What's a VoIP GSM Gateway?

A VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).

Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.

Who offers VoIP GSM Gateways?

See especially (alphabetic order):

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2N TELEKOMUNIKACE

2N VoiceBlue Enterprise - SIP/H323 VoIP GSM Gateway / IP PBX

2N VoiceBlue Enterprise can be used as:

• VoIP to GSM Gateway and vice versa
• Protocol translator between the SIP and H323
• Gateway from your LAN to the Internet
• Small IP PBX

Features:

• SIP/H.323
• SIP proxy
• Up to 4 GSM channels
• Standard SIP/H.323 clients embedded in one hardware unit
• LCR according to free minutes and GSM providers
• SIP proxy Registrar for IP phones included
• Mobility Extension for SIP proxy users
• Intelligent incoming call routing
• DISA for incoming calls from GSM
• Top voice quality (EFR super sound)
• Easy way from Ethernet to Internet over GPRS with "dial on demand" function
• SMS sending and receiving (WEB interface and SMTP/POP3 protocols)
• Simple web based configuration
• CDR buffer for up to 500,000 records
• LOG and statistics saving
• Integrated antenna splitter
• Worldwide use (all frequencies are supported)

Successfully tested and proved as compatible with following systems:

• Cisco Call Manager
• Alcatel Omni PCx Enterprise systems
• Siemens Hi Path 3 000, 3 500, 3 800 PBX system
• Asterisk PBX system
• OnDo IP PBX
• Televentage equipment
• And other

Contact: sales@2n.cz

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2N VoiceBlue Lite - VoIP GSM Gateway

Ideal complementary product to any SIP-based IP PBX. It allows the user to get from the IP to a GSM network and vice versa very cheaply. The Ethernet (RJ 45) connection is a significant advantage in most cases.

Features:

• Up to four GSM channel gateway
• Mobility Extension
• Standard SIP client
• Smart Voice Routing - Least Cost Routing (LCR)
• Intelligent incoming call routing
• SMS server - support for SMS messages sending & receiving
• LCR according to free minutes
• Internal Callback
• SMS Callback support (XAPI server)
• Top voice quality (EFR super sound)
• Worldwide use (GSM 900/1800 MHz and 850/1900MHz)
• DISA with voice navigation, user defined voice message for incoming calls
• AT command based administration
• CDR buffer for up to 500,000 records, Compact flash
• LOG and statistics saving
• Integrated antenna splitter

Contact: sales@2n.cz

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Large GSM Gateway for termination - 2N StarGate ISDN PRI GSM IP gateway with VoIP card

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2N Traffic Control Manager

Unique support solution for traffic termination when using GSM VoIP gateways
It is LINUX based solution which enables successful traffic termination

The device can spread the traffic across a high number of VoIP GSM gateways in the best possible way.

Other added values:

• Centralized download of statistics and CDRs over web interface.

Contact: sales@2n.cz

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Also avalable at:

• Belgium VoIPsolutions (Reseller Prices available) Shipping all of Europ

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Cyber-Telecom.net.

Cyber Telecom GSM Gateway Range



Cyber Telecom is a OEM VoIP Hardware Provider, it is specialized in GSM Gateway. The GSM Gateway range is available from single port to 32 ports with configuration like Dual Band, Tri Band, Quad Band, SMS, GPRS, 3G / UMTS, SIP etc

Cyber-Telecom.net CT-G4S4 is a 4 Channels Quad Band GSM Gateway.
The design of the 4 Channels GSM Gateway providing you with a budget multi channels communication for your PBX or wireless application.

Features:

• Dialing tone with 450hz

• DTMF Dialing

• Bell signal

• Polarity reversal

• The RJ-11 socket which connect the phone or the payment

• Antenna socket (FME)

• IP usage switch

• Working state indicator

• Signal Strength indicator

Technical Specifications

GSM Module Dual band GSM 850mhz/900mhz/1800mhz/1900MHz

Phone interface：supply RJ-11 Phone Interface

Hanging voltage:45V Picking off voltage：30mA/41mA

Dialing tone Frequency：450Hz

Antenna interface ：Antenna amplifying 〉1.5DB

Sensitivity 〈 -104DBM

Transmitting power≤3W

Adapter interface：INPUT：AC 175~260V

OUTPUT：+12V/1000MA

Operation Circumstance:

       Operation temperature：-10℃ ~ 60℃

      Storage temperature：-20℃ ~ 70℃

      Operation humidity：45% —95%

      Atmosphere pressure：86 —106Kpa

       Environment noise：〈 60DB

       Transmitting Power: 2W

       Sensitivity: <-104dbm

       Antenna Amplifying: >1.5db

       Dialing frequency: 450Hz

       Hanging voltage: 45V

       Picking current: 30mA/41mA  

      Packing:
    ○ Product dimension: L455x W280 x H45mm
    ○ Gift box dimension: L510x W170x H64mm
    ○ G.W: 5.0kg

For more information, please refer to our website: http://www.cyber-telecom.net/

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ELGATO Communications

VoIP-GSM router Elgato

VoIP - GSM gate complex ELGATO includes capabilities of GSM and VoiP gates

VoIP - GSM gate complex ELGATO establishes telephone traffic exchange between GSM and IP NETwork.

The application of the given equipment in a corporate network allows to make calls from mobile GSM subscribers to IP network and on the contrary, the mobile subscribers can receive a call from subscribers of a corporate network. The system also can be applied in operator business, for transferring of the traffic from GSM network to IP a network, for example, for organizing international calls throughout networks of the IP-operators.

The transmission of the traffic in VoIP-GSM gate is organized through the protocol SIP, H.323.

A quality of voice transmission depends from audio-codec: G.711, G.726, G.729, G.723.

Device components:

Main block:
-Industrial case Advatech, back plain, 9 ISA, 3-PCI,
-Standard process border CPU ( P3 and higher)
-Flash HDD 128 Mb, RAM min 64Ìá, keyboard, floppy disk

Switchboard components
-Switchboard (support ISDN PRI, DSS1) and two FXS optionally!
-Radio-channel boarder(from 8 till 32 GSM channels)
-SIM matrix (up to 128 SIM-card)

Software
-Built-in
-Service (monitoring, Billing)
-IMEI codes changing program

GSM modules
-GR47 SonyEricsson
-Modules supporting IMEI codes changing

Antenna complex
-Antenna combiner 8/1 (8 GSM channel in 1 direction), signal loss less then 7dB
-One direction antenna YAGI , dual band, 13dB gain, 3m cable
-13m cable for YAGI antenna, signal loss 6dB/m

VoIP module (optionally)
-Cisco MC3800 íà 24 channel (codec G.729)

Computer block Call Centre (optionally)
The basic characteristics:
-Function of a return call - CallBack


With an incoming call the GSM gate determines and compares telephone number of the calling subscriber with numbers of the subscribers, for which the function CallBack is permitted(subscribers database). If such number is found, VoiceBlue breaks off connection and calls back him itself. If there is no such number in database the GSM gate either forbids connection or switches the subscriber according to the given route.


Calls restriction
With the help of prefix device determines the route of call and then compares it with number in database and either allows call or forbids them.

Possibility of recording of a voice greeting DISA
There is an opportunity of recording of a voice greeting in GSM gate ELGATO, voice greeting will be reproduced to the calling subscriber just after connection.

Least Cost Routing Function
The outcoming call is routed according to the prefix mask taken from database.

Automatic telephone number recognition
The function allows to determine number of the calling subscriber and either forbid or permit transfer of SIM number that established in a GSM gate .

Time conversation restriction (Max. of called minutes)
A limit of conversation can be set up for each SIM card

Programming
The GSM gates is programmed through the console or with the help of software working under Windows( CallCentre module). The programming can be implemented both locally and remotely.

Software updating
For updating the version of the software it is possible to use either applied CD or manufacturer(our) WEB-site. Updated software is given for free-of-charge.

Statistics of calls (CDR)
The device stores the detailed information about all outgoing and incoming calls (up to 500000 records)

LOG-file
The information about device condition is written in LOG-file: set up, initialization, arising of mistakes etc.

External antenna combiner
Due to external antenna combiner the signal from 8 GSM channels passes through only one external ( directed) antenna.


Technical characteristics
-from 1 up to 32 GSM channels
-a range of working frequencies - 900/1800
-type of used SIM card - plug-in 3 V (" small ")
-antenna complex YAGI (1 antenna for 8 GSM the channel, combiner, coaxial cable)
-Signal protocol - SIP, H.323
-IMEI code changing support

Voice compression:
-G.711 PCM at 64 kbps
-G.726 and G.727 E-ADPCM at 16 to 40 kbps
-G.723.1 (optional) MP-MLQ/ACELP at 6.3/5.3 kbps

Echo compensation:
-G.168-2000; max. echo length 25ms (15 ms for G.729A)

Connection:
-Ethernet (10BaseT), RJ45
-Dimensions – width 420(19”), height 170, depth 470
-a range of working temperatures - from 0 C up to 40 C

Multichannel GSM-gate ELGATO

Multichannel GSM-gate ELGATO can include from 8 till 30 channels.
It has specialized digital GSM module SonyEricsson GR47
GSM-gate is connected to telephone station by means of ISDN PRI DSS1.

Multichennel GSM-gate provides:
-outcoming calls from mobile telephone to PSTN (public switched telephone network)
-incoming calls from PSTN to mobile phone
-mobile-to-mobile connection
-statistics transaction (time of connection, calls costs, ….)
-protection against the non-authorized access in system
-device is assembled on modern element base and has high reliability (2 years warranty);
-the device constructively is IBM compatible industrial computer with switchboard(PRI board) and GSM channel board.


Flexible client-oriented solutions.

Contact: sale@ltd.com.ua
For more information, please refer to our website: http://elgato.com.ua/en/

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EURODESIGN BG LTD.

MobiGater SIP to GSM Gateway



MobiGater SIP is a model supporting one SIM card. The number of mobile phone numbers that can reach the MobiGater SIP and make SIP calls is unlimited. MobiGater SIP forwards the incoming SIP calls to a single mobile phone number.

Main Features:

• Incoming SIP calls to a mobile phone
• Outgoing SIP calls from different mobile phone numbers
• MobiGater SIP configuration via web interface
• Transfer call function to your mobile phone, during active SIP conversation
• Access your VoIP service from your mobile phone
• Use one headset to make and receive VoIP and GSM calls

MobiGater SIP Software Functions:

• Status Indication (Registered/Unregistered/Talk)
• Call Timer
• Unlimited Speed Dials
• Redial
• Call Transfer
• Call Forward (to GSM)
• Incoming Caller ID
• Unlimited Phone Book Entries
• Call Log- Missed, Received, Outgoing
• Multi-language Support

Networking: SIP, RTP, RTCP, DTMF
Voice: Voice Activity Detection, Acoustic Echo Cancelation
Codecs: G.711 uLaw, G.711 ALaw, G.729a, G.723, GSM

MobiGater SIP combines the best features of both communication tools - Internet calls and mobile phones. You have free VoIP calls and GSM mobility and coverage.

For more information, please refer to our website: http://www.mobigater.com/index.php?p=2&s=3

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HYPERMEDIA SYSTEMS

The VoIP GSM Gateways manufactured by Hypermedia Systems are pre-configured to meet precise customer requirements, and can be easily expanded to meet evolving telephony needs over time.
Hypermedia's range of VoIP GSM Gateways comes with 8 up to 72 GSM ports (and VoIP 3G Gateway with up to 36 ports). These VoIP GSM Gateways provide a tailor-made solution for alternative carriers and call termination and for call origination providers.

VoIP GSM Gateway – for Call Termination 72 GSM ports / 288 SIMs

This VoIP GSM Gateway introduces new techniques for reducing telephony call costs. It provides superior voice technology for connecting IP-based telephone systems, soft switches, and IP-PBXs to the GSM networks. This professional VoIP GSM Gateway connects directly to the IP-based systems using the existing IP/VoIP networks and cost-effectively routes incoming and outgoing fixed line, PSTN, cellular and VoIP calls – all in one box.



Features & Highlights:

• Ideal choice for long distance call / traffic termination operators, corporate and ISP customers, who wish to bridge traffic between VoIP or PRI and the GSM network.
• Compliance to all the relevant ITU-T recommendations.
• Modular, 6U high, with built-in antenna combiner per slot.
• 36 months warranty.
• Comply with Cisco, Quintum, Asterisk and others.
• VoIP – optional (Embedded VoIP)
• Web management and support - remote monitoring and configuration through TCP/IP access.
• The interface of the network side is E1/T1 Digital Interface with PRI ISDN signaling.
• The interface on the user side is up to 30 +2 x GSM channels (128 SIMs per E1 - 4 SIM per channel).
• All 30+2 GSM channels are dual band - GSM 900 MHz / 1800 MHz or GSM 850 MHz / 1900 MHz with built in auto-selection capability.
• Each GSM channel is individually programmable to work with any of the 4 SIMs, according to the user programmed time-of-day schedule.
• Pre-Paid Real-Time Usage Counter with minute roundability option.
• Unique caller ID blocking feature (if network allows)
• Echo cancellation.
• Accurate answer supervision and disconnect supervision.
• Reduces call costs - resulting from complete GSM to E1 PRI ISDN integration.
• Advanced Least Cost Routing (LCR) & Routing Groups
• Fast and easy installation and maintenance.

Contact: sales@hyperms.com

VoIP GSM Gateway - HG-4000

Hypermedia VoIP GSM Gateway bridges between the VoIP network (or IP-based systems) and the cellular GSM networks for reducing telephone call costs. It enables inbound and outbound VoIP and Cellular calls - all in one compact box!
The VoIP GSM Gateway is a complementary product to IP based equipments and supports up to 72 GSM ports. By using HG-4000 there is no need for any external VoIP Gateways such as Cisco or Quintum.

VoIP GSM Gateway - Fast Start Call Termination Kit

The VoIP GSM Gateway - HG-2000V is the ideal solution for anyone who would like to start a new termination business with a minimum investment: just plug it to your broadband carrier, and start selling traffic!
The HG-2000V series enables inbound and outbound VoIP and Cellular calls - all in one compact box.



VoIP GSM Gateway HG-2000V comes complete with:

• The VoIP Gateway and GSM ports (Embedded VoIP, no need for any external VoIP Gateway).
• 16 simultaneous bi-directional VoIP to GSM connections.
• 64 SIM cards – MultiSIM rotation.
• Upgradeable up to 72 GSM Ports / 288 SIMs.
• GSM worldwide use (850/1900 or 900/1800).
• Advanced LCR & Routing Groups.
• IP address & DDI pattern restrictions.
• Remote Pre-Paid SIM recharge.
• Integrated antenna splitter.
• Redundant Power Supply Optional.
• WEB management and control.
• Quick Installation.

Contact: sales@hyperms.com

VoIP GSM Gateway - HG-1600V

The VoIP GSM Gateway- HG-1600 series comes with 8 GSM VoIP ports, connecting IP-based systems with GSM networks. The VoIP GSM Gateway is designed to provide a cost-effective alternative for fixed-to-mobile or mobile-to-fixed telephone calls for small and medium businesses.



VoIP GSM Gateway main features:

• 4 - 8 GSM / VoIP ports
• Up to 32 SIM cards
• Advanced LCR and routing groups
• IP address and DDI pattern restrictions
• Remote pre-paid SIM recharge
• Integrated antenna splitter
• GSM worldwide use
• WEB management and control

Contact: sales@hyperms.com



Hypermeida's range of VoIP GSM Gateways provide:

Key Advantages:

• Fast return on investment (ROI)
• Fast installation and operation
• Flexible and scalable solutions
• Terminate GSM traffic – from VoIP to GSM, 3G/UMTS, and CDMA
• Terminate landline – from VoIP to PSTN, Analog and E1/T1 PRI
• Terminate "mixed" traffic – terminate GSM and landline calls in one box
• Remote web-based management and control
• Online technical support and training
• One point of contact for customer service
• Reliable and proven technology
• Professional carrier-grade gateways

Cost-Saving Advantages:

• Advanced LCR and routing groups
• IP address and DDI pattern restrictions
• 36 months warranty

Other related products:
3G VoIP Gateway
3G UMTS Analog Gateway
GSM Analog Gateway

FOR MORE INFORMATION:
www.hyperms.com
or by mail to Info@hyperms.com

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ICS France

ICS Universal Analog Cellular Gateway.
ICS TopGSM Series Analog GSM Gateway comes with 1/4/8 ports and provides a cost- effective alternative for fixed-to-mobile telephone calls. They are compatible with any PBX system or analog phones.
opGSM Series GSM analog gateway is used for connecting with the FXO ports of an analog VoIP gateway. With the use of the TopGSM gateway, inbound VoIP calls destined for GSM handsets and calls originating from GSM handsets can be sent to or received from the GSM handsets by the VoIP gateway without having to route the calls through the local PSTN thus eliminating the cost of paying the local wireline telephone exchange for the unnecessary switching.
Up to 4 SIM Cards per GSM Channel
The 4 SIM cards feature of the TopGSM gateways is important in markets where there are more than one GSM operator providing GSM services. A common practice of a GSM operator is to offer special rate concession to its subscribers for calling Mobile Numbers that are within the same network of the GSM operator in various forms such as: one flat rate for calling any Mobile Number within the GSM network; or special discounted rates for calling Mobile Numbers that are within the GSM network.
Each GSM Module board contains 4 SIM cards holders. An intelligent SIM card can be automatically switched according to 4 conditions:

Minute parameter --- You can change counting of outgoing calls.
         Count of minutes:  Gateway will counter time of outgoing calls.
         Count of calls: Gateway will counter number of outgoing calls.
By time --- SIM cards will be switched according to standby time
Time of use SIM ---- Interval of using selected SIM card.
Random --- SIM cards will be switched at random time.
             --- Interval of change SIM card. 
In this interval hours: minutes will be randomly switched SIM card. (SIM card will be switched after proceeding call hang-up).

ICS France
38 Rue Dunois, 75013 Paris France
Tel: +33 1 77 69 58 67 Fax: +33 1 56 61 28 33
E-mail: sales@icstel.com Website: www.icstel.com

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ITS Telecom

Convergence of Cellular, PSTN & VoIP Interfaces in One Box

Designed for corporate businesses and cellular operators the VoIP-Cellular-PSTN Gateway lets you seamless handover between cellular to fix calls or to VoIP and vice versa, any combination in any direction is possible.

Extremely beneficial for:

• Cellular operators- Enable them to stream cellular traffic via their network, increase traffic and gain new revenues from incoming calls
• Corporations- Enable them to migrate to the IP world without replacing the existing infrastructure and gain from low-cost VoIP calls and maximize network resources.

CGW-PX One Multi-Access Gateway

ITS Telecom and its multi-access gateway the CGW-PX offers a scalable, high performance gateway with 4 PRI interfaces, up to 32 cellular channels and supports GSM, CDMA and UMTS networks. ITS inherent PRI cards eliminate the need to purchase an additional PRI card for the PBX.

CGW-PX Main Features

• Embedded VoIP, PRI and Cellular interfaces
• Up to 32 cellular channels
• 4 SIM cards per channel
• PRI switching
• Up to 4 PRI trunks
• Supports GSM, CDMA & UMTS networks
• Cost saving LCR table
• CDR support
• Superior audio quality
• Easy installation- no need to reconfigure the PBX

For more Information:

Cellular-to-VoIP solution page: http://www.its-tel.com/category_info.asp?id=103_
CGW-PX Product page: http://www.its-tel.com/info.asp?id=55
More cellular solutions: http://www.its-tel.com/category_info.asp?id=103_144_

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NovaTec

NovaTec now offers the first on the world market TransNova® Internet Pathfinder (TIP) the combiner and advanced application between VoIP, ISDN and GSM. Which integrates your current telephone equipment either in Transparent mode or in SIP Protocol.

With no need to change your ISDN Infrastructure (fax, data and video conferencing), in fact the gaining of more advanced features such as least cost routing for telephony, fixed network, GSM Calls and also internet telephony, and allowing high double figure savings in your telephone bills, this is the killer application to rival the already killer application VoIP.

The ISDN structure is transported fully transparently into IP Data, therefore allowing for example your employees to completely mirror their workplace anywhere in the world where an IP network is available, also allowing you to mirror all your ISDN Features in the workplace direct to your home office, starting from Video door cameras and even video conferencing, and going as far as allowing all calls made to your office, being able to be taken in your home office at the company's expense over the Internet, and for companies with sub-branches abroad can now speak to their employees in these office free of charge, or when used in conjunction with the TMG if into the Mobile network, for the cost of a local Mobile to Mobile call.

NovaTec are the pioneers of GSM Gateways, SIP Gateways,VoIP Gateways and Encryption Gateways , in the digital compact form.

TransNova Mobile Gateways in the PRI range with inbuilt VoIP, these range from the TMG S6 with 4 - 16 GSM channels and can be fitted with up to 4 x E1's and have 4 Simcards per GSM channel, also the possibility to send Email - SMS - Email and a wide range of other features that is listed in the TMG Features list, this can be downloaded from the website at: www.novatec.de these are call through gateways and need no adjustment on already installed PABX's, as this can be used also for least cost routing etc.
The next range is the TMG S20 with from 4 - 60 GSM Channels (Completely Modular and hot pluggable) these can also be fitted with up to 4 x E1's so different interfaces can be connected, e.g. Voip and PABX, with the Least cost routing being done within the TMG. These are also call through so can be fitted before the PABX or after.
Other features include Call back options, from GSM or Fixed network, and also all systems come complete with antenna combining units with no extra costs being charged for this or other features, all are available with real time remote monitoring over IP, ISDN dial up or Local connection. Also realtime ASR monitoring is available with a spill over programm which lets you know if you have enough GSM Channels for the amount of traffic being terminated, if the ASR falls (down to simcard level)below a certain percentage then the system will send a report either by email or SMS to the network controller so ASR rates can be monitored and adjusted.
Antenna rotating systems can also be offered where the Antennas can be moved between base stations, randomly or as timed events, so taking pressure of the network providers when not enough capacity exists on individual base stations (also to hide the effect that the traffic is being terminated through a GSM Gateway).
All GSM Gateways will also be able to be used with the Sea of Sims (Simserver) this is a new application available for the TransNova where any number of GSM gateways can be installed in various locations without Simcards, these are held in a central location (Local SCU) and have a direct IP connection to the terminating units (remote location), the remote gateways request the simcard information required via the IP link and the SCU passes virtual sim information to the Gateway which using this information can terminate the GSM Calls. In the remote location GSM Gateways are local backup Sims for the eventuality that there is an IP breakdown. The SOS then takes away the restriction of how many Simcards can be used per GSM Channel. All Simcards are configurable per time, Budget or amount of minutes, and individual CDR's can be produced per Simcard.



For More information please contact us at info@novatec.de or visit http://www.novatec.de

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PORTech MV-370 / MV-372 / MV-374 / MV-378

PORTech is a manufacturer in taiwan. PORTech major in 31 channels E1 PRI/VoIP GSM Channel Bank(IP GSM Gateway build in VoIP CARD),SIM Server,SIM Bank,1 port /2 ports /4 ports / 8 ports VoIP GSM/CDMA/UMTS(3G) Gateway,GSM Fixed Wireless Terminals(Follow Me GSM Gateway) for 1 port/ 4 ports/ 8 ports and Skype GSM Gateway and Smart Power Monitor,IP Power Switch /PSTN Power Switch(4 Ports /8 ports)-Remote power manage by PSTN and WEB
DMT-GSM Channel Bank: GSM ,CDMA can be combined in one DMT

http://www.portech.com.tw

PORTech VoIP GSM/CDMA/UMTS(3G) Gateway,Follow Me GSM Gateway,GSM Fixed Wireless Terminal,E1 GSM Channel Bank,Skype GSM Gateway,Smart Power Monitor

VoIP GSM Gateway / VoIP CDMA Gateway / VoIP UMTS Gateway

Portech MV-370 - 1 channel VoIP GSM Gateway

MV-370 : 1 channel VoIP GSM Gateway

Look at the MV-370 configuration howto

Portech MV-372 - 2 channels VoIP GSM Gateway

MV-372 : 2 channels VoIP GSM Gateway

Look at the MV-372 configuration howto

Portech MV-374 - 4 channels VoIP GSM Gateway

Portech MV-378 - 8 channels VoIP GSM Gateway

for call termination (VoIP to GSM/CDMA/UMTS ) and origination (GSM/CDMA/UMTS to VoIP). It is SIP based and compatible with Asterisk. It can enable to make 1,2,4,8 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS network to IP phone.
MV serial can Receive SMS and send SMS on web
Allows your program Send/receive SMS with AT Command
Call Back feature

GSM Frequency :

• Dual band:900/1800MHZ
• Tri band:900/1800/1900MHZ
• Tri band:850/1800/1900MHZ.
• Quad band:850/900/1800/1900MHZ
• CDMA 2000 (800mhz)

• 3G/UMTS Version: 3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz

DuMV@PCI: 2 ports GSM/VoIP PCI Card

DuMV@PCI: 2 ports GSM/VoIP PCI Card is Full function (both GSM and VoIP) on board , SIP based and compatible with Asterisk. It can enable to make 2 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS network to IP phone.
Beside Voice,2 ports GSM/VoIP PCI Card support SMS sending and SMS receiving on web directly . Also allows your program Send/receive SMS with all GSM AT Command .
Worldwide use ( 2G-850/900/1800/1900 MHz ,3G compatible with all world and Japan,CDMA 2000)

• Antenna : 1
• SIM Card Hot swapping
• Individual route table- LAN --> MOBILE and MOBILE --> LAN
  • Support one stage diaing
  • You can dial any GSM destination number via the sim from lan phone directly.
  • Support free mode (two stage dialing) and assigned mode (one stage dialing)4.Voice response for setting and status(dial in from mobile).
• Standard SIP(RFC2543，RFC3261) protocol，communicates with other gateway or PC
• Receive SMS and Send SMS On web directly
• Allows your program Send/receive SMS with all GSM AT Command
• call back
• Full function on board and can be set on web.

□PT-110211 Dual band:900/1800MHZ
□PT-110212 Tri band:900/1800/1900MHZ
□PT-110213 Multi band:850/1800/1900MHZ
□PT-110214 Quad band(siemens) :850/900/1800/1900MHZ
□PT-110215 Quad band(simcomm) :850/900/1800/1900MHZ
□PT-110216 3G(World)
□PT-110217 3G(Japan)
□PT-110218 CDMA 2000 (800MHZ)

GSM Fixed Wireless Terminals / CDMA Gateway

MT-350~1 ports GSM Fixed Wireless Terminals,Follow Me GSM Gateway
MT-350C~1 ports CDMA Fixed Wireless Terminals

• MT-350C doesn't have call transfer-follow Me function.

MT-354~4 ports GSM Fixed Wireless Terminals,Follow Me GSM Gateway
MT-358~8 ports GSM Fixed Wireless Terminals,Follow Me GSM Gateway
MT-358_Follow Me GSM Gateway
It has FXS,FXO,GSM interface that connect with PBX,PSTN or VoIP Gateway
Auto-select economic route (LCR)
GSM Prefix No.Setting (50 sets)
Polarity reversing function
Call-transfer: Follow Me
( 1) PSTN to GSM:
     An incoming call of PSTN will be transferred to specify GSM no. 
( 2) GSM to PSTN:
     *An incoming call of GSM will be transferred to specify PSTN no. 
     *GSM in PSTN out, give you a prompt "du" and you can enter any destination number 
GSM Frequency :
Dual band:900/1800MHZ
Tri band:850/1800/1900MHZ
option:MT-350S, Add SMS Function(Short message Service)(dual band)(1 port only)

E1 / T1 PRI / VoIP GSM/CDMA/UMTS Channel Bank

DMT: E1 GSM Channel Bank
DMT-V: 8/16/24/31 VoIP GSM Channel Bank
DMT-GSM Channel Bank: GSM ,CDMA can be combined in one DMT
DMT accepts incoming call from E1/T1 PRI of PSTN or
PBX and choose one GSM channel to dial out according to the prefix of the
destination mobile number. In this way, we can have least cost routing.
DMT can provide Call Detail Record (CDR) for traffic and accounting management.

• ISDN PRI interface: 1 port or VoIP Card*1
• GSM/CDMA/UMTS interface: 31 ports
• Optional external SCE-900: Expands SIM card upto 32 pieces per GSM channel.
• Optional external SCBox: Expands SIM card upto 4 pieces per GSM channel.
• Applicable for telecom service providers
• VoIP application: Gateway's E1/T1 port connects DMT's E1/T1 port
• Calling time: user can assign the calling time for each sheet of the SIM

card.When the time runs out, system will block this GSM port, or switch
to next unused SIM card (in the case that SCE is installed).
Besides PRI to GSM,you may also apply GSM to PRI (two ways).

• Additionally,we also develop DMT-2 that is E1*1 in,GSM*31 and E1*1 out.
• Frequecny:
  • Dual BAND:900/1800 MHZ
  • Tri Band: 900/1800/1900MHZ
  • Tri Band: 850/1800/1900MHZ
  • Quad Band:900/1800/1900/850MHZ
  • CDMA 2000(800MHZ)
  • 3G/UMTS:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz

DMT-E1 GSM Channel Bank

Antenna collecting Box: Base equipment.

 
If you want to connect extra Outdoor directional antenna or Yigi antenna.....,you can purchase option Antenna Combiner(AC-32)

 

PSTN Power Switch/ IP Power Switch

If your control room is usually no body there,you can consider our Smart Power Monitor(4 ports)- Remote power on/ off /reboot by PSTN and WEB..

Skype GSM Gateway

SkyTrunk: Skype to PBX free, PBX to Skype free

SkyTrunk ST-1004~4 ports skype,FXO,FXS,GSM
GSM Frequency :Dual band:900/1800MHZ,Tri band:900/1800/1900MHZ


SkyTrunk ST-2004 ~4 ports Skype,FXS


SkyTrunk ST-3004~4 ports skype,FXO,FXS


SkyServer:~4 ports Skype,FXS (Build in pc)


MV-370 GSM to SIP Gateway


That VOIP GSM gateway can be directly connected to any SIP IPBX.
It is manufactured by Portech, a Taiwanese company.
Main advantage of that gateway is the direct SIP connexion, which makes the voice quality very good.
Another advantage is the price : you can buy it for MV-370 about $220,MV-372 about $360 on ebay.

Firmware and manual are available at http://www.portech.com.tw/eweb/index1.htm.
As the product is still very young there is little info on the net about it.
To avoid instability problems, be sure to flash the latest firmware before any configuration.
If you want to connect that box to Asterisk, I've written a how-to available at http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk

I hope it will make you gain lots and lots of days of setting up and adjustments :-)

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Available at:

• Italy,France,Ukraine,Spain and Portugal http://www.wildix.com

• Belgium VoIPsolutions (Reseller Prices available) Shipping all of Europe

• Portugal Quiosque Virtual - Portech MV-370 - Portugal & Europe
• Portugal Quiosque Virtual - Portech MV-372 - Portugal & Europe

• United Kingdom http://www.westlake.co.uk

• Austria http://www.fastcomp.at

• USA http://www.ecrater.com

• Hungary http://www.gamaxnet.hu

• Russia http://www.mototelecom.ru

• PORTech distributors all world http://www.portech.com.tw/

If you have interest,please contact us

PORTech Communications Inc.
150 Shiang Shung N.Rd,Taichung,Taiwan
http://www.portech.com.tw
E-mail:info@portech.com.tw
http://www.portech.com.tw/eweb/PORTECH_dm.pdf ---- PORTech Products list
You can click the product's picture to read the catalog.

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QuesCom

QuesCom VoIP GSM gateway

QuesCom 300

Technical specifications:

• 8 GSM Channels
• compatible with frequencies Tri-band GSM 850/1800/1900 MHz or 900/1800/1900 MHz or UMTS 850/900/1800/1900/2100 MHz
• SIM Free
• SIM Management by QuesCom SIM Server
• Supports H.323 and SIP simultaneously
• Supports Number Portability Database
• Integrated SMS Gateway

Further information on QuesCom GSM gateways

QuesCom 400

Technical specifications:

• Up to 16 GSM, compatible with Tri-band GSM 850/1800/1900 MHz or 900/1800/1900 MHz
• SIM Free
• SIM Management by QuesCom SIM Server
• Supports H.323 and SIP simultaneously
• Supports Number Portability Database
• Integrated SMS Gateway

Further information on QuesCom GSM gateways

As an addition to your QuesCom GSM gateway solution architecture, the QuesCom SIM Server manages the SIM allocation without the need of human intervention by gathering all SIM cards in a single location. According to the time of the day, calls to GSM are routed to the best SIM card without ever exceeding plans.

Website: http://www.quescom.com

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SIPCPE

FX-300 GSM Call Director(a GSM <--> VOIP/PSTN switch and bridge)


Most of the voip customer premise equipment (Cisco ATA 186, Grandstream ATA 286, Sipura SPA-2000, Motorola ATA, etc) sold or provided for free by the various and numerous voip service providers such as Vonage, AT&T, Time Warner, Packet8, etc do not provide any interconnect to the traditional cellular or landline networks. Thus, users must use a separate handset to make and receive voip calls and they are restricted to using the voip service when at home or in the office.

SIPCPE products such as the FX-300 GSM Call Director are accessory add-on products designed to provide a bridge or interconnect between the customer's voip hardware mentioned above and the traditional cellular (GSM) networks respectively thereby enabling remote access to the customer's voip service.

The most cost effective mobile convergence solution available and the only solution you can use with the popular VOIP services such as Vonage, Packet 8, At&T, Version, Time Warner, etc.

Suggested Retail only $299

Specifications

• Built in Tri-Band GSM Modem 850/1800/1900 or 900/1800/1900
• Both a PHONE port and a LINE port
• Connects to VOIP or PBX or PSTN
• External Antenna

Advanced Features

• Forward unanswered VOIP calls to your GSM phone
• Access you VOIP service from your GSM phones.
• Share your unlimted VOIP service plan with all your GSM phones in your family or business.
• Use one handset to make and recieve your VOIP and GSM calls
• Connect to VOIP, PSTN or PBX
• Set up a private user group up to 15 members

Now you can connect any voip gateway (FXS port) to your GSM network.

SIPCPE Support Forum

VOIP (Grandstream) Connection Illustration

http://www.sipcpe.com//fx300GSM.html
sales@sipcpe.com for inquires or call 1-650-641-1784

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SL SOLUCIJE

SL Gbox: ASTERISK PCI GSM CARD
PCI card with four slots for up to 4 GSM channels on one card for using with Asterisk (with drivers).
Create your own Asterisk gateway with 4 GSM channels or more (using more PCI cards)

Expected date of commercial arrival: July 2007.

Expected prices:

PCI card (for 4 gsm modules): $500
GSM module: $150

Presale prices: PCI card $350, GSM module $140 (PCI card with 4 modules $900)
Presale reservations on: info(x)slsolucije.hr

(In development: standalone rackmount SIP GSM gateway with up to 32 GSM channels and with one CPU/ethernet module - expected price for a system with 32 GSM modules included is: $6500. Additional E1 cards can be installed for E1 connectivity and/or pass through - interception of GSM calls)

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SunComm SC-375 Voip GSM Gateway

This is a product distributed, in Italy, by Altesys, which offer the ability to gateway GSM traffic via SIP protocol and vice versa.
It has an external antenna with about 2.5m of RG174 coax cable, and works with asterisk configuring:
sip.conf
[09]
type=friend
host=dynamic
username=09
secret=12311
mailbox=10
dtmfmode=inband
call-limit=1
insecure=very
context=default

SC-375 configuration
There are several parameter to configure, very normal for people who have already configured other SIP device.
A special note that should be taken into account is the routing parameters to permit LAN to MOBILE and MOBILE to LAN:

• in 'Mobile to LAN settings' set a rule with CID = '*' and URL = '0' or the extensions to enter in the dialplan to reach the IVR menu for the incoming calls from GSM network;
• in 'LAN to Mobile settings' set URL = '192.168.0.1' or the IP number of the asterisk PBX, and Call Num = '#' to enable outgoing callings from asterisk to any number through GSM network.

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TelecomChinaSourcing

TelecomChinaSourcing- WirelessWay3001/3004

This WirelessWay 3004 take advantage of the established worldwide availability of GSM/CDMA Systems for all of customers� communications requirements, including digital voice, analog fax (optional), which can provide access to home, office or any remote location where phone service is needed and GSM/CDMA wireless networks are available. WirelessWay 3004 is 4 lines Terminal, provides wireless access through GSM/CDMA network, connect to normal phone as well as VOIP GateWay. WirelessWay 3004 also provides charging signal and stat. function, available for small/middle size enterprise and public call office.

More information on http://www.telecomchinasourcing.com or by email: sales@telecomchinasourcing.com

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TELES Communication Systems

TELES.VoIPBOX GSM - VoIP GSM Gateway

The TELES Access Gateway products portfolio has been specifically designed to meet and support the changing demands of the small carrier and enterprise markets. TELES' best-in-class range of access gateway products support the wide array of IP network application scenarios, allowing you to grow your business at minimal outlay without changing your technology footprint.

TELES.VoIPBOX GSM is a combined mobile and VoIP gateway solution for carrier networks and corporate customers wanting to connect their PBX to mobile and VoIP services.

This full-featured gateway can be added to analog and IP environments in a cost-effective, quick and convenient manner. TELES.VoIPBOX GSM converts fixed-to-mobile into mobile-to-mobile calls, thereby terminating calls to mobile networks at lower rates than is possible via fixed-net interconnection.

TELES.VoIPBOX GSM

Features:

TELES.VoIPBOX GSM is the ideal mediator for carriers who want to give corporate customers with access to their mobile and VoIP networks. It enables mobile network operators to add fixed-net minutes to their current revenues, thereby reaching out to a completely new customer base.

TELES.VoIPBOX GSM provides extremely flexible least cost routing combined with other superior call-routing and rerouting capabilities. Routing strategies are based on various criteria such as called party, CLIP/CLIR, time-of-day, time quota, service quality and SIM card profile.

On top of the basic call scenario needs, TELES.VoIPBOX GSM provides a diverse range of special inter-networking applications such as email-to-SMS conversion, callback functionality, two-stage dialing and more.

Integrated GSM/CDMA gateway

Integrated VoIP gateway

Analogue support ( 4 FXS)

Best call destination selection engine

Easy installation and remote management

Solutions

TELES.VoIPBOX GSM is connected to the mobile network and directly to the PBX via FXS lines and/or IP. It includes a VoIP gateway with 8 VoIP channels, 4 mobile channels and 4 FXS interfaces. Analog phones, existing PBX, or a SoftPBX can be connected. You can set up TELES.VoIPBOX GSM for several mobile networks and automatically route your call to the network with the lowest rate.

Where to buy

• Interwork Technologies - Serving the US and Canada. Resellers call for pricing!
• Iridiacom Limited - Serving the UK and Ireland. Resellers call for pricing!

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TELEOPEN.NET

TELEOPEN.NET PCI GSM Card
100% X100P Compatible.



100% X100P Compatible.
One GSM channel per card.
Call Start/End detection.
900/1800 MHz networks support.


TELEOPEN.NET IP8GSM, IP16GSM, IP24GSM, IP32GSM
8, 16, 24, 32 port GSM gateways for IP-PBX

Basic features:

• 8,16,24,32 GSM channels
• 8,16,24,32 VoIP channels
• Integrated H.323 server and client
• Integrated SIP server (Registrar and Proxy) and client (User Agent)
• Integrated IAX2 server and client
• Call routing (via routing tables)
• Direct Inward System Acccess (DISA), e. g. DTMF extensions postdialling
• Download of user defined announcements for user guidance
• Call Detail Records (CDR) to MySQL server
• SIM management software
• Remote monitoring via web interface
• GSM 850/900/1800/1900
• Integrated antenna splitters
• Configuration via web interface

Further information: http://www.teleopen.net/equipment.html

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Topex

TOPEX VoIP to GSM gateway - VoiBridge

VoIP to GSM/UMTS small capacity gateway. Its main functionality is to interconnect VoIP networks with mobile networks. It offers up to 12 GSM channels.

More information on http://www.topex.ro/en/products/voibridge_-_voip-gsm_gateway


TOPEX VoIP to GSM gateway - QUTEX

It offers you the possibility to terminate calls from VoIP network to cellular networks and from cellular networks to VoIP.
It supports from 2 to 8 GSM calls and 8 VoIP calls and very important, it is working both on SIP and H323 protocols.

More information on http://www.topex.ro/en/products/qutex_-_media_gateway
?catID=4#prodId38|http://www.topex.ro/en/products/qutex_-_media_gateway]
or at sorin.blideran@topex.ro



TOPEX VoIP to 30 GSM - multiACCESS

multiACCESS is a versatile GSM gateway with support of VoIP communication by tis integrated VoIP card which supports both SIP and H323. It is a flexible and reliable solution designed for industrial use with support of all kind of interfaces like FXS/FXO/CDMA.

The equipment is also available for UMTS networks.

More information on http://www.topex.ro
or by email: sorin.blideran@topex.ro



TOPEX UMTS - VoIP gateway

UMTS gateway with bidirectional connection between one E1 trunk and up to 30 UMTS channels, ideal for cutting your telephonic bills. T he fixed-to-mobile calls are converted into mobile-to-mobile calls with substantially reduced cost.
Using the VoIP card this equipment act as VoIP to UMTS gateway. It features 30 VoIP channels to 30 UMTS channels. This way the equipment is a perfect solution for companies with a considerable international traffic. Instead of 2 equipments (VoIP gateway, UMTS gateway), you only need one equipment which includes the features of the others two.

http://www.topex.ro/en/products/multiaccess_-_pri_gsm_gateway

• Iridiacom Limited - Serving the UK and Ireland. Resellers call for pricing!

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Tresto

http://www.gsmtermination.com/features.html#H323Gw

Tresto develops NGN softswitches and VoIP GSM gateways for Alternative Telecom Service Providers and VoIP Operators offering GSM Softswitch, IP-Centrex and Class 5 services.



Tresto Voip - GSM Gateway Features:

• 8 channel gateway,
• up to 64 simcard/gateway
• Dual Band (900 / 1800 MHz or 850 / 1900 MHz)
• Half rate, full rate, enhanced full rate
• Sending and receiving SMS and USSD messages
• DTMF send/receive
• CLI restriction
• GSM field-strenght monitoring
• Integrated antenna splitter
• H323, SIP
• Voice Codec: g.729, g.723.1, g.711 alaw / ulaw
• Voice Activity Detection/Silence Suppression
• Adaptive de-jitter buffer
• Least Cost Routing
• Load Balancing
• CDR Generation
• Failovering on SIM and GSM channel
• Inteligent routing (Based on Direction/Time,Priorities,Quality,Prices,etc)
• Load Balancing, Least Cost Routing, Priority routing
• Reporting and alerting in emails and SMS's (error/system status/profit reports)
• Supported VOIP protocolls are SIP and H.323
• CallBack, Bulk SMS, Redial, Conferencing and other Class5 features
• Support all kinds of networks: PPPoE (ADSL), ISDN, VPN.
• Blacklists Whitelists, automatic traffic analysis
• Automatic failowering and rebalacing based on quality and prices

Physical Interfaces:
• Ethernet 10/100 Base-T
• FME antenna connector
• Console access

Accessories:
• 900/1800 MHz omni-directional antenna
• antenna cable
• monitor cable




http://www.gsmtermination.com/solutions.html

TRESTO Systems has developed an innovative and sophisticated soft switch solution. The company's soft switch solution has unique capabilities on multiple functional areas. The softswitch contains Class 5 features, IP Centrex solution and GSM Features. By inlcuding the GSM features in this Softswtich, Tresto has developed the first GSM Softswitch.

Tresto GSM Softswitch Features:

• Distributed absolutely fault tolerant system
• Multiple signaling protocoll support
• Load distribution between the operational channels
• No hard limit on the number of simultaneous calls
• High Availability
• No additional Tresto hardware required
• Equipment management
• Channel management
• Sim card management
• Automatic recharge
• Access Control Lists
• Routing
• Billing
• Exploits almost any SIM tariff model
• Number translation
• Support for different currency (embedded currency converter)
• Protocoll encryption
• Media proxy
• Automatic time syncronizations
• H.323/SIP Gateway Topology Hiding
• Embedded firewall
• Enchanged Security (autotomatic detection of flood attacks)
• Web GUI for endusers
• Encrypted communications
• External system supervisor service (email and sms alerts, watchdog can restart failed subsystems)

More information: http://www.gsmtermination.com/

-==-

VIERLING

See detailed VIERLING VoIP GSM gateways product overview:
English: http://www.vierling.de/www_vierling/gsm_gateways_overview-en_135_130_11_f.htm
German: http://www.vierling.de/www_vierling/gsm_gateways_overview-dt_38_37_11_f.htm

The VIERLING ECOTEL VoIP provides reliable connections between IP, S0 and GSM. The number of GSM channels is scalable in steps of two up to a total of eight. Multiple ECOTEL VoIP are cascadeable. The GSM modules are compatible with all types of GSM networks (GSM 850/900/1800/1900/R).

Potential applications range from connection to an existing PBX system or integration into an existing VoIP infrastructure all the way through to distributed installations with multiple gateways and common configuration software. Integrated with a central SIM card server ECOTEL VoIP offers carrier-class standard.

For multiple application examples see:
English: http://www.vierling.de/www_vierling/ecotel_voip-en_779_370_6_f.htm?query=
German: http://www.vierling.de/www_vierling/ecotel_voip-dt_402_369_6_f.htm?query=

ECOTEL VoIP allows intelligent SIM management: The gateway can switch from one SIM to another so the user is able to use up to eight SIM cards per channel (64 SIM cards within each ECOTEL VoIP). Additionally, it is possible to integrate ECOTEL VoIP with the ECOTEL VTM SCS (SIM Card Server) using thousands of SIM cards of different carriers. VIERLING also offers the ECOTEL VTM pro and ECOTEL VTM flex multichannel gateways with 32 GSM channels. It is possible to cascade multiple VTM gateways. ECOTEL VoIP offers the broadest functionality, currently available on the market.

- ECOTEL VoIP for small- and medium-sized companies
- ECOTEL VoIP as distributed solution with SIM card server for enterprise applications and carrier-class applications
- ECOTEL VTM pro or VTM flex for enterprise applications and carrier-class applications
- Gateways for companies with large volumes of mobile radio traffic
- Cost-effective connections (least cost routing, LCR)
- Intelligent SIM management and SIM server integration
- Connection of mobile sites (office containers, ships, RVs, emergency vehicles)
- Last mile concepts
- Internet via GSM/GPRS


VIERLING ECOTEL VoIP



Specifications VIERLING ECOTEL VoIP

Signaling: SIP
GSM frequencies: 850/900/1800/1900/R
Codecs: G.711 PCM for 64 kbit/sG.726 and G.727 E-ADPCM from 16 to 40 kbit/sG.729/Annex A CS-ACELP for 8 kbit/sG.723.1 (optional) MP-MLQ/ACELP for 6.3/5.3 kbit/s
Interfaces: LAN 10/100 Base-T, RJ45, USB 1.1 B type
GSM channels: 2 to 8
VoIP channels: 8
ISDN BRI ports: 2
Remote access: yes
Call detail records (CDR): yes
SMS messages: yes
Callback functions: yes
Internet via GPRS: yes
Configuration: LAN, USB, GSM
Power supply: 100 � 290 V AC, 50 � 60 Hz, 15 V DC
Dimensions: 255 x 185 x 63 mm


VIERLING ECOTEL VTM flex


Further information on the VIERLING ECOTEL GSM gateway portfolio:
English: http://www.vierling-group.com
German: http://www.vierling-group.de

English: alternative product ECOTEL ISDN http://www.vierling.de/www_vierling/mobile_communication-en_134_133_11_f.htm
German: alterantive product ECOTEL ISDN http://www.vierling.de/www_vierling/mobile_communication-dt_37_415_11_f.htm

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Voiptech

VOIP GSM GATEWAY - KTS GSM VoIP gate

The http://www.voiptech.ru 1 GSM VoIP ports gate. The VoIP GSM Gateway provide a low cost fixed-to-mobile or mobile-to-fixed telephone calls for small businesses.

KTS GSM VoIP main features:

1 GSM (GSM900/GSM1800) / VoIP ports
full support SIP protocol
All codecs support (G.711 a/u law, G.723.1 (6,3/5.3 кбит/с), G.726, G.729 a/b, GSM)
Hotline, IVR
VAD/CNG/AGC/AEG
WEB management and control
Use the special configuration utility
Firmware upgrade: Web, FTP, TFTP, KTS-Tools
protocol support : IEEE: 802.3 10 BaseT, ARP, TCP, IP, ICMP, HTTP, DHCP, DNS, UDP, RTP, Telnet

Other information:
KTS GSM VoIP Gateway
KTS SAT VoIP Gateway
KTS FXS VoIP Gateway

FOR MORE INFORMATION:
www.voiptech.ru
sales@voiptech.ru

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VoiSmart

http://www.voismart.com/
VoiSmart GSM IP Voice Gateway

VoiSmart Voice Gateway is an Asterisk-based IP Gateway which allows you to transform fixed-to-mobile into mobile-to-mobile traffic in the simplest and most efficient way. It's a true black box, which can be reached as a SIP client. It allows
multiple RX/TX (from 4 to 32 simultaneously active GSM channels, in different case configurations).
No extra hardware to be added, no installation effort, no technical skills: a real plug-and-play device, designed to make your business immediately running.
Think of it as a standard SIP client, and integration is done. LCR and smart routing of incoming calls are immediately available, as well as all its powerful features.

Professional Asterisk for Professionals

Powerful and versatile, think of it as a pure node in the network. VoiSmart IP GSM Gateway is the building brick on which you will rely to implement high profile solutions, as well as the next door to the world of message communication.
A perfect match for Asterisk architectures, but perfect to be integrated in non-Asterisk environment as well.

Features at a glance

-SIP / IAX supported
-All in one: no extra HW or SW required
-Easy to integrate
-Easy to install
-Simultaneous operation on multiple channels (4 to 32)
-LCR (Least Cost Routing) full support
-Incoming calls routing (AutoCLIP, routes unanswered calls to the right IP phone)
-Multicodec (G.711v, G.711a, ILBC, GSM)
-Echo cancellation
-High efficiency thanks to multiple channels operation

VoiSmart vGSM board

is the first 100% Asterisk compatible PCI bus card, for multiple channel (up to four) GSM communication management. This board allows systems integrators to transform highly priced fixed to mobile voice traffic into a much cheaper mobile to mobile call management, with considerable savings.

No external gateways needed.

100% compatible with EU and US GSM networks (900/1800/1900 MHz) and with Brazil and Japan networks. SMS are handled in native modality, thanks to a new Asterisk application for messages TX and RX, available free together with this board.

VoiSmart Quad GSM board is supplied with GPL open source drivers, it is compatible with all Linux / Asterisk systems, managing kernel versions from 2.6.x onwards.

The antenna is external, and included in the package.

You can find latest news and docs on VoiSmart Open Source website

Drivers for this card (kernel space), providing 4 TTY interfaces for the signalling section and a channel driver for the popular Asterisk OpenSource PBX, are available.

-Bus Type: PCI 2.2 compliant
-IRQ Request levels: Allocated by PCI BIOS
-IO base addr (hex): Allocated by PCI BIOS
-Dimensions in mm: 154mm x 107mm (PCB), 164mm x 129mm including SIM holder, bracket, connectors
-Plug & Play: yes
-External Interface: 4 x SMA/F 50 Ohm antenna connectors
-Antennas: included
-Radio Interface: tri band GSM (900/1800/1900), phase 2/2+ compliant, class: Small MS, TX Power: Class 4 (2W) at EGSM 900 Class 1 (1W) at GSM 1800 and GSM 1900
-SMS: MT, MO, CB, Text and PDU mode
-SIM Interface: Supported SIM card: 3V
-Performance: 4 channels GSM (900/1800/1900) voice interface

Note: Please, be carefull if you plan to buy this card. The software driver for linux, included in "visdn" project, does not support other cologne-chip cards in the same system (hfc-pci y hfc-e1). If you plan to use this card in a system with other simple BRI card or E1 card that uses mISDN or vISDN drivers, this card is incompatible. For more information read visdn mail list archive at: https://mailman.uli.it/pipermail/visdn-hackers/2007-July/thread.html

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WaveTramp Technologies, Inc

Technical specification and manuals are available at http://www.wavetramp.com/products-en.php.
You have another option in choosing your fixed wireless solution: WaveTramp Technologies. WaveTramp design, develop, manufacture and market fixed wireless products targeting china and africa market. Please check www.wavetramp.com for more details.

Contact Patrick Liu
WaveTramp Technologies,Inc
www.wavetramp.com
sales@wavetramp.com
China office: +86-755-26014880
China cell: +86-13005436876
USA phone: +1-781-218-3078


More Information

- PORTech from Taiwan (MV-370/MV-372 VoIP GSM / CDMA Gateway,DMT E1 GSM Channel Bank,MT-350/MT-354/MT-358 GSM Fixed Wireless Terminal http://www.portech.com.tw
- QuesCom from France (QuesCom 400 IP/GSM was one of the earliest products on the market)
- NovaTec from Germany (NovaTec are the pioneers of GSM Gateways) and offer an all in one solution IP/GSM/ISDN no extra equipment needed.
- VIERLING from Germany (offers high quality and highly sophisticated SIM management enabling maximum cost reduction)
- 2N from Czechia (VoiceBlue also was one of the earliest VOIP GSM Gateways on the market; offers SIP AND H.323)
- UK Stockists of VoIP GSM Gateways for Asterisk and VoIP PBX - http://www.westlake.co.uk:Westlake Communications for MV-370, MV-372 and Telecom FM OneStream (SIP & H.323)
- UK OnLine Store of VoIP GSM Gateway http://www.iptelephones.co.uk:Portech MV-370 and MV-372 SIP GSM Gateways in stock for next day delivery in the UK
- GSMDATA - Online Store for VoIP GSM Gateways and GSM Terminals http://www.gsmdata.co.uk
- Discount VoIP GSM Gateways in the UK http://www.discountphonesystems.co.uk:discount VoIP gsm gateways for asterisk and voip pbs phone sytems - UK stockists for Portech MV-370 and MV-372 and the new Telecom FM Onestream Dual SIM VoIP GSM Gateway with PSTN connections

• Asterisk Siemens M20: Thoughts on using a M20 GSM box with Asterisk
• XLink Cell Phone Gateway: Use your cell phone and an ATA (e.g. Linksys SPA3102) to create a cheap 1-line GSM gateway
• GSM
• 3G UMTS
• CDMA

---

See also: Asterisk Connecting to the Cellular Network

CellSocket

Here's a GSM to analog line adapter that is pretty reasonably priced. You should be able to connect your PBX or regular phone to it and make calls out through your GSM provider

- http://www.cellsocket.com/index.htm

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JUNGHANS PCI GSM CARD
The Junghanns.NET GSM PCI series cards provide scalable connectivity to GSM networks for your Linux machine. It comes with fully GPLed drivers for the Linux 2.4.X and 2.6.X kernels. Multiple Junghanns.NET GSM PCI series cards can be interconnected over an external PCM bus. The card's active channel switching capability (to bridge voice channels on the card without latency and not using the host CPU) is also working over the external PCM bus (e.g. voice channels on different cards can be actively switched). The Junghanns.NET GSM PCI series cards can be connected to any other Junghanns.NET card to build a real TDM switched PBX.
http://www.junghanns.net/en/GSM-PCI_produkt.html

EUR900 for PCI card with 1 module, EUR1200 for card with 2 modules.

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GP-710 Bluetooth Mobile SIP VoIP Gateway For Global Systems

GP-710 is a revolution and innovation product.
It Integrated VoIP and 2.4G Bluetooth technical in a device,
this not only offered high cost-effectiveness in mobile communication for mobile user,
but also offered very flexibility in re-building system.

Feature:

1. Use Bluetooth Technology, GSM.WCDMA VoIP don't care Bands & Systems.
2. VoIP and Bluetooth Mobile fully integration.
3. Compatible with SIP RFC543, RFC3261.
4. Support UPLINK , DOWNLINK Routining.
5. Support Voice report and Setting IP function
6. Offered Web Site for inquire or setting.
7. Support SIP PROXY, or point to point application.
8. Offered one stage ,two stage free dialing and call transfer function.
9. Bluetooth with auto pairing and auto searching function.
10.QoS and Digital Transmit.
11.Could use with GSM/CDMA/3G/3.5G various Bluetooth mobile phone.
12.Support invite 183.

Application Diagram:

Specification:

VoIP
Web Browser
IVR Interface
Uplink Route Setting
Downlink Setting
1-Stage, Dialout Called, Free Dial

Network Protocol:

SIP v1 (RFC2543), v2(RFC3261)
IP/TCP/UDP/RTP/RTCP
IP/ICMP/ARP/RARP/SNTP
DHCP Client/ PPPoE Client
DNS Client

NAT Traversal:

STUN

Voice Quality:

VAD: Voice activity detection
CNG: Comfortable noise generator
LEC: Line echo canceller
Packet Loss Compensation
Adaptive Jitter Buffer

Bluetooth:

Bluetooth Specification V2.
Carrier Frequency 2400MHz to 2483.5MHz( USA , Spain ,France)
Modulation Method GFSK,1Mbps,0.5BT Gaussian
Output level, class 2
10 meters working range

Package

Bluetooth VoIP Gateway：147mm X 108mm X 27mm
2.4GHz Antenna
Power Supply：Input 100-240VAC 50-60Hz Output 12VDC 1000mA
1.8m Network Cable


Gempro Technology Inc.
E-mail: service@gempro.com.tw
webs: http://www.gempro.com.tw

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COST-SAVING PROGRAM

Gempro GP-510 Fixed Wireless (PBX) Bluetooth Mobile Terminal for GSM/CDMA/WCDMA & SKYPE

GP-510 uses Bluetooth to connect Bluetooth mobile phone with common telephone or PBX.It makes immediately PBX has a trunk by mobile phone to transmit.It is not only far away the electromagnetic wave, but also unnecessary to change user’s habit of using the table phone. It is a trunk of mobile to mobile, and it also saves expenses.

Feature

1.PBX/phone Bluetooth Mobile Voice is transmitted on the trunk by mobile phone.
2.GP-510 pairs to Bluetooth mobile phone to use directly, it doesn’t relate to telecommunication provider and frequency.
3.Just use phone to dial, it can be far away the electromagnetic wave.
4.Speed Dial Function.
5.The calling will pass to PSTN automatically when power down, mobile phone is unpaired or out of range.
6.Volume of digitization and it adjusts volume and gains.
7.Set local code and automatically add code, and user isn’t necessary to change habit of using the table phone.
8.Bluetooth has functions of pairing and searching automatically.
9.Transmit high quality voice.
10.It is compatible with GSM/CDMA/3G/3.5G/WCDMA Bluetooth mobile phone.
11.Phone/PBX dial out and answer Skype. (already pair to Bluetooth Dongle)as convenient as using common telephone.
12.Use Bluetooth technology, so the connection doesn’t have any wires.
13.Phone(PBX) can dial out and answer Skype. (already pair to Bluetooth Dongle)
14.100 sets phone book Speed Dial & memorize
15.Don't change the dialing habit to skype out

Specification

Metering Signals: 12KHz, 16KHz,Polarity Reversal
A/D ,D/A : G.711:A-law or u-law PCM Mode
DTMF : Detect & Generator
Caller ID Format: DTMF
Tone Programmable: DIAL Tone & Busy Tone
Power : AC110/220V 50/60HZ

Bluetooth

Bluetooth Specification V2.
Carrier Frequency 2400MHz to 2483.5MHz( USA , Spain ,France)
Modulation Method GFSK,1Mbps,0.5BT Gaussian
Output level, class 2
10 meters working range

PACKAGE

Bluetooth VoIP Gateway：147mm X 108mm X 27mm
2.4GHz Antenna
Wire*2
Power Supply：Input 100-240VAC 50-60Hz Output 12VDC 1000mA





Gempro Technology Inc.
TEL:886-4-22292128
FAX:886-4-22290413
E-mail:service@gempro.com.tw
webs: http://www.gempro.com.tw

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Telecom FM Onestream G

The Telecom FM Onestream G is a Dual SIM, Dual Band VoIP GSM Gateway that is ideally used in conjunction with a VoIP PBX as a SIP to GSM Gateway. The Onestream G is the first in the range of VoIP GSM products to be launched in the UK from Telecom FM, who have traditionally supplied Analogue and ISDN2 Fixed Cellular Terminals. The range will quickly expand to offer the Onestream GFX and The Onestream GBRI2 and Onestream GBRI4 multi-purpose VoIP GSM Gateway's. The Onestream GFX will provide VoIP and Dual SIM termination options, together with 6 x FXS and 6 x FXO connections so that it can be connected in-line between the analogue trunk ports of a PBX and the PSTN telephone lines. The Onestream GFX will then be able to route calls from the PBX via GSM, PSTN or VoIP. The Onestream GBRI units will be dual 2 or dual 4 port ISDN2e, again designed to sit in-line between the PBX and the ISDN telephone lines and provide multiple routes over GSM, ISDN and VoIP.

The Onestream range of gateways primarily uses SIP as it's main VoIP Protocol, but can also be set up for H.323, making this one of the lowest priced, multi-functional and easiest to use VoIP GSM Gateways on the market.

The price for the Entry Level Onestream G - Dual SIM VoIp GSM Gateway is just £395 and is available from stock in the UK from GSMSAVE " where a full spec and installation manual are available for additional information.

Where to buy

JmcSoft (IRL) - UK & Ireland TelecomFM distribution partners
Voip4u.ie - For all your Voip needs - secure safe online shopping

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Soundwin Network Inc.

http://www.soundwin.com/
Soundwin GSM Voice Gateway
The V100 Quad-Band GSM over VoIP gateway has been designed for user to make calls and receive calls from a cellular phone via the internet using VoIP (SIP/H.323).


Target Application:

• GSM Connectivity for IP PBX /Asterisk
• Mobile IP PBX
• GSM VoIP Gateway
• GSM Callback Services (Optional)

Features:

• FXS interface (for analog phone or PBX CO line)
• Calls from cellular over VoIP
• Calls from GSM network to the V100 unit provides a VoIP dial tone
• SMS Server for SMS sending & receiving
• Dialed number restriction, evaluation and modification
• Easy & comfortable maintenance, configuration and upgrade
• Block / Allow Policy for termination / origination

IP Specifications:

• H.323 v2/v3/v4 and SIP (RFC 3261) , SDP (RFC 2327), Symmetric RTP, STUN (RFC3489), ENUM (RFC 2916), RTP Payload for DTMF Digits (RFC2833), Outbound Proxy Support.
• Voice Codec: G.711(a-law /μ-law), G.729 AB, G.723 (6.3 Kbps / 5.3Kbps)
• WAN: Support PPPoE client, DHCP client, Fix IP Address, DDNS client

Call Features:

• Voice channels status display
• Direct Dialing Mode : peer to peer call (support IP Address Call or Domain Name Call)
• Register Call Mode : register to SIP Proxy Server or register to H.323 Gatekeeper
• DSP Adjustable volume : - 9 db ~ 9 db
• Silence Compression / VAD
• Auto Dial for speed
• Dynamic Jitter Buffer
• Hot-Line and Warm-Line Support

Configuration & Management:

• Web-based Graphical User Interface
• Remote management over the IP Network
• FTP firmware upgrade
• Syslog client support
• Support Management IP

General Specification

• Compatible with GSM networks (900/1800/850/1900 MHz)
• SIM: supports SIM card (3V)
• 1 WAN port, 1 FXS port
• Radio interface: Quad-Band EGSM 900/1800/850/1900

Contact: sales@soundwin.com

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saving cost project Gempro GP-712 WCDMA/UMTS/GSM VoIP Gateway (2 voice channels) for all systems

GP-712 is a revolution and innovation product.It Integrated VoIP and 2.4G Bluetooth technical in a device, this not only offered high cost-effectiveness in mobile communication for mobile user but also offered very flexibility in re-building system.

Feature
1.VoIP and Bluetooth Mobile fully integration.
2.Compatible with SIP RFC543, RFC3261.
3.Support UPLINK , DOWNLINK Routing.
4.Support VOICE REPORT IP function.
5.Offered Website for inquire or setting.
6.Support SIP Proxy, or point to point application.
7.Offered one stage ,two stage free dialing and call transfer function.
8.Bluetooth with auto pairing and auto searching function.
9.QoS and Digital Transmit.
10.Could use with GSM/CDMA/3G/UMTS various Bluetooth mobile phone.