Introduction
Today, with expanding business horizons,
communication has emerged as a life-line for business sustenance.
Communication that is seamless, faster and cost effective is what the
new-age businesses need. Emergence of new telecom networks and the
advantages they offer have encouraged organizations to re-consider their
existing telecommunication network usage. They need an access to
crucial omnipresent telecom networks mainly VoIP, GSM and POTS to avail
the benefit of cost, convenience and quality of Service. For instance,
an organization with POTS communication backbone, needs connectivity to
VoIP and GSM networks or a business communicating through IP based
platform, needs to connect to GSM and POTS network. However, not all
businesses are equipped with telecom infrastructure flexible enough to
access these networks. On the other hand replacing their existing
telephony can be a costly affair. Under such circumstances a solution
that offers connectivity to these networks using existing communication
infrastructure, is the need of the hour for any business.
Matrix presents SETU VGFX- The Single-box
Gateway solution, offering seamless connectivity between VoIP, GSM and
POTS (FXO and FXS) networks. SETU VGFX supports flexible and intelligent
call routing options to ensure that communication always happens
through the most cost effective network.
Let Matrix SETU VGFX be your bridge to the new world of diverse telephony!
SETU VGFX allows seamless connection between
the VoIP, GSM and POTS network. On VoIP side, it supports (SIP based)
IP interface, allowing it to connect to any existing IP network. On the
GSM side, it supports Quad-band GSM operation, allowing it to work with
any GSM network. On the POTS side, both FXS and/or FXO interfaces are
supported. Incoming call from one network can be routed to another
destination using an appropriate network, depending on the destination
number dialed. Likewise, outgoing calls from FXS port will be routed
through an appropriate network, depending on the destination number
dialed. It can handle calls on all the ports simultaneously allowing
full traffic.
Using the innovative gateway for
multi-branch voice communication, an organization avails full benefit of
the low-tariff internet telephony, by establishing calls directly
between two destinations. Alternately, Matrix SETU VGFX can also act as a
SIP client, that too with the flexibility to register with multiple SIP
Service Providers.
Programmable Access Codes, Automatic Number
Translation, CLI Based Routing, Emergency Number Dialing, Least Cost
Routing, Automatic Mobile Network Selection and other advanced
functionalities ensures operational ease and convenience.
Matrix SETU VGFX is a single-box solution for Small
Office/Home Office, which needs to access VoIP, GSM and POTS networks.
Users can make calls to these networks using standard telephony
instrument connected to SETU VGFX. Additionally, FXS to FXS calls can
also be placed. Ideal to be used as a stand-alone gateway, SETU VGFX
ensures the most cost-effective route for communication.
SETU VGFX can be configured in a stand-alone mode
by connecting standard telephone instruments to its FXS ports and
VoIP-GSM-FXO networks to its respective network ports. Calls over IP are
made either through peer-to-peer or using the Proxy Server of a SIP
service provider.
Peer-to-peer calls eliminate the role of IP PBX or
SIP proxy server to establish calls over internet. It facilitates easy
and low-cost communication between geographically spread, multi-branch
offices. The user can access the GSM and POTS facilities of the remote
branch office to make calls and save long distance call charges borne
otherwise.
The Gateway using Peer-to-Peer Calls for VoIP
To establish Peer-to-Peer calls, both communicating
ends require internet connectivity with fixed IP address. A
Peer-to-Peer table needs to be programmed with the IP addresses of the
locations and corresponding numeric dialing codes. Using the FXO and GSM
ports, SETU VGFX can be connected to local FXO network and GSM network
respectively.
SETU VGFX helps an organization to integrate new
communication technologies to its existing infrastructure. Using SETU
VGFX, a traditional PBX system can enhance its capability and connect to
VoIP and GSM networks. With new networks being integrated,
communication happens using the most cost-effective networks. Features
like Automatic Number Translation, Programmable Access Codes and host of
other advanced functionality, enable users to access new networks
without changing their dialing habits.
GSM-VoIP Gateway for Traditional PBX
SETU VGFX can be connected to a traditional PBX in
two ways. Firstly, the FXS ports of SETU VGFX can be connected to the
FXO ports of the PBX. On dialing prefix assigned in the PBX, the user
can access these FXS ports to make GSM or VoIP calls. Secondly, the FXS
port (SLT port) of the PBX can be connected to FXO port of the SETU
VGFX. PBX users can access VoIP and GSM network connected to SETU VGFX
by dialing the FXS port number of the PBX.
SETU VGFX enables the IP PBX users to make calls
to the GSM and POTS network. It assures users the most cost effective
route of IP for all their calls to GSM and POTS till the last mile of
termination.
GSM-FXO Gateway for IP PBX
SETU VGFX gets registered to an IP PBX as a client.
More than one client can be registered to a single IP PBX, enabling it
to connect GSM and POTS network of various geographical locations.
Allowed and Denied Numbers
SETU VGFX offers flexibility to allow and deny
the dialing of a particular number or a set of numbers. The denied list
restricts a user to dial a number programmed in the denied list.
Automatic Number Translation
This feature allows the Gateway to translate
the number string dialed by the user to a format compatible with the
network through which the call is to be routed. So, the user can dial
numbers freely without worrying about the network through which the call
will be routed.
Call Progress Tones and Rings
The Gateway offers flexibility of programming
call progress tones and ring cadence to match the standards of the
country of installation. Country Specific Call Progress tones like Dial
Tone, Ring Back Tone, and Busy Tone etc. can also be programmed.
Caller Line Identification and Presentation (CLIP)
SETU VGFX can provide Caller Line
Identification Presentation (CLIP) on FXS ports. Analog CLIP protocols
such as DTMF, FSK ITU-T V.23 and FSK Bellcore 202A are supported by the
Gateway.
CLI based Authentication
A caller can be authenticated from the CLI
presented on SETU VGFX. When a call lands on the network ports of the
Gateway, it detects the CLI, verifies its authenticity and routes to the
predefined destination.
Call Detail Record (CDR)
SETU VGFX can store details of 2000 calls in
its memory. Call reports can be generated using filters like source
port, destination port, calling number, called number, date, time and
duration.
Day Light Saving
The Real Time clock (RTC) of the Gateway,
adjusts automatically to be in tune with the Day Light Saving
requirements of the country of installation.
FXO
The Gateway offers FXO port to connect the PSTN
network and route incoming calls to VoIP, GSM or FXS network and
vice-versa. It can also be used to connect extension of a PBX to network
two PBX over VoIP.
International Mobile Equipment Identity (IMEI)
IMEI number is a unique 15-digit code used by
SETU VGFX to identify a GSM device. The Gateway uses IMEI numbers to
identify its GSM Ports. This number also helps to associate a GSM port
of SETU VGFX with a particular GSM network.
Mobile Network Selection
This feature enables the
selection of a GSM network automatically or manually. Automatic network
selection allows the Gateway to select the available network every time
it is powered-on. While, the Manual network selection mode provides user
with the option to select a desired network.
Network Port Parameters
Web Jeeves of SETU VGFX provides a
configuration menu to program its network port parameters to match the
LAN addressing scheme of the installation site. Parameters such as IP
address, subnet mask, connection type, etc., can also be programmed
through a telephone instrument connected to the system.
Peer-to-Peer Calling
SETU VGFX support VoIP calls, between two
locations without going through a proxy server. Fixed IP addresses of
various locations can be programmed in the Peer-to-Peer table of the
Gateway to avail this facility. The Gateway supports 500 entries in the
Peer-to-Peer table. Numerical dialing codes can be defined to simplify
the calling between various locations.
PIN Authentication
The Gateway uses PIN Authentication to verify a
caller’s identity before routing the call from one network to another.
It is an important feature that prevents unauthorized usage.
Prefix to Domain Name Conversion
This feature enables the conversion of a
numerical code dialed by the user to a domain name. It helps the SIP
service provider to understand and route the call to the required
destination.
Remote Held/Transfer
The user of SETU VGFX can keep a called person
at remote end on hold or even transfer the call to a third person. This
feature can be activated only if the device at the called end supports
Call Hold/Call Transfer feature.
Returned call to Original Caller (RCOC)
RCOC is supported on GSM ports of the Gateway.
When an outgoing call is made through the GSM Port and the called party
is found busy or not responding, SETU VGFX will store details including
the caller’s FXS port number. In case the called person returns the
call, the Gateway will automatically route this call to the same FXS
port from where the call was attempted.
SIM Balance Inquiry and Recharge
SETU VGFX facilitates the
user to inquire about SIM balance and recharge the balance from
web-based GUI console without opening or extracting the GSM SIM.
SIP Accounts
Nine SIP accounts can be
programmed and each FXS user can be assigned with one SIP accounts for
outgoing calls. Dynamic allocation of SIP account is also possible using
Dial Plan.
Universal Routing
Universal Routing allows the Gateway to route a
call received from one network to another based on the routing
mechanism programmed in the system. A source port can be programmed for
more than one destination port so that if one port is busy, the call can
be routed through another port.
