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 ATA211 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 ATA211 be your bridge to the new world of IP Telephony!

Matrix SETU ATA211 is a SIP based Analog Telephone 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 ATA211 converts the voice traffic into data packets for transmission over the Internet. When a telephone number is dialed by a user, Matrix SETU ATA211 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. 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 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/FXO 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.

The FXO port allows users to originate a call and terminate it at VoIP network. The routing logic used between FXO and VoIP helps the user to dial a destination PSTN number directly over VoIP network. Using FXO-VoIP routing logic two or more PBX can be networked to offer seamless connectivity between the PBXs

Matrix SETU ATA211 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 ATA211 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 ATA211 is easy to install and operate. It can be configured using its built-in web pages served by the internal HTTP server.

SETU ATA211 can be configured automatically from a central location. The configuration details like Registrar Server Address, Authentication User ID, and User Password are stored in the central server. When user connects SETU ATA211 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.

PSTN can be interfaced to the SETU ATA211 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, it may happen that 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 Auto PSTN Fall Back through which the call will be automatically get routed through the alternate FXO Port.

SETU ATA211 supports multiple port types: FXS, FXO, 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.

Matrix SETU ATA211 supports programmable tones and rings to match those of the country where it is installed.

SETU ATA211 allows users to program the FXS ports for any of the three CLIP protocols - DTMF, FSK ITU-T V.23 and FSK Bellcore 202A.

Matrix SETU ATA211 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 ATA211 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.

When a call is released from the other side of the internet, the Matrix SETU ATA211 can propagate this call release on the FXS in the form of disconnect signal. The device senses this signal and frees the FXS port.

The user can send and receive Fax over SIP account, once a Fax machine is connected to SETU ATA211. The SETU ATA211 supports FoIP using T.38 UDPTL and Pass Through.

SETU ATA211 FXO Port should be connected to the PSTN or PBX so that the user can make PSTN calls from the FXO Port.

Calls arriving from any SIP account can be routed to either FXO or FXS port.

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 ATA211 from any part of the world once it is connected with the IP network.

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 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.

Incoming calls on FXS or FXO ports of the ATA211 can be restricted to a specific caller. The caller has to first prove his authentication before calling to ATA211. This feature is used to avoid the possibility of malicious calls and to avoid misuse of its services.

SETU ATA211 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 ATA211 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.

Matrix SETU ATA211 supports PPPoE client and hence can be used with any xDSL modems

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.

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.

SETU ATA211 allows user to set transmit and receive gain to improve the quality of speech.

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.

SETU ATA211 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.

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.

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.

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

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

Connection	RJ11
Off Hook Impedance	600Ω
Loop Limit	1500Ω
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	DTMF, FSK ITU-T V.23 and FSK Bellcore 202A
Protection	Solid state (Over Voltage and over current) Built-in secondary Protection

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

Input	12VDC @1.25A through External Adaptor (90-265VAC, 47-63Hz)
Connector	DC Power Jack

SETU ATA211

5W Approx.

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

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

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