setu ata2ll system manual
TRANSCRIPT
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SETU ATA2LL
System Manual
MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.MATRIX TELECOM PVT. LTD.
Corporate Office: Factory:
394-GIDC, Makarpura, 39-GIDC, Waghodia-391 760
Vadodara-390 010, India Dist. Vadodara, India
Tel. : +91 265 2630555 Tel. : +91 2668 263172/73
Fax: +91 265 2636598 Fax: +91 2668 262631
E-mail: [email protected] E-mail: [email protected]
URL: www.MatrixTeleSol.com
Version 2 August 2006
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Documentation Information
This is a general documentation and it covers many models with
different specifications. A particular product may not support all the
features and facilities described in the documentation.
Matrix Telecom reserves the right to revise information in this
publication for any reason without prior notice. Information in this
documentation may change from time to time. Matrix Telecom
makes no warranties with respect to this documentation and
disclaims any implied warranties. While every precaution has been
taken in preparation of this system manual, Matrix Telecom
assumes no responsibility for errors or omissions. Neither is anyliability assumed for damages resulting from the use of the
information contained herein.
Matrix Telecom reserves the right without prior notice to make
changes in design or components of the equipment as engineering
and manufacturing may warrant.
Neither Matrix Telecom nor its affiliates shall be liable to the
purchaser of this product or third parties for damages, losses, costsor expenses incurred by purchaser or third parties as a result of:
accident, misuse or abuse of this product or unauthorized
modifications, repairs or alterations to this product or failure to
strictly comply with Matrix Telecom’s operating and maintenance
instructions.
All rights reserved. No part of this system manual may be copied or
reproduced in any form or by any means without the prior written
consent of Matrix Telecom.
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Contents
Section 1: Introduction ................................................................ 5Welcome ...................................................................................... 7
Packing List ................................................................................. 9
Warranty Statement ................................................................... 10
Introducing the System .............................................................. 13
Getting Started ........................................................................... 16
Section 2: Features and Facilities ............................................. 25Dial Plan ..................................................................................... 27
FXS Port .................................................................................... 29
LAN/WAN Explanation ............................................................... 39Jeeves ........................................................................................ 46
LAN Port ..................................................................................... 56
Life Line ...................................................................................... 58
Password .................................................................................... 60
Peer-to-Peer Calling ................................................................... 61
Phone Book ............................................................................... 64
Resetting the ATA ...................................................................... 65
SIP Accounts ............................................................................. 66
Software Upgrade ....................................................................... 71Status ......................................................................................... 72
Supplementary Services ............................................................ 75
System Name ............................................................................ 86
Time Settings ............................................................................. 87
VoIP Basics ............................................................................... 88
WAN Port ................................................................................... 95
Section 3: Appendices ............................................................... 97 Appendix A: Troubleshooting ..................................................... 99
Appendix B: Applications ......................................................... 101
Appendix C: Product Specifications ........................................ 103
Appendix D: Glossary .............................................................. 106
Appendix E: Feature at a Glance ............................................. 112
Appendix F: System Commands .............................................. 113
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Index ......................................................................................... 114
Notes .........................................................................................117
Programming Register ...........................................................119
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Section 1: Introduction
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Matrix
SETU ATA2LL System Manual 7
Welcome
Welcome to the world of telecom solutions from Matrix and thanks
for purchasing a Matrix product.
We want you to get the maximum performance from our product. If
you run into technical difficulties, we are here to help. But please
consult this system manual first.
If you still can’t find the answer, gather all the information or
questions that apply to your problem and with the product close to
you, call your dealer. Matrix dealers are trained and ready to give
you the support you need to get the most from your Matrix product.In fact, most problems reported are minor and can be easily solved
over the phone.
In addition, technical consultation is available from Matrix engineers
every business day. We are always ready to give advice on
application requirements or specific information on installation and
operation of our products.
The system manual is divided in following sections:Section 1: Introduction
Section 2: Features and Facilities
Section 3: Appendices
We suggest the first time users to read this system manual in the
following sequence and then remaining chapters:
• Section 1
• Section 2 (In hierarchy given below)
• Getting Started 16• Jeeves 46
• Life Line 58
• Peer-to-Peer Calling 61
• Supplementary Services 75
• Resetting the ATA 65
• Status 72
• Time Settings 87
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SETU ATA2LL System Manual8
• FXS Port 29
• LAN Port 56
• WAN Port 95
• VoIP Basics 88
• SIP Accounts 66• Dial Plan 27
• Phone Book 64
• Software Upgrade 71
• System Name 86
• Troubleshooting 99
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SETU ATA2LL System Manual 9
Packing List
The ideal sales package for SETU ATA2LL is as mentioned below:
Sr. Accessories Qty.01 SETU ATA2LL 1
02 System Manual 1
03 Adaptor 12V, 1.25Amp. 1
04 Mounting Screw 30/7 2
05 Screw Grip 2
06 Warranty Card set 1
07 Support Card 1
08 RJ11 Cable 3
09 RJ45 Cable 2
10 Mounting Template 1
11 SETU ATA2LL CD 1
• Please make sure that these components are present.
• In case of short supply or damage detection, contact the sourcefor where you have purchased the system.
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SETU ATA2LL System Manual10
Warranty Statement
Matrix Telecom Pvt. Ltd. (Matrix) warrants to its consumer
purchaser any of its products to be free of defects in material,workmanship and performance for a period of 15 months from date
of manufacturing or 12 months from the date of installation which
ever is earlier.
During this warranty period, Matrix will at its option, repair or replace
the product at no additional charge if the product is found to have
manufacturing defect. Any replacement product or part/s may be
furnished on an exchange basis, which shall be new or like-new,
provided that it has functionality at least equal to that of the product,
being replaced. All replacement parts and products will be theproperty of Matrix. Parts repaired or replaced will be under warranty
throughout the remainder of the original warranty period only.
This limited warranty does not apply to:
1. Products that have been subjected to abuse, accident, natural
disaster, misuse, modification, tampering, faulty installation, lack
of reasonable care, repair or service in any way that is not
contemplated in the documentation for the product, or if the
model or serial number has been altered, tampered with,
defaced or removed.
2. Products which have been damaged by lightning storms, water
or power surges or which have been neglected, altered, used for
a purpose other than the one for which they were manufactured,
repaired by customer or any party without Matrix’s written
authorization or used in any manner inconsistent with Matrix’s
instructions.
3. Products received improperly packed or physically damaged.4. Products damaged due to operation of product outside the
products’ specifications or use without designated protections.
Warranty valid only if:
• Primary protection on all the ports provided
• Mains supply is within limit and protected
• Environment conditions are maintained as per the product
specifications
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SETU ATA2LL System Manual 11
Warranty Card:
• When the product is installed, please return the warranty card
with:
• Date, signature and stamp of the customer
• Date, signature and stamp of the channel partner • Matrix assumes that the customer agrees with the warranty
terms even when the warranty card is not signed and returned as
suggested.
The Purchaser shall have to bear shipping charges for sending
product to Matrix for testing/rectification. The product shall be
shipped to the Purchaser at no-charge if the material is found to be
under warranty. The Purchaser shall have to either insure the
product or assume liability for loss or damage during transit.
Matrix reserves the right to waive off or make any changes in its
warranty policy without giving any notice.
If Matrix is unable to repair or replace, as applicable, a defective
product which is covered by Matrix warranty, Matrix shall, within a
reasonable time after being notified of the defect, refund the
purchase price of the product provided the consumer/purchaser
returns the product to Matrix.
In no event will Matrix be liable for any damages including lost
profits, lost business, lost savings, downtime or delay, labor, repair
or material cost, injury to person, property or other incidental or
consequential damages arising out of use of or inability to use such
product, even if Matrix has been advised of the possibility of such
damages or losses or for any claim by any other party.
Except for the obligations specifically set forth in this WarrantyPolicy Statement, in no event shall Matrix be liable for any direct,
indirect, special, incidental or consequential damages whether
based on contract or any other legal theory and where advised of
the possibility of such damages.
Neither Matrix nor any of its distributors, dealers or sub-dealers
makes any other warranty of any kind, whether expressed or
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Matrix
SETU ATA2LL System Manual12
implied, with respect to Matrix products. Matrix and its distributors,
dealers or sub-dealers specifically disclaim the implied warranties of
merchantability and fitness for a particular purpose.
This warranty is not transferable and applies only to the originalconsumer purchaser of the Product. Warranty shall be void if the
warranty card is not completed and registered with Matrix within 30
days of installation.
All legal course of action subjected to Vadodara (Gujarat, India)
Jurisdiction only.
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SETU ATA2LL System Manual 13
Introducing the System
• Matrix offers the “Voice over IP-Analog Telephone Adaptor”, VoIP
ATA or Matrix Phone Adaptor (SETU ATA2LL) which is a SIPbased VoIP product.
• Sending voice signals over the Internet is called Voice over IP or
VoIP. Session Initiated Protocol, SIP is an internationally
recognized standard for implementing VoIP.
• VoIP-ATA is an Analog Telephone Adapter that allows a user to
make voice call over IP network using a conventional telephone
instrument.
• VoIP-ATA has two IP ports, one for WAN connection and other
for LAN connection.• WAN port is used to connect VoIP-ATA to the Internet using
broadband modem, cable modem, router, etc. The VoIP calls are
made through WAN port.
• LAN port is used to connect a Computer to VoIP-ATA. Doing so,
the user can surf the Net as well as make VoIP calls from the
telephone instrument simultaneously.
• VoIP-ATA has a Lifeline port to which a Trunk line from PSTN can
be connected. However, the Lifeline port is not a true FXO port.
• VoIP-ATA has two FXS ports to which conventional telephone
instruments (SLTs) can be connected.
• Call routing mechanisms for IC calls from IP to FXS and OG calls
from FXS to IP are included in VoIP-ATA.
The IC call routing mechanism shall help the user to route the calls
to either or both the FXS ports whereas the OG call routing
mechanism shall help the user to route the call through desired SIP
Account which shall be helpful when services from two differentITSPs have been taken, each having cost benefit in some terms
over the other.
• When a telephone number (A E.164 number or a SIP number
without domain name. i.e. the prefix of the SIP telephone number
e.g. Suppose a user with SIP number SIP: [email protected] is to
be called then the caller should dial 12345) is dialed by the user,
the VoIP-ATA converts this into an IP call using SIP protocol and
makes a call to the callee.
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SETU ATA2LL System Manual14
• Using Phone book, the user can also dial complete SIP number
by just dialing feature code and corresponding index number.
• VoIP-ATA uses SIP protocol (RFC 3261) to make Voice calls
over IP.
• User can make a voice call on PSTN also, but using an accesscode (Prefix).
• Emergency calls can be placed on the Lifeline port. (However,
the caller has to dial the Emergency number twice).
• Refer ‘Feature List’ which includes important features supported
by VoIP-ATA.
Feature List:
• Configuration of the ATA through Web-pages
• Reset the ATA
• User Access with Password
• Synchronization with Time Server
• LAN Port IP Address programming
• Supplementary Services with T.38 Fax capability
• WAN Port IP Address programming
• Use of two SIP accounts, simultaneously
• Routing of Dialed Number as per Dial Plan
• Emergency Number Dialing• Speed dialing using Phone Book feature
• Peer-to-Peer Calling
• Status page display of main parameters
Before proceeding further, please ensure that you have an
Internet access and a SIP Account already set up.
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SETU ATA2LL System Manual 15
SETU ATA2LL Photograph:
Relevant Topics:
1. Glossary 1062. VoIP Basics 88
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SETU ATA2LL System Manual16
Getting Started
Overview
The Matrix ATA allows you to use the analog telephone to make
phone calls over the Internet. The ATA connects to the Internet
through a DSL Modem or Router as shown below:
This chapter gets the ATA up and running quickly. The details which
we have skipped to make this brief can be found elsewhere in the
manual. It is divided into four sections:
1. Getting to know the phone adapter
2. Instruction for connecting the phone adapter
3. Basic steps for configuration
4. Making phone calls
Getting to know the phone adapter The Matrix ATA’s ports are located on side panel and LEDs are on
front panel. Both are explained below:
• Connection of side panel ports
• Front panel LEDs
Side Panel Ports
Following picture shows how the ports are connected to various
interfaces:
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SETU ATA2LL System Manual 17
No. Port Name Connector Description
1 Power DC JackPower Adaptor, 12Volt 1.25Amp.
DC.
2 WANRJ 45 with
two LED’s
Connection to cable modem/
ADSL modem. Ethernet activity
LED (Green) is glowing.
3 LANRJ 45 with
two LED’s
LAN cable connection for co-
located PC.
4 FXS 1 RJ 11To connect analog
telephone/fax.
5 FXS 2 RJ 11To connect analog
telephone/fax.
6Life Line
FXORJ 11
For PSTN bypass/Lifeline
Feature.
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SETU ATA2LL System Manual 19
ONIC Call/OG Call-Speech and Off-
Hook
Fast BlinkingError condition during IC/OG
call/Busy Condition.
Slow BlinkingIC Call-Ring Event/OG Call-Number
being dialed
OFF FXS port On-hook
ONSIP1 Active and ATA registered
with the SIP server
Fast BlinkingSIP1 Active but ATA not registered
with the SIP server
Slow Blinking SIP1 Inactive
ONSIP2 Active and ATA registered
with the SIP server
Fast BlinkingSIP2 Active but ATA not registered
with the SIP server
Slow Blinking SIP2 Inactive
FXS2
SIP1
SIP2
Where,
OFF : Continuously OFF
ON : Continuously ON
Slow Blink ing : 500ms ON, 500ms OFF
Fast Blinking : 200ms ON, 200ms OFF
IC, OG : Incoming, Outgoing
Instruction for connecting the phone adapter
This section describes the instructions on, how to connect the
Matrix ATA to internet network and telephones.
Unpack the box. Get satisfied with the contents and the condition of the parts. Refer to ‘Packing List’. If parts are OK, proceed with
connections as shown below:
• Mount your ATA in a safe and convenient location where cables
for your network and phone system are accessible. For this,
mechanical drawing can be used as a reference, which is
provided elsewhere in the manual.
• Connect a standard analog telephone instrument to FXS1 and
FXS2 ports, using telephone cables provided.
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SETU ATA2LL System Manual20
• Insert a standard telephone cable into the Life-Line port of ATA
and connect the other end of the telephone cable to a wall jack
for the PSTN line.
• Insert one end of the Ethernet cable into the WAN port of ATA and
connect the other end of the Ethernet cable to an uplink port (arouter or a DSL modem), which is used to access internet.
• Insert one end of the Ethernet cable into the LAN port of ATA and
connect the other end of the Ethernet cable to the PC.
• Check the voltage from the power point from where the supply is
to be accessed. It should be between 90-265VAC, 47-63Hz.
Insert the DC output terminal of power adapter into the ‘Power’
jack of the ATA and connect the 230VAC pins of the power
adapter to a wall outlet for 230VAC.
• When you power ON the ATA, observe that all LED’s are ON for few seconds and then SIP1, SIP2 LED’s are flashing.
Basic Steps for Configuration
LAN Port:
• Consult your LAN Administrator to allocate the IP Address for ATA
or read the IP of PC, by clicking on Network Neighborhood →
Properties → TCP/IP Connection → Properties for Windows98.
• Choose the unique IP address of the ATA such that its subnet issame as subnet of IP address of PC.
PLEASE CHECK !
The subnet of your ATA IP Address must be
same as of PC to be connected at LAN port
• For example, if IP address of PC is 192.168.5.10, then LAN port
should be programmed as 192.168.5.X. where ‘X’ can be any
number from 1 to 254, except 10. By doing this, subnet remains
same.
• Pick up the handset of the analog phone connected to ATA.
• Enter the programming access code #19 followed by password
1234. You get programming tone.
• Enter the LAN IP Address, using command 12-<LAN IP
Address>-#*. For e.g. to enter IP Address 192.168.1.120. Enter
command 12-<192168001120>-#*.
• You get confirmation tone followed by system restarting.
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SETU ATA2LL System Manual 21
• Start Internet Explorer (IE6 with SP2, Service Pack) connected to
the LAN port of the ATA. Type in the IP Address programmed
above in the Address field. On the login page, enter default
password ‘1234’.
WAN Port:
• Click on WAN Port from menu on left side of the page.
ABSOLUTELY NEEDED !
Use WAN IP Information from your LAN Admin. or ISP
• IP Address (must)
• IP Mask (must)
• Gateway
• DNS IP Address (must)
• DNS Domain Nmae
• Note that subnet of WAN and Internet (Router) are same and is
different from subnet of LAN IP Address.
• For example, if LAN IP Address is 192.168.A.120, then WAN IP
Address can be 192.168.B.C, where B can be any number from 1
to 254, except A and C can be any number from 1 to 254.
• Now, enter above information in the respective field. If informationof some parameter is not provided, keep the field value as default
or unchanged.
• Click on ‘Submit’. The system starts rebooting.
• Click on SIP Account from the menu provided on left side of the
web page.
Get Information from your ITSP for SIP Account
• SIP ID
• Registrar Server Address• Registrar Server Port
• Authentication User ID
• Authentication Password
• Outbound Proxy Server Address
• Outbound Proxy Server Port
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SETU ATA2LL System Manual22
• Enter above information in SIP Account1 field. If these information
is not available, keep the values as default. For e.g. Registrar
Server Port = 5060.
• Enable SIP Account for SIP1 and also enable Outbound Proxy, if
information is programmed.• If you have a second internet phone Service Account (SIP2), then
use above fields for SIP2 port and enable. It allows you to use
second phone (or Fax) connected at FXS2 port.
• Submit page. If registration is successful, SIP LED will glow
steady.
Note:
The computer need not be dedicated tothe ATA, after installation is complete.
• You can have a look at all above parameters and software version
information by clicking on ‘Status’ from the menu.
• ATA is ready for further IP calling service!
• Refer ‘Troubleshooting’ in the Appendix section or default the
system with the push button switch and start again, if you cannot
establish connectivity from your ATA.
Making Phone Calls
• After SIP1 and/or SIP2 LED is ON, just dial the SIP ID numerical
number e.g. 4567 which is registered to the same ITSP provider..
• If SIP account is [email protected] then the SIP ID number is
4567 and SIP service domain is ‘Provider.com’.
• PSTN calls can be made even if ATA is not registered to ITSP
provider (you get ‘error tone’ in this case).
• PSTN number can be called directly, to the area which your ITSP
Service Provider supports.• Dial ‘##’ to get PSTN dial tone. If the ATA is powered OFF, you
can access the PSTN directly. Dial PSTN number as usual e.g.
02652630555.
• If Emergency Number is programmed, dialing it gives you the
PSTN dial tone. Then dial the same number again.
• You can also use feature of ‘Speed Dialing’ and ‘Peer-to-Peer
Calling’ as explained below:
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SETU ATA2LL System Manual 23
Speed Dialing: The numbers which you need to dial frequently can
be programmed in the ‘Phone Book’ table. Suppose the number you
want to dial is entered at index 05 of the phone book, then go OFF-
Hook and dial #8-05. The number will be dialed!
Peer-to-Peer Calls: Peer-to-Peer number can be dialed even if ATA
is not registered to any service provider (you get ‘error tone’ in this
case). If Peer-to-Peer is enabled and FXS port number is
programmed, you can call another ATA by dialing its FXS port
number. You can also call another ATA or IP-Phone by dialing its IP
Address if Peer-to-Peer is enabled. For e.g. Just dial 192*168*1*21,
to call the IP number 192.168.1.21.
Mechanical dimension of SETU ATA:
1 9 7 . 0
0 m m ( 7 . 7
6 I n c h )
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Section 2: Features and Facilities
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SETU ATA2LL System Manual 27
Dial Plan
• Generally Matrix ATA user can assign two SIP Account numbers
to two FXS Ports, one for each. Hence if Call from FXS1 needs to
use SIP2 Account, then it can be done by assigning SIP2 to
FXS1 in the FXS Port parameters screen.
• But the ATA has also facility to allocate SIP account, as per the
number dialed. The database for this mapping is called Dial Plan.
• Dial Plan has in all 10 entries.
• Each entry has a number and corresponding SIP account which
the system will use to make an OG call.
• If the number string dialed by the user matches with any of the
number string in the Dial Plan, the call is made using thecorresponding SIP Account.
• This flexibility is helpful in following case:
Example:
If Numbers are mapped with SIP Account as given below:
Index Number SIP Account
01 1234 SIP1
02 12 SIP2: : :
10
• Then, a person in the Cyber cafe can call to Africa on ’1267’ even
if another person is talking to US on SIP1 Account on ‘1234’
number. The ATA allocates SIP2 account to the person who
wants to call Africa, based on closest best fit criteria. Thus Dial
Plan works as powerful routing tool.• Similarly, if the user dials a number say 1256783, the call should
be made using SIP2 as this number best fits with the number
string in index 02.
• If entries are not matching or the table is blank, the OG call will
use default SIP1 Account.
• Click on ‘Dial Plan’.
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SETU ATA2LL System Manual28
Dial Plan:
Value/Default Description
Index 1 to 10This is the index number at which
entry is done.
Number
16 digit Maximum is
from 0 to 9.
Default = Blank
Enter the number string which the
user needs to dial frequently.
SIP
Account
SIP 1 and SIP 2
Default = SIP1
Enter the SIP Account Number
provided by ITSP.
Relevant Topics:
1. FXS Port 29
2. SIP Accounts 66
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SETU ATA2LL System Manual 29
FXS Port
Introduction:
For efficient interfacing with Analog telephone connected at FXS1
and FXS2 and for efficient use of many features like Echocancellation, VAD, etc., some parameters are programmed on FXS
port. This Chapter gives detailed procedure of programming.
• The ATA supports only DTMF dialing (Tone Dialing).
• ATA supports four types of Caller Number Presentation viz. None,
DTMF (not in this version), BellCore FSK and V.23 FSK.
• ATA supports Calling Party Control to disconnect the Trunk
which is still busy when remote called party has gone ON-Hook.
It is implemented by : CPC period of 500 Millisecond.• ATA supports programming at Input Gain (-6dB to +6dB), Output
Gain (-6dB to +6dB), Programmable Flash Timer, Forward Error
Correction, Programmable Ring Timer, Programmable Inter-Digit
Timer and fixed First Digit Wait Timer.
• Some parameters are described as below before explaining
programming:
FXS Port Name:
• This name is sent to the callee in the Display Name field in theSIP call.
• The callee shall be able to identify the FXS port from which the
call is made along with the SIP number provided by the ITSP for
the SIP Account.
• To elaborate: Suppose the ITSP has provided a SIP number
[email protected]. When the call is made by FXS1 or FXS2, the
callee gets the same number. Hence he is unable to identify that
who is the actual caller; FXS1 or FXS2. The Station name feature
resolves this problem by assigning ‘xyz’ name only to FXS1 andcalling from port FXS1.
• FXS Port Number: It is used for Peer-to-Peer calls only.
Call Progress tone:
• Different types of tones are played during specific event of
establishing the call, speech duration and termination of the call.
As per ITU standard these tones are of different specification for
different country. For example, Dial Tone for India consists of
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NameTimer
(Sec.)Matures to
Dial Tone 7 Error Tone → Standby
RBT 60 Error Tone → StandbyBusy Tone 30 Error Tone → Standby
Error Tone 7 Error Tone → Standby
Confirmation Tone 3 Dial Tone/Feature Tone
Programming Tone 7 Error Tone → Standby
CCWT 60 Error Tone → Standby
Timer column shows that for how long the tone will continue andlast column ‘Matured to’ shows type of tone after timer is expired. It
goes in stand by after error tone completes. For example, ATA plays
error tone for 7 secs. and then goes in stand by.
Refer Table 2 for tone specification for different countries, at
the end of this chapter. Just select country name from the list of
countries in the Combo Box for FXS1 and FXS2 to program the
Call Progress Tone.
Ring Type:
• The Matrix ATA supports different cadence for the ring when
incoming call lands on FXS1 or FXS2.
• These cadence is standardized for different country. For example,
India supports: 25Hz, 0.4 ON 0.2 OFF, 0.4 ON 2.0 OFF (Sec.).
USA/Canada supports: 25Hz, 2.0 ON 4.0 OFF (Sec.).
• In order to support a specific ring type, user has to just select the
country. The system will select the cadence as per table 3, at the
end of chapter. Default country is India.
First Digit Wait Timer:
• First digit wait timer signifies the time for which the system waits
for the user to dial something once the FXS port goes OFF-Hook.
On expiry of this timer, the system will give error tone to the user.
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Internal Digit-Digit Wait Timer: (Timer of Relevance)
• This is the time during which if user does not dial a digit after
pressing a digit, the system stops collecting further digits and
processes digits dialed so far.
Echo Cancellation:
• Line echo is caused when conversion between 4-wire to 2-wire
line is carried out, by telephony hybrid. And delay exceeds 50ms.
Acoustic echo is caused when our own voice reverberating the
phone-receiver when we talk.
• Echo Cancellation is a technique that allows for isolation and
filtering of unwanted signals caused by echoes.
• The echo canceller-device puts a signal on the return
transmission path, which is equal and opposite of the echo
signal.
• Echo cancellers (digital filters) are used depending on the length
of the echo to be compensated, also depends on the distance
between gateway and line hybrid. (Between 8-32ms for SOHO
products)
VAD:
• Voice Activation Detection (VAD), is a software application thatallows a data network carrying voice traffic over the Internet to
detect the absence of audio and conserve bandwidth by
preventing the transmission of “silent packets” over the network.
• Most conversations include about 50% silence.
• VAD (also called “silence suppression”) can be enabled to
monitor signals for voice activity so that when silence is detected
for a specified amount of time, the application informs the Packet
Voice Protocol and prevents the encoder output from being
transported across the network.
• Click on “FXS”
• FXS Ports
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Parameter Values/DefaultDescript ion (Enter for FXS1
and FXS2)
FXS Port
Name
16 char. Max.
Default = Blank
Enter if name is to be sent
with the dialed number.
FXS Port
Number
4 digits max.
(0,9,*)
Default = Blank
Enter here, the extension
number on which, incoming
Peer-to-Peer call, when dialed
on this number will land. Both
phones at FXS1 and FXS2 will
ring is this number if not
matching.
Listening
Volume
-6 to +6 dB, in 1-
dB Steps
Default = 0-dB.
Select to modify audio level
being output to the device
attached to the FXS.
Speaking
Volume
-6 to +6 dB, in 1-
dB Steps
Default = 0-dB.
Select to modify audio level
entering voice before it is sent
over the network to the remote
VoIP.
Call Progress
Tones
List of countries
refer table 2.
Default=India
Select the country where the
ATA is installed. Refer table 2
for the frequency and cadence
of various tones supported.
Ring Type
Lisr of countries
refer table 3.
Default = India
Select the country where ATA
is installed. Refer table-3 for
the cadence of various ringssupported
Ring Timer 01-99 seconds
Default = 60
Select the time period to play
the ring.
Voice Volume Control
Tones and Rings
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Enable Default = YesClick here, to enable the
feature.
TypeFSK-BellCoreDefault = FSK-
ITU (V.23)
Select the CLI Type as per thePhones connected at FXS1
and FXS2.
Send Caller
IDDefault = Yes
Click if CID is to be sent
during an OG call. If NO, CID
will be Blocked.
Enable Default = NoWhen called party goes OFF-Hook caller gets 'Polarity
Reversal'.
Enable CPC Default = No
Click if CPC feature required.
When called party goes ON-
Hook, CPC is detected for
500msec.
Flash Timer
100 to 900
Millisecond in
steps of 100 ms.
Default = 600 ms.
Select as per the Flash period
standards of the terminal or
phone.
Enable Default = Yes Click if the feature required
Tail Length
8 msec
16 msec
32 msec
Default = 32
msec.
Select such that delay is
minimum so that Voice quality
is improved.
Calling Party number Presentation
Polarity Reversal
Call Party Control
Caller ID
Flash Timer
Echo Cancellation
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Condition Battery Polarity
Idle Negative
Off-hook Toggle to Positive
Remote Party Answers Toggle to Negative
On-hook Retain to Negative
Table 2:
Freq. Cadence Freq. Cadence Freq. Cadence Freq. Cadence
Hz second Hz second Hz second Hz second
Australia 425*25 cont. 400*250.4on 0.2off
0.4on 2.0off 425
0.375on
0.375off 425 0.375on 0.375off
Argenti na 425 cont. 425 1.0on 4.0 off 425 0.3on 0.2off 425 0.3on 0.4off
Brazil 425 cont. 425 1.0on 4.0 off 425 0.25on 0.25off 425 0.25on 0.25 off China 450 cont 450 1.0on 4.0off 450 0.35 on 0.36off 450 0.7on 0.7off
Egypt 425*50 cont 425*50 2.0on 1.0off 425*50 1.0on 4.0off 450 0.5on 0.5off
France 440 cont 440 1.5on 3.5off 440 0.5on 0.5off 440 0.25on 0.25off
Germany 425 cont 425 1.0on 4.0off 425 0.48on 0.48off 425 0.24on 0.24off
Greece 4250.2on 0.3off
0.7on 0.8off 425 1.0on 4.0off 425 0.3on 0.3off 425 0.15on 0.15off
India 400*25 cont 400*250.4on 0.2off
0.4on 2.0off 400 0.75on 0.75off 400 0.25on 0.25off
Indonesia 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.25on 0.25off
Iran 425 cont 425 1.0on 4.0off 425 0.5on 0.5off 425 0.25on 0.25off
Israel 400 cont 400 1.0on 3.0off 400 0.5on 0.5off 400 0.25on 0.25off
Italy 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.2on 0.2off
Japan 400 cont 400*380 1.0on 2.0off 400 0.5on 0.5off 400 0.25on 0.25off
Kenya 425 cont 4250.67on 3.0off
1.5on 5.0off 425
0.2on 0.6off
0.2on 0.6off 425 0.2on 0.6off
Korea 350+440 cont 440+480 1.0on 2.0off 480+620 0.5on 0.5off 480+620 0.3on 0.2off
Malaysia 425 cont 4250.4on 0.2off
0.4on 2.0off 425 0.5on 0.5off 425 0.5on 0.25off
Mexico 425 cont. 425 1.0on 4.0off 425 0.25on 0.25off 425 0.25on 0.25off
New Zealand 400 cont. 400+4500.4on 0.2off
0.4on 2.0off 400 0.5on 0.5off 400 0.25on 0.25off
Phillippines 425 cont. 425+480 1.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off
Poland 425 cont. 425 1.0on 4.0off 425 0.5on 0.5off 425 0.5on 0.5off
Portugal 425 cont. 425 1.0on 5.0off 425 0.5on 0.5off 450 0.33on 1.0off
Russia 425 cont. 425 0.8on 3.2off 425 0.4on 0.4off 425 0.25on 0.25off
Saudi Arabia 425 cont. 425 1.2on 4.6off 425 0.5on 0.5off 425 0.25on 0.25off
Singapore 425 cont. 425*240.4on 0.2off
0.4on 2.0off 425 0.75on 0.75off 425 0.25on 0.25off
South Africa 400*33 cont. 400*330.4on 0.2off
0.4on 2.0off 400 0.5on 0.5off 400 0.25on 0.25off
Spain 425 cont. 425 1.5on 3.0off 425 0.2on 0.2off 425 0.25on 0.25off
Thailand 400*50 cont. 400 1.0on 4.0off 400 0.5on 0.5off 400 0.3on 0.3off
Turkey 450 cont. 450 2.0on 4.0off 450 0.5on 0.5off 4500.2on 0.2off
0.6on 0.2off
UAE 350+440 cont. 400+4500.4on 0.2off
0.4on 2.0off 400
0.375on
0.375off 400
0.4on 0.35off
0.225on 0.525off
UK 350+440 cont. 400+4500.4on 0.2off
0.4on 2.0off 400
0.375on
0.375off 400
0.4on 0.35off
0.225on 0.525off
USA/Canada 350+440 cont. 440+480 2.0on 4.0off 480+620 0.5on 0.5off 480+620 0.25on 0.25off
Country
Dial Tone Ring Back Tone Busy Tone Error Tone
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Remarks:
Frequency notations shall be as:
• f1*f2: f1 is modulated by f2.
For All CountriesConfirmation Tone : 400Hz, 0.1 on 0.1 off,
Programming Tone : 400Hz, 0.1 on 0.9 off,
CCWT : 400Hz, 0.3 on 6.0 off
Table 3:
ON 1 OFF 1 ON 2 OFF 2
01 Australia 25 0.4 0.2 0.4 2.002 Brazil 25 1.0 4.0
03 China 25 1.0 3.0
04 Egypt 25 2.0 4.0
05 France 25 1.5 3.5
06 Germany 25 3.5 5.5 0.79 1.1
07 Greece 25 1.0 4.0
08 India 25 0.4 0.2 0.4 2.0
09 Israel 25 2.0 3.010 Italy 25 1.0 4.0
11 Japan 25 1.0 2.0
12 Korea 25 1.0 3.0
13 Malaysia 25 2.0 3.0
14 New Zealand 25 2.0 3.0
15 Poland 25 2.0 3.0
16 Portugal 25 1.0 5.0
17 Russia 25 1.0 3.0
18 Singapore 25 0.4 0.2 0.4 2.0
19 South Africa 25 0.4 0.2 0.4 2.0
20 Spain 25 1.5 3.0
21 Thailand 25 2.0 3.0
22 UAE 25 2.0 3.0
23 UK 25 0.4 0.2 0.4 2.0
24 USA/Canada 25 2.0 4.0
Code CountryFrequency
(Hz)
CADENCE (In Seconds)
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Relevant Topics:
1. Peer-to-Peer Calling 61
2. SIP Accounts 66
3. Supplementary Services 75
4. Dial Plan 27
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LAN/WAN Explanation
What is this?
• This chapter explains programming aspects of LAN and WANports, in detail. If you have already started programming using
chapter ‘Getting Started’, do not use this information. ATA has
facility to program both, LAN and WAN ports by normal phone as
well as the computer.
• For this, “DHCP server used in the LAN’ is also discussed. For
more details on IP and DHCP refer Glossary and Jeeves topics.
How to program?
• ATA has a WAN port whose IP address is 192.168.10.10 (Bydefault) and a LAN port with IP address 192.168.1.10 (By
default).
• The WAN port of ATA can be connected to one of the ports of the
LAN Switch/Hub/Router (in Corporate Offices) Or it can be
connected to the Ethernet port of the Broadband modem (in
SOHO).
• When connected to LAN of the organization, the IP address of
the WAN port should be changed according to the LAN
addressing scheme. The LAN might be using static addresses or
dynamic addresses. If static addresses are used, the SE should
take the static address of the WAN port from the LAN
administrator and program it as the IP address for the WAN port
of ATA.
• If DHCP server is used in the LAN, then DHCP server should
be enabled in the product using the embedded Web server.
Doing so, the DHCP server in the LAN switch shall assign an IP
address dynamically to WAN port of the ATA whenever ATA isrestarted. If required, the IP address assigned dynamically to the
WAN port of ATA by the DHCP server can be known by logging in
the ATA from the LAN port.
• The IP address of the WAN/LAN port can be configured using a
computer as well as conventional instrument.
• Also, WAN IP address is requested from server, if PPPoE server
is enabled and user name and password are entered.
• Programming methods are explained with 4 options as mentioned
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below:
• Configuring the IP Address of WAN port of ATA, using computer.
• Configuring the IP Address of WAN port using conventional
phone instrument.
• Configuring the IP Address of LAN port of ATA using computer.• Configuring the IP Address of LAN port using conventional
phone instrument.
Configuring the IP address of WAN port of ATA using
Computer
• Make the connections as shown in the figure given below:
WAN PortFXS1
FXS2
LAN PortLifelinePort
SETU ATA2LL
Computer
• Get the IP address for the WAN port of ATA from the LAN
administrator.• Open Web browser (Internet Explorer or Netscape) on the
computer connected to the LAN port of the ATA.
• Enter the default LAN port address (192.168.1.10) in the URL
field and press Enter. (Note: PC and LAN port should be on same
subnet: 192.168.1.XXX)
• Embedded Web server takes the SE to the Home page.
• The SE has to Log in into the ATA using password (default
password = 1234).
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• The SE will have to go to ‘WAN port settings’ link and change the
IP address of the WAN port from default to that given by the LAN
administrator. The SE may change the Subnet Mask, Gateway
address and DNS address if required.
Note: WAN port and LAN port must be on different subnets.
Configuring the IP address of WAN port of ATA using
Conventional Phone Instrument
• Make the connections as shown in the figure given below:
WAN PortFXS1
FXS2
LAN PortLifelinePort
SETU ATA2LL
• Get the IP address for the WAN port of ATA from the LAN
administrator.
• Pick up the Handset of the Conventional Phone.
• You get error tone, since you are not registered.
• If you are registered, you shall get dial tone. (This is less likely
when configuring the ATA for the first time)
• Enter the Programming Access Code ‘#19’ followed by Password
1234.
• You get Programming tone.
• Enter the WAN IP address using following command:
11-WAN IP Address-#*
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WAN PortFXS1
FXS2
LAN PortLifelinePort
SETU ATA2LL
Computer
• Get the IP address for the LAN port of ATA from the LAN
administrator.
• Open Web browser (Internet Explorer or Netscape) on the
computer connected to the WAN port of the ATA.
• Enter the default WAN port address (192.168.10.10) (if not
changed) in the URL field and press Enter (Note: PC and LAN
port should be on same subnet).
• Embedded Web server takes the SE to the Home page.
• The SE should Log in into the ATA using password (default
password = 1234).
• The SE should go to ‘LAN port settings’ link and change the IP
address of the LAN port from default to that given by the LAN
administrator. The SE may change the Subnet Mask if required.
• The SE should note the changed IP address of the LAN port of the ATA so that he can access the Embedded Web Server using
this IP address. (This shall be useful if dynamic addresses are
assigned to the WAN port of the ATA by the DHCP server)
Note: WAN port and LAN port must be on different subnets.
Configuring the IP address of LAN port of ATA using
Conventional Phone Instrument
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• Make the connections as shown in the figure given below:
WAN Port FXS1
FXS2
LAN Port
Lifeline
Port
SETU ATA2LL
• Get the IP address for the LAN port of ATA from the LAN
administrator.
• Pick up the Handset of the Conventional Phone.
• You get dial tone.
• Enter the Programming Access Code ‘#19’ followed by Password
1234.
• You get Programming tone.
• Enter the LAN IP address using following command:
12-LAN IP Address-#*
Where,
LAN IP Address is of 12 digits. Each octet is of three digits. If
only single digit is used then it should be converted to three digits
and then entered.
For e.g. To enter IP address 192.168.9.10, enter command
12-192168009010-#*.
• You get confirmation tone for 3 seconds followed by programming
tone.
• Enter ‘0’ to get out of programming mode.
• The SE can dial any programming command during confirmation
tone.
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Important Point:
• When SIP is not registered or Ethernet cable is not connected,
error tone is coming on both FXS ports, but programming by
phone is still possible.
Relevant Topics:
1. Jeeves 46
2. Glossary 106
3. Supplementary Services 75
4. LAN Port 56
5. WAN Port 95
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Sub Net
• It is a mechanism that is used to split a network into a number of
smaller sub networks. It can be used to reduce traffic on each
sub network by confining traffic to only the sub networks for
which it is intended, thereby eliminating issues of associatedcongestion on other sub networks and reducing congestion in the
network as a whole.
• Makes entire network more manageable.
• Each sub network functions as though it was an independent
network, keeping local traffic local, and forwarding traffic to
another sub network only if the address of the data is external to
the sub network. Such decisions are made on the basis of
routing-tables contained within the various routers, with each
table comprising an IP address table.
• Subnet is a portion of the network, which may be a physically
independent network, which shares a network address with
other portions of the network and is distinguished by a subnet
number.
Default Gateway
• A default gateway is a node on a computer network that serves
as an access point to another network.• In enterprises, the gateway is the computer that routes the traffic
from a workstation to the outside network that is serving the Web
pages.
• In homes, the gateway is the ITSP that connects the user to the
internet.
• A default gateway is used by a host when an IP packet’s
destination address belongs to someplace outside the local
subnet.
• The default gateway address is usually an interface belonging tothe LAN’s border router.
Domain Name System (DNS)
• DNS is a system that stores information about host names and
domain names in a type of distributed database on networks,
such as the Internet.
• Of the many types of information that can be stored, most
importantly it provides a physical location (IP address) for each
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domain name, and lists the mail exchange servers accepting
e-mail for each domain.
• The DNS provides a vital service on the Internet as it allows the
transmission of technical information in a user friendly way.
• While computers and network hardware work with IP addressesto perform tasks such as addressing and routing, humans
generally find it easier to work with hostnames and domain
names (such as www.example.com) in URLs and e-mail
addresses.
• The DNS therefore mediates between the needs and preferences
of humans and of software.
Dynamic Host Configuration Protocol (DHCP)
• The DHCP is a client-server networking protocol.
• A DHCP server provides configuration parameters specific to the
DHCP client host requesting, generally, information required by
the host to participate on the Internet network.
• DHCP also provides a mechanism for allocation of IP addresses
to hosts. The DHCP protocol provides three methods of IP
address allocation:
• Manual Allocation
• Automatic Allocation• Dynamic Allocation
How to configure ATA?
To start basic VoIP configuration first enter the default LAN IP
address of the ATA, obtained from SE in the ‘Address’ bar of the
Web-browser screen as explained below:
• Launch the Web browser as explained below:
• Double Click on Internet Explorer icon
Or Click on Start Æ Programs Æ Internet Explorer
• Enter IP Address at Address Bar after confirming from SE. Let us
say IP Address 192.168.5.1 is entered at Address Bar on opening
page of Microsoft Internet Explorer .
• ATA displays Welcome page with a field for entering ‘Password’.
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2. After Login, the welcome page displays the time after which login
expires as shown below:
3. Click on ‘SIP Accounts”. Enter SIP ID Numbers 978 and 651
provided by ITSP and enable SIP accounts as shown below:
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4. Click on ‘FXS Port’ and select ‘Dial Plan for FXS1’ and select
‘SIP Account2’ for FXS2. Thus any number dialed from phone at
FXS1 will use the SIP number mentioned in the ‘Dial Plan’ screen
or SIP1 as default. But any number dialed from FXS2 will be ‘SIP
Accounts2’ only as shown below:
5. Click on ‘Dial Plan’. Enter the required number string that will be
dialed from FXS1 and use the specific SIP account as shown
below:
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Example 2:
1. Click on ‘SIP Accounts’ and select ‘Route to BOTH’ for incoming
calls field, as shown below:
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Example 3:
• To program Ring Type for country, click on FXS port and then
select the name of country from the list of the combo box of the
‘Ring Type’, as shown below:
Relevant Topics:
1. Password 602. Software Upgrade 71
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LAN Port
The Matrix ATA has two Ethernet ports:
• LAN
• WAN
This chapter describes, LAN Parameters.
• When trying to program the ATA from LAN port, the SE should
ensure that the computer which he is using to configure the ATA
is in the same subnet as ATA. However, this shall be required in
very few cases. The IP of the computer can be changed by going
to the properties of TCP/IP connection. This can be reached
using the path:Network Neighborhood J Properties J TCP/IP connection J
Properties (for Windows-98).
• Normally the MAC address will not be changed. It needs to be
changed only in one case i.e. when more than one ATA are to be
connected to the same LAN.
• Refer Glossary for more details on MAC and Web Jeeves for IP
Address and Subnet Mask.
• Click on ‘LAN’ to program these parameters:
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Life Line
Introduction:
• This is one of the Ports of the Matrix ATA for interfacing with
PSTN, called the ‘Life Line Port’. It is not a true FXO port.
• The basic purpose of this port is to provide connectivity to the
PTSN in following cases:
• When a number from a list of numbers is dialed. This list can
have maximum of 5 numbers. Generally, Emergency numbers
can be stored by the user in this list. By default, this list is blank.
• When the user wishes to dial a number using PSTN. The user
prefers this option when he is unable to achieve toll quality
speech through IP due to some reasons.• The user has to dial access code ‘##’ to access the Life Line.
The access code is fixed and not programmable.
• It is required to dial emergency number twice to make a call from
PSTN port.
How it works:
• On receipt of the number from the user dialed from FXS port, the
system compares it with the numbers in the number list and if a
perfect match is found it connects the user to the PSTN. Theuser will listen to the PSTN dial tone if number match is found in
emergency number list and dial plan.
• The user will dial the number again to actually dial it on the PSTN.
• Click on ‘Lifeline’
Emergency Numbers Table
Value/Default Description
Index 1 to 5This is the index number atwhich an emergency number
is stored.
Number
4 digit Maximum is
from 0 to 9.
Default = Blank
Enter emergency number
likeHospital, Fire, etc.
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SETU ATA2LL System Manual60
Password
For system security, it is advisable to change the default ‘System
Password’ after configuring the parameters. Old Password, New
Password and Retyping of the new-password are required to be
entered. The ‘Password’ will not be visible. Only ‘*’ will be seen for
every character of the password. The fields are as shown in the
Web Page:
• Click on ‘Password’
Value/Default Description
Password 4 digitsMax/1234
Type the default password or the
existing password you use toaccess the system, 1234.
New
Password4 digits Max Type the new password
Retype to
confirm4 digits Max Type the new password again
By default, Password is Blank.
Only numerals (0 to 9) are allowed up to four digits maximum. Thisis because same password should be used to enter SE mode from
SLT.
Relevant Topic:
1. Jeeves 46
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SETU ATA2LL System Manual 61
Peer-to-Peer Calling
What’s this?
• ATA can call another user agent (ATA or Softphone), which is in
the same LAN (figure 2) or which can be in another network,which is connected by virtual LAN provided by ITSP (Figure 1).
For example, ITSP provides VLAN services to connect two offices
situated in different cities.
• ATA can call to an extension number of some other ATA or soft
phone without going through any proxy. This is known as Peer-
to-Peer Calling. For this, the extension number (FXS Port
number), which is to be called, is programmed in the number field
and IP address of the called ATA is programmed in ‘IP Address’field of the Peer-to-Peer Call Table.
• Each entry has two parameter viz. ‘Number’ and ‘IP Address’.
• For example, if IP address of ATA which is to be called, is
192.168.1.12 and extension number 3005, then at Index 01, 3005
is programmed in ‘Number’ field and 192.168.1.12 is entered in
‘IP Address’ field and screen is ‘Submitted’. For Peer-to-Peer OG
Call, user has to follow following steps:
• Enable Peer-to-Peer in FXS Port screen.
• Dial 3005.• For incoming Peer-to-Peer call, the FXS port number on which
call is to be received, is programmed in ‘FXS Port’ number field
on the screen ‘FXS Port’. For example, if 4001 is programmed in
‘FXS Port’ screen in FXS1 column then any IC call on 4001 will
land on FXS1 port. But, if incoming call doesn’t have perfect
match and only IP Address of Matrix ATA is called, then call land
on both FXS1 and FXS2.
• ATA also supports to call the ATA by an IP number. After lifting
the handset, user dials digits with ‘*’ in place of ‘.’. This is called‘Direct IP Calling’ feature.
• This feature is very much useful in case, Party’s IP Address only
is known and not ‘SIP-ID’.
• For example, a PC phone, with its IP Address 192.167.1.21, as
shown in figure 1 can be called by just dialing 192*167*1*21, after
lifting the handset.
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SETU ATA2LL System Manual62
Figure 1:
SETU ATA2LL
SETU
ATA2LL
SETU ATA2LL
SETU ATA2LL
Internet
192.168.1.10
192.168.1.11
192.168.1.12
192.168.1.13
2001
2002
2004
2005
192.168.1.15
3001
3002
3005
3006
192.167.1.20
192.167.1.21
192.167.1.23
192.167.1.22
Router Router
VLAN
Services
Baroda Office Mumbai Office
192.167.1.24
PC Phone
PC PhonePC Phone
PC Phone
Figure 2:
SETU
ATA2LL
SETU
ATA2LL
SETU
ATA2LL
SETU
ATA2LL
192.168.1.10
192.168.1.11
192.168.1.12
192.168.1.13
2001
2002
2004
2005
192.168.1.15
3001
3002
3005
3006
192.167.1.20
192.167.1.21
192.167.1.23
192.167.1.22
Router Router
First Floor Second Floor
192.167.1.24
PC Phone
PC PhonePC Phone
PC Phone
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SETU ATA2LL System Manual64
Phone Book
• It is the Database for storing Full URI SIP number, name and SIP
account number.
• Maximum 99 numbers can be stored in a phone book.
• Each entry has three parameters viz. the SIP number, name and
SIP Account number.
• The number is dialed out using the SIP Account assigned to it or
as per the Dial Plan.
• The Access Code for dialing Phone book numbers is #8-Index
Number . The user shall go off-hook and dial #8 followed by the
Phone book index number (from 01 to 99). e.g. #8-64, to dial the
number, stored at Index ‘64’.
• Click on ‘Phone Book’.
Value/Default Description
Index 01 to 99This is the index number at
which an entry is done.
Number
SIP number
maximum 16
characters.Default = Blank
Enter the numberof person to be
called. e.g. 12345678.
Name
Maximum 16
characters Default
= Blank
Enter name of person, whose
SIP number is stored in the
phone book.
SIP
AccountSIP 1 and SIP 2
Enter SIP Account number on
which the OG call will be made,
when dialing the specified
number.
Relevant Topic:
1. SIP Accounts 66
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Resetting the ATA
What is this?
• ATA has the provision for changing the password. Hence if you
have forgotten the Password or you may need to default/reset the
system, ‘System Default’ feature is required.
• The system will loose all previous configuration.
How it works?
• Keep pressed the ‘reset’ button, till you see the LED sequence
for reset cycle. (Approx. time for pressing is 3 secs.)
Relevant Topic:1. Getting Started 16
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STUN:
• Enable: Click, if there is a NAT router between the ATA and the
ITSP’s SIP server.
By default, Enable STUN is Blank.
• Server Address: Enter the STUN server’s address. Allow IP
address also. Allow extended ASCII characters.
Valid range is N.N.N.N, Where N=0 to 255 or maximum 40
characters.
By default, STUN Server Address is Blank.
• Server Por t: enter the STUN server’s listening port for SIP. SE
may change this field if the ITSP provides a STUN Server portnumber other than the default.
Valid range is from 1025 to 65535.
By default, STUN Server Port is 3478.
NAT Keep Alive:
• Enable: Click Yes to enable. The ATA’s SIP messages will use
NAT router’s public IP address and SIP port number. The NAT
should be also be configured with this port number to forward the
traffic with this port number.NAT eliminates the need for STUN.
By default, disable.
• Interval: Assign the time after which the ATA should send SIP
notify messages to the SIP server.
By default, NAT keep Alive interval is 120 seconds.
Incoming Call-Routing:
• Incoming Cal ls : Select the port on which you want theincoming calls on SIP Account 1 to be routed and to land.
Similarly specify the port for SIP Account 2 calls.
Valid range is FXS1, FXS2 or Both.
By default, FXS1 for SIP1 and FXS2 for SIP2.
QOS (Layer 3):
• SIP QOS: This field specifies the QOS type viz. TOS
(Precedence, also called Priority), or DiffServe for voice traffic.
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SETU ATA2LL System Manual72
Status
What’s this?
• ATA user may require to view remotely for LAN, WAN Addresses
and registration status. Matrix ATA has this facility. The
registration time is provided by the server. User can quickly have
a glance for LAN and WAN address and can change it from
respective page, if required.
• It also provides ‘Registration Status’ which is useful to know
whether ‘SIP’ is registered or not. For example, if the Ethernet
cable for Internet gets disconnected than after some time,
‘Registration Status’ is displayed on the ‘Status’ screen (if
updated) and user can take corrective action, just by connectingthe cable instead of restarting the system.
• Click on ‘STATUS’ page. Following basic parameters are
displayed:
STATUS
Parameters Display Description
System NameMatrix SETU
ATA2LL
Display the name of
systemVersion/
RevisionVxRy
Display the current version
'x' and Reviosn 'y' of
software
IP Address 192.168.50.96Display the programmed
IP Address
Subnet Mask 255.255.255.0display the programmed
subnet mask
MAC Address 00.50.C2.55.B0.18Display the programmed
MAC Address
IP Address 192.168.1.98
Subnet Mask 255.255.255.0
MAC Address 00.50.C2.55.B0.12
WAN Port:
LAN Port:
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SETU ATA2LL System Manual74
Public IPN.N.N.N.
N = 001 to 255
When STUN is enabled
'Pulblic IP' is displayed
here.
Relevant Topics:
1. LAN Port 56
2. WAN Port 95
3. SIP Accounts 66
4. System Name 86
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Supplementary Services
This chapter contains following sections:
• Introduction• Programming Type
• Programming of Web Pages
• Programming using Phones
• Important Points
Introduction:
ATA provides following call features based on supplementary
services provided by the ITSP:
• Making Second Call• Call Hold
• Call Waiting
• Call Toggle
• Call Transfer-Blind
• Call Transfer-Attended
• 3-Party Conference
• Call Forward-Unconditional
• Call Forward-On Busy
• Call Forward-On No Reply
• DND
• Caller ID
These supplementary services should be supported by the service
provider.
Making Second Call
• Telephone user can make second call while talking to first caller
by keeping the first caller on hold. The user should keep thefirst caller on hold and then dial the destination number to which
he wants to talk. On completing the talk, ask the called party to
hang-up. When the called party hangs up, the user gets
connected to the first caller. Also, while in speech with second
call, he can talk to first user by using call toggle access code
(except type 3 users). The user can also have conference
between three parties by dialing conference access code.
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Call Hold
• Call Hold allows one party to maintain an idle (non-talking)
connection with another party while receiving another call (Call
Waiting) or while initiating another call (Call Transfer), or while
performing some other call management functions.
Call Waiting
• Call Waiting notifies an engaged caller of an incoming call and
allows them to receive a call from a third party while the party
with whom they have been speaking is put on hold.
Call Transfer (Blind):
• Call Transfer allows one party to reconnect the party with
whom, they have been speaking; to a third party. The first party
is disconnected when the third party becomes connected. Call
transfer without consulting to the party on whom call is to be
transferred is “Blind Transfer”.
Call Transfer-Attended
• Telephone user dials Call Hold access code followed by the
Transfer target’s number to transfer the caller (Transferee) to
the Transfer target. In case of, Call transfer with consulting or Call Transfer-attended the user gets ring back tone followed by
speech or Busy Tone. He may talk to transfer target and hang
up.
Call Toggle
• To attend to the second call, phone user has to use this feature
by dialing ‘Call Toggle Access Code’.
• When the user dials the Call Toggle access code when a party
is held, user is connected to the waiting caller. If no party isheld, then user gets error tone.
• The user is connected to the current party when he goes ON-
Hook, gets ring and again goes OFF-Hook.
• Type 3 user cannot use this feature.
3-Party Conference
• Telephone user dials Conference access code to have
Conference between the User and the two callers.
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Blind Call
Transfer
Flash-*1-
Number-ON
Hook
Flash-*98-
Number-ON
Hook
Flash-#90-
Number-ON
Hook
Attended Call
Transfer
Flash-Number-
Speech-OnHook
Flash-Number-
Speech-OnHook
Flash-Number-
Speech-OnHook
Conduct
ConferenceFlash-0 Flash-3 Flash
Conference to
ToggleFlash-1 Flash-2 --
Call Toggle Flash-1 Flash-2 --
Call Toggle
Call Transfer
Conference
Programming of Web Pages:
Programming through Jeeves is as given below:
• Click on “Supplementary Services”
• Type Selection:
Description
Type1 Type2 Type3
Feature
Code
Type
Value/Default
Default = Type 1
Select Type from Type1, Type2
and Type3 as per the country in
which ATA is installed. Select
Type1 for Matrix System, Type2
for European System and Type3
for American System.
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Procedure of Calling:
Call Forward
• Call Forwarding (Always)
Incoming calls may always be forwarded to another designated
party, and call forwarding founder can hear the notified rings.
Example:
1 A calls B. NO TALK.
2
B responds: it is
unavailable. So it returns a
forwarding destination C.
User at A is
unaware of this.
3
A is automatically forwarded
to C. Talk between A, C.
• Call Forwarding (Busy):
Incoming calls can be forwarded to another designated party if
the phone of the called party is busy.
1 A calls B. NO TALK.
2 B responds, (Busy state).
B is in busy state and
the new forwarding
state is C.
3 A is automatically
forwarded to C.Talk between A, C.
• Call Forwarding (No-Answer):
Incoming calls can be forwarded to another designated party
when there is no response, from the original called party.
1 A calls B.B is ringing, A can hear
ring back tone.
2
B responds to A, after 45
secs. (programmable) with
no one picking up the
phone.
No answer and A needs
to contact C.
3 A is automatically
forwarded to C.Talk between A, C.
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Call Transfer (for Type 1):
• Call Transfer without consultation (Blind transfer):
Assuming that call party A and B are in conversation. A wants to
Blind Transfer B to C: To transfer incoming call to another phone:
1 Speech ON. A and B under talk.
2 Dial Flash on SLT-A. Feature tone
3 Dial * 1-Station-#.Ring back tone on B
and C is ringing.
4 Note1. B and C talking.
Note 1 is:
(a) A gets confirmation tone followed by Dial tone if transfer issuccessful.
(b) A gets busy tone for 2 seconds followed by speech, if
transfer target is busy.
(c) A gets continuous busy tone, if no response comes from
transfer target.
(d) A gets error tone, if transfer isn’t successful due to either
timeout or invalid codes have been entered.
By default, Matrix Flash key Command
• Call transfer with consultation (Attended Transfer)
Event 1: When A&B are in communication, A puts B on hold and
then dials Number of C plus the pound key #.
Event 2: After A and C are in communication, A goes ON-Hook
and C gets connected to B.
Call Waiting (For Type 1):
1C calls B, when A and B are
talking.
B hears waiting tone, C can hear
ring back tone.
2
B press “Flash 2” or
B press “Flash 3” or just
hang-up
B press “Flash 1”
Reject the second call.
Disconnect first call and answer
second call.
A is on hold and B can answer
to C.
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3-Party Call Conference (For Type 1):
• To make 3 party call conference.
1 Speech on.
2 Dial Flash.Feature tone (Second dial tone
B on hold).
3 Dial Station-#.Ring Back Tone on B. C is
ringing.
4 A presses Flash-0. A and B and C are talking.
5 A presses Flash-1.For conference toggle only A
and B are takling.
• Event 1: When A and B are in communication, A puts B on hold
then dials C’s number plus the pound key #.• Event 2: After A and C are in communication, press Flash-0
button again to initiate the 3 party call conference.
• Event 3: During conference if A goes ON-Hook, B gets
transferred to C.
• Event 4: Press Flash-1 to separate activated 3 party call
conference into two individual connections (One is online the
other is on hold).
Call Hold (For Type 1):
• Caller can use hook flash key to hold the call.
1 A calls B Talk is ON.
2 A presses Flash. A, hears second dial tone, B
hears nothing. B is on Hold.
3 A can press again
Flash.Talk between A, B.
Programming using phone:
• Call Forward and DND:
Use following commands:
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Feature Command Description
Set Call Forward-Unconditional
51-SIP Account-Destination Number-#*
Enter SIP Account
number 1 or 2 for ‘SIP
Account’. After issuingthe command user
gets Confirmation
tone.
Cancel Call
Forward-
Unconditional
52-SIP Account-#* -Do-
Set Call Forward-
When Busy
53-SIP Account-
Destination Number-#*
-Do-
Cancel Call
Forward-When
Busy
54-SIP Account-#* -Do-
Set Call Forward-
When No Reply
55-SIP Account-
Destination Number-#*-Do-
Cancel Call
Forward-When No
Reply
56-SIP Account-#* -Do-
Set Call Forward
Ring Timer
57-SIP Account-Timer
(Time in seconds is from
00-99)
Enter value in
seconds for XX, for
how long Ring should
continue. Default=60
secs.
Set DND 61-SIP Account-#*
Use command if DND
feature required on a
SIP Account
Cancel DND 62-SIP Account-#*
Use command if DND
feature is not required
on a SIP Account. 1
for SIP1 and 2 for
SIP2.
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SETU ATA2LL System Manual86
System Name
• The System Name is for identification purpose.
• This screen allows user to change the system name. This
feature helps the user identify the system. This is required when
multiple ATA systems are connected in the same network. Now
when accessing the ATA for configuring it, the name programmed
in this field should appear as the system name.
This shall help SE to confirm that he is programming the correct
system.
• Click on ‘System Name’.
Value/Default Description
System Name
24 char. Max. may be
ASCII. Default = Matrix
SETU ATA2LL.
Enter the name to
identify ATA
Relevant Topic:
1. FXS Port 29
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SETU ATA2LL System Manual 87
Time Settings
• Matrix ATA can be configured for correct time based on the
Simple Network Time Protocol (SNTP).
• SNTP is a protocol built on top of TCP that ensures accurate
local time-keeping with reference to radio and atomic clocks
located on the Internet.
• The ATA gets synchronized with the specified NTP server (IP
Address), which is a global server and gets the date and global
time. Based on the time zone, settings, Matrix ATA shows
correct time for that country. For example, if time shown by
ntp1.cs.wisc.edu server for India (Culcutta, Chennai, Mumbai,
New Delhi) is 10:44:50AM, then current time shown for Kabulis 09:44:50AM. Because for India GMT is ‘GMT +5.30’ but for
Kabul it is ‘GMT +4.30’ (1 hour behind India). The date is
displayed as 17May2006.
• Click on ‘Time Settings’
Field Name Value/Default Description
Current Date DD-Month-YYYY ATA displays current
system DateCurrent Time HH:MM:SS
ATA displays current
system Time
Time Server
Address
ntpl.cd.wise.edu
time.windows.com
time.nist.gov
Select the suitable
Time server
Time Zone
List of places as
per windows
software of PC.Default = Calcutta
Select the places
name, based on
country where ATA isinstalled
Relevant Topics:
1. FXS Port 29
2. Supplementary Services 75
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• User agent client (UAC)-A client application that initiates the
SIP request.
• User agent server (UAS)-A server application that contacts the
user when a SIP request is received and that returns a
response subject to user’s input.
Network Servers:
There are three types of network servers in a SIP network: proxy
server, redirect server and registrar server.
Proxy Server: A SIP proxy receives a request, makes a
determination about the next server to send it to, and forwards the
request, possibly after modifying some of the header fields. Assuch, SIP requests can traverse many servers on their way from
UAC to UAS. Responses to a request always travel along the same
set of servers the request followed, but in reverse order.
Redirect Server: The redirect server does not forward requests to
the next server. Instead it sends a redirect response back to the
client containing the address of the next server to contact.
SIP Registrar: SIP Registrar is an entity where SIP users can getthemselves registered. Registrar imparts mobility to the SIP users.
A SIP user can register himself with a registrar. If the user changes
his location, he has to register again with the registrar stating his
latest contact information. Whenever a call is to be delivered to that
user, Registrar can provide the information about the location where
the user was active recently.
RTP:
After a VoIP call is established using SIP, the RTP (Real TimeProtocol) is used to handle voice data transfer in the packets form.
Refer RFC 1889 for more details.
NAT and STUN:
NAT (Network Address Translation-RFC 1631), is the translation of
IP address of a host in a packet. For example, the source address
of an outgoing packet, used within one network is changed to a
different IP address known within another network.
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For using this feature, two terms are explained as below:
a. NAT
b. STUN
• One network is designated the inside network and the other is
the outside. Typically, a company maps its local inside networkaddresses to one or more global outside IP addresses and
unmaps the global IP addresses on incoming packets back into
local IP addresses. This helps ensure security since each
outgoing or incoming request must go through a translation
process that also offers the opportunity to qualify or authenticate
the request or match it to a previous request.
• NAT also conserves on the number of global IP addresses that a
company needs and it lets the company use a single IP addressin its communication with the world.
Types of NAT:
Four types of NAT’s can be used:
• Full Cone NAT-This type of NAT maps all requests from the
same internal IP address and port to the same external IP
address and port. Any external host can send a packet to the
internal host only if the internal host had previously sent a packet
through the NAT.• Restricted Cone NAT-This type of NAT maps all requests from
the same internal IP address and port to the same external IP
address and port. An external host (with IP address X) can send
a packet to the internal host only if the internal host had
previously sent a packet to IP address X.
• Port Restricted Cone NAT-This type of NAT maps all requests
from the same internal IP address and port to the same external
IP address and port. An external host (with IP address X and
source port P) can send a packet to the internal host only if theinternal host had previously sent a packet to IP address X and
port P.
• Symmetric NAT-In a symmetric NAT, all requests from the same
internal IP address and port to a specific destination IP address
and port are mapped to the same external IP address and port. If
the same host sends a packet with the same source address
and port, but to a different destination, a different mapping is
used. Furthermore, only the external host that receives a packet
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SIP Call flow:
A simple example of how SIP requests and responses are sent will
help understanding SIP and RTP.
When A calls B, the steps for the set up and tear down of a SIP call
is explained as below: A B
1. INVITE
2. Ringing
3. OK
4. ACK
5. Voice Data-RTP
6. BYE
7. OK
1. A sends SIP INVITE request to B. This message is an invitation
to B to participate in a SIP telephone call.
2. B sends a response that it is ringing.
3. A answers the call. B responds it by ‘OK’
4. B also answers the call and A sends acknowledgment by sending
‘ACK’ message.
5. Now, voice session starts between A and B in form of RTP (Real
Time Protocol).
6. After the talk, A hangs up and sends a BYE request.
7. Finally, B sends the reply by sending ‘OK’ response and the call
is terminated.
Important Voice Parameters:
Jitter Buffer:• The clarity of voice heard from the called party is also distorted
due to non-uniform delay introduced by the Telephony network
known as ‘Jitter’.
• In order to reduce the effect of ‘jitter’, some sort of ‘filters’ are
used in the voice-channel called ‘Jitter Buffer’. The ATA has built-
in adaptive buffer that helps to smooth out variations in delay
(jitter) for voice traffic. This will ensure good voice quality. It is not
programmable.
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Voice Compression:
• Voice compression is a technique to reduce the band width of
voice channel, using suitable codec or Coder-Decoder. The codec
converts voice signals from analog form to digital signals and vice
versa. It uses the compression alongwith, CELP.• Compression-decompression is a voice compression-
decompression algorithm that defines the rate of speech
compression, quality of decompressed speech and processing
power requirements. It is called companding.
• Codec of suitable bandwidth is selected during the speech
reception or transmission, to minimize the digital voice channel
bandwidth.
QoS:
Quality of Service, QoS is the capability of a network to provide
better service to selected network traffic over various technologies,
including Ethernet and 802.1 networks, and IP-routed networks. ATA
supports Type of Service tagging and Differentiated Services tagging
as mentioned below: (This allows ATA to tag voice frames so they
can be prioritized over the network)
1. Type of Service (ToS)
2. DiffServe
ToS:
• Network traffic can be classified by setting the ToS values at the
data source (at the ATA) so that a server can decide best method
of delivery, such as the least cost, fastest route etc.
• This parameter allows you to configure Type of Service (ToS) bits
by specifying the precedence and delay of audio and signaling
IP packets
DiffServe:
DiffServe is differentiated Services, which defines a Class of Service
model that makes packets so that they receive per hope treatment
at DiffServe-compliant network services along the route, based on
the application types and traffic flow. Packets are marked with
DiffServe Code Point (DSCP) indicating the level of service desired.
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WAN Port
Matrix ATA is receiving and transmitting data to the Internet Network
through Wide Area Network (WAN) port.
• There are three ways to obtain WAN IP-address as mentioned
below. Depending upon type of Internet connection, one of this
is used. e.g. if Internet connection is through xDSL (Digital
Subscriber Loop), PPPoE is selected.
• By manually entering a static IP address
• By configuring the ATA in DHCP mode
• By configuring the ATA in PPPoE mode
• Point-to-Point Protocol over Ethernet (PPPoE) defined in RFC
2516, is a means of connecting one premise to Internet ServiceProvider.
• PPPoE is a method of encapsulating data for transmission to a
far point.
• Its advantage is that, it eliminates need for the ITSP to manage
the allocation of IP address.
• If DHCP or PPPoE is enabled, IP address is allocated
dynamically by the respective server and static IP address need
not be entered.
• The IP address (Refer Glossary)• Click on ‘WAN’
WAN Port:
Parameters Value/Default Description
Enable DHCP Default = DisableClick, if DHCP is supported
by the ITSP
Enable PPPoE Default = Disable
Click, if PPPoE is
supported by the ITSP
User Name
Maximum 16
characters. Allow
ASCII characters
Enter Name e.g. Matrix
Password
Maximum 16
characters. Allow
ASCII characters
Enter Password
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Enable Static IP
AddressDefault = Enabled
Selete if static address
available from ITSP
IP Address
N.N.N.N
Default =
192.168.10.10
Ener IP address provided
by SE or ITSP while
Network installation at your
site. Do not enter if DHCP
is enabled.
IP Mask
N.N.N.N.
Default =
255.255.255.0.
Not required to change if
other IP addresses are with
same subnet
GatewayN.N.N.N
Default = BlankProvided by ITSP
MAC AddressDefault =
00.50.C2.55.B0.18
Matrix registered address.
Change, if multiple ATA are
used in the network.
IP Address
N.N.N.N = 1.0.0.1 To
255.255.255.254.
Default = Blank.
Provided by ITSP
Domain Name
Blank (40 char.).
Allow Extended
ASCII set.
Provided by ITSP
DNS:
Relevant Topics:
1. Glossary 106
2. VoIP Basics 88
3. Supplementary Services 75
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Appendix A: Troubleshooting
This section describes trouble shooting procedures for the Matrix
ATA. Some most common Problems and Corrective Action are
described below:
Symptom Probable Corrective Action
The Power LED does notlight up
Go through this checklist :
• Unplug the power adapter from the Phone Adapter.Wait for five seconds. Thenplug the power adapter intothe Phone Adapter again.
•
Make sure the power adapter you are using is the oneincluded with the ATA.
• Make sure that power sourceis turned ON.
After Power ON, SIP1, SIP2LEDs are fast blinking
Check that SIP Account isenabled and Registration withSIP Server is successful
After Power ON, SIP1, SIP2
LEDs are slow blinking
Check that on SIP Account page,
SIP1 and SIP2 are not disabledJeeves:
Matrix ATA cannot beaccessed from the ‘LANPort’ using Jeeves
• Make sure that computer’s IPaddress is in the samesubnet as ATA’s IP address.
• Ping the ATA. Click StartÆ Programs Æ Accessories Æ Command Prompt. Type‘ping’ in the command
prompt followed by ATA’s IPaddress and ‘Enter’. ATAshould respond. Otherwisemake sure that computer’sEthernet adaptor is ok.
• Make sure that ‘Password’ iscorrectly entered. Otherwisedefault the system with‘Reset’ button
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Password was changed butnow it isforgotten
No other way, except to restorethe default configuration file byresetting ATA.
Cannot change the
Password
Ensure that ‘retyping of new
Password’ is correct.Telephone port :
No dial tone, Error tone isheard.
• Error tone may be due to‘SIP Account not registered’.Check Phone portconfiguration.
• Ensure that it is the samephone port which is assignedthe SIP Account number
from the ISP.• Make sure the telephone is
plugged in appropriate port,Phone1 or Phone2.
• Verify the SIP ID number,entered is correct andactive/enabled.
Only Error Tone is heardwhen user of any phoneconnected at FXS1 or FXS2, goes OFF-Hook
Check that, on SIP Accountpage, SIP1 and SIP2 are enabledand on FXS port page for OGCalls, SIP1 assigned to FXS1and SIP2 is assigned to FXS2
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Appendix B: Applications
Typical deployment of “SETU ATA2LL” is explained for following
applications:
1. Making normal VoIP calls
2. Making calls via PBX and connecting multiple ATA
Normal VoIP calls:
• Today the Internet Connection from the service provider is
available at most of the places. Hence, Matrix ATA can be used
widely in a home or small offices, for making and receiving VoIP
calls through the ITSP.
Service Provider Network Operations
PSTNVoIP
Gateway
IP Network
Telephone or Fax Machine
IP Network Access(Modem or Router)
SETU ATA2LL
Telephone or Fax Machine
SETU ATA2LL
IP Network Access(Modem or Router)
• The above figure shows how the IP call is routed to the
destination. The analog phone and the ATA convert the call into
VoIP and sends to the Internet and ITSP’s SIP server through the
Modem or router. The server forwards it to the IP phones through
the Internet or to analog phones through the VoIP Gateway and
PSTN.
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Appendix C: Product Specifications
Configuration and Capacity:
Application No. of Parts
Connector Type
Maximum FXS
Port
For connecting Analog
Phone or Fax Machine2 RJ 11
PSTN Life Line
PortFor accessing PSTN line 1 RJ 11
LAN PortTo connect PC for
programming ATA1
RJ 45, 10/100
Base T
WAN Port
To connect Ethernet
cable router provides by
the service provider
1RJ 45, 10/100
Base T
Push Button
Switch
To default the system by
pressing a switch1 --
DC Jack
To connect the power
adapter for DC supply
voltage
1 --
VoIP and Networking Protocol:
• MAC Address (IEEE802.3)
• IPv4-Internet Protocol Version 4 (RFC791) Upgradable to V6
(RFC1883)
• ARP-Address Resolution Protocol
• DNS-A record (RFC1706), SRV Record (RFC 2782)
• DHCP Client-(RFC 2131)
• ICMP-Internet Control Message Protocol (RFC729)
• TCP-Transmission Control Protocol (RFC793)
• UDP-User Datagram Protocol (RFC768)
• RTP-Real Time Protocol (RFC1889) (RFC1890)
• RTCP-Real Time Control Protocol (RFC1889)
• DiffServe (RFC2475), Type Of Service TOS (RFC791/1349)
• SNTP-Simple Network Time Protocol (RFC 2030)
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Voice Functionalities:
• SIP-Version 2
• Two SIP Accounts can be used simultaneously
• DTMF: Inband and Outband (RFC2833)
• Call Progress Tone Generation: Select as per the country• QoS: RTP QoS and SIP QoS using DiffServ and ToS
• Full Duplex Audio
• Echo Cancellation: G.168, with programmable tail length of 8, 16
and 32msecs.
• Forward error correction (FEC)
• Voice Activity Detection (VAD)
• Fax using T.38 and Pass Through
• Speech Volume Setting-Transmit and Receive
• Flash Timer Programming
Voice Codecs: G.711 (a-Law and u-law), G.729, G.723
Telephony features:
• Voice Calls using SIP proxy and Voice calls without using SIP
proxy (Peer-to-Peer Calling)
• Call Waiting and Cancel Call Waiting
• Caller ID Blocking• Call Forwarding
• 3-Party Conference
• Call Transfer
• Caller ID: Generation and Display
• Calling Party Control (CPC)
• Speed Dialing with Phone Book feature
PSTN Life Line Support:
• Send Emergency Call to PSTN• Make PSTN Call by Dialing Prefix Number
• PSTN Relay when Device Power Down
• Receiving PSTN Phone calls
Time Settings: Synchronizing with specific Time Server
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Firewall Features;
• Outbound Proxy
• STUN
• NAT Keep Alive
Provisioning, Administration and Maintenance:
• Programming Using Web Page
• Software Upgrade using FTP
• Phone Programming of some parameters like:
• LAN IP Address
• WAN IP Address
• DND
LED Indication:
1 LED for Power
1 LED for each FXS Port
1 LED for each SIP Account
Security : Password Protected Administrator
Dimension (W x H x D) : 130x197x45mm (5.12” x 7.76” x 1.77”)
Power Supply:
External Adaptor : [email protected].
Power Consumption : 12W
Environmental:
Operating Temperature : -10°C to 50°C (140° to 122°F)
Operating Humidity : 5-95% RH, Non-Condensing
Storage Temperature : -40°C to 85°C (-40°F to 185°F)
Storage Humidity : 0-95% RH, Non-Condensing
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Appendix D: Glossary
10BaseT
10-Mbps base band Ethernet specification using two pairs of twisted-pair cabling (Categories 3, 4, or 5): one pair for transmitting
data and the other for receiving data. 10BASET, which is part of the
IEEE 802.3 specification, has a distance limit of approximately 328
feet (100 meters) per segment.
A-l aw
It is an ITU-T companding standard, used in the conversion between
analog and digital signals in PCM systems. A-law is used primarily
in European telephone networks and is similar to the North American μ-law standard. See also companding and μ-law.
CELP
It is the Code Excited Linear Prediction compression. It is the
compression algorithm used in low bit-rate voice encoding. Used in
ITU-T Recommendations G.728, G.729, G.723.
Codec
It is ‘Coder decoder’. In Voice over IP, Voice over Frame Relay, and
Voice over ATM, a DSP software algorithm used to compress/
decompress speech or audio signals.
Companding
Contraction derived from the opposite processes of compression
and expansion. Part of the PCM process whereby analog signal
values are rounded logically to discrete scale-step values on a
nonlinear scale. The decimal step number then is coded in itsbinary equivalent prior to transmission. The process is reversed at
the receiving terminal using the same nonlinear scale. Compare with
compression and expansion. See also a-law and μ-law.
Compression
The running of a data set through an algorithm that reduces the
space required to store or the bandwidth required to transmit the
data set. Compare with companding and expansion.
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DHCP
It is the Dynamic Host Configuration Protocol. It provides a
mechanism for allocating IP addresses dynamically so that
addresses can be reused when hosts no longer need them.
DNS
It is the Domain Name System. System used on the Internet for
translating names of network nodes into addresses.
Firewall
Router or access server or several routers or access servers,
designated as a buffer between any connected public networks and
a private network. A firewall router uses access lists and other
methods to ensure the security of the private network.
FTP
File Transfer Protocol - A protocol used to transfer files over a TCP/
IP network.
FXO
It is Foreign Exchange Office. An FXO interface connects to the
public switched telephone network (PSTN) central office and is theinterface offered on a standard telephone. Matrix-ATA’s FXO
interface is an RJ-11 connector that allows an analog connection at
the PSTN central office or to a station interface on a PBX.
FXS
It is Foreign Exchange Station. An FXS interface connects directly
to a standard telephone and supplies ring, voltage, and dial tone.
Matrix-ATA’s FXS interface is an RJ-11 connector that allows
connections to basic telephone service equipment and PBXs.
G.711
Describes the 64-kbps PCM voice coding technique. In G.711,
encoded voice is already in the correct format for digital voice
delivery in the PSTN or through PBXs. Described in the ITU-T
standard in its G-series recommendations.
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G.723
It is an ITU standard for speech codecs that uses the ADPCM
method and provides toll quality audio of 20 and 40 Kbps.
G.729Describes CELP compression where voice is coded into 8-kbps
streams. There are two variations of this standard (G.729 and G.729
Annex A) that differ mainly in computational complexity; both
provide speech quality similar to 32-kbps ADPCM. It is described in
the ITU-T standard in its G-series recommendations.
HTTP
Hyper Text Transport Protocol-The communications protocol used to
connect to servers on the World Wide Web.
IP
It is the ‘Internet Protocol’, Network layer protocol in the TCP/IP
stack offering a connectionless internetwork-service. IP provides
features for addressing, type-of-service specification, fragmentation
and reassembly and security. It is defined in RFC 791.
MACMedia Access Control Address - The unique address that a
manufacturer assigns to each networking device.
• MAC is Media Access Control address in the form of 48-bit
number, which is unique to the LAN NIC (network interface card).
• It is programmed into the card at the time of manufacture. IEEE
registration authority administers MAC address scheme for all
LANs which conform to IEEE, 802stds. Including both Ethernet
and token ring.
• Consists of two parts: 24-bit company id (manufacturer ID0 and24-bit Extension id (board id)
• Destination and source MAC names are contained in the header
of the LAN packet and are used by various devices like hubs,
bridges.
• A VoIP Service Provider will typically have its subscribers register
with the Service Provider’s VoIP service server before starting a
subscriber’s service. Often, an ITSP will require the registration of
the MAC addresses of any devices directly connected to their
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network.
NAT
It is the Network Address Translation. Mechanism for reducing the
need for globally unique IP addresses. NAT allows an organizationwith addresses that are not globally unique to connect to the
internet by translating those addresses into globally routable
address spaces, also known as Network Address Translator.
NAT-Traversal
It is a method of enabling specialized applications, such as Internet
phone calls, video and audio, to travel between your local network
and the Internet. STUN is a specific type of NAT traversal.
NAT Server
It is the Network Address Translation, an Internet standard that
enables a local-area network (LAN) to use one set of IP addresses
for internal traffic and a second set of addresses for external traffic.
NTP
It is the Network Time Protocol. Protocol built on top of TCP that
assures accurate local time-keeping with reference to radio andatomic clocks located on the Internet. This protocol is capable of
synchronizing distributed clocks within milliseconds over long time
periods.
Ping
Packet Internet Groper-An Internet utility used to determine whether
a particular IP address is online.
Proxy Server It is an intermediary program that acts as both a server and a client
for the purpose of making requests on behalf of other clients.
Requests are serviced internally or by passing them on, possibly
after translation, to other servers. A proxy interprets, and if
necessary, rewrites a request message before forwarding it.
QoS
It is the ‘Quality of Service’, the capability of a network to provide
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UDP
User Datagram Protocol-A network protocol for transmitting data
that does not require acknowledgment from the recipient of the data
that is sent.
VAD
It is the Voice activity detection, when enabled on a voice port or a
dial peer, silence is not transmitted over the network, only audible
speech. When VAD is enabled, the sound quality is slightly
degraded but the connection monopolizes much less bandwidth.
VoIP
It is the Voice over IP, the capability to carry normal telephony-style
voice over an IP-based Internet with POTS-like functionality,
reliability, and voice quality. VoIP enables a router to carry voice
traffic (for example, telephone calls and faxes) over an IP network. In
VoIP, the DSP segments the voice signal into frames, which then
are coupled in groups of two and stored in voice packets.
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Appendix F: System Commands
Meaning Command
To program the WAN IP Address 11-WAN IP Address-#*
To program the LAN IP Address 12-LAN IP Address-#*
Set Call Forward-Unconditional51-SIP Account-Destination
Number-#*
Cancel Call Forward-Unconditional 52-SIP Account-#*
Set Call Forward-When Busy53-SIP Account-Destination
Number-#*
Cancel Call Forward-When Busy 54-SIP Account-#*
Set Call Forward-When No Reply55-SIP Account-Destination
Number-#*
Cancel Call Forward-When No Reply 56-SIP Account-#*
Set Call Forward Ring Timer 57-SIP Account-Timer
Set DND 61-SIP Account-#*
Cancel DND 62-SIP Account-#*
Block Caller ID (per call) 65-FXS-#*
Send Caller ID (per call) 66-FXS-#*
Send Caller ID (all subsequent calls) 67-FXS-#*
Disable Call Waiting beeps (per call) 71-FXS-#*
Enable Call Waiting beeps (per call) 72-FXS-#*
Disable Call Waiting beeps (all
subsequent call)73-FXS-#*
Cancel all features 99-#*
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Index
A
Applications 101
Authentication 66
B
C
Call Progress Tones 33
Caller ID 34
Call Party Control 34
Calling Party Number Presentation 34Call-Routing 69
Call Waiting 75
Call Hold 75
Call Toggle 75
Call Transfer 75
Call Forward 75
Conference 75
DDial Plan 27
Default 65
Do Not Disturb 75
DTMF Option 68
E
Echo Cancellation 31
Emergency Number 58
F
Flash Timer 29
Forward Error Correction 29
FXS Port Name 29
G
H
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I
Incoming Call 36
J
Jeeves 42
K
L
LAN IP Address 52
Lifeline 58
M
MAC Address 56
N
Number Dialing Timer 35
New Password 60
NAT Keep Alive 69
O
Outgoing Calls 35Outbound Proxy 68
P
Polarity Reversal 34
Peer-to-Peer 35
Q
QoS 69
R
Ring Types 33
Registrar Server 66
Re-Registration Timer 66
S
SIP Account 66
SIP ID 66
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Status 72
SIP Protocol 13
STUN 69
TTones and Rings 33
Time Settings 87
U
Upgrade 71
V
VAD 111
Volume Control 33
Vocoder 67
W
WAN Port 95
X
Y
Z
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Notes
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Notes
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R e g i s t e
r o f C h a
n g e s
S.N. Date Major Programming Changes made
Programming Register
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fC h a
n g e s
Programming Register
S.N. Date Major Programming Changes made