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 ATA1S 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 ATA1S be your bridge to the new world of IP Telephony!

Matrix SETU ATA1S 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 ATA1S converts the voice traffic into data packets for transmission over the Internet. When a telephone number is dialed by a user, Matrix SETU ATA1S 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.

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 ATA1S 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 ATA1S 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 ATA1S is easy to install and operate. It can be configured using its built-in web pages served by the internal HTTP server.

SETU ATA1S 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 ATA1S 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.

SETU ATA1S 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.

Matrix SETU ATA1S supports programmable tones and rings to match those of the country where it is installed.

SETU ATA1S 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 ATA1S 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 ATA1S 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 ATA1S 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 ATA1S. The SETU ATA1S supports FoIP using T.38 UDPTL and Pass Through.

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 ATA1S 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 SETU 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.

SETU ATA1S 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 ATA1S 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 ATA1S 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 ATA1S 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 ATA1S 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)	Answer Signaling
Auto Configuration	Automatic Number Translation
Called Party Number Table	CLIP (DTMF, FSK-ITU-T V.23, Bellcore 202A)
Called Party Number Table	CLIP (DTMF, FSK-ITU-T V.23, Bellcore 202A)
Comfort Noise Generation	DHCP Client/ PPPoE
Dialed Number Table	Digest Authentication
Disconnect Signaling	Echo Cancellation
Fax over IP-T.38 and Pass Through	Flash Timer
Forward Error Correction (FEC)	Full Duplex Audio
Incoming Call Routing	LED Indications
MAC Cloning	Multiple Gateway Support
Multistage Dialing	Password Protection
PCAP Trace	Peer-to-Peer Calling
Polarity Reversal	PPPoE
Programmable Call Progress Tones and Rings	SIP over TCP
Speech Volume Setting (Transmit and Receive)	Speed Dialing
STUN	Symmetric RTP
Supplementary Services	Syslog Client
Voice Activity Detection	VLAN Tagging
Web based GUI for Configuration	Call Forward Unconditionally
Call Forward on Busy	Call Forward on No Reply
Call Hold	Call Toggle
Call Waiting	Caller ID
Call Transfer-Blind	Call Transfer-Attended
Conference 3 Party	Do Not Disturb (DND)
Making Second Call