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  SETU ATA211
SIP VoIP to FXO and FXS Gateway
 
 
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Matrix Telecom Solutions
Introduction

VoIP Adaptors with FXO, FXS and Multiple SIP Accounts

Internet Telephony offers intrinsic benefits of cost and flexibility. At the same time legacy telephony infrastructure and habits cannot be replaced overnight. People desire the best of both worlds - lower cost of VoIP and convenience of using existing telephony products and methods.

Matrix SETU ATA Range of Products is designed to meet this requirement of converting VoIP network to traditional telephony interfaces and vice-versa. It handles all the complexities of VoIP technology internally and provides simple telephone interfaces to make and receive calls.

Let Matrix SETU ATA be your bridge to the new world of IP Telephony!

Internet Telephony offers intrinsic benefits of cost and flexibility. At the same time legacy telephony infrastructure and habits cannot be replaced overnight. People desire the best of both worlds - lower cost of VoIP and convenience of using existing telephony products and methods.

Matrix Setu ATA is designed to meet this requirement of converting VoIP network to traditional telephony interfaces and vice-versa. It handles all the complexities of VoIP technology internally and provides simple telephone interfaces to make and receive calls.

Let Matrix Setu ATA be your bridge to the new world of IP Telephony!

Matrix Setu ATA is a SIP based Analog Terminal Adaptor (ATA), it interfaces legacy telephone devices with IP-based networks. It is specially designed for SOHO users to offer them the advantages of low-tariff Internet Telephony for long distance calls, international calls and Peer-to-Peer calls. It can be used with any existing PBX providing users access to VoIP trunks. It can also be used in a stand-alone mode.

Matrix Setu ATA converts the voice traffic into data packets for transmission over the Internet. When a telephone number is dialed by a user, Matrix Setu ATA converts it into an IP call using the SIP protocol and initiates a call to the dialed number in any part of the globe. Using an appropriate VoIP service provider, long distance call charges can be reduced significantly or eliminated through peer-to-peer calling on the IP network, connecting inter branch using your existing broadband connection.

Making an outgoing call is as easy as from a normal telephone. Call progress tones like Dial Tone, Ring Back Tone and Busy Tone are fed to the caller as per the called number status. The FXS ports can make outgoing calls on a common or two different SIP accounts. In addition, number based SIP account selection is provided to select the most economical SIP account for a given outgoing number.

An incoming call from a SIP account can be routed to any one or both FXS ports. All different CLIP protocols are supported so that the user can identify the caller before answering the call.

Once a call is established, features like Call Hold, Call Toggle, Call Transfer, Call Wait and Conference are supported to manage two calls from the same FXS port. Call forward in different conditions and Do Not Disturb are also provided.

The FXO port allows the user to dial numbers on the PSTN. This line can be used to make calls only when the internet connectivity is available. The FXO Port can be used to Network two PBX on different locations through the VoIP, and as the FXO Port is connected on the extension of the PBX it gives transparency of using features of other PBX.

Matrix Setu ATA provides two Ethernet ports - one for WAN and the other for LAN. The user can connect his PC on the LAN port and browse the Internet or check his emails while talking on VoIP calls.

Matrix Setu ATA can also be used with any PBX without changing its existing infrastructure. PBX users can make voice calls on IP to avail of the low-tariff of VoIP calls. The users can continue to make and receive calls without worrying on which network their calls are routed. Matrix Setu ATA is easy to install and operate. It can be configured using its built-in web pages served by the internal HTTP server.

Application Diagrams  

Residential Application 

 

Business Application

 

Peer-to-Peer Calling Application

Key Features  

Auto Configuration
SETU ATA can be configured automatically from a central location. The configuration details like Registrar Server Address, Authentication User ID, User Password are stored in the central server. When user connects SETU ATA to the network, it automatically downloads its configuration using TFTP. This plug-n-play feature requires the user to enter only the server address provided by the service provider.

Automatic Number Translation
Setu ATA supports multiple port types, FXO, FXS and SIP. Whenever a number is dialed from any of these ports, gateway routes the call to the desired destination port as per the routing mechanism defined for that port. In certain cases, the dialed number string is not understood by the network through which the call is to be routed, so by using Automatic Number Translation the dialed number string is translated into a number that is understood by the network or ITSP to reach the desired destination port.

Auto PSNT Fall back
Setu ATA can be interfaced to the PSTN using FXO Port. This port is used to dial out numbers to the PSTN Network. When Routing the calls from PSTN number to SIP trunk, the Ethernet Link may go down or the SIP Account used is not registered. So the call will not be routed through SIP and you will get error tone to avoid this you can use this feature Auto PSTN Fall Back through which the call will be automatically get routed through the alternate FXO Port.

Calling Party Control (CPC)
CPC is required to prevent hanging of the FXS port when it is connected to a device like an answering machine, a voice mail system, etc. When a call is released from the other side of the Internet, the Matrix SETU ATA can propagate this call release on the FXS in the form of Calling Party Control (CPC) signal. The device senses this signal and frees the FXS port.

Call Progress Tones and Rings
Matrix SETU ATA supports programmable tones and rings to match those of the country where it is installed.

CLIP
SETU ATA allows users to program the FXS ports for any of the three CLIP protocols - DTMF, FSK ITU-T V.23 and FSK Bellcore 212A.

Dial Plan
Matrix SETU ATA provides a list of programmable numbers or part-numbers with the preferred SIP account for each entry. When the user dials a number, the SETU ATA finds the matching number using the “best-fit” logic. It then uses the SIP account given against this matching number to make that call. This ensures lowest cost for all the outgoing calls.

Fax over IP (FoIP)
The user can send and receive Fax over SIP account, once a Fax machine is connected to SETU ATA. The SETU ATA supports FoIP using T.38 UDPTL and Pass Through

FXO
SETU ATA FXO port should be connected to the PSTN or PBX so that the user can make PSTN calls from this.

MAC Cloning
When replacing the existing hardware with other, you can simplify the installation process by copying the MAC Address of your existing PC. In such case, you do not need to delay the SETU process by informing your service provider of newly installed equipment.

Multi-Stage Dialing
Multi-Stage dialing is useful for ATAs connected to a SIP Server used for networking PBX of multiple sites. The user can dial the entire number string, both the destination number and extension number of the destination PBX together. The ATA will split the string into two stages, and dial out the destination number first and on receiving the answering signal the extension number. This ensures hassle free access to PBX extension

Incoming Call Routing
Calls arriving from any SIP account can be routed to either one or both FXS ports.

Jeeves (Web Based Programming Tool)
Flexible and user friendly windows based software, Jeeves, helps in programming the features through web browser. This web based programming feature helps users to configure the SETU ATA from any part of the world once it is connected with the IP network.

Peer-to-Peer Calling
SETU ATA can make and receive calls from other VoIP users without any Registrar or Proxy server. Numbers and IP addresses can be assigned to the other VoIP users to provide direct access across the network. For Peer-to-Peer calling, SETU ATA provides two options - (i) Peer-to-Peer Number Dialing (ii) IP Address Dialing. Organizations having multiple locations like branch offices and factories can use this feature to provide direct dialing between these end-points.

Phone Book
Frequently used numbers can be programmed in the internal phone book with 99 entries. The user can dial these numbers by using short codes in place of the complete, long numbers.

PPPoE
Matrix SETU ATA supports PPPoE client and hence can be used with any xDSL modems.

PIN Authentication
Authentication required to connect from one network to another, is called PIN Authentication, and is used to authenticate the caller to prove his identity before the call is being processed by SETU ATA to avoid the possibility of malicious calls and to avoid misuse of its services

Quality of Service (QoS)
Matrix SETU ATA supports TOS and DiffServe to facilitate improved voice quality.

Router
Basic routing capabilities are provided so that LAN port packets can be transferred on WAN port. This allows the user to browse the Internet and check his emails while making and receiving VoIP calls.

SIP Accounts
Two SIP accounts can be programmed and each FXS user can be assigned one of the SIP accounts for outgoing calls. Dynamic allocation of SIP account is also possible using Dial Plan.

Speech Volume Setting
SETU ATA allows user to set the transmit and receive gain to improve the quality of speech.

STUN
This capability allows Matrix SETU ATA to work behind asymmetrical NAT.

Supplementary Services
SETU ATA supports supplementary service like Call Hold, Call Waiting, Call Toggle, Call Transfer, Call Forward, Conference, Caller ID, DND and Making Another Call. These are the Service Provider dependant features.

Surface Mount Technology (SMT)
The Surface Mount Technology is the current semi-conductor packaging technology that offers reduction in real estate resulting in less heat generation and low power consumption. This is in turn improves reliability.

System Log
Syslog is one of the protocols used extensively for sending debug messages on IP network. It is a client/server protocol that uses UDP as transport protocol for debugging process. SETU ATA has in-built syslog client that enables it to send the debug messages. Logging has several benefits which include troubleshooting, security and system administration. Debug messages are sent to remote server on IP network for finding and reducing the number of bugs or defects from a system.

Features List
Auto Configuration Remote Programming
Calling Party Control (CPC) Speech Volume Setting (Transmit and receive)
CLIP (FSK-ITU-T V.23, Bellcore 212A) Symmetric RTP
CLIP to Caller Supplementary Services
Comfort Noise Generation
  Call Forward On BusyCall
Forward On No Reply
Call Forward Unconditionally
Call Hold
Call Toggle
Call Waiting
Caller ID
Call Transfer-Blind
Call Transfer-Attended
Conference 3 Party
Do Not Disturb (DND)
Making Second Call
STUN
Voice Activity Detection
DHCP Client
Dial Plan
Echo Cancellation (Programmable Tail Length- 8/16/32ms)
Fax over IP-T.38 and Pass Through
Flash Time (Programmable from 100-900ms)
Flexible Incoming Call Routing
Forward Error Correction (FEC)
Full Duplex Audio
LED Indications
MAC Cloning
Password Protection
Peer-to-Peer Calling  
Phone Book  
PPPoE  
Programmable Call Progress Tones and Rings  
Technical Specifications
VoIP    
     
VoIP Protocols
:
SIP v2, SDP, RTP, RFC 2833
Network Protocol
:
IPv4, TCP, UDP, DHCP, SNTP, STUN, HTTP, PPPoE
SIP
:
2 SIP Accounts
Out Bound Proxy Support
Display Name, User Name, Password, URL, Proxy URL, Registrar URL, Registrar Interval
NAT
:
STUN and NAT Keep Alive
Voice CODECS
:
G.711 A-Law, µ-Law, G.723, G.729A, G.729B
Line Echo Cancellation
:
G.168 with 8/16/32ms Tail Length
Call Progress Tones
:
Dial Tone, Ring Back Tone, Busy Tone, Error Tone
Voice
:
Dynamic Jitter Buffer (Adaptive), Comfort Noise Generation and Voice Activity Detection
Fax
:
T.38 and Pass Through
Quality of Service
:
Layer 3 DIFFServ and TOS
Data Network
:
WAN Port (RJ45), Auto MDIX 10/100 BaseT
Security
:
Password Protected Administration
 
 
FXS Port
Connection
:
RJ11
Off Hook Impedance
:
600
Loop Limit
:
270 (Max) Excluding Telephone Set
Loop Feed
:
39mA (Max)
Ringing Voltage
:
55Vrms @25Hz, 3REN
Pulse Dialing
:
10 PPS and 20PPS @ 1:2, 2:3 and 1:1
DTMF Dialing and Reception
:
ITUT Q.23 and Q.24
Caller ID Presentation (CLIP)
:
FSK ITU-T V.23 and FSK Bellcore 212A
Call Maturity
:
Polarity Reversal
Protection
:
Solid state (Over Voltage and Over Current) built-in Secondary Protection
     
FXO Port    
Connection
:
RJ11
Off Hook Line Impedance
:
600W
Loop Limit
:
1500W
Pulse Dialing
:
10 PPS and 20 PPS @ 1:2, 2:3 and 1:1
DTMF Dialing and Reception
:
ITUT Q.23 and Q.24
CLI Reception
:
DTMF, FSK ITU-T V.23 and FSK Bellcore 202A
Call Maturity
:
Polarity Reversal
Protection
:
Solid state (Over Voltage and Over Current) built-in Secondary Protection
     
Power Supply    
Input
:
12VDC @1.25A through External Adaptor (90-265VAC, 47-63Hz)
Power Consumption
:
5W (Typical)
Connector
:
DC Power Jack
     
Mechanical    
Dimensions (WxHxD)
:
7.9x10.5x2.7cm (3.1”x4.1”x1.1”)
Unit Weight
:
0.45Kgs (1.10lbs) Approx.
Shipping Weight
:
1.00Kgs (2.20lbs) Approx.
Material
:
ABS Plastic
Installation Mounting
:
Wall and Table-Top
     
Environmental    
Operating Temperature
:
-10°C to +50°C (-14°F to +122°F)
Storage Temperature
:
-40°C to +85°C (-40°F to +185°F)
Operating Humidity
:
5-95% RH (Non-Condensing)
Storage Humidity
:
0-95% RH (Non-Condensing) at 40°C
System Resources
Hardware
No. of Ports
Maximum FXS Ports
1
Maximum FXO Ports
1
LAN Port 1
WAN Port
1
DC Power Jack 1
VoIP Products Range
SETU ATA1S
SIP based Analog Telephone Adaptor with 1 FXS  Port and 2 Ethernet Ports
SIP based Analog Telephone Adaptor with 2 FXS Ports and 2 Ethernet Ports
SIP based Analog Telephone Adaptor with 1 FXO, 1 FXS Port and 2 Ethernet Ports
SIP based VoIP Gateway with 4 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port
SIP based VoIP Gateway with 8 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port
Multi-Port VoIP Gateway with 8 VoIP Channels, 4 GSM Channels, 2 FXO and 2 FXS Ports
Multi-Port VoIP Gateway with 8 VoIP Channels, 4 GSM Channels and 4 FXO Ports
Multi-Port VoIP Gateway with 8 VoIP Channels, 4 GSM Channels and 4 FXS Ports
Executive IP-Phone with 2 Lines x 24 Characters LCD Display
Executive IP-Phone with 2 Lines x 24 Characters LCD Display and PoE
Executive IP-Phone with 2 Lines x 24 Characters LCD Display and PoE
Executive IP-Phone with 6 Lines x 24 Characters LCD Display and PoE
SAPEX IPXP
All-Integrated IP PBX for 100 Users
   
 
   
  Copyright © Matrix Telecom 2009-2010. All rights are reserved.
Due to continuous technology upgradation, product specifications are subject to change without notice.^ Top