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 ATA211G 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 ATA211G be your bridge to the new world of IP Telephony!

Matrix SETU ATA211G is a SIP based Analog Telephone Adaptor (ATA), it interfaces legacy telephone devices with IP-based networks. In addition to the standard connectivity of IP and PSTN, it offers unique facility by providing connectivity between IP, GSM and PSTN 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 ATA211G converts the voice traffic into data packets for transmission over the Internet. When a telephone number is dialed by a user, Matrix SETU ATA211G 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 or inter-office call charges can be reduced significantly or eliminated through peer-to-peer calling on the IP network.

Making an outgoing call is as easy as from a normal telephone. SETU ATA211G automatically translates the dialed number matching to the format that can be understood by the destination (IP, GSM or PSTN) network. 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 or GSM ports can make outgoing calls on a common or 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 the FXS/GSM 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 is also provided.

Matrix SETU ATA211G 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 ATA211G 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 ATA211G is easy to install and operate. It can be configured using its built-in web pages served by the internal HTTP server.

Application Diagrams

Business Application
Business Application

Key Features

Auto Configuration

SETU ATA211G can be configured automatically by retrieving its configuration file from an Auto-Configuration Server (ACS). ATAs can be auto-configured in various power ON options or can be synchronized periodically from ACS .When user connects SETU ATA211G to the network; it automatically downloads its configuration files using TFTP, HTTP or HTTPS. This plug-n-play feature requires the user to enter only the server address provided by the service provider.

Automatic Number Translation

SETU ATA211G supports multiple port types: FXS, GSM, Ethernet 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.

Call Progress Tones and Rings

Matrix SETU ATA211G supports programmable tones and rings to match those of the country where it is installed.


SETU ATA211G allows users to program the FXS ports for any of the three CLIP protocols - DTMF, FSK ITU-T V.23 and FSK Bellcore 202A.

Dial Number Table

Matrix SETU ATA211G 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 ATA211G 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.

Disconnect Signaling

When a call is released from the other side of the internet, the Matrix SETU ATA211G can propagate this call release on the FXS in the form of disconnect signal. The device senses this signal and frees the FXS port.

Fax over IP (FoIP)

The user can send and receive Fax over SIP account, once a Fax machine is connected to SETU ATA211G. The SETU ATA211G supports FoIP using T.38 UDPTL and Pass Through.

International Mobile Equipment Identity (IMEI) Number

International Mobile Equipment Identity (IMEI) number provided on SETU ATA211G GSM engine is a unique 15 digit code to identify the GSM port. This number can be used to associate and tie the equipment with a particular GSM network.

Incoming Call Routing

Calls arriving from any SIP account can be routed to the FXS/GSM port.

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 ATA211G from any part of the world once it is connected with the IP network.

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 installation 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 it dial extension number. This ensures hassle free access to PBX extension.

Network Selection

SETU ATA211G provides flexibility to register with a GSM network manually or automatically. This is useful when the installation is close to a state or national border where local and foreign GSM networks overlap. Programming the ATA to work only with selected network prevents it from registering with an overlapping foreign network.

Peer-to-Peer Calling

SETU ATA211G 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 ATA211G 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.

PIN Authentication

Incoming calls on FXS, GSM or SIP ports of the ATA211G can be restricted to a specific caller .The caller has to first prove his authentication before calling to ATA211G. This feature is used to avoid the possibility of malicious calls and to avoid misuse of its services.


Matrix SETU ATA211G supports PPPoE client and hence can be used with any xDSL modems

Quad-Band Support

Matrix SETU ATA211G supports Quad-Band for 2G Network

Returned Calls to Original Caller (RCOC)

Matrix SETU ATA211G maintains records of all the unsuccessful calls on GSM and IP network. When a call is returned, it routes the call to the original caller comparing the called number, caller’s number and the system port details to the entries stored in its RCOC table. Thereby a returned call can be landed to the extension which placed the call, hence, saving valuable time.


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.

Signal Strength Indication

SETU ATA211G gives the indication of signal strength available for communication. Thus the possibility of network availability can be found. The signal strength indication is shown on the LCD of the telephone instrument supporting FSK CLIP.


SETU ATA211G allows user to program 4 digits PIN number (Personal Identification Number) which prevents the SIM against unauthorized use. User has to enter PIN for making any operations. User can change the SIM PIN as and when required. SIM gets blocked if PIN is entered wrongly thrice in a row.

SIP Accounts

Multiple 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 ATA211G allows user to set transmit and receive gain to improve the quality of speech.

Speed Dialing

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.

Supplementary Services

SETU ATA211G 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 built in protocol used extensively for sending debug messages on IP network. This client/server protocol uses UDP as transport protocol for debugging process. 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

100 Rel/PRACK (RFC 3262) IMEI Number
Auto Configuration LED Indications
Answer Signaling MAC Cloning
Automatic Number Translation Making Second Call
Called Party Number Table Mobile Network Selection
Comfort Noise Generation Multiple Gateway Support
CLIP (DTMF, FSK-ITU-T V.23, Bellcore 202A) Multi-Stage Dialing
Call Hold Polarity Reversal
Call Waiting Password Protection
Call Toggle Peer-to-Peer Calling
Call Transfer-Blind PIN Authentication
Call Forward on Busy Programmable Call Progress Tones and Rings
Call Transfer-Attended Quad-Band Support
Call Forward Unconditionally Return Call to Original Caller(RCOC Speech Volume Setting (Transmit and Receive))
Call Forward on No Reply Signal Strength
Conference 3-Party STUN
Do Not Disturb (DND) SIP over TCP
Dialed Number Table Speed Dialing
DHCP Client/ PPPoE Symmetric RTP
Disconnect Signaling Syslog Client
Digest Authentication Supplementary Services
Echo Cancellation Voice Activity Detection
Flash Timer VLAN Tagging
Full Duplex Audio Web based GUI for Configuration
Forward Error Correction (FEC)
Fax over IP-T.38 and Pass Through

Technical Specification


VoIP Protocols SIP v2, SDP, RTP, RFC 2833
Network Protocol IPv4, TCP, UDP, DHCP, SNTP, STUN, HTTP, PPPoE
SIP Multiple 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
Call Progress Tones Dial Tone, Ring Back Tone, Busy Tone, Error Tone, Waiting Tone
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, LAN Port (RJ45), Auto MDIX 10/100 BaseT
Security Password Protected Administration

FXS Port

Call Maturity Polarity Reversal
Connection RJ11
Off Hook Impedance 600Ω
Loop Limit 270Ω (Max) Excluding Telephone Set
Pulse Dialing 10 PPS and 20 PPS @ 1:2, 2:3 and 1:1
DTMF Dialing and Reception ITU-T Q.23 and Q.24
CLI Reception Polarity Reversal
Protection Solid state (Over Voltage and over current) Built-in secondary Protection

GSM Port

GSM Band Quad-Band: GSM850, EGSM900, DCS1800, PCS1900
SIM Card One SIM
SIM Interface 1.8V, 3V
Transmission Power Class 4 (2W) at GSM 850 and EGSM900 MHz band
Class 1(1W) at DCS1800 and PCS1900 MHz band
RF Sensitivity Better than -102dBm at GSM850/EGSM900/DCS1800/PCS1900
External Antenna Gain: Dipole= 2.5 dBi
Type of Antenna: Dipole/Whip Fixed/Omni Directional Antenna Roof- Top Antenna with flexible cable of 3 mtrs.(optional)
Antenna Connector: SMA (Female), 50Ω Impedance.
Speech Gain (Transmit- Receive) Programmable

Power Supply

Input 12VDC @0.75A through External Adaptor (90-265VAC, 47-63Hz)
Connector DC Power Jack

Power Consumption

SETU ATA211G 6W Approx.


Dimensions (W×H×D) 7.9×10.5×2.7cm (3.1"×4.1"×1.1")
Unit Weight 0.45Kgs (1.10lbs) Approx.
Shipping Weight 1.00Kgs (2.20lbs) Approx.
Material ABS Plastic
Installation Mounting Wall and Table-Top


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

Matrix SETU Range of VoIP Products

ETERNITY IP-PBX The IP-PBX with Universal Connectivity and Seamless Mobility
SAPEX All-in-One Embedded IP-PBX Server
VYOM CCX High-Density SIP Gateway
ETERNITY The Universal Telephony Gateway
SETU VGFX Multi-Port SIP based VoIP to GSM-FXO-FXS Gateway
SETU VFXTH Multi-Port SIP based VoIP to FXO-FXS Gateway
SETU VFX SIP based VoIP Gateway with 4/8 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port
SETU ATA211G SIP based Analog Telephone Adaptor with 1 FXS, 1 GSM and 2 Ethernet Ports
SETU ATA211 SIP based Analog Telephone Adaptor with 1 FXO, 1 FXS and 2 Ethernet Ports
SETU ATA2S SIP based Analog Telephone Adaptor with 2 FXS Ports and 2 Ethernet Ports
SETU ATA1S SIP based Analog Telephone Adaptor with 1 FXS Port and 2 Ethernet Ports
SETU VP248PE The High-Definition IP-Phone with 6 Lines x 24 Characters LCD Display and PoE
SETU VP248SE The High-Definition IP-Phone with 2 Lines x 24 Characters LCD Display and PoE
SETU VP248P The High-Definition IP-Phone with 6 Lines x 24 Characters LCD Display
SETU VP248S The High-Definition IP-Phone with 2 Lines x 24 Characters LCD Display
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