255.255.0 192.168.1.1

:

5




CHAPTER 5 Configuring SETU ATA1S

In this chapter, you will learn the configuration of various parameters of ATA1S.

Admin Password

Admin is a person who installs, configures and maintains the SETU ATA1S. Logging into Jeeves using Admin

Password will allow you to access all the parameters of ATA1S.

Thus, the system allows you to change or modify or default the Admin Password.

How to program?

Open the Web Jeeves and click on 'Admin Password' link. (Only Admin can access this Webpage). The Webpage

will be as shown below:

Change the Admin Password using Jeeves:

You can change Admin Password by following the steps given below:

• In 'Old Password' field enter the current password; enter new password in 'New Password' field and retype

new password in 'Re-Enter to Confirm' field.

• Password can be of maximum 4 digits. Only digits 0-9 are allowed.

• Whenever Admin changes the Admin Password from Jeeves, the Admin will be logout from the Jeeves as

a confirmation that in ATA1S, the password has been changed.




Now, if you forget the password, you will be required to default the password. This can be done by following:

• Default the Admin Password using Access Code

• Default the Admin Password using Jeeves

Default the Admin Password using Access Code:

It is possible to default Admin password using access code. It can be done by following the steps given below:• Dial the access code '#***'.

• You will get dial tone after successful execution of this Access Code.

• This will default the User Password to factory set value.

• Please note that this will default the Admin Password only if, the option 'Default Admin password when

User defaults the ATA?' is set to 'Yes' on Webpage 'System Parameter'. If the option for 'Default Admin

password when User defaults the ATA?' is set 'No', issuing command '#***' will not default the Admin

password. Refer topic “System Parameters” .

Default the Admin Password using Jeeves:

• If Admin wants to default the Admin password, using Jeeves, he can do so by clicking on the link for

'Default the ATA1S'.

• Refer topic “Reinstate the Default Settings”.

Relevant Topics:

1. “User Password” 117

2. “Reinstate the Default Settings” 129

3. “System Parameters” 110




Auto Configuration

Overview

This feature has been designed to support ITSPs, who can generate and store the configuration files of each

ATA1S from a centralised location. Auto-Configuration Server (ACS) serves as the centralised location which

can be configured using either TFTP or HTTP.

When ATA1S connects to the IP network and customer starts ATA, it is the Auto Configuration feature which

enables ATA to get automatically configured by retrieving its configuration file from ACS and thus, reducing the

need of customers of any external technical assistance.

To configure Auto-Configuration:

1. Open Web Jeeves of ATA1S. Login as an Admin.

2. Click Auto-Configuration link. This opens the Auto-Configuration page as shown below.

3. As shown in above figure, six options are available for Auto-Configuration: Never , At Each Power ON, At

Next Power ON, Resync Periodically, At Each Power ON & Resync Periodically and At Next Power

ON & Resync Periodically. Optionally, select any of the options required for Auto-Configuration of ATA1S.

• Never: Select this option if you do not want Auto-Configuration process to be done during power ON.

• At Each Power ON: Select this option if you want ATA1S to perform Auto-Configuration at each power

ON.

• At Next Power ON: Select this option if you want ATA1S to perform Auto-Configuration only during

next power ON and once Auto-Configuration is done successfully, the option gets changed to Never .

• Resync Periodically: Select this option when ATA1S does not need to auto configure during power

ON, however, needs to resync periodically with ACS to check whether there is any update in the




configuration or not. If you select this option, then, Auto-Configuration will be performed periodically

after expiry of every Resync Timer.

• At Each Power ON & Resync Periodically: Select this option when ATA needs to auto configure

during power ON and also to resync periodically with ACS to check whether there is any update in the

configuration or not. If you select this option then Auto-Configuration process will be performed at each

power ON and also resync periodically after the expiry of every Resync Timer.

• At Next Power ON & Resync Periodically: Select this option when ATA needs to auto configure only

during next power ON but needs to resync periodically with ACS to check whether there is any update

in the configuration or not. If you select this option then Auto-Configuration process will be performed

only during next power ON and once Auto-Configuration process has been performed successfully,

Auto-Configuration flag gets changed to Resync Periodically option. Later on, ATA will resync

periodically after expiry of every Resync Timer.

Protocol

Auto-Configuration feature supports three types of protocols: TFTP, HTTP and HTTPS. In Protocol

parameter, select the required protocol.

• TFTP: Select this option, if you want to perform Auto-Configuration process using TFTP protocol. TFTP

is the preferred method when Auto-Configuration is to be done in Local Area Network (LAN).

• HTTP: Select this option, if you want to perform Auto-Configuration process using HTTP protocol.

HTTP is the preferred method when Auto-Configuration is to be done in Wide Area Network (WAN).

• HTTPS: Select this option, if you want to perform Auto-Configuration process using HTTP over SSL.

HTTPS is the preferred method when security is the concern during Auto-Configuration.

Server Address

The Auto-Configuration Server Address resembles the IP Address/Domain of ACS where all configuration

files of ATA are stored. ACS address (TFFT/HTTP/HTTPS) can also be retrieved using DHCP.

• If ACS address is configured manually, then, ATA uses the manually configured address for Auto-

Configuration.

• If ACS address is not configured manually and DHCP is configured in WAN Port, then, ATA uses

the TFTP or HTTP address provided by DHCP server.

• In Server Address field, enter the Auto-Configuration Server Addres, of maximum 40 characters.

Config Folder Path

Config Folder Path specifies the path where the configuration files of specific ATA gets stored. When ATA1S

gets auto configured at configured time, ATA1S will send the request for the configuration fi les to the selected

server i.e. TFTP/HTTP/HTTPS server and searches for the files in the configured folder.

• In Config Folder Path field, enter the path of the folder provided by ITSP in which the configuration

files are stored, of maximum 96 characters.

Timers

This parameter displays the timers relevant to Auto-Configuration viz. Request Timeout, Retry Timer and

Resync Timer .

• Request Timeout: Request Timeout is used when ATA tries to connect to the Auto-Configuration

Server for TCP/TLS binding. This timer specifies for how long ATA should wait for successful TCP/TLS

binding.




• In Request Timeout field, enter the required time in seconds. The range of Request Timeout is 01-

99 seconds. The default value is 60 seconds.

• Retry Timer : This timer is used if the connection is not established with Auto-Configuration Server

before the expiry of the Request Timeout. Then, in such case, ATA will try to reconnect automatically

after this time interval.

• In Retry Timer field, enter the required time interval in seconds. The range of Retry Timer is 001-

999 seconds. The default value is 10 seconds.

• Resync Timer: Resync Timer is the time after expiry of which ATA will send the resync request to

check if there is any change in the configuration files. This timer is applicable only when Resync

Periodically or At Each Power ON & Resync Periodically or At Next Power ON & Resync

Periodically options of Auto-Configuration are selected.

• In Resync Timer field, enter the required resync timer in hours. The range of Resync Timer is 001-

999 hours. The default value is 24 hours.

4. Click Submit to save the configurations.




Automatic Number Translation

• Automatic Number Translation (ANT) feature translates the number string dialed by the user to the number

string needed to be dialed on the ITSP (SIP) network.

• ANT is applicable on Destination Port (SIP1 or SIP2 or SIP3) only and when the call is made from FXS

(source port).

• Two types of number Lists are programmed for this feature: Dialed number list and Substitute Number List.

• In the Dialed number list the numbers are programmed which are to be dialed by the user and in the

Substitute number List, the numbers are programmed which will be actually dialed out by ATA while out

dialing the number.

• You can program the Number List with the Number String which is to be dialed. You can program

maximum of 24 such number strings in one Number List, called 'Dialed Number List'. As per your

requirement, you can program different Dialed Number Lists ranging from 1 to 6. Assign this list number as

'Dialed Number List' to SIP trunk for using ANT feature.

• You can program the Number List with the Number String which the ATA will dial out when the user dials

the number programmed in the corresponding Index of the Dialed Number List. You can program

maximum of 24 such number strings in one Number List called 'Substitute Number List'. As per your requirement, you can program different Substitute Number Lists ranging from 1 to 6. Assign this list

number as 'Substitute Number List' to SIP trunk for using ANT feature. Only SIP ID will be replaced using

ANT.

• This feature will be used for ANT applications like, multi-stage dialing. Refer chapter “Using SETU ATA1S”.

To elaborate:

While making international calls over IP, it is usually required to dial the same number as dialed when making call

using PSTN e.g. User dials the number 001 479 29785434 on PSTN to make a call to U.S. User also dials the

same number when he wants to make call over IP as per the services provided by ITSP. But few ITSP restrict user

to dial only country code. This means, the user should only dial 1 479 29785434 to make a call to U.S.

Now, since the user is habitual with dialing a complete number, he may dial the complete number with prefix (00)

and finally, this call will be rejected by ITSP. Hence using ANT feature, the user can manipulate the number string

as required by ITSP to complete the call.

For above example, program the Dialed Number List-1 and Substitute Number List-2, for ANT feature as shown

below:

Assign these List numbers to program the ANT feature in web page 'SIP Trunks' and enable 'ANT'. Thus, '001 479

29785434' will be replaced by '1 479 29785434' when the number is dialed out by ATA using 'Automatic Number

Translation' (ANT).

Number List Index01 Index02 .. Index24

1 001

2 1

:

6




How to Program?

Open the Web Jeeves and click on 'Automatic Number Translation' link. (Only Admin can access this Webpage).


You can program different Number lists from 1 to 6.

Index:

You can program max. 24 number strings in each number list at the Index01, Index02….Index24, respectively.

In the Number List, program the number strings of 24 characters maximum at different Index. ASCII characters are

allowed.

You can prepare different number lists as per your application. e.g. Allowed Number List, Denied Number List,

Dialed Number List and Substitute Number List.

Refer topic “SIP Trunks” for ANT on SIP trunk.

The ATA1S will consider character '~' as Wait for Answer and character '^' as Pause when they are programmed in

the Substitute number List and will provide suitable delay before dialing next digit after this.

These characters are used for multi-stage dialing application. Refer chapter “Using SETU ATA1S” for more details.

By default all the Number Lists are Blank.




Date and Time Settings

The Network Time Protocol (NTP) is commonly used for time and date synchronization needs. NTP is a protocol

designed to synchronize the clocks of computers over a network. For providing actual accurate time and date

ATA1S uses Simple Network Time Protocol (SNTP). It is a simplified adaptation of the NTP, to get time from a

reliable, public Internet Time Server.

The ATA1S allows you to select from three free, reliable public internet time servers run by the University of

Wisconsin-Madison, Microsoft, and the National Institute of Standards and Technology (NIST), to obtain date and

time. You can also configure an NTP time server of your preference, other than these.

These public internet time servers provide time offset from the Greenwich Mean Time (GMT), and you can select

the time according to the time zone of the country you are installing the SETU ATA1S. For instance, if your ATA1S

is installed in India, you can select the time zone for India (GMT+5.30 Calcutta, Chennai, Mumbai, and New Delhi).

The time for India is offset from GMT by +5.30 hours. Similarly, if your ATA1S is installed in Hawaii, select the time

zone for Hawaii, which is offset by GMT -10:00 hours, to set the correct time and date.

How to Program?

Open the Web Jeeves and click on 'Date-Time Settings' link. (Only Admin can access this Webpage). The Web

page will be as shown below.Time Server

Time Server

• You can select one of the following free Internet Time Servers provided in the combo box:

• ntp1.cs.wisc.edu

• time.windows.com

• time.nist.gov

• By default, 'ntp1.cs.wisc.edu' is selected.

OR

• You can set the NTP server address manually, by entering the IP address/domain of any other time server.Select the blank NTP address box and enter the IP address/domain manually.




• Only characters (0-9, A to Z,a to z and .) are allowed and you can also use the combination of alphabets or

numerals separated by "-".

• By default it is blank.

Time Zone

Select the Time Zone where the ATA1S is located, e.g. if you are in Honolulu Hawaii, select '(GMT-10:00) Hawaii'

from the list of time zones provided. The ATA1S will display correct time after considering GMT hours.

By default, '(GMT+05:30) Calcutta, Chennai, Mumbai, New Delhi' is selected.

Current Date: Default = 01January 2009

Current Time: Default = 00:00:00

After clicking on the 'Submit', the page will be refreshed and the time and date will appear in the fields Current Date

and Current Time as per the time zone you have selected.

The fields for 'Current Date' and 'Current Time' cannot be edited. They only display the date and time

according to the Time Zone you have selected. The date is displayed in Date-Month-Year (DD-MM-YYYY)

format, while the current time is displayed in Hour: Minute: Seconds (HH:MM:SS) format.

When ATA1S gets defaulted, time parameters will not be set to default values.




Daylight Savings Time Adjustments

• In many countries clocks are advanced at the beginning of the Daylight Saving Time (DST) and delayed at

the end of Day Light Saving Time. ATA1S supports to maintain the clock accordingly for such changes.

• Program the parameters as per the DST rules of the Country where the ATA1S is installed. Disable

Daylight Saving Adjustment if not applicable in the country.

• DST can be set according to Day-Month Wise or Date-Month Wise.

• DST is changed on a particular day and month every year for some countries. The DST is changed on

exact date and month every year for some other countries and some countries don't follow DST.

• If "Day-Month Wise' is selected, the DST is effective on the programmed day and the month whereas, if

'Date-Month Wise' is selected, the DST is effective on the programmed date and the month.

• The configuration of DST parameters is explained with Examples, after explaining the programming of

DST.

When ATA1S gets defaulted, DST parameters will not be set to default values.

How to program?

Open the Web Jeeves and click on 'Daylight Saving Time Adjustments' link. (Only Admin can access this

Webpage). The Web page will be as shown below:




Forward Time Adjustments:

• Type: Select the type from following options:

• None: Program this option for a country where DST is not applicable.

• Day-Month wise: Program this option only for a country where DST is observed as per Day-Month wise

every year.

• Date-Month wise: Program this option only for a country where DST is observed as per Date-Month wiseevery year.

• Default = None.

Day-Month Wise:

Program 'day' and 'month' on which DST will be applied every year. For this, program Ordinal, Day and Month. For

this option, 'Type' should be selected as 'Day-Month wise'. Default = 1st Sunday January.

• Ordinal: Program 1st, 2nd, 3rd, 4th, or 5th. For example if DST starts on first Sunday of January every

year, then program '1st' here.

When the DST of a particular country starts or ends on the Last Sunday or any other day e.g. the last

Tuesday, last Friday of the month, always set the Ordinal Number as '5th'.

• Day: Program the day on which DST starts. Select from: Sunday, Monday, Tuesday, Wednesday,

Thursday, Friday, Saturday

• Month: Program the month on which DST starts every year. Select the month from January to December

of the year.

• Change Time From: It is the current time settings at which DST will start to change. The time is in HH:MM

(Hours-Minutes) format, where HH=00 to 23 and MM=00 to 59. Default = 00:00.

• To: It is the time to which the DST is advanced. Enter the time in HH:MM (Hours-Minutes). Default =

00:00.

Date-Month Wise:Program 'date' and 'month' on which DST will be applied every year. Default = 01January.

• Date: Program the date on which DST starts. Select from '01' to '31' of the month. For example if first

January is the DST date, then select '01'. Default = 01.

• Month: Program the month on which DST starts every year. Select the month from January to December.

Default = January.

• Change Time From: It is the current time settings at which DST will start to change. The time is in HH:MM

(Hours-Minutes) format. Default = 00:00.

• To: It is the time to which the DST is advanced. Enter the time in HH:MM (Hours-Minutes). Default =

00:00.

Backward Time Adjustments:

• Select this option 'Backward Time Adjustments' to set the time back (i.e. end DST and begin standard

time).

• Follow the same steps described above (for 'Forward Time Adjustments) to set the day/date, month, hours

and minutes except, here you must set these parameters according to the time when DST ends.

If for some country, the Backward or Forwarded Time Adjustments is at 00:00 hours, use previous date

with "from" time = 23:59 and "to" time as required.

The following are different examples for DST details, if you were installing SETU ATA1S, in the European Union,

Chile and Israel:




Example 1:

If you are installing ATA1S in a country in the European Union, as per the European Summer Time, the DST would

start on the last Sunday in March and end on the last Sunday in October each year. Clocks are advanced by one

hour at 01:00 hours GMT at the start of DST and set back by one hour at 01:00 hours GMT when DST ends.

• Go to the option 'Forward Time Adjustments' to advance the time when DST starts.

• Select 'Day-Month Wise' as the 'Type' of forward time adjustment

• Select '5th' as the 'Ordinal'

• Select 'Sunday' as the 'Day'

• Select 'March' as the 'Month'

• Change Time 'from': Hour: 01, Minute: 00

• Change Time 'to'; Hour: 02, Minute: 00

• Now go to the option 'Backward Time Adjustments' to set the time back as DST ends.

• Select 'Day-Month Wise' as the 'Type' of backward time adjustment.

• Select '5th' as the 'Ordinal'

• Select 'Sunday' as the 'Day'

• Select 'October' as the 'Month'

• Change Time 'from': Hour: 02, Minute: 00• Change Time 'to'; Hour: 01, Minute: 00

DST Adjustment in a European Union member country.




Example 2: Santiago, Chile (mainland)

In Santiago, mainland Chile, the DST for 2008-2009 starts from October 12, 2008 at 00:00 hours (midnight), with

clocks turned forward by one hour. DST ends on 15 March, 2009 at 00:00 hours, with clocks turned back by one

hour.




Example 3: Tel Aviv, Israel

The DST changes on the last Friday before the 2nd of April and ends on the Sunday that falls between the Jewish

festivals Rosh Hashanah and Yom Kippur. The dates of the two festivals vary each year. In the year 2008, the last

Friday before the 2nd April is March 28, Rosh Hashanah and Yom Kippur fall on September 29 and October 8,

respectively. The Sunday between these two dates is October 5. DST starts at 03:00 hours and ends at 02:00

hours.

SETU ATA1S gives you the flexibility to set the 'Forward DST Adjustment' according to Date-Month, while

the Backward DST Adjustment according to Day-Month. Similarly, the reverse is also possible, i.e.

Forward DST may be set according to Day-Month, while the Backward DST may be set as Date-Month.

This flexibility is particularly useful for setting DST of countries where the start of DST is defined by date

and month, e.g. March 11, but the end of DST is defined by Day and Month, e.g. the last Sunday of

October (as observed in Cuba).




Debug

The ATA1S supports a feature by which the Admin can view the system debugs on the 'Syslog server' over IP

network. The parameters 'Syslog Server Address' and 'Server port' number can be programmed in the ATA. The

'Syslog Client' in the ATA1S supports multiple debug levels: e.g. system debug, SIP debug etc.

Each debug message includes the MAC Address of ATA1S which is sending debug messages in 'syslog server'.

ATA1S supports following different Debug levels:

• APPL

• CONFIG

• SYS

• SIP

• CALL

• DSP

• REG

• STACK• NAT

• AUTO CONFIG

• SNTP

As per the level selected, debug log will be generated. For example: if debug log of the calls is required, enable

'CALL' level and disable all other debug levels.

How to program?

Open the Web Jeeves and click Debug link. (Only Admin can access this Webpage). The Web page appears as

below:




Enable Debug?

Select 'Yes' to enable or 'No' to disable the feature. By default it is 'Disable'.

If Debug flag is 'Disable', syslog client will not send any debug to 'syslog server address'.

If Debug flag is 'Enable', syslog client will send debug to 'syslog server address' and the 'Debug Level' which is

enabled, can be viewed on the syslog server. The Server address and port number are to be programmed if Debug

is enabled.

Syslog Server Address

Program the IP Address of Syslog Server.

Server Port

Program the listening port of the Syslog Server from 1024-65535, 514. By default it is '514'.

Debug Levels

Select the parameter for which debug-log is required. The list of levels is as mentioned in this chapter. By default,all are enabled.

Relevant Topic:

1. “PCAP Trace” 87




Dialed Number Table

ATA supports the routing of the call to the specific destination route, based on which 'number' you have dialed for

making OG call. This feature is called 'Dialed Number Based Routing'.

This feature is useful for Least Cost Routing (LCR), application as explained in following scenario:

Suppose, the ATA is using SIP trunk ITSP 1 (SIP1) to make OG calls, which provides cheaper calling rates for

calling to US and at the same time, should use ITSP 2 (SIP2) which provides cheaper calling rates for calling to

Gulf countries. Thus this feature can be used for automatic routing of the call through a cheaper Service Provider.

When OG call is made, the dialed number can be routed to destination port, using following options for the

destination port: SIP1/SIP2/SIP3.

For this feature, a Dialed Number Table is configured in the ATA, using following parameters:

• Number

• Minimum Digit

• Maximum Digit• Destination Route

In this Table, the specific Number to be dialed and the Destination route, corresponding to each Number are

programmed.

Also, the destination route is specified, for the numbers to be dialed from FXS port, which are not matching with the

numbers programmed in the Dialed Number Table.

To dial only the required number of digits which will be valid for the interfacing PBX or device, ATA supports

programming of Maximum number of digits and Minimum number of digits in the Table.

Significance of Maximum number of digits and Minimum number of digits:

If the user has dialed the number of digits which is less than 'Minimum number of digits' programmed in the Table,

user will get error tone, when he has made OG call on this number.

If the user has dialed the number of digits as equal to specified Maximum number of digits programmed in the table,

the ATA will consider it as 'End of dialing' and will dial out the number.

End of Dialing will also be considered by the system, in following cases:

• If Inter-digit wait timer expires while dialing the digits

• If End of Dialing digit is dialed while dialing the digits. (Default 'End of dialing digit' is '#')

• Fixed Destination number is dialed.

• Call forwarded number is dialed.

Following points are relevant for configuring the Table:

• Maximum 100 numbers which are to be dialed can be programmed in the Dialed number table.

• Dialed number table is applicable when the call is made from: FXS Port.

• Each entry is programmed with Number and corresponding Minimum & Maximum digits with destination

route (SIP1/SIP2/SIP3) which will be used to make an OG call.

• If the number of dialed digits matches with the programmed 'maximum digits', ATA will consider it as 'End

of Dialing' and call will be made through corresponding destination route.

• The dialed number is a valid number only if dialed number of digits is equal to or more than the 'minimum

digits' programmed for that number prefix.

• If the user has dialed the 'number of digits', more than 'maximum digits' (programmed for that number

string prefix in dialed number table), the excess digits will not be dialed out. (This may be used for application like multi stage dialing).




If the user has dialed numbers not matching in the Table ('No match found'):

The destination route for the dialed numbers which do not match with the numbers programmed in the Table can

be configured as explained below:

• The destination route is specified in the first Index, which will be used by ATA for the numbers dialed,

which are not matching with the numbers programmed in the Dialed Number Table ('No Match Entry').

• If dialed number string does not match with entries in Dialed Number Table, and if the call was made from

FXS Port, the call routing option can be set from options: None/ SIP1/SIP2/SIP3. If 'None', is selected

caller will get error tone.

Following is the example for programming of the Dialed Number Table:

• Suppose dialed number table is programmed as shown below:

In this Table:

• If dialed number string does not match with entries in Dialed Number Table, the call will be routed to the

programmed destination route 'SIP1', which is programmed in the Table in Index001.

• The prefix '2' for the numbers to be dialed from Service provider-a, are programmed in Index002,

destination route is programmed as 'SIP2' and Minimum digits to consider the number as valid number

is programmed as 7. Also the maximum number of digits is programmed as 7. Hence all numbers,

starting from '2' will be routed from SIP2. For e.g. if you dial '2630555', it will be routed from SIP2.

• The prefix '3' for the numbers to be dialed from Service provider-b, are programmed in Index003,

destination route is programmed as 'SIP3' and Minimum digits to consider the number as valid number

is programmed as 7. Also the maximum number of digits is programmed as 7. Hence all numbers,

starting from '3' will be routed from 'SIP3'. For e.g. if you dial '3065571', it will be routed from 'SIP3' port.• The prefix '98' for the numbers to be dialed from Service provider-c (e.g. Mobile numbers) are

programmed in Index004, destination route is programmed as 'SIP1' and Minimum digits to consider

the number as valid number is programmed as 10. Also the maximum number of digits is programmed

as 10. Hence all numbers, starting from '98' will be routed from SIP1. For e.g. if you dial '9898074306',

it will be routed from SIP Trunk1.

• The prefix '00' for the numbers to be dialed from Service provider-d (e.g. ISD numbers) are

programmed in Index005, destination route is programmed as 'SIP2' and Minimum digits to consider

the number as valid number is programmed as 12. Also the maximum number of digits is programmed

as 12. Hence all numbers, starting from '00' will be routed from SIP2. For e.g. if you dial ISD number, it

will be routed from SIP Trunk2.

Emergency Number Dialing:

To dial Emergency Number, configure Dialed Number Table with the Emergency Number and desired

destination SIP trunk. Configure routing option as 'Dialed Number Table' for the SIP trunk in web page 'SIP

Trunks'.

Index Number Minimum Digit Maximum Digit Destination Route

001 No match found 01 24 SIP1

002 2 7 7 SIP2

003 3 7 7 SIP3

004 98 10 10 SIP1

005 00 12 12 SIP2




How to program?

Open the Web Jeeves and click on 'Dialed Number Table' link. (Only Admin can access this Webpage). The


You can configure maximum of 100 entries in the Table at different 'Index' from 00 to 99.

• Number: Program different numbers which will be required to be dialed. The 'Number' for Index 001 is not

allowed to be programmed. You can program the Number using maximum 24 characters. ASCII

characters are allowed.

Default = as shown in Default 'Dialed Number Table', at end of chapter.

• Minimum Digit: Select the number for Minimum digit for the valid number to be dialed by ATA

corresponding to the 'Number' programmed above. Range = 01-24, Default = 01 for all Index.

• Maximum Digit: Select the number for Maximum digits for the Number to be dialed. Range = 01-24,

Default = 24 for all Index.

• Destination Route at Index001: Configure destination route at Index001, for the numbers to be dialed

which are not programmed in the 'Dialed Number Table'. Select the destination route from:

• None

• SIP1

• SIP2

• SIP3.Default = SIP1.

• Destination Route at Index002 to Index100: Select the port on which ATA should route a call when the

programmed number prefix is dialed. The options, for the route are SIP1/SIP2/SIP3. Default = SIP1.

Default 'Dialed Number Table':


001 No match found 01 24 SIP1

002 0 01 24 SIP1

003 1 01 24 SIP1

004 2 01 24 SIP1

005 3 01 24 SIP1




006 4 01 24 SIP1

007 5 01 24 SIP1

008 6 01 24 SIP1

009 7 01 24 SIP1

09 8 01 24 SIP1

010 9 01 24 SIP1

011 Blank 01 24 SIP1

: : : : :

100 Blank 01 24 SIP1





Digest Authentication

Sometimes it is required to receive the incoming SIP calls by the users (such as the users at 'corporate offices'),

only from the specific callers. Other callers should not be allowed to make call to these users.

To achieve this, feature 'Digest Authentication' is supported in ATA1S.

This feature can also be used as security feature for ATA1S while receiving important calls though IP network.

To enable the 'Digest Authentication', select the option 'Yes' for parameter; 'Enable Digest Authentication?' for the

SIP trunk, in the web page 'SIP Trunks'. Refer topic “SIP Trunks”. Configuration of other parameters is explained in

this topic.

SIP provides the challenge based authentication service called SIP Digest Authentication, to authenticate the

caller. Using digest authentication, the SIP call initiated by the caller will be challenged in form of SIP message to

convey the callers to prove their proper credentials. These credentials are:

• User ID

• User Password

If the 'User ID' and 'User Password' sent by the IP device of the caller, are same as the corresponding parameters

programmed in the 'Digest Authentication Table' of your ATA1S, then only the incoming call from that caller will be

processed further. Else the incoming call will be rejected.

Your ATA can receive incoming call with any invalid digest authentication parameters, only once. After this attempt,

if the caller again calls ATA with invalid parameters, the ATA will reject that call.

How to configure ATA1S for 'Digest Authentication'?

• Enable the 'Digest Authentication' feature on SIP Trunk by configuring the page 'SIP Trunks' in your

ATA1S, by enabling the parameter 'Digest Authentication' on the desired SIP trunk.

• In the web page 'Digest Authentication', program the parameters; 'User ID' and 'User Password' of the IP

device from which incoming call is to be allowed.

How to program?

Open the Web Jeeves and click on 'Digest Authentication' link. (Only Admin can access this Webpage). The Web

page will be as shown below:




Program the parameters as explained below:

You can program up to 100 entries in the Digest Authentication Table in Index 001 to Index 100.

• User ID: Program 'User Id' of caller IP device which is to be allowed to make IC call on ATA1S on the SIP

Trunk. This can be programmed using maximum of 40 characters.

The only characters (0-9,A to Z,a to z,-,_,.,!,~,* and ') are allowed.

Default = Blank.

• User Password: Program the 'User Password' of caller IP device which is to be allowed to make IC call on

ATA1S on the SIP Trunk.

This can be programmed using maximum of 24 characters. Only characters (0-9, A to Z, a to z, - , _, ., !, ~,

*, ', =, +, $ and ,) are allowed.

Default = Blank.




FXS Port

Analog Telephone instruments can be connected at FXS port of ATA1S, through which calls can be made and

received. Also the FXS port will be required to be interfaced to FXO port of the PBX, for the required application.

Hence some of the parameters of the FXS port will be required to be configured as explained in this chapter.

How to program?

Open the Web Jeeves and click FXS Port link. (Only Admin can access this Webpage). The Webpage will be as

shown below:




FXS port parameters are configured as follow:

FXS Port Status

• The FXS port can be enabled or disabled. Select 'disable' if port is not to be used.

• If FXS port is disabled, it can not be used to make OG calls. Also incoming calls can not be received. This

flag will be helpful when FXS port gets faulty due to some reasons. The 'Admin' can just disable this port

without changing all the settings to keep away the FXS port in the Call routing logic.

Default = Enabled.

FXS Port Name

• A name can be assigned to FXS Port of ATA1S. This name will be displayed on the SLT of the called party

if the instrument of the called party supports display name functionality (and the received call is not

anonymous call). FXS Port Name can be of maximum 24 characters.

• The only characters (0-9, A to Z, a to z, - , _, .,!,~,*) are allowed or you can also enter the desired

characters between quoted string e.g. "%@#$".

Default = Blank.




FXS Port Number

• In case of peer to peer calls, this number will be displayed on the SLT of the called party if the SIP ID is

programmed as '*' for the corresponding peer to peer trunk and instrument of the called party is compatible

to display the number.

• It can be programmed using maximum 24 characters. The digits 0-9, '*' and blank are allowed.

Default = Blank.

Outgoing Calls

Allow OG Calls?

Select the option from Yes/No. Select 'No' if OG call is not to be allowed from ATA. By default, Yes is selected.

OG Call Route

Select the SIP Trunk number from where you want to route the outgoing call when you make the OG calls from

FXS port. Following option can be selected:

• SIP1: If Fixed Routing Option 'SIP1' is selected, all outgoing calls will be made using SIP Trunk1.



• Dialed Number Based Routing: If this option is selected, all outgoing calls will be routed to the

destination port as per the entries made in Dialed Number Based Table. Please refer topic “Dialed Number

Table”.

By default, SIP1 is selected.

Refer “Making Calls” topic in Using ATA1S chapter, to know how to make OG calls on the selected trunk using

access code.

Class of Service• You can enable/disable following features from Class of Service (COS): Call Hold, call Waiting, Attended

Transfer, Conference, Call Forward, Call Toggle, Blind Transfer and Do Not Disturb (DND).

• The feature will not work if it is not allowed from COS. (Refer topics “Managing Calls” and “Call Features”

in chapter 'Using SETU ATA1S' for more information about features and programming using phone)

Default = Enabled (all features mentioned above).

Subscriber Type

Subscriber Type signifies the type of services FXS Port user wants to access. Subscriber Type has two

options: Gateway and Network.

• Gateway: It is the type in which the FXS Port user wants to use local features of SETU ATA1S and

eventually the features supported by Matrix Eternity IPPBX. If subscriber type is set to Gateway for the

FXS Port then the FXS Port user can use the local features of ATA1S and also the features of remote

server which requires Flash using access code of Using Supplementary Services of Service

Provider . Select this option when you want to use the local features supported by ATA1S and also the

remote server features which requires ATA1S to out-dial Flash during speech.

• Network: It is the type in which the FXS Port user wants to use the remote end features i.e. features

supported by IPPBX which requires ATA1S to out-dial Flash during speech. You will require using the

option only when ATA1S is connected with Matrix Eternity IPPBX (or any other PBX which requires

Flash to support supplementary features) and you want to use the advance features supported by

Matrix Eternity i.e. Call Hold, Call Waiting, Call Transfer, etc. If subscriber type is set to Network for




FXS Port then the FXS Port cannot use any local features of SETU ATA1S that requires Flash. Select

this option only when you are using ATA1S only with IPPBX which requires Flash to support

supplementary features.

Calling Party Number Presentation (CLIP)

• 'Calling Party Number Presentation' feature enables the presentation of the number of the calling party on

the display of your phone connected to FXS port.

• ATA1S supports CLI types: None, DTMF, FSK- Bellcore and FSK-ITU (V.23). Select the type of CLI

supported by your phone instrument. Select 'None' if CLI display is not required to be displayed on the

phone.

By default, FSK-ITU (V23) is selected.

Answer Signaling Type

• In general, ATA kind of products is mostly connected with PCO machines in Cyber Cafe's with the PBX.

When call is made through ATA and called party answers the call, it is required to generate maturityinformation on FXS port of ATA so that the device connected to FXS port of ATA viz. PCO machine or

similar telecom equipment can start billing. In absence of maturity information on FXS port, call will not be

considered as matured and no billing will be generated.

Select the Answer Signaling Type from the options:

• None

• Polarity (Battery) Reversal

• None: This option is used when no answer signaling is to be generated on the FXS port.

• Polarity Reversal: This option is used when answer signaling is to be generated in the form of Polarity Reversal on the FXS port. In case of polarity reversal, the Battery polarity of the FXS port

will get reversed when the called party answers the call.

Default = Polarity Reversal.

Disconnect Signaling Type

• As a general application, telecom equipment like PCO machine is connected to the FXS port of the ATA.

Now whenever the called party (remote party) disconnects or goes on-hook, it is required to inform this

event (by some signaling), to the FXS port of ATA, so that the PCO machine can consider the call as

complete and stop billing. In absence of this signal, the call is considered as complete when the caller

goes on hook. But this will result in inaccurate billing.• To solve such problems, the system supports 'Disconnect Signaling'. When a call is made from FXS to SIP

or SIP to FXS, if remote party disconnects, Disconnect Signal is generated on the FXS port.

You can select Disconnect Signaling Type from following options:

• None

• Polarity Reversal

• Open Loop Disconnect

• None: It is used when no signaling is to be generated on FXS for call disconnection.

• Polarity Reversal: It is used when the call disconnection is to be signaled in the form of Polarity

Reversal. If this option is selected, the Battery polarity of the FXS port will be reversed and user will

get Error tone, when the call is disconnected by the remote end.




• Open Loop Disconnect: It is used when call disconnection is to be signaled in the form of Open

Loop Disconnect signal. During this, the Battery voltage on FXS port will be removed for time of

Open loop disconnect timer programmed for that FXS port and will be restored again. If this option

is selected, user gets Error tone, when the call is disconnected by the remote end.

Default = Polarity Reversal.

• Open Loop Disconnect Timer: Open Loop Disconnect Timer is applicable only if Open Loop

Disconnect option is selected for the Disconnect Signaling Type.

When open loop disconnect signal is generated, the Battery(Current) voltage on FXS port is

removed for Open loop disconnect timer programmed for the FXS port and the voltage will be

restored again after this period. But the Polarity of the FXS port battery voltage will not be changed.

You will get Error tone after disconnect signal is generated on the FXS port. Default = 500 msec.

Ring Timer:

• It is the time duration for which conventional Phone connected to FXS port will 'ring', when IC call on SIP is

routed to FXS port of ATA.

• Program the suitable value of 'Ring timer' from the range of 01 to 99 seconds.

• Default = 45 seconds.

Flash Timer

Flash timer signifies the time period for which the loop current breaks. ATA uses this event for various features like

Call Hold, Call Transfer etc. Flash timer is programmable. Select it from 100 msec to 900 msec in steps of 100

msec.

Default = 600 msec.

Echo Cancellation

Echo Cancellation is a technique that allows isolation and filtering of unwanted signals caused by echoes.

Sometimes an echo is produced while talking on phone due to which it becomes difficult to hear and understand

the speech of the remote party. Echo cancellation puts a signal on the return transmission path, which is equal and

opposite of the echo signal.

• Enable Echo Cancellation?: Select this as 'Yes' to enable Echo Cancellation or 'No' to disable it. The tail

length should be selected for programming Echo Canceller if this is set as 'Yes'. Default = 'Yes'.

• Tail Length: Program this parameter if Echo Cancellation is enabled. Select tail length depending on the

length of the echo.Valid range of tail length is 8 msec to 32 msec. Default = 32 msec.

AC Impedance

This feature allows you to program the AC impedance of FXS port.

Generally, the different communication equipment connected to FXS port of ATA has different AC impedance.

ATA1S supports programmable AC impedance for interfacing with other equipment, with following values:

• 600

• 900

• 270 + (750 || 150 nF)

• 220 + (820 || 115 nF)• 370 + (620 || 310 nF)




• 320 + (1050 || 230 nF)

• 350 + (1000 || 210 nF)

• 200 + (680 || 100 nF)

Default = 600

Volume Control

You can adjust the Speaking Volume (Transmit) on FXS port so that your voice will be clearly heard by the remote

user. You can also adjust the Listening Volume (Receive) on FXS port so that you can hear clearly the remote

user's voice.

• Speaking Volume: The ATA1S supports programmable Speaking volume from values -6dB to +6dB in

steps of 1 dB, viz. -6, -5,-4,-3,-2,-1,0,+1,+2,+3,+4,+5,+6 (dB).

Default = 0dB.

• Listening Volume: The ATA1S supports programmable Listening volume from values -6dB to +6dB in

steps of 1 dB, viz. -6, -5,-4,-3,-2,-1,0,+1,+2,+3,+4,+5,+6 (dB).

Default = 0dB.




LAN Port

LAN Port of ATA1S is used to connect ATA to a computer or to one of the Ethernet ports of LAN switch according to

the application for which it is to be used. Programming of basic LAN parameters is explained in this chapter.

The user should get the IP address for the LAN port of ATA1S from the LAN administrator which will be required to

be configured on LAN port of ATA1S.

While connecting the LAN port, to the PC, it may be required to change the IP address of the LAN port of the

ATA1S so that the user can access the Embedded Web Server using this IP address also.

The subnet of LAN IP address and IP address of PC should be different. Also the subnet of LAN should be deferent

from subnet of WAN address.

All the SIP calls will be routed only through WAN port as there is no SIP stack in LAN port of ATA.

How to program?

Open the Web Jeeves and click on 'LAN Port' link. (Both Admin and User can access this Webpage).The Webpagewill be as shown below:

LAN Port parameters can be configured as follow:

IP Address:

• An 'IP Address' is a 32-bit number with 4 octets, having 8-bits in each octet. Ask your network

administrator for valid IP address.

• By default, the LAN IP Address of ATA1S is 192.168.002.16. You can change the IP address as per your

LAN addressing scheme. Please note that the IP address of LAN and WAN port should be in different

subnet. Else the routing functionality of ATA1S will be affected. Also, if LAN port of ATA1S is connected to

the LAN switch or computer, IP address of all the LAN port and computers should be in the same subnet.




Subnet Mask:

• 'Subnet Mask' is a 32-bit number with 4 octets, having 8-bits in each octet. Ask your network administrator

for valid Subnet Mask.

• By default, the subnet mask of LAN port of ATA1S is 255.255.255.0. You can change subnet mask as

per your LAN addressing scheme.

The LAN IP address and LAN Subnet Mask can also be changed using commands through conventional

phone connected to the FXS port. Refer topic “Programming Options” in chapter 'Knowing SETU ATA1S'.

• MAC Address: This is the unique MAC address of LAN Port. You can not change this field.

The system will restart when you change either IP Address or subnet mask of LAN port of ATA1S.




PCAP Trace

• 'PCAP Trace' enables you to capture network packets. The ATA supports this feature, whereby the

Ethereal packets can be captured without the help of Hub. Generally, this tool will be useful when you find

any problem in the field, related to SIP trunk e.g. SIP trunk is not getting registered.

• The specific 'PCAP library' is configured in the ATA, for capturing the network packets.

• Following parameters are supported for this feature: (refer 'How to program?' topic later, for more details)

• Filter Settings

• Promiscuous Mode

• Start, Stop and Save Trace File buttons

• 'Filter Setting' and 'Promiscuous Mode' are parts of the Flash memory of the system. Hence, these settings

will not be cleared when you switch off the system.

• Refer 'How to use the feature?' at the end of this topic for details of using 'PCAP Trace'.

How to program?

Open the Web Jeeves and click on 'PCAP Trace' link. (Both Admin and User can access this Webpage). The Web







How to use the feature?

• Open the web jeeves page for 'PCAP Trace'.

• Set the filter as per the packets to be captured.

• After entering the Filter settings, start capturing of the packets by pressing the 'Start' button.

• Press 'Stop' button to stop capturing the packets (or system will stop the packet capturing once maximum

allotted memory is utilized viz. 1 MB (RAM).

• Once, capturing is stopped, press button 'Save Trace File' to save the trace file (i.e. packets captured by

the system) on your PC or some other PC by giving the path.

• Now, you can open the trace file using 'Ethereal' or 'Wire Shark' or some other software which supports to

open this trace file.




Peer-to-Peer Call Table

A call made between two IP clients without the intervention of a Proxy Server is defined as Peer-to-Peer Call (P2P

Call).

If the SIP trunk which is used for routing logic of the FXS port is programmed with Registrar Server Address,

ATA1S will consider it as SIP Proxy trunk. If the Registrar Server Address is not programmed, ATA1S will consider

it as SIP Non-Proxy (P2P) trunk. This trunk will also be used for receiving P2P call. Refer topic “SIP Trunks” to

enable SIP trunk and configure SIP Id.

Peer-to-Peer feature (P2P feature), enables the system to make an OG call to any SIP device over IP network,

whose IP Address/Domain is known to system. By using this feature, the user gets away of dialing long and

uncomfortable IP address. Thus, it also serves as abbreviated dialing of IP numbers.

For this feature to work you need to program the number (prefix), you want to call and IP Address/Domain of that

SIP device, with required minimum and maximum length of called number in the Peer-to-Peer Table. Refer topic

“Dialed Number Table” for more information about minimum and maximum digits.

Refer topic “Special Features” in chapter 'Using SETU ATA1S' for more information about working of P2P feature.

Refer topic “Making Calls” in chapter 'Using SETU ATA1S' for more information about making P2P call.

How to program?

Open the Web Jeeves and click on 'Peer-to-Peer Numbers' link. (Only Admin can access this Webpage). The Web


Total 500 entries can be programmed in the Peer-to-Peer table. Each entry should be programmed using following

parameters:




Number:

• Program the Number string, which is to be dialed as peer to peer. If the number to be dialed out is

<dialednumber@destination address>, then program the 'Dialed number', in this field.

• You can also program the prefix of number string.

• Index-001 signifies the 'No Match Found' condition which is selected when dialed number does not match

best with any other index.

Maximum of 24 characters can be used. ASCII characters are allowed.

Default = Blank

Name:

• Program the Name of the contact person which helps to identify the programmed number string of called

party. This is just a 'Tag' and will not be used in SIP signaling. Maximum 24 characters can be

programmed. ASCII characters are allowed.

Default = Blank

Minimum Digit:

• This parameter specifies the least length required for number string to be out dialed.

• If the number of digits dialed is less than this, the call will be rejected by the system.• Entry at Index-001 signifies the 'No Match Found' condition which is selected when dialed number does

not match best with any other index.

Range = 01 to 24

Default = 01

Maximum Digit:

• This parameter specifies the maximum length allowed for number string to be out dialed.

• Entry at Index-001 signifies the 'No Match Found' condition which is selected when dialed number does

not match best with any other index.

• If the dialed digits are exceeded, the additional digits will be stripped off.

Range = 01 to 24

Default = 24

Destination Address:

• This parameter specifies the domain or IP address where the number is to be called.

• If the number to be dialed out is <dialednumber@destination address>, then program the 'destination

address', in this field. It can be in form of Address: Port number.

• 'Destination Address' at Index-001 signifies the 'No Match Found' condition which is selected when dialed

number does not match best with any other index.

• Maximum of 40 characters can be used. Only characters (0-9,A to Z,a to z,: and .) are allowed and you can

also use the combination of alphabets or numerals separated by "-" (The character '&' is not allowed).

Default = blank.




Restart ATA1S

This feature enables the Admin to restart the ATA1S, without physically making the ATA1S, OFF and ON using

Power switch.

When Admin has restarted the ATA1S, all the current calls will be disconnected.

How to restart?

Open the Web Jeeves and click on 'Restart' link. (Only Admin can configure this Webpage).

The Webpage will be as shown below:

• When Admin clicks on the link, system will provide the alert message as shown above. Click on OK to

restart your ATA. Else click on 'Cancel'

After 'Restart' of the system, the parameters of 'ATA1S', will not get default values!




SIP Trunks

While configuring ATA1S, one of the most important steps is to configure the SIP parameters as you can have

maximum three SIP trunks. Thus the ATA1S can take the services of maximum three ITSP's.

While taking services from ITSP, you should take all the important information such as SIP ID, Authentication ID,

Authentication Password, Registrar Server Address, Registrar Server Port, Outbound Server Proxy Address and

Outbound Server Port from the ITSP and configure the same in 'SIP Trunks' page in the web Jeeves.

Proxy and P2P Trunks:

For Proxy trunk, you should program the Registrar Server Address and SIP Id for the relevant SIP trunk to be used.

SIP Id differs from call to call which will be authenticated by the proxy.

For P2P trunk you need not configure the Registrar Server Address and SIP Id for the relevant SIP trunk to be

used. You should program '*' as SIP Id for P2P trunk which will be used for P2P call.

This type of call is generally made by the users who have their own private VPN or leased line connected to the

different branch offices.

How to program?

Open the Web Jeeves and click on 'SIP Trunks' link. (Only Admin can access this Webpage).The Webpage will be

as shown below:







SIP Trunks parameters should be configured for maximum of three SIP Trunks as per your requirement, as

explained follow:

SIP Trunk - Enable/Disable

• Select 'Yes' to enable SIP trunk.

• If SIP Trunk is disabled, it can not be used to make OG calls. Also incoming calls can not be received. This

flag will be helpful when a SIP trunk not valid or can not get registered due to some reasons. The 'Admin'

need not change all the settings to keep away a SIP trunk in the Call routing logic and can just disable it.

Default = No.

SIP ID

• Enter the SIP ID provided by the ITSP. This will be a number or text with Maximum of 40 characters. The

only characters (0-9,A to Z,a to z,-,_,.,!,~,* and ') are allowed. Blank is not allowed.

• Suppose the SIP URI provided by the ITSP is [email protected], then enter '12345' in this field.Default = *.

Registrar Server Address

• This specifies the SIP Registrar server's address. It can be IP address or domain. You should keep it blank

for P2P trunk. It should be configured for Proxy trunks.

• Max. 40 characters can be used. Only characters (0-9,A to Z,a to z and .) are allowed and you can also

use the combination of alphabets or numerals separated by "-".

Default = Blank.




Registrar Server Port

• This specifies the Registrar Server's listening port for SIP.

• 'Admin' can change this field if the ITSP provides a SIP port number other then the default. Valid range for

this field is 1024-65534.

Default = 5060.

Add 'rinstance' in REGISTER?

• The parameter 'rinstance' is any random value which can be used by the ATA , to fetch its own contact

binding i.e. to know the registration expiry timer, assigned by server.

• While sending the REGISTER message, the ATA will check whether 'Add 'rinstance' in REGISTER

message?' flag is set or not for the SIP trunk.

• If the flag is set, the ATA will generate any random value of 'rinstance' and include in the REGISTER

message. Thus, the ATA will use the registration expiry timer of that contact binding.

• If the flag is not set, the ATA will send the REGISTER message without adding the 'rinstance' and will use

the registration expiry timer of relevant contact binding.

Default = Yes

Send REGISTER?

• This feature allows ATA not to send REGISTER message from the Proxy SIP trunk which is enabled in

ATA. To achieve this, the flag is provided as 'Yes' or 'No' for 'Send REGISTER?'.

• If the flag is set as 'Yes' the ATA will send the REGISTER message to Registrar proxy or Outbound proxy

as applicable, when SIP Trunk is enabled and configured with relevant information.

• If the flag is set as 'No' the ATA will not send the REGISTER message to Registrar proxy or Outbound

proxy as applicable when SIP Trunk is enabled and configured with relevant information. ATA will start

sending NAT Keep Alive message (if applicable) only when call is initiated.

• When 'Send REGISTER?' flag is set to 'No' for Proxy SIP trunk, SIP trunk will be considered as 'Active' (or

registered). In this case, the 'Status' page of Web Jeeves will display the registration status as 'Active' for

the relevant SIP trunk.

• For proxy SIP Trunk where 'Send REGISTER?' is set to 'Yes', if 'SIP Over TCP' is disabled in 'WAN Port'

but 'Transport Configuration' on corresponding SIP Trunk is set to either 'TCP' or 'TCP (fallback to UDP),

SIP trunk will be considered as Failed. For such case, the 'Status' page of Web Jeeves will display the

registration status as 'Failed' and OG/ IC calls will not be allowed by ATA. Also, 'Reason for registration

failure' will be displayed as 'TCP Disabled'. Refer topic “Checking the Status” for details about status

parameters.

• The Yes or No flag will not affect processing the incoming calls by ATA.

Default = Yes

Allow Outgoing Calls Without Registration?

• This feature allows ATA to make OG calls even when SIP Trunk is not registered. To achieve this, the flag

is provided as 'Yes' or 'No' for 'Allow Outgoing Calls Without Registration?'. The feature is applicable only if

flag for 'Send Register?' is set as 'Yes'.

• If the flag is set as 'Yes', the ATA will allow making OG calls even if the relevant SIP trunk is not registered.

• If the flag is set as 'No', the ATA will not allow making an OG calls even if the relevant SIP trunk is not

registered. This flag will not affect processing of the incoming calls by ATA.

Default = No




Re-Registration Timer

• This timer signifies the time after which the ATA will send Registration request again to get registered

• Range = 00001 to 65535 seconds.

Default = 3600 seconds.

Registration Retry Timer

• After sending the REGISTER request, if registration attempt fails, the ATA will again send the registration

request after the expiry of Registration Retry Timer. This registration retry timer increases by 1x, 2x …5x,

where 'x' is time for first retry.

• Range = 00001 to 65535 seconds.


Authentication User-Id

• This is the Authentication Id for authentication of the SIP trunk. Max. 40 characters can be programmed.The only characters (0-9,A to Z,a to z,-,_,.,!,~,* and ') are allowed. This is relevant when SIP User ID and

Authentication User ID are not same.

Default = Blank.

Authentication Password

• This is the password associated with the Authentication Id, programmed above. Max 24 characters can be

used. Only characters (0-9,A to Z,a to z,-,_,.,!,~,*,',=,+,$ and ,) are allowed.

Default = Blank.

Outbound Proxy-Enable/Disable

• Select 'Yes' to enable this field if the ITSP has a SIP outbound server to handle voice calls. Else, select

'No'.

Default = No.

Outbound Proxy-Server Address

• This specifies the Outbound Proxy servers' address. It can be IP address or domain.

• You can use 40 characters max. Only characters (0-9, A to Z,a to z and .) are allowed and you can also

use the combination of alphabets or numerals separated by "-".Default = Blank.

Outbound Proxy -Server Port

• This specifies the Outbound Proxy Server's listening port for SIP. Admin can change this if the ITSP

provides a Outbound Proxy Server port number other than the default.

• Range = 1024-65534.

Default = 5060.




Source Port IP Address

You can select either of the options mentioned below for each SIP Trunk:

• Use WAN port IP Address

• Use IP Address fetched using STUN

• Use Router's Public IP Address

Default = Use WAN Port IP Address.

Use WAN Port IP Address:

• Select this option if ATA1S is connected directly to the public internet or behind the NAT, but you do not

need to make call to Public Internet.

Use IP Address fetched using STUN:

• STUN is the most widely used protocol by SIP devices when located behind the NAT router. STUN is used

to map the public IP address and port of the NAT router behind which the SIP client is located. Select this

option if ATA1S is located behind the NAT router other than Symmetric. This option will work only if

Outbound is disabled on the SIP trunk. If you have selected this option, program the STUN server address

and port in WAN Port Parameters page of Web Jeeves.

Use Router's Public IP Address:

• Select this option if ATA1S is located behind the NAT router (any type). This option will work only if

Outbound is disabled on SIP trunk. If you have selected this option, program the Router's Public IP

Address field in the WAN Port Parameters page of Web Jeeves.

You can click on the link 'Status' in Web Jeeves to view your WAN port and SIP configuration status.

Incoming Calls

Allow incoming calls?

• This can be enabled or disabled.

• Enable this parameter for the SIP Trunk, if IC calls are to be allowed on that SIP Trunk. Disable the feature

on a SIP Trunk, if IC calls are not to be entertained by the ATA1S on that SIP Trunk.

Default = Enable.

Incoming Call Route

This feature signifies whether IC call on SIP trunk should be allowed and how the incoming call on SIP trunk, will be

routed to the FXS port.

If the flag to 'Allow IC Calls?', is not selected for the SIP trunk, then ATA will block the IC calls on that SIP trunk.

If you have selected the flag to allow IC calls ATA will entertain the IC calls. Now you can select the suitable routing

type from the options mentioned:

Routing options

• 'Route to FXS'

• 'Collect number and route to FXS'

Default = 'Route to FXS'

Different routing options for IC calls on SIP are explained below:

'Route to FXS'

• This option is generally required when user wants to receive the all IC calls on FXS Port.




• If 'Route to FXS' option is set, the IC call will be placed on FXS port and SLT or Phone connected to FXS

will 'ring' for Ring Timer (default = 45 sec ). The IC call will be rejected after the expiry of ring timer, if the

Phone user does not attend the call.

'Collect number and route to FXS'

• This option is generally required when FXS port of ATA1S is connected to FXO port of PBX and caller

wants to reach the extension of PBX by simply dialing the PBX extension number (The PBX should be

properly configured for routing this type of call).

• If this option is set, the caller will get dial tone on receipt of the IC call on SIP trunk. When the caller has

dialed the number, the ATA1S will route the call to FXS port and the conventional Phone connected to

FXS will 'ring', for Ring Timer (default = 45 sec). The IC call will be disconnected after the expiry of ring

timer, if the Phone user does not attend the call.

Send 183 Session Progress?

• This feature allows the ATA to send the SIP message; either '183 Session Progress' or '200 OK' when the

ATA answers the IC call on the SIP trunk (viz. when caller gets dial tone) and speech is to be established.

• The feature is applicable for following IC call routing option in the ATA:

• 'Collect number and route to FXS'

• Select 'Yes' if you want the ATA to send '183 Session Progress'.

• Select 'No' if you want the ATA to send '200 OK'.

Default = Yes

Transport Configuration

The ATA always accepts the SIP messages transported over UDP. However, you can restrict the ATA to accept SIP

messages over TCP connection. Moreover, you can program the default transport to be used for outgoing SIP

messages.

Default Transport For Outgoing Messages:

The ATA supports following three options for outgoing SIP messages.

• TCP: If it is selected, TCP will be used as default transport protocol for outgoing messages provided 'SIP

over TCP' flag in web page 'WAN Port' is enabled. Else the messages will not be sent or call will be

dropped. Refer topic “WAN Port” to enable SIP over TCP.

• UDP: If it is selected, UDP will be used as default transport protocol for outgoing messages.

• TCP (with fallback to UDP): If it is selected, TCP will be used as default transport protocol for outgoing

messages provided 'SIP over TCP' flag in web page 'WAN Port' is enabled. Else the messages will not besent or call will be dropped. Refer topic “WAN Port” to enable SIP over TCP. However, if TCP connection

to remote peer fails due to any reason, the ATA will send message over UDP.

• Default = UDP.

DNS SRV

• In Use DNS SRV? option,

• Select Yes to send DNS SRV query to the configured domain server.

• Select No to send DNS A query to the configured domain server. By default, No is selected.




Enable Digest Authentication

• SIP provides the challenge based authentication service called SIP Digest Authentication, to authenticate

the caller. Using digest authentication, the SIP call initiated by the caller, will be challenged in form of SIP

message to convey the caller to prove their proper credentials. These credentials are SIP ID and User

Password.

• This feature is required when the incoming calls on SIP is to be allowed only from specific callers.• If 'Digest Authentication' is enabled, the Authentication will be applicable for all IC calls on SIP trunk.

• If 'Digest Authentication' is disabled, the Authentication will not be applicable for IC call on SIP trunk.

• Refer topic “Digest Authentication” for configuring the Digest Authentication Table.

• Default = No.

Automatic Number Translation

• This feature allows the system, to replace the dialed number string to desirable number string which is

needed to be dialed on IP.

• Refer topic “Automatic Number Translation” for more details about the ANT feature.

• Apply ANT?: Program 'Yes' for 'Apply ANT?' to enable the feature. Program 'No' to disable the feature.

Default = No.

• Dialed Number list: Select from 1 to 6, default = 1.

• Substitute number list: Select from 1 to 6, default = 1.

Anonymous Call Rejection

• This feature is used to reject the incoming calls which do not have the identity of the caller and is useful for

security purpose.

• The Anonymous calls are the calls without any Calling Party information.

• Enable Anonymous call Rejection?: Select 'No' to disable the feature. When disabled, Anonymous calls

will be accepted and routed/answered by ATA. Select 'Yes' to enable the feature. When enabled,

Anonymous calls will be rejected.

• Default = No.

CLI Number on FXS Port

• This parameter is useful to provide the information to the user about the 'Number' of the calling party or

called party.

• The 'Type' to be displayed can be programmed with two options:

• Received Calling Party

• Received Called Party• If 'Received Calling party' option is selected, 'calling party number' will be displayed as CLI on FXS port

and call will be routed further.

• If 'Received Called Party' option is selected, 'called party number' will be displayed as CLI on FXS port and

call will be routed further.

Default = 'Received Calling Party'.

Caller ID

• Send Caller ID: If this parameter is set 'No', all subsequent OG calls will be considered as 'Anonymous

calls' and caller's number will not be sent. If it is set as 'Yes', the ATA1S will send the Caller's number.

Default = 'Yes' for all SIP Trunks.




This feature can be configured using Phone command also. (Refer topic “Call Features” in chapter 'Using

SETU ATA1S' for more information about feature and programming using phone).

Vocoder Preference

• The ATA supports following Vocoders:• Preference1: G.711 (A-law)/G.711(Mu-law)/G.729/G.723. Default = G.711 (A-law)

• Preference2: G.711 (Mu law)/G.729/G.723/G.711(A-law). Default = G.711 (Mu law)

• Preference3: G.729/G.723/ G.711 (A-law)/G.711(Mu-law). Default = G.729

• Preference4: G.723/G.711 (A-law)/G.711(Mu-law)/G.729. Default = G.723

• This parameter signifies the Vocoder in preferential order which can be requested to the remote user while

making OG call.

• For IC call, the system will select the codec from the types offered by the IP device of the caller and if it is

same as any of the vocoder type, configured in the preference vocoders, the call will be established. Else

the IC call will be rejected.

• Default = Sequence of Vocoders, mentioned above.

G.723 bit rate

• The ATA supports following bit rates for the G.723 codec:

• 5.3 kbps

• 6.3 kbps

• The ATA supports the above programmed bit rate only for sending the RTP packets if G.723 codec is

negotiated during call on the SIP Trunk. However, while receiving the RTP packets from the remote end if

G.723 codec is negotiated, the ATA will accept any bit rate.

• The ATA also supports switch over of bit rates (i.e. 5.3kbps to 6.3kpbs or 6.3kpbs to 5.3kpbs) during the

call.

Default = 6.3 kbps

Enable Symmetric RTP?

• Using this feature, it will be possible for a SIP device to send the RTP on the same connection on which it

is listening for RTP. This will be done only for peer to peer trunks.

• This parameter can be set as 'Yes' or 'No'. Select the option 'Yes' if the ATA1S is located on a public IP

and you want to make an outgoing call to the SIP client located behind the NAT router or if you need to

receive incoming calls from the SIP client located behind the NAT router.

Default = No.

DTMF Option

• Following options are used to signify, how the DTMF digits will be sent over IP:

• Inband

• Outband

• SIP (INFO)

• This option is used while making OG calls

Default = 'Outband'




DTMF Dial Timers

DTMF ON Time:

• Program this parameter for the time for which the DTMF digit which is to be outdialed is required to be ON.

• You can program this timer from values 100, 150 or 200msec.

By default = 100msec.

DTMF OFF Time:

• Program this parameter for the time for which the DTMF digit which is to be outdialed is required to be

OFF.

• You can program this timer from values 100, 150 or 200msec

Default = 100msec.

• DTMF ON/OFF time parameter will be used when DTMF option is selected as 'RFC 2833 (Outband)'.

• However, if DTMF Option is selected as 'SIP INFO', only DTMF ON Time will be applicable. DTMF OFF

Time will be fixed viz. 500 msec. These timers are used for multi-stage dialing application. Refer

chapter “Using SETU ATA1S” .

Pause Timer

• This timer is required, to insert the delay while the digits of a number string are out dialed by ATA1S.

• When ' '̂ character is configured in the number string, it will be considered as Pause. One of the

applications is multi-stage dialing. Refer chapter “Using SETU ATA1S”.

• You can program this timer from 1-9 seconds.


Call Hold Method

• Select RFC: This feature allows the ATA to select the method for Call Hold, so that the ATA can be

integrated successfully with other IP devices. The ATA supports following methods for Call Hold for each

SIP trunk:

• RFC 2543

• RFC 3261

• Default = RFC 3261

Program the Call Hold Method = RFC 3261 if the ATA should send following parameters in the SIP

message (SDP):

• Connection Information: IP Address (Local IP or STUN mapped IP or Router Public IP)

• Media attribute (a) : sendonly

Program the Call Hold Method = RFC 2543 if the ATA should send following parameters in the SIP

message (SDP):

• Connection Information: 0.0.0.0

• Media attribute (a) : sendonly

FAX Option

• User can send/receive the FAX by normal procedure when a FAX machine is connected to FXS port of the

ATA1S.

• Following FAX options are supported by ATA:

• T.38(UDPTL)

• Pass Through




• The option selected here signifies how FAX messages will be handled by the ATA.

• Select Pass-through option if you need to send fax message over G.711. The peer device should also use

G.711.

• While using T.38 to send fax messages, the peer device should also support T.38.

Default = T.38 (UDPTL).

If the FAX sent using T.38, is rejected, then ATA1S will use Pass Through option for sending the FAX (as a

Fall Back Mechanism).




Speed Dialing

ATA supports this feature by which the user need not dial all the digits of the number string to make an OG call. The

user can use abbreviated number for the contacts stored in the system, for dialing the desired number.

To achieve this feature the Table is required to be programmed in the ATA, using following parameters:

• Number of the contact

• Name of the contact

• Destination Trunk which will be used to dial the number by the ATA.

The user can dial the number using speed dialing, only from FXS port, during dial tone or during 'Held' stage of the

call.

The number can be dialed by dialing short access code followed by the Index number at which the number is

programmed in the Table. The number will be routed through the 'SIP trunk' assigned to it.

When you dial the Access code and Index-number from FXS port, the ATA1S will check the Table to fetch

the number and destination trunk corresponding to index-number dialed. You will get error tone if thedialed number is not programmed in the Table.

How to use?

Making Calls using Speed Dialing:

Go OFF-Hook Dial #9-Index Number Talk

Where,

Index Number is the Index for the Number to be dialed, configured in the Speed Dial Table (00 to 99)

ATA will dial the number from the destination trunk configured in relevant Index dialed.

How to Program?

Open the Web Jeeves and click on 'Speed Dialing' link. (Both Admin and User can access this Webpage). The web


• You can program up to 100 entries for the Table at Index 00 to 99.




Number:

• 'Number' signifies the desired contact which is to be dialed using conventional phone connected to the

FXS port. The complete SIP URI/ IP Address can be entered here e.g. [email protected].

• If you need to program 'SIP-ID' in 'Number' field, you can program either Peer to Peer trunk or Proxy trunk

as destination trunk. The call will be routed as per FXS port routing logic. Refer topic “FXS Port”.

• If you need to program 'IP-Address' in 'Number' field, you should program only Peer to Peer trunk as the

destination trunk.

• If you need to program 'SIP URI' in 'Number' field, you should program only Peer to Peer trunk as the

destination trunk. You need not program the complete URI, if the ATA is registered to the same domain;

only SIP-ID is required to be programmed.

• You can use maximum of 40 characters. All ASCII characters allowed.

Default = Blank.

Name:

• 'Name' identifies the person, whose number is stored in the table. 'Name' programmed here will not be

sent in the SIP message while the ATA dials the corresponding 'Number' using speed dialing.

• You can use maximum of 24 characters. All ASCII characters allowed.

Default = Blank.

Destination Trunk:

• 'Destination trunk' signifies that using which trunk the call should be routed while the ATA out dials the

corresponding 'Number'. Program the destination trunk as (SIP1 or SIP2 or SIP3) corresponding to each

number.

Default = SIP1.




Static Routing Table

This feature helps you routing the calls between point to point sites (connected through MPLS, Frame Relay etc)

and to public internet at the same time.

Thus you can call to ATA1S at different locations through Multiple Gateway application. Refer topic “Multiple

Gateway Application” in chapter 'Using SETU ATA1S' for more details.

How to program?

Open the Web Jeeves and click on 'Static Routing Table' link. (Only Admin can access this Webpage). The


'Static Routing Table' parameters can be configured as follow:

You can program up to 5 entries for the Table at Index 1 to 5.

• Destination Address: It is the network address where SIP device resides.

• Subnet Mask: It is the mask to be applied on destination address. It can be of maximum 15 characters.

• Gateway Address: It is the IP address of LAN interface of your router from where you want to route the

call for specific destination.




Supplementary Services

ATA1S supports various Supplementary call Services. These can be configured using Web Jeeves as well as

phone command.

For Programming Using Commands and more details on supplementary call features, refer topics

“Managing Calls” and “Call Features” in Using SETU ATA1S. Refer “Class of Service” in topic 'FXS Port' to

enable the feature.

How to program?

Open the Web Jeeves and click on 'Supplementary Services' link. (Both Admin and User can access this

Webpage). The Web page will be as shown below.

Following 'Supplementary Services' features can be configured on this page:

• Call Forward

• Call Waiting

• Do Not Disturb

Call Forward:

This feature enables you to forward calls to desired number when you are busy or not available at your place. You

can forward all incoming calls on SIP trunk only if the call is placed on your FXS port (routing type is configured to

route the call to FXS).

Select the required Call forward (CF) type from following options and enter the number or IP address (destination

address) in the corresponding field, on which you want the call to be forwarded. By default, all types of CF are

disabled.

• Call Forward-Unconditional

• Call Forward-When Busy• Call Forward-When No Reply




Call Forward Timer

• The call will be forwarded to the programmed destination, on expiry of the timer called 'Call Forward

Timer'.

• This timer is applicable only when option 'Call Forward-When No Reply' is selected. The range of Call

Forward Timer is 01 to 99 seconds. By default it is 45 seconds.

Call Waiting:

• This feature notifies you of another incoming call, when you are busy on a call. You will get beeps during

speech when you have waiting call.

• Waiting call is the incoming call, received during an active call. Select 'Set' for this if you want to get

indication (beeps), for the waiting call. Select 'Cancel' if you want to disable the feature. If the feature is

disabled, you will not get beeps for the waiting call and the waiting call will get rejected. By default it is 'Set'

(enabled).

Do Not Disturb:

• DND feature is useful when you do not want to receive any incoming calls. When DND feature is set on

FXS port, all IC calls on FXS port will be rejected by ATA1S, but OG calls can be made as usual.• Select the option 'Set' to enable DND. If DND is disabled, all incoming calls will be allowed by the ATA. By

default, the feature 'DND' is disabled.




Supplementary Services of Service Provider

If you are using SETU ATA1S with Matrix Eternity IPPBX, you can use various advanced features of Matrix Eternity

IPBPBX like Call Hold, Call Transfer, Call Waiting, etc. To use these features of Matrix Eternity, SETU ATA1S

supports with the facility of dialing Flash during speech on SIP. However, if you are also using ATA1S with someservice provider (along with Matrix Eternity) where you want to use the local features of ATA1S, you will be required

using the access code of Using Supplementary Services of Service Provider .

You may have installed SETU ATA1S in following scenarios:

1. Network Mode: If you are running SETU ATA1S in Network Mode, you will not be able to use any local

features supported by SETU ATA1S. You will probably run ATA1S in this mode when ATA1S is connected

only with Matrix Eternity IPPBX. During this mode, any activity performed on FXS port (dialing DTMF digit,

Flash) will be send transparently on SIP and will not be interpreted locally by ATA1S.

During this mode, there is no need to use the Using Supplementary Services of Service Provider

access code.

2. Gateway Mode: If you are running SETU ATA1S in Gateway Mode, you will be able to use all local

features supported by ATA1S. Eventually, if you are using ATA1S also with Matrix Eternity IPPBX, you will

be required to use the Using Supplementary Services of Service Provider user code.

To use the Supplementary Services of Service Provider , Class of Service (CoS) for Call Hold feature

must be enabled on the respective FXS Port .

• Access Code assigned for Using Supplementary Services of Service Provider is #0 . Dialing this

access code enables you to use the ATA1S in the same manner as used during Network Mode.

To use access code for Using Supplementary Services of Service Provider

1. Dial Flash during speech. This will place the active call on hold and you will get a feature tone.

2. To use the network features (e.g.: Call Waiting feature of Matrix Eternity IPPBX), dial Using

Supplementary Services of Service Provider access code i.e. #0 .

3. On dialing #0, you will again get in speech with the last held call and the system starts the Service

Provider Access Code Wait Timer (the timer is of fixed value i.e. 10 seconds). Before expiry of Service

Provider Access Code Wait Timer, any activity performed on FXS Port will not be interpreted by ATA1S

locally and will send to remote server transparently.

4. After expiry of Service Provider Access Code Wait Timer, ATA1S will again work normally and any activity

performed on FXS Port will be interpreted locally by ATA1S which is the default behaviour of Gateway

Mode.




System Parameters

System Parameters will allow the administrator to change the various system configurations.

You can get the details of current Software Version/Revision number (VXRY) and Kernel Date from the Status

page, where X is version number and Y is revision number.

How to program?

Open the Web Jeeves and click on 'System Parameter' link. (Only Admin can access this Webpage). The





System Name

This will allow Admin to identify the particular ATA unit when multiple ATA's are connected in LAN network. Enter

the system name here. Name can be of maximum 24 characters. All ASCII characters are allowed. By default,System Name is 'Matrix SETU ATA1S'.

Language Selection

• This feature allows both Admin and User, to select the desired Language for the web pages (Jeeves).

• The 'Admin' OR 'User' who wants to login the ATA, may desire to access the Jeeves pages, in his own

language which may not be 'English' language. For such users, ATA supports Web GUI viz. Jeeves in

multiple languages as mentioned below:

• Admin Language: Admin can select desired language from following options, for login and further

accessing the web pages:

• English

• Italian

• Spanish

• French

• German

• Portuguese

Once Admin changes the language, all strings in Jeeves will be displayed in the programmed Admin

language when login next time.




Admin can also click on desired language on welcome/login page from options on right top side of the

page. But this will not change the Admin language programmed here. Refer chapter “Knowing SETU

ATA1S”.

By Default Admin Language is 'English'.

• User Language: Since the 'User' can not access this 'System Parameters' page, only Admin can change

the Language option for the 'User', from following options:

• English

• Italian

• Spanish

• French

• German

• Portuguese

When the User enters the Web Jeeves from any web browser and clicks on 'User' link, the Login page will

be displayed as per programmed User Language. User can enter required Password in relevant field toget the Home page. All further pages (including Home page) will be displayed in the programmed User

Language only.

User can also click on desired language on welcome/login page from options on right top side of the page.

But this will not change the User language programmed here. Refer chapter “Knowing SETU ATA1S”.

By Default User Language is 'English'.

Call Progress Tone Generation (CPTG)

• The ATA1S plays various tones while using configuring or using ATA1S for making and receiving calls.The ATA1S supports country specific Call Progress Tones Generation (CPTG).

• Country wise call progress tones provide you the flexibly to simulate the same tones as given by PSTN.

You can select the country where this system is installed.

By default, INDIA is selected. (Refer Table1 for the list of countries and various tones supported by those

countries.)

• The ATA1S supports following call progress tones: Dial Tone, Ring Back Tone, Busy Tone, Error

Tone1, Error Tone2, Confirmation Tone, Programming Tone, CCWT, Routing Tone.

• Dial Tone: ATA plays the Dial Tone when it is ready to accept the first digit to make an outgoing call. It

is normally played to user when that port goes OFF-Hook.

• Ring Back Tone: ATA plays the Ring Back Tone when the called device sends alert to the callee about

receipt of a call from the callee.• Busy Tone: ATA plays the Busy Tone when the callee is busy.

• Error tone2: Error tone2 is played during following call conditions.

• Transfer Failure

• Programming Error.

• In all the remaining states, Error Tone1 is played.

• Confirmation Tone: ATA plays 'Confirmation Tone', to indicate that valid command is received by the

system.

• Programming Tone: ATA plays 'Programming Tone' which prompts the user to enter fresh commands

during programming.

• CCWT: ATA plays the CCWT (CO Call Waiting Tone) when an incoming call arrives while the user is

busy with one active call.




Table 1: CPTG (Refer Note1 below Table)

Note1:

Cont. = continuous

f1*f2: f1 is modulated by f2.

f1+f2: the juxtaposition of two freq. f1 and f2 without modulation.

Please note that frequency and cadence for following tones are same and fixed for all countries:

• Confirmation 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

• Routing Tone: 400Hz, 0.1 on 1.9 off

Code Country

Dial Tone Ring Back Tone Busy Tone Error Tone1 Error Tone2

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

Hz second Hz second Hz second Hz second Hz second

01 Australia 425*25 cont. 400*25 0.4on 0.2off 0.4on 2.0off

425 0.375on0.375off

425 0.375on 0.375off 425 1on 1off

02 Argentina 425 cont. 425 1.0on 4.0 off 425 0.3on 0.2off 425 0.3on 0.4off 425 1on 1off

03 Belgium 425 cont. 425 1.0on 3.0off 425 0.5on 0.5off 425 0.167on 0.167off 425 1on 1off

04 Brazil 425 cont. 425 1.0on 4.0 off 425 0.25on 0.25off 425 0.25on 0.25 off 425 1on 1off

05 China 450 cont. 450 1.0on 4.0off 450 0.35 on0.36off

450 0.7on 0.7off 450 1on 1off

06 Egypt 425*50 cont. 425*50 2.0on 1.0off 425*50 1.0on 4.0off 450 0.5on 0.5off 450 1on 1off

07 France 440 cont. 440 1.5on 3.5off 440 0.5on 0.5off 440 0.25on 0.25off 440 1on 1off

08 Germany 425 cont. 425 1.0on 4.0off 425 0.48on 0.48off 425 0.24on 0.24off 425 1on 1off

09 Greece 425 0.2on 0.3off 0.7on 0.8off

425 1.0on 4.0off 425 0.3on 0.3off 425 0.15on 0.15off 425 1on 1off

10 India 400*25 cont. 400*25 0.4on 0.2off 0.4on 2.0off

400 0.75on 0.75off 400 0.25on 0.25off 400 1on 1off

11 Indonesia 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.25on 0.25off 425 1on 1off

12 Iran 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.25on 0.25off 425 1on 1off

13 Israel 400 cont. 400 1.0on 3.0off 400 0.5on 0.5off 400 0.25on 0.25off 400 1on 1off

14 Italy 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.2on 0.2off 425 1on 1off

15 Japan 400 cont. 400*20 1.0on 2.0off 400 0.5on 0.5off 400 0.25on 0.25off 400 1on 1off

16 Kenya 425 cont. 425 0.67on 3.0off 1.5on 5.0off

425 0.2on 0.6off 0.2on 0.6off

425 0.2on 0.6off 425 1on 1off

17 Korea 350+440 cont. 440+480 1.0on 2.0off 480+620 0.5on 0.5off 480+620 0.3on 0.2off 480+620 1on 1off

18 Malaysia 425 cont. 425 0.4on 0.2off 0.4on 2.0off


19 Mexico 425 cont. 425 1.0on 4.0off 425 0.25on 0.25off 425 0.25on 0.25off 425 1on 1off

20 New Zealand 400 cont. 400+450 0.4on 0.2off 0.4on 2.0off


21 Philippines 425 cont. 425+480 1.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off 480+620 1on 1off

22 Poland 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.5on 0.5off 425 1on 1off

23 Portugal 425 cont. 425 1.0on 5.0off 425 0.5on 0.5off 450 0.33on 1.0off 450 1on 1off

24 Russia 425 cont. 425 0.8on 3.2off 425 0.4on 0.4off 425 0.25on 0.25off 425 1on 1off

25 Saudi Arabia 425 cont. 425 1.2on 4.6off 425 0.5on 0.5off 425 0.25on 0.25off 425 1on 1off

26 Singapore 425 cont. 425*24 0.4on 0.2off 0.4on 2.0off


27 South Africa 400*33 cont. 400*33 0.4on 0.2off 0.4on 2.0off


28 Spain 425 cont. 425 1.5on 3.0off 425 0.2on 0.2off 425 0.25on 0.25off 425 1on 1off

29 Thailand 400*50 cont. 400 1.0on 4.0off 400 0.5on 0.5off 400 0.3on 0.3off 400 1on 1off

30 Turkey 450 cont. 450 2.0on 4.0off 450 0.5on 0.5off 450 0.2on 0.2off 0.6on 0.2off

450 1on 1off

31 UAE 350+440 cont. 400+450 0.4on 0.2off 0.4on 2.0off

400 0.375on0.375off

400 0.4on 0.35off 0.225on 0.525off

400 1on 1off

32 UK 350+440 cont. 400+450 0.4on 0.2off 0.4on 2.0off

400 0.375on0.375off

400 0.4on 0.35off 0.225on 0.525off

400 1on 1off

33 USA/Canada 350+440 cont. 440+480 2.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off 480+620 1on 1off




If the ATA1S is defaulted, CPTG parameters will not get default values.

Rings

This feature allows you to program the ring cadence country wise.

During an incoming call, the system will play the ring over SLT as per the ring cadence specified for the country.User can select the name of the country from the list of countries. By default, INDIA is selected. For list of country

specific 'Ring', refer 'Table-2'.

Table 2: (Ring Cadences)

Where,

T1ON, T1OFF, are ON and OFF period for first ring cadence and T2ON, T2OFF, are ON and OFF period for

second ring cadence.

Waveform for 'ring', is Sine Wave.

If the ATA1S is defaulted, 'Ring' parameters will not get default values.

Play Routing Tone?

This feature signifies whether the system will play routing tone when the call is initiated from FXS and is routed fromany 'SIP Trunk.

Code CountryFrequency

(Hz)

CADENCE (In Seconds)

T1ON T1OFF T2ON T2OFF

01 Australia 25 0.4 0.2 0.4 2.0

02 Belgium 25 1.0 3.0

03 Brazil 25 1.0 4.0

04 China 25 1.0 3.0

05 Egypt 25 2.0 4.0

06 France 25 1.5 3.5

07 Germany 25 3.5 5.5 0.79 1.1

08 Greece 25 1.0 4.0

09 India 25 0.4 0.2 0.4 2.0

10 Israel 25 2.0 3.0

11 Italy 25 1.0 4.0

12 Japan 25 1.0 2.0

13 Korea 25 1.0 3.0

14 Malaysia 25 2.0 3.0

15 New Zealand 25 2.0 3.0

16 Poland 25 2.0 3.0

17 Portugal 25 1.0 5.0

18 Russia 25 1.0 3.0

19 Singapore 25 0.4 0.2 0.4 2.0

20 South Africa 25 0.4 0.2 0.4 2.0

21 Spain 25 1.5 3.0

22 Thailand 25 2.0 3.0

23 UAE 25 2.0 3.0

24 UK 25 0.4 0.2 0.4 2.0

25 USA/Canada 25 2.0 4.0




It can be configured as Yes or No for the parameter 'Play Routing Tone?'. If this parameter is programmed as

"Yes", ATA will play routing tone to the FXS port user when call is made through the SIP trunk. If this parameter is

programmed as "No", ATA will not play the routing tone to the FXS port user when call is initiated on SIP trunk. The

caller will get Busy tone or RBT or Error tone as per the response received. By default it is 'Yes'.

First Digit Wait Timer

First digit wait timer signifies the time for which the system waits for receiving a first digit after going OFF-Hook from

FXS Port. On expiry of this timer, the system will give error tone to the user. It is programmable from 01 to 99

seconds. By default, the Fist Digit Wait Timer in ATA1S is set to 15 seconds.

Inter Digit Wait Timer

Inter Digit Wait Timer signifies the time period between two consecutive digits while the system is receiving the

digits from caller/user. If the user does not dial the next digit before expiry of this inter digit wait timer, then ATA1S

will process the digits dialed so far by the user. Refer topic “Dialed Number Table” for information about End of

Dialing while dialing minimum or maximum numbers. The range of Inter Digit Wait Timer is 01 to 99 seconds. By

default, it is 5 seconds.

Call Release Timer

This feature allows the system to release the call if made from FXS to SIP and SIP to FXS, if that call is not

released before specified time. Sometimes the call remains live till remote user disconnects the call, even if FXS

port goes on-hook. For such case, the Call Release Timer is useful to release the call e.g. Answer signaling on FXS

port. Refer topic “FXS Port” for more details on answer signaling.

On expiry of call release timer, any type of call will be released.

Enable Call Release Timer?:

• Select this as Yes/No. Select 'Yes' if any type of matured call is to be disconnected by the system, after

specified time. Else select 'No'. By default, it is 'No'.

Call Release Timer:

• This timer will be applicable only if you have enabled the Parameter 'Enable call release timer?'. This

timer will be useful to restrict long duration calls. You can program the timer from 001 to 999 Minutes.

By default it is 999 Minutes.

VoIP Silence Disconnect Timer

This feature allows disconnecting the call if silence is detected during speech, for more then specified duration.

The ATA supports this feature to detect the silence during VoIP calls. If this silence duration exceeds the specified

value, the call will be disconnected. It is applicable for following types of calls: FXS to SIP and SIP to FXS.

Enable VoIP Silence Disconnect Timer?: Select this as Yes/No. Select it as 'Yes' if this feature is required

for the ATA.

By default it is 'Yes'.

VoIP Silence Disconnect Timer: This timer is applicable if you have enabled the parameter 'VoIP Silence

Disconnect Timer'. You can program the timer from range 01 to 99 Minutes.

If the silence duration exceeds this programmed value, the call will be disconnected.

By default, it is 99 Min.




Transfer Notification Timer

This timer signifies for how much time ATA will wait for the call transfer notification from the remote peer.

Refer topic “Call Transfer” in topic 'Managing Calls' for more details.

User can select the timer value from 01 to 99 seconds.

By default it is 60 seconds.

Enable Forward Error Correction?:

Forward Error Correction (FEC) is used for data transmission whereby ATA1S adds redundant data to its

messages, which allows the receiver to detect and correct errors without asking the sender for additional data.

Select 'Yes' (enable) if FEC is required for better voice quality at the cost of higher bandwidth, for digital signal

processing.

By default "Forward Error Correction" is No.

Enable Voice Activity Detection?:

Voice Activity Detection (VAD) is a software application which allows 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.

• Enable VAD?: Select 'Yes' if ATA is required to support VAD (also called 'Silence Suppression') feature.

Else select 'No'. By default, it is No.

Default 'Admin' password when 'User' defaults the ATA1S?:

When user (who has login with User Password) defaults ATA1S, only limited parameters get defaulted. These are

the parameters which User is allowed to access through Jeeves. Parameters which are not accessible by 'User' do

not get defaulted. However, this parameter, provides flexibility to default even the Admin password also, when the'User' defaults the ATA.

ITSP should set this flag carefully so that ITSP- customer (user) will not be able to default the 'Admin

Password' and will be bound to access only limited settings of ATA1S.

• Select 'Yes' for 'Default 'Admin' password when User defaults the ATA?', if you need the system, to default

the Admin password also whenever User defaults the ATA.

• Select 'No' for 'Default 'Admin' password when User defaults the ATA?', if you do not need the system to

default the Admin password whenever User defaults the ATA.

By default, the flag is 'Yes' for "Default 'Admin' password when 'User' defaults the ATA1S?".

This flag is also applicable when the user uses the Access Code to default the Password. Refer topics

' “Admin Password” and ' “User Password” .






Default the User Password using Jeeves:

• After login as 'User', if the user default the ATA1S using Webpage 'Reinstate the Default Settings', the

system will also default the 'User' password. Refer chapter “Reinstate the Default Settings”.




WAN Port

ATA1S has a WAN port which is used for receiving and transmitting data over Wide Area Network. WAN port of

ATA1S can be connected to the Ethernet port of the broadband modem/router or to one of the Ethernet port of LAN

switch.

All the SIP calls will be routed only through WAN port as there is no SIP stack in LAN port of ATA.

How to program?

Open the Web Jeeves and click on 'WAN Port' link. (Both Admin and User can access this Webpage). The










MAC Address Setting

• The ATA1S provides unique MAC address to its WAN port.

• The ATA1S also facilitates the changing of MAC address at the WAN port by the process of 'MAC Address

Cloning'.

• Thus, you can select MAC address from two options:

• MAC Address• Clone MAC Address.

• MAC Address: If 'MAC Address' is selected, ATA1S will use pre-programmed MAC address as source

MAC address in all Ethernet frames. By default pre-programmed MAC Address is selected.

• Clone MAC Address: Select this if you want ATA1S to use MAC address other then 'MAC address',

pre-programmed. If this option is selected, program the desired MAC address manually. Only

Hexadecimal characters are allowed. By default, it is blank.

To prevent the possible collision and loss of data, do not clone the MAC address which is already used by

other device in the same network.

Enable 100rel/PRACK?

• This parameter is programmed to Enable/disable 100rel SIP extension. It is useful, where it is required to

support reliable transmission of (SIP) provisional responses.

• Select 'Yes' if you want the ATA, to use 100rel SIP extension for reliable transmission of SIP (provisional)

responses and to use PRACK (Provisional Acknowledgement). By default, it is 'No'.

SIP Timers

• ATA1S supports three programmable SIP timers viz. SIP INVITE Timer, SIP Provisional Timer and

General Request Timer.

• SIP INVITE Timer: It is the time for which ATA1S waits for a response from the called party after

sending INVITE message. This timer starts after sending INVITE message to the called party and stops

on receipt of provisional response or final response or when the user goes on-hook. On expiry of the

timer, call is disconnected and user gets error tone. The range of SIP INVITE Timer is 010 to 180

seconds. By default, it is 30 seconds.

• SIP Provisional Timer: It is the time for which ATA1S waits for final response after receiving

provisional response from the called party. This timer starts on receipt of provisional response from the

called party and stops on receipt of final response from the called party or when the user goes on-hook.On expiry of timer, call is disconnected and user gets error tone. The range of SIP Provisional Timer is

010 to 180 seconds. By default, it is 60 seconds.

• General Request Timer: It is the time for which the ATA1S waits for the response of a transaction

request. This timer starts on initiating a transaction. This timer stops on receipt of a response for the

request. On expiry of timer, the ATA1S clears the transaction. The range of General Request Timer is

10 to 60 seconds. By default, it is 20 seconds.

Enable SIP over TCP?

• This feature allows ATA to send or receive SIP messages over UDP or TCP connection. It is applicable for

both Proxy and Peer to Peer calls. To achieve this, flag is provided as 'Yes' or 'No' for parameter 'Enable




SIP Over TCP?'. It is significant when the option for 'Default Transport for Outgoing Messages' is set as

'TCP' or 'TCP (with fallback to UDP)'. Refer topic “SIP Trunks” for details about default transport for OG

messages.

• If the flag is set as 'Yes', then only ATA will try to send or receive SIP messages over TCP. Else, the ATA

will send or receive SIP messages over UDP only. By default, it is 'Yes'.

UDP NAT Keep Alive

When the ATA is connected behind the NAT router and SIP messages are transported over UDP then UDP NAT

Keep Alive messages are to be sent to refresh the UDP binding in the NAT router. UDP NAT Keep Alive settings

can be done as follows:

Enable UDP NAT Keep Alive?: Select 'Yes' if you want the ATA to send keep alive messages periodically to

refresh the binding in NAT router. By default, it is No.

Interval (Seconds): Select the time period after which the ATA should send keep alive messages. This time

period should be less than the UDP binding timer of the router. The valid range is 001 to 999 seconds. By

default, it is 120 seconds.

Message Type: Select message type if UDP NAT Keep Alive is enabled. Select either REGISTER or

NOTIFY. By default, it is NOTIFY.

TCP NAT Keep Alive

When the ATA is connected behind the NAT router and SIP messages are transported over TCP then TCP NAT

Keep Alive messages are to be sent to refresh the TCP binding in the NAT router. The ATA sends CR-LF (Carriage

Return-Line Feed) message as TCP NAT Keep Alive message to refresh the TCP binding. TCP NAT Keep Alive

settings can be done as follows:

Enable TCP NAT Keep Alive?:

Select 'Yes' if you want the ATA to send Keep Alive messages periodically to refresh the binding in the NAT router.

By default, it is No.

Interval (Seconds): Select Time period after which the ATA should send Keep Alive messages. This time period

should be less than the TCP binding timer of the router. The valid range is 0001-9999 seconds.

By default, it is 120 seconds.

STUN

• Enable: Enable STUN (Simple Traversal of UDP through NAT) if ATA1S is located behind the NAT router.

STUN server facilitates traversing through most NATs, except symmetric NATs. If your router has

Symmetric NAT, do not use STUN.

• Server Address: Enter STUN Server Address if STUN is enabled. Server Address can be of maximum 40

characters.

• Only characters (0-9, A to Z, a to z and .) are allowed and you can also use the combination of alphabets

or numerals separated by "-". By default, STUN server address is blank. Refer topic “SIP Trunks”.

• Server Port: Enter STUN Server's listening port. Valid range of Server Port is from 1024 to 65535. By

default, the value is 3478.

• Use SIP Port fetched using STUN?:




If the ATA1S is located behind the NAT router and you have forwarded the SIP listening port of the ATA1S

in the router to the ATA1S, you do not need to use the port provided by STUN server as SIP listening port.

Hence, set this field to 'No' in those scenarios. Else, set this field to 'Yes' if you have not forwarded the 'SIP

port in the router' to the ATA1S. When the flag is set to 'No', the ATA will use the Local Programmed SIP

listening port in SIP signaling.

When the flag is set to 'Yes', the ATA will use the port provided by the STUN server as SIP listening port in

SIP signaling.

By Default it is programmed as 'Yes'.

Router's Public IP Address:

• 'Router's Public IP Address' is the public IP Address of the NAT router, behind which the ATA1S is located.

• Program this parameter only if 'Use Router's Public IP Address' option is selected for parameter "Source

Port IP Address" programmed in the Web page 'SIP Trunks'. The 'Router's Public IP Address' is also

displayed in 'Status' page. Refer topics “SIP Trunks” and “Checking the Status”.

By default, it is blank.

VLAN/CoS (Layer2):

VLAN is defined as: 'When switch creates multiple broadcast domains by putting some of its interfaces in one

broadcast domain and other interfaces into other broadcast domain, then each broadcast domain is referred to

Virtual LAN (VLAN)'.

This feature allows the ATA to add VLAN header (Layer2 VLAN header), in the packets leaving the WAN port of the

ATA. VLAN Tag will be added in all the packets relayed from LAN to WAN port of ATA. The VLAN header includes

following information:• VLAN ID

• Class of Service (CoS) (Layer2 or L2)

VLAN ID is the Identifier, assigned to each physical port of switch, to segment the LAN into Virtual LAN.

Quality of Service (QoS) on Layer2 is referred to 'Class of Service (CoS)' which is defined by IEEE 802.1P. ATA

supports CoS for SIP and RTP packets. The system user should program different CoS bits for SIP packets and

RTP packets, since priory of SIP and RTP packet is different due to real time nature of RTP packets.

To achieve this feature, flag is provided to enable/disable VLAN/CoS (Layer2).

The system user should configure following parameters for VLAN feature in ATA.

• Enable VLAN/CoS?: It allows enabling or disabling the, 'VLAN /CoS' feature for the ATA.

Select 'Yes' if the system should add VLAN header in packets generated by ATA so that user can use the

VLAN feature. Select 'No' if the system should not support VLAN feature. By default it is No.

Admin or User can also use command to enable or disable VLAN/CoS. It will be useful in following

scenario:

Suppose, you have enabled the VLAN and set the VLAN ID to 2, this will result in configuring the Jeeves

from PC which resides in VLAN 2. Now if you have no PC which resides in VLAN 2, you will not be able to

access Jeeves from WAN Port for further configuration. Hence using this command, you can first disable

L2 VLAN/CoS and then access Jeeves.




Both Admin and User can issue the command from SLT to enable/ disable L2 VLAN/ CoS as explained

below:

Use the following command to enable/ disable L2 VLAN/ CoS on WAN port:

16-Code-#*

Where,

By default, it is disabled.

• VLAN ID: It signifies the ID which should be added in all the packets relayed from LAN to WAN.

ATA1S will receive the packets on its WAN port only if tagged with programmed VLAN-ID irrespective

whether packets are destined for WAN or any host located on LAN port.

Generally, the 'Group' of users which works together is assigned the same VLAN ID.

You can configure maximum of 4094 number of VLANs in switch which means that you can have 4094

groups in a single LAN.

The range for 'VLAN ID' is 0-4094. By default it is 1.

• SIP CoS: It signifies the 'L2 CoS (priority) bits' which should be added in Layer2 VLAN header of all SIP

packets generated by the system when VLAN feature is enabled.

The range for 'SIP CoS' is 0-7.

Program the CoS bits for all the SIP packets relayed from LAN to WAN as '0'.

The CoS bits to be programmed with respect to traffic type are as follows:

By default it is 3.

• RTP CoS: It signifies the 'L2 CoS (priority) bits' which should be added in Layer2 VLAN header of all RTP

packets generated by the system when VLAN feature is enabled.

The range for 'RTP CoS' is 0-7.

Program the CoS bits for all the RTP packets relayed from LAN to WAN as '0'.

Code Connection Type

0 Disable

1 Enable

L2 CoS (priority) bits for SIP Traffic Type

0 Best Effort

1 Background

2 Spare

3 Excellent Effort

4 Controlled Load

5 Video

6 Voice

7 Network Control




The CoS bits to be programmed with respect to traffic type are as follows:

By default, it is 6.

QoS (Layer 3)

Quality of Service (QoS) refers to priority of IP packets on network layer. QoS is programmed for both

signaling (SIP) and media (RTP).

• SIP DiffServe: This field defines the priority bits for SIP messages. ATA will send all SIP messages using

DiffServe SIP QoS (Quality of Service) setting. Range for SIP Diffserve is from 00 to 63. By default, SIP

DiffServe is 26.

• RTP DiffServe: This field defines the priority bits for RTP packets. ATA will send all the RTP packets using

DiffServe RTP QoS setting. Range for RTP Diffserve is from 00 to 63. By default, RTP DiffServe is 46.

SIP/RTP Port:

SIP UDP Port: SIP UDP Port defines the port on which the ATA listens for SIP messages transported over

UDP. The ATA also uses this port as source port in SIP messages while sending SIP messages to the remote

peer. The range of SIP UDP Port is 1024-65534. By default, it is 5060.

SIP TCP Port: SIP TCP Port defines the port on which the ATA listens for SIP messages transported over

TCP.

The ATA also uses this port as source port in SIP messages while sending SIP messages to the remote peer.

The range of SIP TCP Port is 1024-65534. By default, it is 5060.

RTP Listening Port: RTP Listening port defines the port on which the ATA listens for RTP packets. The ATA

also uses this port as source port in RTP packets while sending RTP to remote peer. The range of RTP

Listening Port is 1024-65530. By default, it is 8000.

L2 CoS (priority) bits for RTP Traffic Type

0 Best Effort

1 Background

2 Spare

3 Excellent Effort

4 Controlled Load

5 Video

6 Voice

7 Network Control




CHAPTER 6 Software Upgrade

The SETU ATA1S is provided with an FTP server that allows you to upgrade the software (new firmware,

configuration files, drivers, etc.).

You can update configuration settings and upgrade firmware only via Web Jeeves.

Open the Web Jeeves and click on 'Software Upgrade' link. (Only Admin can access this Webpage).





When you enter the Admin password (default is 1234) and click on 'Log on', you will get new FTP server window as

shown:

As per files distribution in the folders, the Web page will display following folders:

1. system

2. config

3. driver

4. web

5. dhcp

6. ppp

'system' folder includes the current firmware/software used by ATA.

'config' folder includes the files for the feature configuration files of ATA.

'driver' folder includes the files for current drivers used by ATA.

'web' folder includes files for the html files.

'dhcp' folder includes files for DHCP configuration.

'ppp' folder includes files for PPPoE configuration.

You can click on any of the above folders to view the relevant files that the ATA is currently using. You can

delete/copy/paste the relevant files.

Now, to update the files on the FTP server, copy and paste the desired updated file in the related folder in the

FTP server window. The existing file will be replaced by the new one.

OR

Delete the file that you want to update; copy and paste the new file you want to replace it with.

Switch off the ATA and start again. The new configuration/driver/firmware you have copied and pasted on the

FTP server will take effect only during the next power ON.

Before you upgrade software, you can save the existing application/configuration/driver/ web pages

folders and files as back-up for retrieval, later. For example: If you want to retain the existing Configuration

folder for future record, you can save the folder on your PC. When required, you can copy and paste the

same folder back on the FTP server.




CHAPTER 7 Reinstate the Default

Settings

Matrix ATA1S enables the user to default the system parameter or to reset the system by using 'System Default'

feature. This feature will set parameters to factory set values but setting default values to the parameters will

depend on the login type i.e. login by 'User password' or 'Admin password'.

ATA supports following methods to default the system:1. Default System through Web Jeeves

2. Default System using SLT command

Default System through Web Jeeves

For Resetting ATA1S through the Web page, you should follow the steps shown below:

• Open the Web Jeeves of ATA1S and click on 'Factory Defaults' link. (Both Admin and User can access this

Web page).

• Alert Window for the message will appear as shown above. Click on 'OK' or 'Cancel'.

• If you click on 'OK' the parameters will be assigned default values and if you click on 'Cancel' the

parameters will not be assigned default values.

Default System using SLT command

• Enter the programming mode with Admin or User Password. For relevant commands refer “Configuring

using Phone” in chapter "Knowing SETU ATA1S".

• To default the system using SLT, use following command:

21-Reverse Admin/User Password-#*

Enter reverse 'User' password if you have logged in as 'User'.

Enter reverse 'Admin' password if you have logged in as 'Admin'.

• In case you forgot the password, use the Access Code '#***' to default the 'Admin' and 'User' Passwords.

Refer topics “Admin Password” and “User Password” for detailed information.




If you login using 'User Password'

• Now, if you have login using 'User' Password, then only following Webpages will get default values: (Also if

you have login using 'User' password, then you can access only following pages of Web Jeeves, all

remaining pages are accessible only by Admin):

a. Factory Defaults

b. LAN Port

c. PCAP Trace

d. Speed Dialing

e. Status

f. Supplementary Services

g. User Password

h. WAN Port

The Admin Password will also be defaulted, if the option 'Default Admin password when User defaults the ATA?' is

selected as 'yes' in the 'System Parameter' page.

If you login using 'Admin Password'

• If you login using Admin Password, you can access all the Web pages. If you default the ATA1S, allparameters will get default values except following parameters:

a. Time Parameters (as explained in chapter 'Date and Time settings').

b. Daylight Saving Time (DST) Parameters (as explained in chapter 'Daylight Savings Time Adjustments).

c. CPTG and Ring (as explained in chapter 'System Parameters').




Appendix

Frequently Ask Questions (FAQs)

Q.1 Which internet telephony (VoIP) protocols are supported by ATA1S?

Ans. ATA1S supports Session Initiation Protocol (SIP) to make/receives calls on Internet network.

Q.2 Can I make a call to another SIP terminal (e.g. Soft Phone running on PC) in my LAN network?

Ans. Yes. You can make a call to any SIP terminal in your LAN just by dialing an IP address of the SIP terminal

only from FXS port of ATA1S. If you are not comfortable in dialing an IP address, you can use 'Peer to

Peer' feature which allows you to dial the number instead an IP address for such type of applications.

Q.3 I have only one Internet connection for Internet surfing. Do I need another internet connection for making

VoIP calls using ATA1S?

Ans. No. You do not need another internet connection to use ATA1S. ATA1S has inbuilt router which enables

you to surf internet and making VoIP calls at the same time.

Q.4 I am using the ATA1S to make VoIP calls and surf internet. I have connected my PC to LAN port of ATA1S.

Can I still surf the internet when ATA1S is power down?

Ans. No. You can not surf Internet if ATA1S is power down. You must power ON the ATA1S to enable Internet

surfing on your PC. Else, connect the PC directly to the internet.

Q.5 I am using the ATA1S to make VoIP calls and surf internet. I have connected my PC to LAN port

of ATA1S. Can I still make VoIP calls when my PC is shut down or not connected?

Ans. Yes. You can still make VoIP calls when your PC is shutdown or not connected.

Q.6 My Internet Service Provider (ISP) supports only PPPoE for internet services. Can I use ATA1S with my

ISP?

Ans. Yes. You need to configure PPPoE User-Id and PPPoE Password in ATA1S provided by your ISP.

Q.7 My internet Service Provider (ISP) supports only DHCP to provide internet services. Can I use ATA1S with

my ISP?

Ans. Yes. You need to enable the DHCP client in ATA1S.

Q.8 My internet service provider (ISP) provides me only dynamic IP address. Can I use ATA1S to make VoIP

calls and surf Internet at the same time?

Ans. Yes. Enable DHCP or PPPoE client in ATA1S as used by your ISP to provide internet services.

Q.9 I do not have PC to access Web Jeeves for configuration. Can I change the IP address and subnet mask

of LAN/WAN port?

Ans. Yes. You can change the IP address and subnet mask of LAN/ WAN port of ATA1S if you do not have PC

to access Web Jeeves. You need to issue commands from the FXS port to change the IP address and




subnet mask.

Q.10 I do not have PC to access Web Jeeves for configuration. Can I change the internet connection type of

ATA1S to DHCP or PPPoE or Static?

Ans. Yes. You can change the connection type by issuing command 37-#* from the FXS port of ATA1S and

configuring necessary parameters for the new connection type.

Q.11 I do not have PC to access Web Jeeves for configuration. Can I know whether ATA1S is registered with

ITSP or not?

Ans. Yes. You can know the status of your SIP registration using LED provided on the enclosure of ATA1S or by

command to display registration status on the Phone connected to FXS port.

Q.12 I have connected my PC to LAN port of ATA1S. But, I am not able to access the Web Jeeves for

configuration.

Ans. Please ensure that the IP address of LAN port of ATA1S and IP address of your computer should be in

same subnet to enable Web Jeeves from your computer. If both the IP address is in the same subnet,

check for any error in the Ethernet cable which connects your computer to LAN port of ATA1S.

Q.13 Can I upgrade my ATA1S for new firmware?

Ans. Yes. You can upgrade the ATA1S for new firmware. To upgrade ATA1S for new firmware, please refer

“Software Upgrade”.

Q.14 Can I upgrade the ATA1S for new firmware remotely?

Ans. Yes. If you know the public IP address of WAN port of ATA1S, you can use the FTP server of ATA1S to

upgrade the firmware. Please refer “Auto Configuration” and “Software Upgrade”.

Q.15 If I forgot the password, can I still use the Web Jeeves for configuration?

Ans. Yes. But you need to set the password value to factory default.

There are two important passwords for the configuration of ATA1S viz. Admin Password and User

Password. Admin password protects the access to whole configuration while User password protects the

access to only few configuration of ATA1S.

If you forgot the user password but you still have access to admin password, you can login to Web Jeeves

and change the user password. But, if you forgot the admin password, you need to default the Password

using access code '#***' from the Phone, connected to the ATA1S.

Refer “Reinstate the Default Settings”, “Admin Password” and “User Password” for more details.

Q.16 I am not able to access the Web Jeeves since I forgot the IP address of LAN port of ATA1S. How can I

know the IP address of LAN port?Ans. You can use the system commands to know the IP address of LAN/WAN port of ATA1S. Please ensure

that you have the SLT (compatible for CLI type) connected to the FXS port.

Q.17 Can I change the configuration of ATA1S to factory default?

Ans. Yes. You can change the configuration to factory default. Please refer “Reinstate the Default Settings” for

this information.

Q.18 Can I select the SIP Trunk to make OG call?

Ans. Yes. You can dial first the access code for the SIP1, SIP2 or SIP3 and then dial the required number for

OG call.




Q.19 Can I use the ATA1S for Least Cost Routing (LCR) application?

Ans. Yes. You can do this by selecting 'Dialed Number based Routing' for parameter 'OG Call Route' in Web

page for 'FXS Port'. After selecting option for routing, program in the Webpage 'Dialed Number Table', the

numbers to be dialed and corresponding SIP trunk as 'Destination Trunk', which is cheaper for the call to

be made. Refer topics “FXS Port” and “Dialed Number Table” for more information.

Q.20 Can I route the Incoming calls on SIP?

Ans. Yes. You can configure suitable routing option to route the IC call on SIP to the FXS port, by selecting the

parameter for 'Incoming Call Route' in the Webpage 'SIP Trunk'. Refer topic “SIP Trunks”.




Acronyms

ASCII American Standard Code for Information Interchange

CLI Caller Line Identification

CPT Call Progress Tone

CPTG Call Progress Tone Generation

CCWT CO Call Waiting Tone

DHCP Dynamic Host Configuration Protocol

DND Do Not Disturb

DNS Domain Name Server

DST Daylight Savings Time

DTMF Dual Tone Multi-Frequency

FEC Forward Error Correction

FTP File Transfer Protocol

GMT Greenwich Mean Time

IC Incoming call

IP Internet Protocol

ISP Internet Service Provider

ITSP Internet Telephony Service Provider

ITU International Telecommunication Union

LAN Local Area Network

LED Light Emitting Diode

MAC Media Access Control

ms Milliseconds

MPLS Multi Protocol Label Switching

NAT Network Address Traversal

NTP Network Time Protocol

OG Outgoing Call

PBX Private Branch Exchange

PSTN Public Switched Telephone Network

P2P Peer to Peer

PPPoE Point to Point Protocol over Ethernet

QoS Quality of Service

RBT Ring Back Tone

RFC Request for Comments

RTP Real Time Transport ProtocolSIP Session Initiation Protocol

SLT Single Line Telephone

SNTP Simple Network Time Protocol

SOHO Small Office Home Office

STUN Simple Traversal of UDP over NAT

TCP Transmission Control Protocol

TFTP Trivial File Transfer Protocol

UDP User Datagram Protocol

URI Uniform Resource Identifier

URL Universal Reference/Resource Locator VAD Voice Activity Detection




WAN Wide Area Network

VoIP Voice Over Internet Protocol




Features at a Glance

DESCRIPTION FEATURE CODE

To enter programming mode for Admin #19-Admin Password

To enter programming mode for User #18-User Password

To exit programming mode 0

Speed dialing #9-Index Number

To default User/Admin Password #***

Selective Trunk Access-SIP1 #84






Product Specifications

Note1: ATA1S supports maximum of 3 SIP trunks.

VoIP and Networking Protocol

SIP (RFC 3261)IPv4- Internet Protocol Version 4 (RFC 791)

ARP- Address Resolution Protocol (RFC 0826)

ICMP- Internet Control Message Protocol (RFC 0792)

TCP- Transmission Control protocol (RFC 793)

UDP- User Datagram Protocol (RFC 768)

RTP- Real Time Protocol (RFC 1889)

RTCP- Real Time Control Protocol (RFC 1889)

SNTP- Simple Network Time Protocol

PPP- Point-to-Point Protocol (RFC 1661)

DNS- Protocol (RFC 2136)

Outbound Proxy

STUN

NAT Keep Alive

VLAN- Virtual LAN (Layer2)

Voice Functionalities

SIP Trunks: 3

DTMF: Inband, Outband and SIP (INFO)

Call Progress Tone Generation: Select as per the country

QoS (Layer3): RTP QoS and SIP QoS using DiffServe

Full Duplex Audio

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 (UDPTL) and Pass Through

Flash Timer Programming

Voice Codec's

G.711 (a-Law and u-Law), G.729, G.723

PORT NAME (Refer

Note1 below Table)APPLICATION NO. OF PORTS CONNECTOR TYPE

FXS Port For connecting Analog Phone or Fax Machine 1 RJ 11

LAN Port To connect a Computer 1 RJ 45, 10/100 Base T

WAN PortTo connect Ethernet cable router provided by

service provider 1 RJ 45, 10/100 Base T

DC JackTo connect the power adapter for DC supply

voltage1 DC Power Jack




Telephony features

Voice Calls using SIP proxy and Voice calls without using SIP proxy (Peer-to-Peer Calling)

Call Waiting

Caller ID Blocking

Call Forwarding

3-Party conferenceCall Transfer

Call Hold

Caller ID: Generation and Display

Answer signaling

Disconnect signaling

Speed Dialing

Do Not Disturb (DND)

Polarity Reversal

Call Toggle

Time Settings

Synchronizing with specific Time Server

Provisioning, Administration and Maintenance

Auto Configuration

Programming using Web Jeeves

Software Upgrade

PCAP Trace

Programming of parameters using commands from conventional Phone, such as:

• LAN IP Address

• WAN IP Address

LED Indication

LED for Power

LED for FXS

LED for SIP1

LED for SIP2

LED for SIP3

Security

Password Protected Administrator and User settings

Dimension (W x H x D)

104mm x 27mm x 80mm (4.09" x 1.06" x 3.15")

Power SupplyExternal Adaptor: 12VDC@ 1.0Amp.






System Commands

Command Meaning Allowed to?

11-WAN IP Address-#* To program the WAN IP Address Admin & User

12-LAN IP Address-#* To program the LAN IP Address Admin & User

13-Subnet Mask-#* To program the Subnet Mask of WAN port Admin & User

14-Subnet Mask-#* To program the Subnet Mask of LAN port Admin & User

15-Code-#* To program the connection type of WAN port.

Where, Code is 1 for DHCP, 2 for PPPoE and 3 for Static.

Admin & User

16-Code-#* To enable/disable 'L2 VLAN/ CoS' on WAN port.

Where, Code 0 is for Disable and 1 is for Enable.

Only Admin

21-Reverse Admin/User

Password-#*

To default the ATA1S Admin & User

31- #* To display the WAN IP Address on SLT Admin & User

32-#* To display the LAN IP address on SLT Admin & User

33-#* To display the WAN Subnet Mask on SLT Admin & User 34-#* To display the LAN Subnet Mask on SLT Admin & User

35-#* To display the gateway address on SLT Admin & User

36-#* To display the Primary DNS address on SLT Admin & User

37-#* To display the connection type of WAN port on SLT Admin & User

38 - SIP Trunk - #* To display the registration status of SIP.

Where, SIP Trunk is 1 for SIP1, 2 for SIP2 and 3 for SIP3.

Admin & User

39-#* To display the Secondary DNS address on SLT Admin & User

51-Destination Number-#* To program Call Forward-Unconditional number Admin & User

52-Code-#* Set/Cancel Call Forward-Unconditional

Where, Code is 1 to set and 0 to cancel Call Forward -

Unconditional

Admin & User

53-Destination Number-#* To program Call Forward-When Busy number Admin & User

54-Code-#* Set/Cancel Call Forward-When Busy number

Where, Code is 1 to set and 0 to cancel Call Forward - When

Busy

Admin & User

55-Destination Number-#* To program Call Forward-When No Reply number Admin & User

56-Code-#* Set/Cancel Call Forward-When No Reply

Where, Code is 1 to set and 0 to cancel Call Forward - No Reply

Admin & User

57-Timer-#* Set Call Forward Ring Timer Admin & User

61-Code-#* Set/Cancel DND Where, Code is 1 to set and 0 to cancel DND Admin & User

66-SIP Trunk-Code-#* Send/Block Caller ID for SIP1, SIP2, SIP3Where, SIP Trunk is 1 for SIP1, 2 for SIP2 and 3 for SIP3.

Where, Code is 1 to block and 0 to send Caller ID

Only Admin

72-Code-#* Set/Cancel Call Waiting

Where, Code is 1 to set and 0 to cancel Call Waiting

Admin & User

99-#* Cancel all features Admin & User




Warranty Statement

Matrix Comsec 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 the property 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 bycustomer 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.

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 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 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 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.




Open Source Licensing Terms and Conditions

• The firmware of this product also includes some of the Open-Source software released under GNU

General Public License (GPL) Version 2. Terms of this license is printed in full below.

• The source of the open source software used in this product is available on CD, upon written request from:

R&D Team

Matrix Comsec Pvt Ltd

394, Makarpura GIDC,

Vadodara - 390 010

Gujarat

India.

Customer shall bear the shipping and handling charges.

GNU GENERAL PUBLIC LICENSE

Version 2, June 1991

Copyright (C) 1989, 1991 Free Software Foundation, Inc.,

51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

Everyone is permitted to copy and distribute verbatim copies

of this license document, but changing it is not allowed.

Preamble

The licenses for most software are designed to take away your

freedom to share and change it. By contrast, the GNU General Public

License is intended to guarantee your freedom to share and change free

software--to make sure the software is free for all its users. This

General Public License applies to most of the Free Software

Foundation's software and to any other program whose authors commit tousing it. (Some other Free Software Foundation software is covered by

the GNU Lesser General Public License instead.) You can apply it to

your programs, too.

When we speak of free software, we are referring to freedom, not

price. Our General Public Licenses are designed to make sure that you

have the freedom to distribute copies of free software (and charge for

this service if you wish), that you receive source code or can get it

if you want it, that you can change the software or use pieces of it

in new free programs; and that you know you can do these things.

To protect your rights, we need to make restrictions that forbid

anyone to deny you these rights or to ask you to surrender the rights.These restrictions translate to certain responsibilities for you if you

distribute copies of the software, or if you modify it.

For example, if you distribute copies of such a program, whether

gratis or for a fee, you must give the recipients all the rights that

you have. You must make sure that they, too, receive or can get the

source code. And you must show them these terms so they know their

rights.

We protect your rights with two steps: (1) copyright the software, and

(2) offer you this license which gives you legal permission to copy,

distribute and/or modify the software.

Also, for each author's protection and ours, we want to make certain




that everyone understands that there is no warranty for this free

software. If the software is modified by someone else and passed on, we

want its recipients to know that what they have is not the original, so

that any problems introduced by others will not reflect on the original

authors' reputations.

Finally, any free program is threatened constantly by softwarepatents. We wish to avoid the danger that redistributors of a free

program will individually obtain patent licenses, in effect making the

program proprietary. To prevent this, we have made it clear that any

patent must be licensed for everyone's free use or not licensed at all.

The precise terms and conditions for copying, distribution and

modification follow.

GNU GENERAL PUBLIC LICENSE

TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION

0. This License applies to any program or other work which contains

a notice placed by the copyright holder saying it may be distributedunder the terms of this General Public License. The "Program", below,

refers to any such program or work, and a "work based on the Program"

means either the Program or any derivative work under copyright law:

that is to say, a work containing the Program or a portion of it,

either verbatim or with modifications and/or translated into another

language. (Hereinafter, translation is included without limitation in

the term "modification".) Each licensee is addressed as "you".

Activities other than copying, distribution and modification are not

covered by this License; they are outside its scope. The act of

running the Program is not restricted, and the output from the Program

is covered only if its contents constitute a work based on the

Program (independent of having been made by running the Program).

Whether that is true depends on what the Program does.

1. You may copy and distribute verbatim copies of the Program's

source code as you receive it, in any medium, provided that you

conspicuously and appropriately publish on each copy an appropriate

copyright notice and disclaimer of warranty; keep intact all the

notices that refer to this License and to the absence of any warranty;

and give any other recipients of the Program a copy of this License

along with the Program.

You may charge a fee for the physical act of transferring a copy, and

you may at your option offer warranty protection in exchange for a fee.

2. You may modify your copy or copies of the Program or any portion

of it, thus forming a work based on the Program, and copy and

distribute such modifications or work under the terms of Section 1

above, provided that you also meet all of these conditions:

a) You must cause the modified files to carry prominent notices

stating that you changed the files and the date of any change.

b) You must cause any work that you distribute or publish, that in

whole or in part contains or is derived from the Program or any

part thereof, to be licensed as a whole at no charge to all third

parties under the terms of this License.

c) If the modified program normally reads commands interactivelywhen run, you must cause it, when started running for such






If distribution of executable or object code is made by offering

access to copy from a designated place, then offering equivalent

access to copy the source code from the same place counts as

distribution of the source code, even though third parties are not

compelled to copy the source along with the object code.

4. You may not copy, modify, sublicense, or distribute the Programexcept as expressly provided under this License. Any attempt

otherwise to copy, modify, sublicense or distribute the Program is

void, and will automatically terminate your rights under this License.

However, parties who have received copies, or rights, from you under

this License will not have their licenses terminated so long as such

parties remain in full compliance.

5. You are not required to accept this License, since you have not

signed it. However, nothing else grants you permission to modify or

distribute the Program or its derivative works. These actions are

prohibited by law if you do not accept this License. Therefore, by

modifying or distributing the Program (or any work based on the

Program), you indicate your acceptance of this License to do so, andall its terms and conditions for copying, distributing or modifying

the Program or works based on it.

6. Each time you redistribute the Program (or any work based on the

Program), the recipient automatically receives a license from the

original licensor to copy, distribute or modify the Program subject to

these terms and conditions. You may not impose any further

restrictions on the recipients' exercise of the rights granted herein.

You are not responsible for enforcing compliance by third parties to

this License.

7. If, as a consequence of a court judgment or allegation of patent

infringement or for any other reason (not limited to patent issues),

conditions are imposed on you (whether by court order, agreement or

otherwise) that contradict the conditions of this License, they do not

excuse you from the conditions of this License. If you cannot

distribute so as to satisfy simultaneously your obligations under this

License and any other pertinent obligations, then as a consequence you

may not distribute the Program at all. For example, if a patent

license would not permit royalty-free redistribution of the Program by

all those who receive copies directly or indirectly through you, then

the only way you could satisfy both it and this License would be to

refrain entirely from distribution of the Program.

If any portion of this section is held invalid or unenforceable under

any particular circumstance, the balance of the section is intended to

apply and the section as a whole is intended to apply in other

circumstances.

It is not the purpose of this section to induce you to infringe any

patents or other property right claims or to contest validity of any

such claims; this section has the sole purpose of protecting the

integrity of the free software distribution system, which is

implemented by public license practices. Many people have made

generous contributions to the wide range of software distributed

through that system in reliance on consistent application of that

system; it is up to the author/donor to decide if he or she is willing

to distribute software through any other system and a licensee cannot

impose that choice.

This section is intended to make thoroughly clear what is believed to




be a consequence of the rest of this License.

8. If the distribution and/or use of the Program is restricted in

certain countries either by patents or by copyrighted interfaces, the

original copyright holder who places the Program under this License

may add an explicit geographical distribution limitation excluding

those countries, so that distribution is permitted only in or amongcountries not thus excluded. In such case, this License incorporates

the limitation as if written in the body of this License.

9. The Free Software Foundation may publish revised and/or new versions

of the General Public License from time to time. Such new versions will

be similar in spirit to the present version, but may differ in detail to

address new problems or concerns.

Each version is given a distinguishing version number. If the Program

specifies a version number of this License which applies to it and "any

later version", you have the option of following the terms and conditions

either of that version or of any later version published by the Free

Software Foundation. If the Program does not specify a version number ofthis License, you may choose any version ever published by the Free Software

Foundation.

10. If you wish to incorporate parts of the Program into other free

programs whose distribution conditions are different, write to the author

to ask for permission. For software which is copyrighted by the Free

Software Foundation, write to the Free Software Foundation; we sometimes

make exceptions for this. Our decision will be guided by the two goals

of preserving the free status of all derivatives of our free software and

of promoting the sharing and reuse of software generally.

NO WARRANTY

11. BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY

FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN

OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES

PROVIDE THE PROGRAM "AS IS" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED

OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF

MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS

TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE

PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,

REPAIR OR CORRECTION.

12. IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING

WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR

REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,

INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING

OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED

TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY

YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER

PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE

POSSIBILITY OF SUCH DAMAGES.

END OF TERMS AND CONDITIONS

How to Apply These Terms to Your New Programs

If you develop a new program, and you want it to be of the greatest

possible use to the public, the best way to achieve this is to make it

free software which everyone can redistribute and change under these terms.




To do so, attach the following notices to the program. It is safest

to attach them to the start of each source file to most effectively

convey the exclusion of warranty; and each file should have at least

the "copyright" line and a pointer to where the full notice is found.

<one line to give the program's name and a brief idea of what it does.>

Copyright (C) <year> <name of author>

This program is free software; you can redistribute it and/or modify

it under the terms of the GNU General Public License as published by

the Free Software Foundation; either version 2 of the License, or

(at your option) any later version.

This program is distributed in the hope that it will be useful,

but WITHOUT ANY WARRANTY; without even the implied warranty of

MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the

GNU General Public License for more details.

You should have received a copy of the GNU General Public License along

with this program; if not, write to the Free Software Foundation, Inc.,51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.

Also add information on how to contact you by electronic and paper mail.

If the program is interactive, make it output a short notice like this

when it starts in an interactive mode:

Gnomovision version 69, Copyright (C) year name of author

Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.

This is free software, and you are welcome to redistribute it

under certain conditions; type `show c' for details.

The hypothetical commands `show w' and `show c' should show the appropriate parts

of the General Public License. Of course, the commands you use may be called

something other than `show w' and `show c'; they could even be mouse-clicks or

menu items--whatever suits your program.

You should also get your employer (if you work as a programmer) or your

school, if any, to sign a "copyright disclaimer" for the program, if

necessary. Here is a sample; alter the names:

Yoyodyne, Inc., hereby disclaims all copyright interest in the program

`Gnomovision' (which makes passes at compilers) written by James Hacker.

<signature of Ty Coon>, 1 April 1989

Ty Coon, President of Vice

This General Public License does not permit incorporating your program into

proprietary programs. If your program is a subroutine library, you may

consider it more useful to permit linking proprietary applications with the

library. If this is what you want to do, use the GNU Lesser GeneralPublic License instead of this License.



Index

Numerics

100rel/PRACK 1223-Party Conference 44

A

AC Impedance 83 Admin 2 Admin Password 57

New Password 57Old Password 57

Anonymous Call Rejection 100 Answer Signaling Type 82 Authentication Password 97 Authentication User-Id 97 Automatic Number Translation 100

B

Backward Time Adjustments 67

C

Call Forward

Unconditional 46When Busy 46

Call Hold 42Call Hold Method 102Call Progress Tone Generation 112Call Release Timer 115

Call Toggle 43Call Transfer-Attended 43Call Transfer-Blind 43Call Waiting 44Calling Party Number Presentation (CLIP) 82Class of Service 81Current Date 65Current Time 65

D

Daylight Saving Time (DST) 66Dialed Number Based Routing 73Digest Authentication 77

Direct IP Calling 40Disconnect Signaling Type 82

DND (Do Not Disturb) 48

E

Echo Cancellation 83End of dialing 73

F

FAX Option 102First Digit Wait Timer 115Flash Timer 83Forward Error Correction (FEC) 116Forward Time Adjustments 67FXS port 79FXS Port Status 80

I

Incoming Calls 98Inter Digit Wait Timer 115

K

Kernel Date 37

L

LAN Port 85Language Selection 111Least Cost Routing (LCR) 73LEDs 6

M

MAC Address 122Making Second Call 42Multi Stage Dialing through SIP 52

O

Outbound Proxy 97Outgoing Calls 81

P

P2P trunk 93PCAP Trace 87Proxy trunk 93



Q

QoS (Layer 3) 126

R

Remote Held 42restart 92Ring Timer 83Rings 114rinstance 96Router's Public IP Address 124Routing Tone 113RTP Listening Port 126

S

SIP ID 95SIP over TCP 122SIP Trunk 95Speed Dial Table 104Static Routing Table 106Status 36STUN 123Symmetric RTP 101Syslog Client 71System Default 129

T

TCP NAT Keep Alive 123Time Server 64Time Zone 65Transfer Notification Timer 116Transport Configuration 99

U

UDP NAT Keep Alive 123upgrade the software 127User 2User Password 117

New Password 117Old Password 117

V

VLAN/CoS (Layer2) 124Vocoder Preference 101Voice Activity Detection (VAD) 116VoIP Silence Disconnect Timer 115Volume Control 84

W

WAN port 119Web Jeeves 8





Documents

Setu Ata1s v1 System Manual