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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

Allowed and Denied Numbers	Attended/Blind Call Transfer
Automatic Number Translation	Answer and Disconnect Signaling on FXS Port
Answer and Disconnect Supervision on FXO Port	Call Detail Record
Call Progress Tones	Caller Line Identification and Presentation (CLIP)
Conference	CLI on FXS Port
CLI based Authentication	Day Light Saving
Date and Time Settings	Day Light Saving Mode
Default the Configuration (System Default)	Destination Number Determination Method
Destination Port Determination Method	Digest Authentication
Do Not Disturb	Emergency Number Dialing
Fax over IP (T.38 and Pass-Through)	Hotline
International Mobile	Equipment Identity (IMEI)
MAC Cloning	NAT and STUN Support
Peer-to-Peer Calling Table	PIN Authentication
PPPoE	Prefix to Domain Name Conversion
Programmable Access Codes	Quad-Band Support
Remote Held (SIP Accounts Only)	Remote Transfer (SIP Accounts Only)
Returned Call to Original Caller (RCOC)	SIM PIN
Speech Gain Setting	Supplementary Services Call Forward Call Hold Call Waiting
System Log Client	VLAN Tagging
Web based Programming

Signaling	Loop Start
Connector	RJ11
Off Hook Line Impedance	600Ω/900Ω Complex
Number of Long Loop Extension	1
Loop Limit	1500 (Max) Excluding Telephone set
Loop Feed	39mA (Max)
Ringing Voltage	Trapezodial-60 Vrms @25Hz 3REN Sinusoidal-52 Vrms
Pulse Dialing	10 PPS and 20 PPS @ 1:2, 2:3 and 1:1
DTMF Dialing and Reception	ITUT Q.23 and Q.24
CLIP	DTMF, FSK ITU-T V.23 and FSK Bellcore 202A
Flash Timer	83-999 ms.(Programmable)
Protection	Solid state (Over Voltage) Built-in Secondary Protection
Answer Signaling on FXS	Polarity Reversal
Disconnect Signaling on FXS	Polarity Reversal and Open Loop Disconnect

Signaling	Loop Start
Connector	RJ11
off Hook Line Impedance	600Ω/900Ω Complex
Loop Limit	1200Ω
Pulse Dialing	10 PPS and 20 PPS @1:2, 2:3 and 1:1
DTMF Dialing and Reception	ITUT 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
Answer Signaling on FXO	Polarity Reversal
Disconnect Signaling on FXO	Polarity Reversal and Open Loop Disconnect

GSM Band	Quad Band: GSM850, EGSM900 DCS1800, PCS 1900
Compliant	ETSI GSM Phase2/2+
SIM Card	One SIM per GSM Port
SIM Interface	1.8V, 3V
Transmission Power	Class 4 (2W) at GSM850 and EGSM900 MHZ band Class 1 (2W) at DCS800 and PCS1900 MHZ band
RF Sensitivity	Better than -102 dBm
Antenna (Panel Mount)	1 Antenna per 4 GSM Ports, 3.0 dBi, 50 SMA(Male) Connector

VoIP Protocols	SIP v2, SDP, RTP (RFC 2833)
Network Protocol	IPv4,TCP, UDP, DHCP, SNTP STUN, HTTP
SIP	9 SIP Accounts Out Bound Proxy Support
NAT	STUN and NAT Keep Alive
Voice CODECS	G.711 A-Law, µ-Law, G.723.1, G.729A, G.729B, GSM-FR, GSM-EFR
Line Echo Cancellation	G.168 with 128ms 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
Data Network	WAN Port RJ45 Auto MDIX 10/100 BaseT
Security	Password Protected Administration
Physical Connector	RJ45

Input	12VDC @1.25A through External Adaptor (90-265VAC, 47-63Hz)
Power Consumption	12W (Maximum)
Connector	DC Power Jack
LED Indications	1 each for Power and Status 1 each for GSM Port 1 each for FXO and FXS Port

Dimensions (WxHxD)	23.0 x 5.5 X 16.3cm (9.1" x 2.2" x 6.4")
Unit Weight	1kg (2.2lbs)
Shipping Weight	1.4kg(3.08lbs) (Please change as per given by manju for brochure changes)
Material	ABS Plastic
Installation Mounting	Wall and Table-Top

Operating Temperature	-100C to +500C (-14F to +122F)
Storage Temperature	-400C to +850C (-40F to +185F)
Operating Humidity	5-95% RH (Non-Condensing)
Storage Humidity	0-95% RH (Non-Condensing) at 40°C

System Resources	Description	SETU VGFX Configurations
		8422	8440	8404
VoIP Channels	To Make VoIP (SIP) Calls Using Internet or Intranet	8	8	8
GSM Ports	To Connect to GSM Network	4	4	4
FXO Ports	To Connect Two Wire Trunk Lines	2	4	-
FXS Ports	To Connect Analog Telephones	2	-	4

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