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.

1. Warm Welcome

2. Introduction
SETU VGFX
Multi-port VoIP-GSM-FXO-FXS Gateway
 Gateway offering seamless connectivity between VoIP, GSM, FXO and FXS networks
 Gateway offering VoIP and GSM network connectivity to traditional PBXs
 Gateway offering GSM and FXO network connectivity to an IP PBX
 Gateway offering networking and universal call routing

3. Variants
SETU VGFX
FXO Ports
GSM/3G Ports
FXO Ports
FXS Ports
VGFX8422
8
4
2
2
VGFX8440
8
4
4
0
VGFX8404
8
4
0
4

4. Target Customer

5. Applications

6. Stand-alone Application
GSM
TWT
FXS
POTS
IP
VoIP
SETU VGFX

7. VoIP-GSM Gateway for TDM PBX
GSM/3G
IP
Users
TDM PBX
SETU VGFX
Existing Infrastructure

Make cost-effective calls over VoIP

Save on fixed to mobile call cost

Maintain existing PBX infrastructure and dialing habits

Reduce dependency on a single network to make calls

8. VoIP-GSM Gateway for IP-PBX
GSM/3G
IP
PSTN
Users
IP PBX
Existing Infrastructure

Have access to legacy PSTN network

Make GSM calls from analog phones

Deployable in all sip based VoIP networks
SETU VGFX

9. Peer-to-peer Application
User dials 505
IP
SETU VGFX
SETU VGFX
Ext 202
TDM PBX
Location A
TDM PBX
Ext 505
Location B

Make IP calls without involving any SIP proxy server

Make free calls between multiple branch offices over IP

Short dial codes for 500 IP extensions , no need to remember IP addresses

10. Multi-Site Network Connectivity
PSTN
Access Local PSTN Network
of Paris Office From Sidney
SLT
Paris
Make Calls to Multiple
Office Locations Over IP
Mumbai
IP
Make Calls to
Teleworker
Teleworkers Any Where In
s
World Over IP
Access GSM Network of
Mobile
Mumbai From Sidney
Workers
Employees working
from Home
GSM/3G Mobile
Extensions
Sidney
GSM/UMTS(3G)
Network

11. > > >
Internet Accessibility over 3G/HSPA
SETU VGFX
LAN Users
Internet network
gets down
Internet
3G/ HSPA/EDGE
Conventional
Internet
connection

Access high speed internet over 3G/HSPA/EDGE

Take benefits of 3G/HSPA data plans offered by various service providers

Ideal for locations where there is no wired internet connectivity available

12. Software Features

13. Features
 Access Codes
 Allowed and Denied Numbers List
 Auto Provisioning for Mass Deployments
 Auto PSTN Fallback
 Automatic Number Translation
 Call Transfer (Attended/Blind)
 Call Detail Records
 Call Progress Tone and Rings
 DHCP Server and Client
 Do Not Disturb
 Dynamic DNS
 Emergency Number Dialing
 Hotline
 Internet Access over 3G/HSPA
 Message Wait Indication
 Peer-to-peer Calling
 Port Forwarding and DMZ
 SIM Balance Inquiry and Recharge
 SNMP Monitoring
 TLS and SRTP Support
 VLAN Tagging
 PBX Functionality
 Call Hold
 Call Transfer
 Call Forward
 Call Pickup
 Conference

14. Allowed and Denied Numbers

Allow or deny dialing of predefined outgoing numbers.

Avoid misuse and restrict unproductive calls

Program separately for VoIP, GSM, FXO and FXS ports
Call Detail Record (CDR)

Monitor and keep record of incoming and outgoing calls

Analyze calls with parameters like Call Duration, Calling Number, Port Type

Store details of 2000 calls

Download and print the call detail records - as and when required

Calls can be filtered on the basis of different Ports, Date, Day, Time, PIN Number,
Called and Calling Party, etc.

15. Automatic Number Translation
SETU VGFX adds +91
before dialing it out.
GSM/3G
User dials
23551111
SETU VGFX

Translates the number to match with numbering pattern of the destination network

Dial without worrying about the network through which the call will be routed

Supported on VoIP, GSM, FXO and FXS ports

16. Auto PSTN Fallback
User dials 201
Call automatically gets routed
through PSTN
SETU VGFX
PSTN
IP gets down
Ext. 201
User
IP

In case of IP outage, IP calls automatically get routed through PSTN
 Provides continuous accessibility of network
 Reduces dependency on IP
 More reliable communication

17. Call Progress Tones and Rings
 Different countries have different tones to indicate the process of call
activities such as Dial Tone, Ring Back Tone, Error Tone, Busy Tone, etc.
 Specific cadence can be programmed to match the tone used in each country
 call progress tones can be programmed country wise or can be customized as
per the user requirement

18. Conference
 Built-in feature
 3-Party conference
 Applicable if the call is originated from and terminated on FXS port
 External and internal parties are possible

19. Caller Line Identification

Display calling party’s name and number on FXS Port

Supports CLI on FXO, Mobile and SIP ports

CLIP on call transfer
Digest Authentication

Ensures secured communication over IP

Allow or restrict calls between specific group of users

Automated authentication mechanism

20. Day Light Saving
 Daylight saving is the procedure of setting clocks ahead at a particular time of
the year to make way for additional hour of daylight in the evening
 Real Time Clock (RTC) moves backward or forward automatically in tune with
the daylight saving requirement of the country
 DST can be forwarded or set backward according to day-month wise or datemonth wise as per the requirement

21. Do Not Disturb
 Do Not Disturb (DND) feature enables user to have privacy of not receiving the calls
for particular time period
 Outgoing calls can be made when Do Not Disturb (DND) is enabled
 Applicable on FXS ports

22. Emergency Number Dialing
 Allows the caller to contact local emergency services
 Program up to 5 emergency numbers
 Emergency number assigned on mobile port can be dialed out even if the port is not
registered or a SIM card is not present
Enhanced Voice Quality
 Toll Quality Voice with all Industry Standard Codecs Support
 G.711 A-Law, µ-Law, G.723, G.729AB, GSM-FR Support
 Voice Activity Detection (VAD) Offers Proper Bandwidth Utilization
 Comfort Noise Generation (CNG) and Echo Cancellation Ensures
Superior Voice Quality

23. FAX over IP

Send and Receive Fax using Efficient Media like Internet & ISDN BRI

T.38 Fax and Pass -Through for FoIP

ISDN G3 Fax

24. Hotline

Hotline is supported on FXS ports

Each FXS port can have different hotline number

Provision to impose delay before hotline activation

Delay timer can be minimum 1 and maximum 9 seconds
PIN Authentication
 PIN Authentication is to authenticate a caller to prove identity to proceed the call
from one network to another
 Supported on FXO, GSM and VoIP ports
 Support 500 PIN authentication entries

25. Least Cost Routing
 Different service providers have different tariffs for different regions and countries
 System automatically selects the most economical route to place the call
 Route selected depends on the destination number dialed
 Ensure each call at least possible cost

26. Message Wait Indication
Message wait tone
User
ITSP/PBX User
Voice mail box
SETU VGFX

Provides quick and easy information of voice message

Visual and tone based indication:
 Message wait LED indication on DKP and SLT
 Shuttered dial tone
 Voice message before dial tone
Voice message
indicator blinks

27. Network Selection
 Enables to choose GSM network automatically and manually
 Each GSM port can be programmed for specific service providers network
 Each GSM port can be assigned priority of maximum 9 networks if set on manual
mode

28. Return Call to Original Caller
Ext no. 301 dial
number
Called party is busy
or no replying
301
System stores called party’s
no., calling party’s no. with its
extension no.
234
When the user calls back, call gets
routed to the original extension
 Call attempted from mobile port of SETU VGFX is not answered or found busy
 System keeps the record in the database from which FXS port the number was dialed
to called party number
 System detects called party number on the returned call and will route the call to
original caller who had attempted to call

29. Real Time Clock
 Date and time are very important parameters for some features like Call Detail
Record (CDR), Daylight Saving Time, etc.
 Real time clock uses the Simple Network Time Protocol (SNTP) to get time from the
time server
 Flexibility to choose one from three pre-configured free reliable time servers
 Flexibility to program time server address of their preference

30. System Log
 System log protocol is used for sending debug messages on IP network
 It is a Client/Server protocol
 Uses UDP as transport protocol for debugging process
 Logging has several benefits
 Easier and faster troubleshooting
 Security enhancement
 Better system administration

31. SIM PIN
 SIM PIN is used for security on GSM network to protect the data stored on SIM
card
 PIN (Personal Identification Number) is stored in SIM
 If PIN is programmed, network asks user to provide PIN on every power on
 SIM PIN Number can be minimum 4 and maximum of 8 digits

32. Specifications
SPECIFICATIONS
Description
SETU VGFX8422
SETU VGFX8440
SETU VGFX8404
Number of VoIP Channels
8
8
8
Number of GSM/3G Ports
4
4
4
Number of FXO Ports (RJ11)
2
4
0
Number of FXS Ports (RJ11)
2
0
4
WAN Port
1
1
1
LAN Port
1
1
1
GSM /3G Band
Tri Band and Quad Band Operation
Power Supply
External Adaptor 12VDC/1.25A (Universal Input Range 90-265VAC, 47- 63Hz)
Power Consumption
12 Watt (Maximum)
RF Sensitivity
Better than -102dBm
LED Indications
Power Supply, FXO/FXS – One per Port, GSM – One per Port
Dimensions (W x H x D)
23.0 x 5.5 x 16.3 cms (9.1”x 2.2”x 6.4”)
Unit Weight
1Kg with Wall Mount and Table Top

33. Matrix GATEWAY RANGE OF
PRODUCTS
SETU VGFX
SETU VGB
SETU VFXTH
SETU VFX
SETU ATA211G
SETU ATA211
SETU ATA2S
SETU ATA1S
Multi-Port SIP based VoIP to GSM-FXO-FXS Gateway
Multi-Port SIP based VoIP to GSM and BRI Gateway
Multi-Port SIP based VoIP to FXO-FXS Gateway
Multi-Port SIP based VoIP to FXS Gateway
SIP based Analog Telephone Adaptor with 1 FXS, 1 GSM and 2
Ethernet Ports
SIP based Analog Telephone Adaptor with 1 FXO, 1 FXS and 2
Ethernet Ports
SIP based Analog Telephone Adaptor with 2 FXS Ports and 2
Ethernet Ports
SIP based Analog Telephone Adaptor with 1 FXS Port and 2
Ethernet Ports

35.  Type of Presentation: Product Presentation
 Number of Slides: 35
 Revised On: 1st March 2013
 Version-Release Number: V1R2
For Further Information Please Contact:
Email ID: Info.Telecom@MatrixComSec.com
Visit us at www.matrixcomsec.com