Introduction
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.
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.
CLIP
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.
PPPoE
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.
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.
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.
SIM PIN
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.
