VoIP GSM Gateways

Sep. 02, 2024

VoIP GSM Gateways

What&#;s a VoIP GSM Gateway?

A VoIP GSM Gateway enables direct routing between IP, digital, analogue 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).

If you want to learn more, please visit our website chinaskyline.

Through the least cost routing the gateways select the most cost-effective 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?

Page Contents

2N TELEKOMUNIKACE

2-channel version:

4-channel version:

Follower of 2N® VoiceBlue Lite offers you brand new hardware, intuitive setting via web interface, PoE (Power over Ethernet), SMS sending and receiving directly from the browser and connection of remote SIMs.

2N® VoiceBlue Next is CISCO and Elastix certified VoIP GSM gateway with perfect match for any SIP based IP PBX. You can check the list of supported and tested IP PBX system here.

Features:

2 or 4 channel GSM/UMTS gateway
VoIP: SIP, G.711a/u, G.729
Support of external number portability database
Virtual SIM support &#; Remote SIM management
Elimination of roaming fees with Callback
Employee mobility &#; Split the call to desk and mobile
Power over Ethernet (PoE)
Number portability tone detection
Significant reduction in costs (LCR)
Free minutes counter with up 4 minute packages per SIM
Intelligent incoming call routing
Worldwide use (GSM 850/900//MHz, UMTS (850//MHz)
Top voice quality
DISA with voice navigation, user defined voice message for incoming calls

Contact us for more details

Acecom Teles

Teles IGate GSM Gateway E1 Channel Bank 32 SIMs (Only for Sale in AsiaPacific)

Price around USD 13K includes:

iGATE GSM 32 VoIP 32 channel mobile gateway (UMTS MHz, GSM 850/900// MHz), 32 VoIP channels, 4U 19&#;&#; housing, antenna splitter 8:2
iGATE SIM4-carrier
SIM card carrier for 4 SIM cards &#; 1 SIM per channel&#;
IGATE attena Omni-Directional

Overview

VoIP gateways designed for carrier and enterprise market
Support for digital and analog ports
Available in 2 to 180 channels configuration
Wide selection of signaling protocols including ISDN, SS7, CAS/R2, SIP, H.323, MGCP
Toll-grade voice quality
Support for T.38 and G.711 fax transport
Field proven interoperability with large number of equipment providers
Minimized OPEX with TELES remote management and diagnostic system
Systems installed worldwide

The iGATE connects carrier networks to mobile networks and enhances the functionality of a corporate PBX.

Converts fixed-to-mobile calls into cost-saving mobile-to-mobile calls
Integrated Mobile Number Portability ends costly cross-network calls
Integrated VoIP gateway
PSTN support (PRI and BRI, SS7)
Powerful Least Cost Routing
Seamless integration into existing telecom infrastructures
Fixed network backup via mobile
Fully vGATE SIM Server compatible
Field-proven Mobile Gateways

Invest in outstanding value and benefit from cost saving mobile-to-mobile calls. The premium suite of features ensures complete compatibility with your current carrier equipment and total interoperability with all third-party VoIP equipment.

Save time and effort during installation as a result of built-in, designed compatibility. The TELES iGATE and vGATE SIM Server are the only team of products designed from scratch to interoperate. You avoid complex and unstable third-party to third-party gateway to SIM server configurations.

Flexible and Versatile Connectivity
Benefit from outstanding versatility. The iGATE includes a VoIP gateway with up to 32 VoIP channels, 32 mobile channels, 2 E1/&#;T1 interfaces, and a LAN connection. Independent of the network type, the iGATE can connect to different local or remote VoIP and mobile networks &#; including GSM, 3G UMTS and CDMA &#; and can switch between each single incoming to any outgoing channel.

Least Cost Routing
Save money on every call. iGATE&#;s LCR (Least Cost Routing) features automatically route calls from your PBX &#; via the least expensive route &#; to each destination, whether that is over a standard fixed-line network, a mobile network, or over the Internet. Plus, LCR combines with other superior call-routing capabilities that are based on, for example, called party, CLIP/CLIR, time-of-day, time quota, service quality, and SIM card profile.

Guaranteed Voice Quality
Ensure superior voice quality. In addition to all the standard VoIP features &#; like echo cancellation, silence suppression and comfort noise &#; iGATE&#;s premium feature set includes the best available DSPs, traffic shaping, and multi-level fallback. A variety of voice speech compression codecs reduce bandwidth need and RTP multiplexing technology enables even further bandwidth reduction.

Contact: [ protected]
Contact: +

What is ANTRAX?

ANTRAX solution is a complex of software and hardware which is designed specially for the termination of voice traffic from VoIP (Voice over IP) to GSM networks (networks of mobile operators). Due to the module-based structure, flexible configuration which allows the product to meet any needs and requirements, ANTRAX solution can be implemented for effective calls termination either in large or in small volumes.

The Flames Group SIA company is a manufacturer and official supplier of ANTRAX &#; equipment for voice traffic termination ANTRAX. ANTRAX &#; a unique hardware and software complex for traffic from VoIP (Voice over IP) networks into GSM-network (mobile network operators). Due to the modular structure of the universal solution, it will be the best choice for GSM termination in any volume.

Our ANTRAX 2U solution was specially created for those, who values the simplicity and compactness. It already has on-board PC, which helps the customer to avoid the problems connected with obtaining and placing the separate one. Having 2 on-board GSM modules, it still provides the users with three additional universal slots, which can be filled by 2-6 GSM modules or 1-60 SIM cards.

FEATURES

Case 3 universal slots for placing sim-cards or gsm-boards;
On-board computer;
Up to 3 additional SIM-boards (with ability to accommodate up to 20 SIM cards on each board);
Up to 3 additional GSM-boards (with 2 GSM-channels per board).
Secure VPN connection
As a gift to a set ANTRAX 2U, our campaign provides two built-in GSM module!

PC Specification

CPU: lntel Atom D
RAM: 4 GB
Hard drive: 320 GB
Ethernet: 1 x Ethernet 10/100 Base-T RJ-45

For more information about Antrax 2U mini rack, please visit our website
http://en.antrax.mobi/products/minirack

VoIP GSM-gateway (GSM bridge) &#; modular unit that transmits voice traffic from Internet to mobile network and vice versa.
This gsm gateway is a unique solution to do professional voice call termination and significantly reduce the cost of international calls and make the price equal to local tariff.
The client can choose individual structure of gateway, equipping it with the required number of GSM channels (2 to 30). This feature also means the flexibility in GSM gateway price setting.
The modular architecture allows ANTRAX users to place SIM cards apart from GSM gateways and connect them via IP. This feature provides a huge advantage in terms of safety and removes limitations which could be applied to the topology of your GSM termination system.

VoIP GSM-gateway (GSM bridge) &#; modular unit that transmits voice traffic from Internet to mobile network and vice versa.This gsm gateway is a unique solution to do professional voice call termination and significantly reduce the cost of international calls and make the price equal to local tariff.The client can choose individual structure of gateway, equipping it with the required number of GSM channels (2 to 30). This feature also means the flexibility in GSM gateway price setting.The modular architecture allows ANTRAX users to place SIM cards apart from GSM gateways and connect them via IP. This feature provides a huge advantage in terms of safety and removes limitations which could be applied to the topology of your GSM termination system.

Features

Quad band GSM frequency bands;
IMEI change
Hot swap boards
Provides 1-15 hot-swappable boards providing 2-30 GSM channels
ACD modules giving not less than 4 minutes;
Possibility of SMS termination.

For more information about Antrax gsm gateway, please visit our website http://en.antrax.mobi/products/gsmgateway

SIM-box (also called SIM-bank or SIM card reader) &#; is one of hardware modules of ANTRAX Solution for GSM termination.
SIM box allows you to install and manage from 20 to 300 SIM cards of different mobile operators that enables work of several GSM gateways placed in different locations. You can keep your SIM cards away from the GSM gateway. It allows you to attach a separate SIM card for GSM/VoIP gateway module, enabling the same card to call from different points thereby reducing the risk of blocking.

SIM-box (also called SIM-bank or SIM card reader) &#; is one of hardware modules of ANTRAX Solution for GSM termination.allows you to install and manageof different mobile operators that enables work of several GSM gateways placed in different locations. You can keep your SIM cards away from the GSM gateway. It allows you to attach a separate SIM card for GSM/VoIP gateway module, enabling the same card to call from different points thereby reducing the risk of blocking.

Features

Sub-rack with fifteen universal slots which provides power and cool the system
1-15 SIM-boards that comprises 20 SIM cards
Operate the gateway without Sim-card inside it;
Store up to 300 SIM-cards of different operators in one place;
Each client has opportunity to choose individual structure of SIM box and use from 300 SIM cards during GSM termination.
It is possible to combine GSM and SIM boards in one rack, so it is gsm &#; sim box.

For more information about Antrax SIM BOX, please visit our website
http://en.antrax.mobi/products/simbox

Our Sim Server provides the ability to:

Performs complete management and control all elements of the system;
Provides automatic control of the account,
Complete simulation of the real behavior of the subscriber;
Protect from automatic calls from operator.

For more information about Antrax SIM-server, please visit our website
http://en.antrax.mobi/products/simserver

Graphical User Interface provides the ability to:

Monitors and controls the entire system in a real time
Maintain Drag and Drop function;
View real-time performance and CDR calls ASR, ACD;
Combining the SIM card and channels in certain GSM and SIM groups.

For more information about Antrax, please visit our website http://www.antrax.mobi

The ABILIS-GSM Box is an add-on for the ABILIS Router Gateway which provides an interface between the GSM and the fixed (private/public) network. The GSM-BOX produces high savings and enables personnel to be mobile while keeping themselves linked to the company&#;s network at zero tariff.

Cost Saving Effect:

Any call passing through the Abilis can be routed to the GSM networks and any call reaching the GSM-BOX from the GSM network can be routed to any destination through the Abilis network.
Currently, most of the GSM carriers offer zero or almost zero tariffes for calls within a group of SIMs. Thanks to the Abilis GSM-BOX, mobile phones can be considered as &#;mobile extensions&#; of the companies system. Additionally, the ABILIS &#;Least Cost Router&#; permits to select the cheapest route for any call.

USB interface:

The GSM-BOX connects to the Abilis by means of a USB interface, thus requiring no additional telecom port in the ABILIS router and preservinging digital voice quality.

Dial Through:

Once connected to the GSM-BOX and identified as an authorised user, the caller is invited &#; by means of a customizable message- to dial the desired destination, which can be local or remote. Calls entering from the public network can be automatically switched to the &#;mobile extension&#; if the desk one does not answer within a given time.

Other features:

Two simultaneous channels
Works even without SIM cards !
High-fidelity voice
Call-back
Unlimited Least Cost Routing (LCR)
DTMF dial-in with voice message
Call data records
SMS to
SMA connector with SMA detachable antennas

PRODUCT SUMMARY

The BF400XGSM is used for small enterprises who need up to 4 concurrent GSM calls. It can be used for mobile convergence, LCR and SMS.
The berofix Gateways are compatible and tested with numerous SIP Server&#;s like 3CX, Asterisk, Freeswitch, FreePBX, Trixbox, Elastix, Gemeinschaft and many more

Dinstar 1/4/8/16/32-port GSM/CDMA VoIP Gateways DWG series with SIMBank & SIM Cloud

PRODUCT SUMMARY

The DWG series SIP/GSM Gateways from Dinstar are a cost-effective SIP/GSM gateway for SOHO, SMEs, service provider and system integrators. The Dinstar GSM/VoIP gateways allow a SIP based IP PBX system or VoIP line to be connected directly to the GSM cellular network, which can dramatically lower the cost of mobile calls by taking advantage of lower mobile-to-mobile tariffs. The DWG gateways are Quad-band GSM VoIP gateway that support 1-8 GSM channels / SIM cards. The DWG gateways are fully compatible with Asterisk® based IP PBXs like Elastix®, trixbox®, and AsteriskNOW®, as well as other Open Source and proprietary SIP based IP PBXs, Switch, IVR, and VoIP gateway applications.

TARGET APPLICATIONS

SIP/GSM gateway for SIP based IP PBX (e.g. Asterisk®) or VoIP line
Back up solution in case of a PSTN/VoIP failure
Mobile IP PBX (e.g. for trade shows, temporary offices, disaster recovery, etc.)
SMS gateway for batch sending/receiving text messages (e.g. appointment reminders, customer order updates, remote/field worker communications, etc.)
GSM callback services

TECHNICAL SUMMARY

Voice codecs supported include G.723.1, G.729A/B, G.711 (a-law and u-law)
Support for Fixed IP, DHCP, Automated NAT traversal using IETF STUN
Protocols supported include SIP RFC, TCP/UDP/IP, RTP/RTCP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TELNET, PPPoE, STUN
Quad-Band GSM 850/900// MHz
Send/receive batch SMS up to 300 characters, support for AT commands
Unstructured Supplementary Services Data (USSD) support
Adjustable TX/RX gain: -6db~+6db
Offhook Auto-Dial Hotline service
Built-in PPPoE client for establishing DSL link connection with ISP
DTMF Signaling via RFC or SIP Info
LEDs provide operational status information
Support for external antenna
PIN support for locked SIM cards
Enhanced (Carrier Grade) Echo Cancellation: ITU Rec. G 168, up to 128 ms tail size
Adaptive Jitter Buffer, Packet loss concealment (PLC), Comfort noise generation (CNG) ,Voice activity detection (VAD)
HTTP Web configuration support, Auto Provisioning support
Certificates: CE, FCC, ROHS
How to set elastix/asterisk with DWG GSM/CDMA gateway, check from: https://www.dinstar.com/service/Training.aspx

MODELS

DWG-1G &#; 1 Channel VoIP GSM Gateway

1 External hot-plug SIM card slot for easy access
LAN interface: 1 x RJ45 10/100Mbps autosensing
WAN interface: 1 x RJ45 10/100Mbps autosensing
Weight: Gateway only &#; 185g, Package as shipped &#; 0.5Kg
Dimensions (L x W x H): 112mm x 76mm x 24mm
Universal Switching Power Adaptor: Input: 100-240VAC; 50-60 Hz, Output: 12VDC, mA
Power Consumption: 5W

DWG200E-8G &#; 4/8 Channel VoIP GSM Gateway

Rack mountable 1U chassis
4 hot-plug SIM card slots with external cover for easy access
LAN interface: 3 x RJ45 10/100Mbps autosensing
WAN interface: 1 x RJ45 10/100Mbps autosensing
Weight: Gateway only &#; 4Kg, Package as shipped 4.85Kg
Dimensions (W x D x H): 440mm x 305mm x 44mm
Power: 100-240VAC; 50-60 Hz
Power Consumption: 30W

DWGF-16G &#; 8/16 Channel VoIP GSM Gateway

Rack mountable 1U chassis
8 hot-plug SIM card slots with external cover for easy access
LAN interface: 3 x RJ45 10/100Mbps autosensing
WAN interface: 1 x RJ45 10/100Mbps autosensing
Weight: Gateway only &#; 4.25Kg, Package as shipped 5.1Kg
Dimensions (W x D x H): 440mm x 305mm x 44mm
Power: 100-240VAC; 50-60 Hz
Power Consumption: 30W

For More Information

Home page: https://www.dinstar.com
: [ protected] / [ protected]
: +86 755

VoDroid App

Using an Android as a GSM VoIP Gateway

VoDroid App realizes the dream that build VoIP Gateway into Android .
It makes your Android upgrade to movable VoIP Gateway immediately.
You can not only have Call Termination, but also send/receive SMS and check USSD remotely.
If your cell is 3G/4G, it turns to 3G/4G VoIP Gateway instantly.
The most important thing is that you can achieve it without ROOT permission,
and the only thing you need is a cheap circuit to attain Call Termination.

Features:

Don&#;t need ROOT permission
Call Termination
Answer Signal Detect
Send/receive SMS remotely
Check USSD remotely
Check call logs remotely
Set up the time interval of every call and daily call duration
Set up the prefix rule of number

Specifications&#;

Compatible with SIP RFC543, RFC
Codec&#;G.711-Ulaw, G.711-Alaw, GSM, SPEEX
Support Asterisk, SIP Proxy Server&#;

: [ protected]
Facebook:https://www.facebook.com/VoDroid-/
Skype: nanodroid_6

GEMPRO

GEMPRO GP-630-3G/GP-632-3G 3G/ GSM VoIP Gateway

GP-630-3G/ GP-632-3G is the 3G GSM VoIP Gateway. It connects 3G/GSM Network with VoIP Network directly, and has function of two-way communicating by 3G/GSM to VoIP or VoIP to 3G/GSM.

Feature:

Compatible with SIP RFC543, RFC.
Compatible with Asterisk.
Call termination (VoIP to 3G/GSM and 3G/GSM to VoIP).
High-quality voice.
Support UPLINK (GSM to VoIP), DOWNLINK (VoIP to GSM) Routing..
Upgrade Firmware by Internet.
Web browser for status checking and settings.
Support SIP Proxy, or point to point application.
GP-630 with 1 port; GP-632 with 2 ports.
0.GP-630/GP-632: GSM 850/900// MHz.
GP-630-3G/GP-632-3G :UMTS /900MHz, GSM 850/900// MHz.
Supports 3CX, Asterisk.

GP-710/712/630/632/630-3G/632-3G work with 3CX system Video Configuration Guide:
http://www.youtube.com/watch?v=6W6i_OLozDs

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
webs: http://www.gempro.com.tw

GEMPRO GP-630/GP-632 GSM SIP VoIP Gateway For Remote SIM Card Access

GP-630 & GP-632 are GSM VoIP Gateway. It connects GSM Network with VoIP Network directly, and has function of two-way communicating by GSM to VoIP or VoIP to GSM. GP-63x provides local and remote access function to
register to GSM system provider: In remote access function, there are two ways to register. First way,
insert SIM Card into general PC/SC Smart Card Reader, plug Smart Card Reader into PC&#;s USB port and
connect to remote GP-63x by Internet. Another way is that you can use GP-060 and Android Smartphone
to remote access to GP-63x via Wi-Fi Network and achieve mobilization. Both ways are not only convenient
for remote management, but also saves expensive international roaming bills.

Feature:

Two-way communication by VoIP to GSM or GSM to VoIP
Compatible with SIP RFC 543,RFC
Local & Remote SIM Card Access
Use general PC/SC smart Card Reader, or GP-060 (with Android Smartphone)
to remote access to GP-63x.
Save and international roaming bills
IP or SIM Card ID verifies Access authority
Support UPLINK (GSM to VoIP),DOWNLINK (VoIP to GSM)Routing
Support VOICE REPORT IP function.
Upgrade Firmware by Internet
Offer Website for inquire or setting
Support SIP Proxy, or point to point application
Fix or Free dialing function
Dialout Number from IP Call Number
850/900// MHz
GP-632 2 Voice Channel, 2 GSM Modules, 2 SIM
Support 3CX, Asterisk.

GP-710/712/630/632/630-3G/632-3G work with 3CX system Video Configuration Guide:
http://www.youtube.com/watch?v=6W6i_OLozDs

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
webs: http://www.gempro.com.tw

GP-514 4 Ports Fixed Wireless Terminal for all Cellular systems

GP-514 is a 4 ports Fixed Wireless Terminal.
It uses Bluetooth technology to connect Bluetooth mobile with common or PBX,
and makes PBX immediately able to dial out and answer mobile call. User can not only get far away
from the electromagnetic wave, but also save expenses without changing their habits of using the table .

Feature:

Makes PBX able to direclty dial out and answer mobile call.
Use Bluetooth to pair mobile ; it has nothing to do with telecommunication provider or frequency.
Just using table to dial to keep users far away from the electromagnetic wave.
Digital volume and gain adjustment.
Transmit high quality voice.
Compatible with GSM/CDMA/3G/3.5G/WCDMA Bluetooth mobile .
Each channel has 30 sets of dial plan, and can add or remove digits automatically. There is no need for users to change their dialing habit.
30 sets book Speed Dial & memorize.

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
webs: http://www.gempro.com.tw

GP-710/GP-712 Bluetooth Mobile SIP VoIP Gateway For Global Systems!!

The GP-71x is a revolutionary and innovative product. It Integrates VoIP and 2.4G Bluetooth
technology in a device. This makes the connection free from the limitation of frequency and
telecommunication provider to become more convenient and widely use. Therefore, it can
also be widely used as GSM/CDMA/WCDMA/3G/3.5G VoIP Gateway.

Feature:

1.VoIP and Bluetooth Mobile full integration.
2.Compatible with SIP RFC543, RFC.
3.Support call Origination (BT to VoIP), call Termination (VoIP to BT).
4.Integrated web server for status and settings.
5.Support SIP Proxy, or point to point application.
6.Option of one stage, fix number ; only GP-712 has free dial function.
7.Bluetooth with auto pairing and auto searching function.
8.QoS and Digital Transmit.
9.Could use with GSM/CDMA/3G/UMTS/4G various Bluetooth mobile phones.
10.2 Bluetooth and 2 VoIP channels.

GP-710/712/630/632/630-3G/632-3G work with 3CX system Video Configuration Guide:
http://www.youtube.com/watch?v=6W6i_OLozDs

Application Diagram:

Specification:

Web Browser
IVR Interface
Uplink Route Setting (BT to VoIP)
Downlink Route Setting (VoIP to BT)

Network Protocol:

SIP v1 (RFC), v2(RFC)
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 MHz to .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 mA
1.8m Network Cable

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
webs: http://www.gempro.com.tw

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

GP-510 uses Bluetooth technology to connect Bluetooth mobile with common or PBX.
It makes PBX immediately own a trunk which has mobile routing. User can not only get far away from the
electromagnetic wave, but also save expenses without changing their habits of using the table .

Feature

1.Using Bluetooth to make PBX immediately own a trunk which has mobile routing.
2.GP-510 directly pairs mobile via Bluetooth; it has nothing to do with telecommunication provider or frequency.
3.Just using table to dial to keep users far away from the electromagnetic wave.
4.Speed Dial Function.
5.The calling will be transferred to PSTN automatically when power failure, Bluetooth mobile is unpaired or out of range.
6.Digital volume and gain adjustment.
7.Local code setting and code add automatically. There is no need for users to change their dialing habit.
8.Bluetooth pairing and searching automatically.
9.Transmit high quality voice.
10.Compatible with GSM/CDMA/3G/3.5G/WCDMA Bluetooth mobile .
11.Using Bluetooth technology, so the connection doesn&#;t need any wires.
12. (PBX) can dial out and answer Skype. (Dongle & API have to support VoIP)
13.100 sets book Speed Dial & memorize.
14.No need to 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 MHz to .5MHz
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 mA

The attachment id given is not valid.

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
webs: http://www.gempro.com.tw

GP-530 Bluetooth Wireless Fixed Terminal

For 3G/GSM/Skype PSTN//PBX Trunk

GP-530 uses Bluetooth technology to connect Bluetooth mobile with common or PBX.
Users can not only get far away from the electromagnetic wave, but also save expenses without changing
their habit of using the table . Meanwhile, GP-530 has the functions to connect Bluetooth mobile
with PSTN. That is, incoming PSTN calls can be forwarded via Bluetooth mobile . Incoming
mobile calls can also be forwarded via PSTN or picked up by table .

Feature:

1.The database can be set to select the best trunk to dial out from Bluetooth mobile or PSTN automatically.
2.GP-530 directly pairs mobile via Bluetooth. It has nothing to do with telecommunication provider or frequency.
3.Just using table to dial to keep users far away from the electromagnetic wave.
4.Speed Dial Function.
5.The calling will be transferred to PSTN automatically when power failure, Bluetooth mobile is unpaired or out of range.
6.Digital volume and gain adjustment.
7.Local code setting and code add automatically. There is no need for users to change their dialing habit.
8.Bluetooth pairing and searching automatically.
9.Transmit high quality voice.
10.Compatible with GSM/CDMA/3G/3.5G/WCDMA Bluetooth mobile .
11.Using Bluetooth technology, so the connection doesn&#;t need any wires.
12. (PBX) can dial out and answer Skype. (Dongle & API have to support VoIP)
13.100 sets book Speed Dial & memorize.
14.No need to change the dialing habit to Skype out.
15.With fixed or free dial function in PSTN forward to Mobile System.
16.With fixed or free dial function in Mobile forward to PSTN.
17.The number of rings from 0-9 can be set to forward by Mobile System automatically if the incoming PSTN call is not answered.
18.The number of rings from 0-9 can be set to forward by PSTN automatically if the incoming Mobile call is not answered.
19.Dial out from side can be selected automatically to dial from PSTN or Mobile system by the setting of Database. 30 sets of Route Databases can be set up.
20.With dial limited function to avoid device seizure in PSTN to Mobile System.

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
webs: http://www.gempro.com.tw

Gempro GP-712A GSM/3G/4G SIP VoIP Gateway for all systems (2 ports)

The GP-712A is a revolutionary and innovative product. It Integrates VoIP and 2.4G Bluetooth technology
in a device. This makes the connection free from the limitation of frequency and telecommunication provider
to become more convenient and widely use. Therefore, it can also be widely used as GSM/CDMA/WCDMA/3G/3.5G VoIP Gateway.

Feature
1.VoIP and Bluetooth Mobile full integration.
2.Compatible with SIP RFC543, RFC.
3.Support call Origination (BT to VoIP), call Termination (VoIP to BT).
4.Integrated web server for status and settings.
5.Support SIP Proxy, or point to point application.
6.Option of one stage, fix number ; only GP-712 has free dial function.
7.Bluetooth with auto pairing and auto searching function.
8.QoS and Digital Transmit.
9.Could use with GSM/CDMA/3G/UMTS/4G various Bluetooth mobile phones.
10.2 Bluetooth and 2 VoIP channels.

GP-710/712/630/632/630-3G/632-3G work with 3CX system Video Configuration Guide:
http://www.youtube.com/watch?v=6W6i_OLozDs

Specification

VoIP

Web Browser
Call Orgination Setting (BT to VoIP)
Call Termination Setting (VoIP to BT)

Network Protocol

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

Codec

GSM
G.711: 64k bit/s (PCM)
G.723.1: 6.3k / 5.3k bit/s
G.726: 16k / 24k / 32k / 40k bit/s (ADPCM)
G.729A: 8k bit/s (CS-ACELP)
G.729B: adds VAD & CNG to G.729

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 MHz to .5MHz
Modulation Method GFSK,1Mbps,0.5BT Gaussian
Output level, class 2
10 meters working range

Gempro Technology Inc.
6F., No.314, Sec. 3, Fusing Rd., South District, Taichung City 402,Taiwan (R.O.C.)
:886-4-
FAX:886-4-
: [ protected]
webs: http://www.gempro.com.tw

GP-060 SIM Card Reader for Android

GP-060 is Wireless SIM Card Reader. It can directly use Android Smartphone to register SIM Card
to remote GP-63x GSM VoIP Gateway via Wi-Fi Network. Since it doesn&#;t need PC to complete register,
it can meet the purpose of mobilization. With SIP VoIP software in Smartphone, it connects VoIP
Network and GSM Network to dial or answer GSM calls via the GP-63x Gateway.

Feature
1.Don&#;t need PC, and could use Smartphone Remote SIM Access directly.
2.Don&#;t need to get ROOT Access and don&#;t change Smartphone firmware.
3.Using Bluetooth interface.
4.Using with our GSM Gateway products only.

Specification

Support 5V,3.3V SIM Card
1 Status LED
USB Port (Power Only)

Smartphone requirements

Android 2.1 or higher
Bluetooth V2.0
Wi-Fi function
Install SIP VoIP software

Bluetooth

Bluetooth Specification V2
Carrier Frequency MHz to .5MHz
Modulation Method GFSK,1Mbps,0.5BT Gaussian
Output level, class 2
10 meters working range

Gempro Technology Inc.
6F., No.314, Sec. 3, Fusing Rd., South District, Taichung City 402,Taiwan (R.O.C.)
:886-4-
FAX:886-4-
: [ protected]
webs: http://www.gempro.com.tw

Gempro GP-708 GSM/3G/WCDMA VoIP PBX for all systems

GP-708 is a revolutionary and innovative product with Bluetooth Interface built inside the IP-PBX.
With this technology you can connect GP-708 by the Bluetooth cell .
Meanwhile it can be independent from the limitation between frequency and Telecommunication Provider.
GP-708 becomes the VoIP PBX support with GSM/CDMA/WCDMA/3G/4G systems.
GP-708 is the best solution which is economized and easy to load.

Feature

1. Web based management
2. Built-in SIP/IAX Server
3. 8 Ports of Bluetooth trunks
4. SIP/IAX Extensions
5. Code: G.711-Ulaw, G.711-Alaw, G.726, G.729, GSM, SPEEX
6. Bluetooth Specification V2.0
7. Bluetooth working range: 10 meters

Package

GP-708 measurement : 183mm x 105mm x 30mm
2.4G Antenna : 8
Power Supply : Input 100-240VAC 50-60Hz/Output 12VDC mA

Gempro Technology Inc.
:886-4-
FAX:886-4-
:[ protected]
http://www.gempro.com.tw

Hypermedia Systems Ltd.

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 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 / MHz or GSM 850 MHz / 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.

Hypermedia VoIP GSM Gateway bridges between the VoIP network (or IP-based systems) and the cellular GSM networks for reducing 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- there is no need for any external VoIP Gateways such as Cisco or Quintum.

A cloud-based SaaS for GSM Gateway monitoring, which provides HG- owners and users with real-time information about the performance of their systems.
Don&#;t miss the opportunity for a 30 day free trial. Pre-register now!!

Check out our latest video on How to register and use our new HyperTam service

An advanced built in application for generating several types of cellular traffic:

GSM calls, inbound and outbound
SMS messages, incoming and outgoing
Calls forwarding, to voice mailboxes, etc.
Pre-set scenarios of Calls and SMS

Currently supported by VCG+ are all HG3xxx,HG4xxx and HG7xxx systems which include a PC board

NEW: Economical 320 SIMs Gateways 320 SIM-based box for SMS/ VoIP Best performance ever!

Complete GSM/SMS device with three times more SIM storagecapacity and SIM Management features in one box.
Easy to deploy and quick to implement, this is the perfect all-in-one product solution which allows you to solve all your telephony needs with NO NEED for an additional SIM server!

Enables to maximize SIM price plans usage and special dynamic tariffs.
10 SIM cards per port with individual settings that can be configured for EVERY SINGLE SIM card!

Ready to order? Be among the first 20 buyers and Get SIMGuard free

Meet with us at WWC Check our new Termination Systems!

See our Call Termination System video

The VoIP GSM Gateway &#; HG- 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- series enables inbound and outbound VoIP and Cellular calls &#; all in one compact box.

VoIP GSM Gateway HG- 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/ or 900/).
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.

Hypermedia Systems is now introducing the opportunity to upgrade its VoIP GSM Gateway HG- to include SMS server for offering wholesale SMS termination service. The SMS server comes with an easy-to-use API / SDK which enables third-party applications to send and receive SMS messages.

The VoIP GSM Gateway- HG- 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 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

Hypermedia&#;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:
Contact: Contact Hypermedia
: [ protected]
Website: www.hyperms.com

ICS

TopGSM Series Analog GSM Gateway comes with 1/4/8 ports and provides a cost- effective alternative for fixed-to-mobile calls. They are compatible with any PBX system or analog phones.

TopGSM 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 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 TopGSM & TopCDMA Channel 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.

iQsim

GSM Gateway

ANTRAX GSM Gateway (VoIP gateway) consists of up to 15 GSM Boards per one universal 3U sub-rack.
Each GSM board supports 2 channels, i.e. simultaneous calls.
GSM modules used in VoIP GSM gateway support all band of GSM frequencies, GPRS, IMEI change and enables a hot swap of boards in sub-rack.

MAIN FEATURES:

Module architecture allows flexible configuration and pricing
Premium-level technology guarantees high-quality call routes
Supplied with Sim Server and GUI software, allowing remote control
We provide help with setup and business launch, plus our own VoIP traffic

SIM Box

The client can choose individual structure of gateway, equipping it with the required number of channels (2 to 30). This feature also means the flexibility in GSM gateway price setting. The modular architecture allows ANTRAX users to place SIM cards apart from GSM gateways and connect them via IP.

This feature provides a huge advantage in terms of safety and removes limitations which could be applied to the topology of your call termination system.It is important to mention that the module-based structure of our product is a part of anti-anti-fraud system, which allows you to reduce the load on every single base station, placing GSM gateways in different locations, without compromising the overall capacity of your GSM termination system.

Group association allows you to associate more than one SIM card to every individual GSM module in GSM modem. It increases the reliability of the system and ensures its smooth operation when one or more SIM cards fail for some reason. In case of fail, the system will automatically replace SIM cards from the SIM array, which was previously associated with this GSM module, saving system capacity that could be temporarily lost in different situation.

MAIN FEATURES:

SIM box includes the following elements:
Sub-rack with fifteen universal slots which provides power and cool the system
SIM-board that comprises 20 SIM cards
Each client has opportunity to choose individual structure of SIM box and use from 300 SIM cards during GSM termination. It is possible to combine GSM and SIM boards in one rack, so it is gsm sim box.

Module-based structure of the hardware of this product opens up a wide range of possibilities to its users, such as:

SIM cards could be placed separately from GSM modules (this option will require high-quality Internet connection between GSM-gateway and SIM box);
SIM cards which take part in termination of voice traffic and sms termination to different destinations/countries could be placed in the same spot, making it easy to control their activity (it is important when you have several GSM gateways located in different countries or in different regions of the same country and all sim cards involved in termination, may be located in one or more SIM boxes placed in the same spot), etc.

MAJOR ADVANTAGES:
SIM cards rotation. One of the optimization algorithms of the system is the so-called Antrax &#;SIM-rotation&#;. SIM cards within each SIM box can be divided into groups, each of these groups can be assigned to a separate GSM-module of VoIP-gateway. Over time, the system is able to make changes within each group, changing SIM card which is responsible for making voice calls from one to another. This not only allows you to optimize resource consumption of every single &#;Sim&#;, but also gives a possibility to reduce their workload and, consequently, the suspicion of the mobile operators.

SIM cards migration. Antrax system is capable of registering the SIM cards on different GSM-modules with a specified frequency. If you have several GSM gateways located in different parts of the city, system will make SIM card to make calls from every gateway by turn, creating an illusion of subscriber movement, thus helping to protect your cards from blocking by the mobile operator.

Contact ANTRAX now!

RELATED PRODUCTS:
VoIP GSM Gateway
Call Termination Business
Premium quality soft for GSM gateways

IRON SIM Server One; All-in-One SIM Server with SIM Storage

iQsim&#;s flagship product, IRON Suite, is redundant and scalable software architecture. Especially designed to manage easily a large number of SIM cards with its intuitive and powerful interface, this solution provides you a lot of benefits connecting virtually devices separated physically.

Key benefits:

Ready-to-Go SIM Server (no additional server required)
IRON Technology: Bonus/Promotion, Credit Synchronization
Integrated Capacity of 416 SIMs cards
Protect your Investment: Secure & Scalable product

Main features:

SIM STORAGE

Hot-swappable compact-PCI board
32 SIM cards per board
Up to 416 SIMs

SIM CONTROL

Easy and powerful web based management
Multi-criteria allocation rules by SIM card plans
SIM travel simulation and IMEI control by SLID
SIM Cards Recharge (USSD, SMS, Transfer, IVR)
SIM Cards Credit Synchronization
Advanced Price Plan : SIM Daily, Weekly and Monthly control
Scalable solution for SIMs and more
Prepaid SIM card management

SIM PROTECTION (HBS)

Random/Unpredictable Simulation
SMS/Call/Voice Box query management
SIM Manager web interface
Integrated with SIM Manager price plans
Web based dashboard

&#;For more information, please refer to our website: http://www.iqsim.com/iron_sim_server_one.htm
or contact us at [ protected]&#;

IRON SIM Manager for SIM Control & Protection

Key benefits:

SIM CONTROL

Always use the right SIM at the right time
Up to 416 SIM cards per SIM Rack
Highly scalable architecture : Up to SIM Cards
Specifically designed for multi-operator and cross country deployment
Maximize SIM price plan usage

SIM PROTECTION

Random/Unpredictable simulation
SMS/Text, Call and Voice Box queries
Web dashboard to manage events
Configuration integrated with IRON Suite

Main features:

SIM CONTROL

Easy and powerful web based management
Multi-criteria allocation rules by SIM card group
SIM travel simulation
Dynamic SIM swap
Prepaid SIM card management

SIM PROTECTION

Text/SMS: Incoming/Outgoing/internal, User-defined destination numbers, User-defined SMS messages
Calls: Incoming/Outgoing/Internal, User-defined destination numbers
Voice Box Management: User-defines DTMF based scenario to query SIM voice box

&#;For more information, please refer to our website: http://www.iqsim.com/iron_sim_manager.htm
or contact us at [ protected]&#;

IRON Call Manager for Call Control

Key benefits:

Call routing with the most appropriate SIM
Real-time GSM port supervision
Centralized CDRs
GSM gateway configuration time reduced by 70%
Call processing time reduction (PDD)

Main features:

Real time GSM ports status visibility
Dynamic and auto-learning central blacklisting
Dynamic provider call repartition
White and grey route management
Mobile portability number management

&#;For more information, please refer to our website: http://www.iqsim.com/iron_call_manager.htm
or contact us at [ protected]&#;

Smart SIM Rack for Mobile QoS Testing

SIM Racks are especially designed for SIM cards storage. Whether you work in QoS, M2M or Mobile Termination, we have the product you are looking for!

Key benefits:

XML API for remote SIM control
Interface with any type of M2M devices
Advanced SIM price plan and tariff configuration
Intuitive Web interface
Integration with Quality of Service (QoS) Test Platform

Main features:

Centralize SIM cards in one place
Manual, Automatic or API managed SIM Allocation
Web-based configuration
SIM card storage : Up to 160 SIM card (extensible with additional SIM Rack)
Multi country support

&#;For more information, please refer to our website: http://www.iqsim.com/smart_sim_rack.htm
or contact us at [ protected]&#;

SIM Rack for Mobile Termination

Key benefits:

Remote SIM access over IP
Hot swappable SIM Rack board
Up to 416 SIM cards in a 6U rackable chassis
Support GSM SIM and USIM cards
SIM Storage capacity for Call & SMS Termination

Main features:

IP connection monitoring
Dynamic IP reconnection
SIM travel simulation
Dynamic SIM swap
Prepaid SIM card management

&#;For more information, please refer to our website: http://www.iqsim.com/sim_racks.htm
or contact us at [ protected]&#;

Mobile Probe for Mobile Network Testing

GSM Devices are geographically dispersed and connected virtually to the SIMs through IRON architecture. Whether your start in business or not, iQsim will suit your needs!

Key benefits:

Install your own QoS/test application
XML API for wireless modules and SIM management
Virtual SIM technology included
Compatible with iQsim SIM manager
Can contain up to 8 wireless modules 2G and/or 3G
Easy integration with existing environment

IRON Mobile Gateway

Key benefits:

Ready-to-go gateway
16/32 GSM ports with up to 1:13 SIM ratio
Dynamic IP support and NAT transversal
IRON Suite Technology Powered
Technical support included
Bonus Credits and Promotions Management

Main features:

16 or 32 Quadband GSM modules
Support simultaneously SIP and H.323
VOIP Codecs : G.723, G.729 and G.711
64 or 128 SIMs capacity to any ports (1:4 ratio)
Extensible up to 416 SIMs cards (option)
SIM Allocation : Time, Usage (Calls, Daily Usage)
Prepaid SIM card management : Credit Check, Top-up, Transfer, Activation
SIM cards Bonus and Promotion support

GSM Gateways

Key benefits:

Tri-Band,Quad-Band or UMTS modules
From 8 to 16 ports in 1U
Dynamic IP Support
Fully compatible with IRON Suite SIM Management solution
Easy configuration and deployment with IRON Call Manager
GSM Gateways fully supported by iQsim&#;s support team

Main features:

Easy and powerful web based management
Multi-criteria allocation rules by SIM card group
SIM travel simulation
Dynamic SIM swap
Prepaid SIM card management

&#;For more information, please refer to our website: http://www.iqsim.com/umts_gsm_gateways.htm
or contact us at [ protected]&#;

ITS Telecom

CGW VIP: The Next Generation of VoIP to GSM Gateways

The CGW-VIP is a unique first-of-its-kind cellular gateway, based on pure-IP technology. The CGW-VIP provides a resilient platform specifically designed to handle both voice and managed data traffic, with the ability to ensure priority and quality of service for voice calls. The CGW-VIP has a friendly user interface for easy management and operation using common web browsers, enabling both local and remote management access.

The ultra-high performance CGW-VIP connects to the office IP network, efficiently routing all calls from the IP network to the cellular network and vice versa. This capability drastically reduces communication costs by enabling the use of inexpensive VoIP calls and letting companies take full advantage of their VPN arrangement on the cellular network. The CGW-VIP is especially cost-effective for companies with extensive use of cellular and long distance calls.
Aside from providing significant cost savings, offices can connect branches, remote offices, or teleworkers and establish an efficient and free-of-charge virtual voice private network. Customers can easily add redundancy and resiliency for their key services in case of traditional network outage. CGW-VIP can also serve as a last mile solution, providing rural telephony in places where no terrestrial infrastructure is available.

Extremely beneficial for:

Offices with extensive use of cellular calls
Remote or in-field users (Voice VPN)
Long distance or international calls
Remote branches
Business continuity, back up of
PSTN and landline VoIP

CGW-VIP Unique VoIP-to-GSM 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-VIP Main Features

Supports VoIP, SIP, and GSM connections
Interoperates with all standard IP-PBX/soft switch equipment
Embedded web server
Local and remote management via user-friendly web interface
Local and remote upgrades of configuration files and firmware
Supports registration of SIP extensions
Enables creation of a virtual office with remote extensions
Built-in firewall and NAT
Built-in Least-Cost-Routing (LCR)
CLIP/CLIR

For more Information:
Cellular-to-VoIP solution page: http://www.its-.com/category_info.asp?id=103_
CGW-VIP Product page: http://www.its-.com/info.asp?id=63
More cellular solutions: http://www.its-.com/category_info.asp?id=103_144_

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

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

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, EUR for card with 2 modules.

MCS &#; Mixe Communication Solutions B.V.

For more than 10 years MCS&#; AS5xx-GSM Gateways are available through distributors and resellers. &#;Evolution of Proven Technology&#;: German quality products ranging from ISDN2 to ISDN30 to SIP. Successful SIP implementations based on extensive typetesting with most IP-PBX platforms:

3CX System
Aastra Ascotel 300
Aastra (Ericsson) MX One
Alcatel OXE
Alcatel OXO
Altigen
Asteriks 1.4
Avaya IP Office
Cisco Callmanager (Express)
Interactive
Innovaphone
LG-Nortel IPEC
Microsoft OCS
Mitel ICP
PMN platform
NEC Philips
Nortel BCM
Nortel CS
Scopserv
Shoretel
Siemens HiPath
Siemens HiPath
Siemens OpenOffice
Swyx
Panasonic KX-TDE100E
Vertical Televantage IP-PBX
Xelion
Zyxel IPBBX X

Hosted platforms:

Broadsoft
OneXS
RoutIT
TeleSpectrum TMVS
Natural Convergence / Silhouette
uNet

MCS Gateways Features & Highlights:

2 &#; 30 GSM / VoIP channels
Building Blocks available:
Up to 5 SIM cards per GSM Channel (SimSwitch versions)
Integrated antenna combiner
Siemens/Cinterion Quadband modules with digital audio
Remote configuration
Powerful on-board tracing and Wireshark analysis

19&#; Master & Slave:

For more information, please visit gsm gateway 16 port.

19&#; SimSwitch Master:

Desktop version:

KTS

VoIP GSM Gateway Series &#; KTS4SMi/KTS4Mi

Destination
For 4 or 8 GSM to VoIP connection; Quad band 900 850 MHz; SIP protocol, Asterisk full compatilbe; support all NAT, with change IMEI, with local SIM server.

Fully compatible with:

3CX System
Aastra (Ericsson)
Alcatel
Asteriks all
Avaya IP Office
Cisco Callmanager
LG-Nortel IPEC
Microsoft OCS
NEC Philips
Nortel BCM
Nortel CS
Siemens HiPath all
Siemens OpenOffice
Panasonic KX-TDE all
Zyxel IPBBX X

Major Function:

VoIP (SIP)- GSM conversion
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP (RFC, RFC) protocol
Support SMS to MAIL function
To be setup on web

Specification:

SIP protocol: (RFC, RFC)
TCP/IP: IP/TCP/UDP/RTP/RTCP/&#;CMP/ARP/RARP/SNTP/SNMP/SMTP/POP DHCP/DNS Client&#;
IEEE802.1P/Q&#;NAT Traversal&#;STUN&#;uPnP&#;IP Assignment&#;Static IP&#;DHCP
Codec: G.711 u-Law&#;G.711 a-Law&#;G.723.1 (5.3k) G.723.1 (6.3k)&#;G.729A&#;G.729A/B&#;Voice Quality VAD&#;
CNG&#;AEC&#;LEC&#;Packet loss
GSM Frequency: Quad band 900 850 MHz

4-channel version with separate antenna:

4-channel version with separate big antenna:

4-channel version with antenna splitter and small antenna:

4-channel version with antenna splitter and big antenna:

4-channel version with local SIM server and separate antenna:

4-channel version with local SIM server and separate big antenna:

4-channel version with local SIM server and antenna splitter and small antenna:

4-channel version with local SIM server and antenna splitter and big antenna:

VoIP GSM Gateway Series &#; KTS8SMi

8-channel version with antenna splitter and big antenna:

NICHERONS SpoGSM GSM Gateway & IP-PBX

System Users:

Small and medium enterprises ( SMB )
Law firms
Brokerage offices
Insurance offices
Medical offices
Travel agencies
Apartments
Universities and Schools
Small Hotels, Motels
Home users / Home businesses ( SOHO)
Remote users
Branch offices

System Specification:

Form Factor: Small form factor
Dimensions (WxDxH): 330mmx130mmx37mm
Weight: 1.5kg
Processor: 32-bit 400Mhz RISC processor
Onboard Flash: MB
Onboard SDRAM:64MB
Connectivity: 2x RJ45; 4x GSM slots; Power connector
Power: AC 100-240 V ( 50/60 Hz ) adapter In
OS: uClinux 2.6x;
PBX software: Asterisk 1.4.x

You don&#;t need to be a expertise. We are also behind the product to ensure it up & running!

Easy to Setup and Manage
Cost effective
Easy to use &#; Friendly GUI for 1st time user

Everything you need to cover your needs & MUCH More.
Commercial QUALITY at no compromise, Built to last!!!

P.S. &#; All the Configuration and Management is done via WEB based GUI.

System Key features

4 x GSM Analog Lines (other combination of FXO / FXS ports are available)
Base on Asterisk
Configuration by Web
Built-in SIP/IAX server
Static/DHCP/PPPoE network access
Codec: G7.11 ulaw, G711 alaw, GSM, SPEEX, G.726, G729 (no additional license fee required)
SIP/IAX Trunk (use with voip trunk operator)
Zap trunk (use with PSTN &/or GSM)
Zap Extensions (connect with Analog ) when equipped with FXS ports
SIP/IAX extensions (connect with ip )
voice mail server
Flexible dial plan
Follow me
Ring groups
Local and / or Remote IP Extensions with Voice Mail, Music on Hold
Branch office support
As Many as you need VOIP Extensions (No Per User Licenses)
Simple, Reliable and Economic solution for your needs
Everything you need to place VOIP calls
WEB based intuitive configurator (user friendly and super easy to use)
Call Park
Call Logs
Support ip with key function
FAX T.38 support
3-way calling
Call waiting
Caller ID
Paging & Intercom
Call forward (on busy or an un-answer call or unregistered extensions)
WEB Voice mail with notification
Call reporting
Conference Calling &#; Multiple Conference Bridges supported
Time Based Rules control
Remote Backup/Restore through Internet Browser GUI
Music On Hold
Call Transfer
BLACKLIST
Call Queue
Conference recording
Call feature codes (* codes)
DISA support
Off Premise Extension &#; Remote Extension
Auto Attendant (IVR- Interactive Voice Recording)

Nicherons International Inc.
www.nicherons.com
[ protected]
Skype Toll Free: sales.nicherons
+1 (905)

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 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 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 &#; SMS &#; 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.

TMG S20

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 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 [ protected] or visit http://www.novatec.de

OpenVox Hot-Swap VOIP GSM Gateways

OpenVox VoxStack Series GSM Gateway is an industry 1st open source asterisk-based GSM VoIP Gateway solution for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway.Also secondary development can be completed through AMI (Asterisk Management Interface).

Independent System for Each Module
Affordable Price with Superior Performance
Hot-swap Design for both Sim cards and Modules
Compatible with Asterisk, Elastix, 3CX, FreeSWITCH Sip Server
Available for OEM
3-Month &#;No Questions Asked&#; Return Policy
Lifetime Warranty

OVERVIEW

There are two GSM Gateway models with VoxStack series GSM Gateway, the VS-GW-4G and VS-GW. The OpenVox GSM/VoIP gateways allow a SIP based IP-PBX system or VoIP line to be connected directly to the GSM cellular network, The Modular Design GSM Gateways are ranging from 4 up to 20 GSM channels, developed for interconnecting a wide selection of codecs, including G.711A, G.711U, G.729A, G.723.1, G.722, G.726, GSM, to the GSM cellular networks to quickly reduce telecommunication expenses and maximize cost-savings. With the unique design of the VoxStack Gateway, it can support hot-swap for both SIM cards and GSM gateway modules. Users can simply add or remove the modules for hardware expansion or exchange. Each GSM gateway module runs an independent asterisk system inside.

The VoxStack gateway designs with a Lan Switch board that provides stackability on the hardware upgrade. It supports SMS messages sending and receiving and group sending and SMS to . The GSM Gateways will be 100% compatible with asterisk, Elastix, trixbox, VoipSwitch, VOS, 3CX and FreeSWITCH etc. sip server

VS-GW &#; 4/8/12/16/20 Channels

4/8/16/20 GSM Channels
1 SIM card per GSM channel
Quad-Band GSM/GPRS 850/ 900// MHz
2 LAN ports
Storage temperature range: -40~125&#;
Operation temperature range: 0~50&#;
Operation humidity range: 10%~90% non-condensing
Power source: 1U ATX power souce,100-240V/1-2A
Max power: 46W
Dimension: 44cm × 4.5cm × 30cm
Weight: VS-GW&#;without GSM module)&#;g, VS-GWM400G&#;module): 214g

Home page: http://www.openvox.cn
: [ protected] / [ protected]
: +86-755- / +86-755- / +86-755-

For Asterisk® / FreePBX® developers requiring GSM connectivity for wireline replacement, Least Cost Routing (LCR), or SMS notification, PIKA now offers a GSM module for the WARP Appliance for Asterisk. The GSM module provides two GSM channels/radios per module and uses two externally accessible SIM cards. The module can work in combination with other WARP Appliance expansion modules (FXO, FXS, BRI).

Applications & Benefits

Cost savings with Least Cost Routing (LCR) &#; For calls between landline and GSM or between different GSM networks, which can be very expensive, the FreePBX dial plan can be configured with a prefix that, when dialed, will route calls placed from a SIP or analog over a GSM trunk.
Back-up &#; Should the landline network go down, GSM can be used as a cost-effective backup.
Land Line Displacement &#; For wireless carriers looking to build incremental traffic on their networks by displacing traditional company land lines, Warp with GSM trunk connectivity provides a unique small/medium business system option.
Remote and Temporary Installations &#; Warp with GSM is an excellent option for the installation of business systems in remote areas where traditional, wired lines may not be available, or may have extended installation intervals. It can also be a great option for temporary installations.
Increased productivity with SMS voicemail notification &#; No longer does a user need to continually check voicemail for that important message. Upon receipt of a voicemail, the owner of the mailbox can receive an SMS message that he or she has a new voicemail.

Features

Two quad-band GSM channels/radios per module
Support for 2 SIM cards &#; externally accessible
Two modules per WARP
GSM module can be combined with analog trunks (FXO), analog stations (FXS) or BRI trunks
No external power supply required
Average power consumption = 1W per module
3V, 1.8V SIMs supported
Supports incoming and outgoing voice calls for PBX, Auto Attendant, IVR and other telephony applications
Supports SMS send
Supports incoming caller ID
WARP Appliance LCD display shows trunk status (circuit type and status: busy, available, ringing)
DTMF detection for Auto Attendant, IVR and voicemail applications
FreePBX GUI support
Full text-based Asterisk dial plan supported

Polygator

Features

GSM-card provides communication between your application under OS Linux and mobile operator GSM-network.
Compatible and certified with Elastix

Sphere of application:

GSM VoIP gateway (POLYGATOR Linux software or Asterisk)
SMS-gateway (under POLYGATOR Linux software (direct AT-command support rfc via TCP-ports) or under Asterisk (via SQL))

Technical details:

4/8 GSM-modules (850/900// Mhz), WCDMA(3G)
1 SIM-card per channel
DTMF receiver
Digital sound processing (with possibility of hardware compression: G.729, G.726, G.723)
Light-emitting diode indication of channels work
PCI or PCI-express

Bundling:

GSM-card for 4/8-channels
4/8 vertical antennas

FOR MORE INFORMATION: www.polygator.com
or by mail to [ protected]

PORTech Communications lnc.

PORTech is the original manufacturer in Taiwan with 37 years; therefore we&#;ll do our best to offer the best service and support.

PORTech offer

E1 /T1 GSM Channel bank
1/2/4/8/16/32 Ports VoIP GSM Gateway /GSM VoIP Terminal /3G UMTS Gateway
SIM Server
SBK-32/SBK-128:Remote SIM Bank
SMS Gateway,SMS Server
Free Roaming Gateway
VoIP Adapter
1/4/8 ports GSM Fixed Wireless Terminals(Follow Me GSM Gateway)
GSM Booster,Antenna Combiner
IP/PSTN Power Switch
3 way conference box
IP Speaker
IP Broadcasting System
Worldwide use ( 2G,3G with all world and Japan,CDMA )

http://www.portech.com.tw
Portech Distributor in Germany & EU http://www.portech.de

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

MV-374 : 4 channels VoIP GSM Gateway

Portech MV-378 &#; 8 channels VoIP GSM Gateway

MV-378 : 8 channels VoIP GSM Gateway

Portech MV- : 32 ports VoIP GSM Gateway

Portech MV- : 16 ports VoIP GSM Gateway

Economical solution with competitive price
MV- : 32 channels VoIP GSM Gateway
MV- : 16 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 1,2,4,8,16,32 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS network to IP .
MV serial can Receive SMS and send SMS on web
Allows your program Send/receive SMS with AT Command
Call Back feature

Portech SIM Server

SS-128: 128 sims SIM Server (manage 4 SBK-32 sim banks)
SS-256: 256 sims SIM Server (manage 8 SBK-32 sim banks)
SS-512: 512 sims SIM Server (manage 16 SBK-32 sim banks)

Features:

Set Talk Time per SIM
Set GSM Group (Assign several SIMs Per GSM Port)
Set Day of week
Set Time Range

Portech SBK-32 : 32 SIMs Remote SIM Bank

Remote sim access that connect with serial MV-37X via internet.
SBK-32 : 32 SIMs Remote SIM Bank

Portech SBK-128 : 128 SIMs Remote SIM Bank

Remote sim access that connect with serial MV-37X via internet.
SBK-128 : 128 SIMs Remote SIM Bank

With SBK-32/SBK-128,SIM cards are no longer need to be installed in MV-37X anymore; you can deploy your GSM gateways in different locations and countries. New concept of using SBK-32 solution, you can centralize and supervise all sims in one place.

In that case, SIM management will be easy, quick and safe as well. All SIM cards can be stored and control in one secure place. Once in a while, MV-37X must change SIM cards to get the best rate on different GSM operators. With SBK-32/SBK-128, you can place physical sims in office or change SIM corresponding port directly on web any second.MV-37X connects with SBK-32/SBK-128 to read the SIM information and register it on the GSM network.

Key Specifications/Special Features

Handle 32/128 SIM cards per SBK-32/SBK-128
Central management and secure storage of all SIM cards
Control and Supervision of your SIMs from your office
Allow you to choose the best GSM operators price plan
Immediate control of all your SIM cards from your office
Easy and powerful web based management, Easy to configure convenience any time!
Switch SIM cards between connected gateways, configuration via remote client
Save huge costs on technicians and traveling to the installation sites

Solution suggest

If you need the solution for 32 GSM ports,1 SIM per GSM port, you can buy
MV-*1,SBK-32*1,SS-128
If you need the solution for 32 GSM ports,4 SIMs per GSM port, you can buy
MV-*1,SBK-128*1,SS-128
If you need the solution for 64 GSM ports,4 SIMs per GSM port, you can buy
MV-*2,SBK-128*2,SS-256

Any GSM solution you need, please contact us soon

GSM Fixed Wireless Terminals /Follow Me GSM Gateway

MT-350~1 ports GSM Fixed Wireless Terminals,Follow Me GSM Gateway
option: MT-350S, Add SMS Function(Short message Service)(1 port only)

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

DMT E1 GSM Channel Bank

E1 PRI/T1 PRI
DMT: E1 GSM Channel Bank

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
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).

DMT-E1 GSM Channel Bank

Antenna collecting Box: Base equipment.

If you want to connect extra Outdoor directional antenna or Yigi antenna with long cable-10m,20m,30m,you can purchase option Antenna Combiner(AC-32)

GSM Booster: BT-921/ BT-918

BT-921 (GSM + 3G) is Dual Band GSM Booster
BT-918 (GSM + DCS) is Dual Band GSM Booster

PORTech BT-921 provides and improves the voice and cellular signal, and also high data rate communication like HSDPA / HSUPA / HSPA+ (mobile internet).

It&#;s high quality with superior amplification, simple installation for office, home and indoor location need
GSM Booster: BT-921/ BT-918

If you have interest,please contact us

PORTech Communications Inc.
150 Shiang Shung N.Rd,Taichung,Taiwan
http://www.portech.com.tw
:[ protected]
https://www.portech.com.tw/index.aspx?lang=US&#;- PORTech Products list

QuesCom

QuesCom VoIP GSM gateway

QuesCom gateways

QuesCom is a certified member of major PBX vendor programs as 3COM, Aastra Matra, Alcatel, Avaya, Cisco, Innovaphone, Mitel & Swyx.

Technical specifications:

Up to 16 GSM channels in one GSM gateway
Connection to the PBX either in IP or in ISDN
Quad band GSM (850/900// MHz)
Dual band CDMA (800/ MHz)
Tri band UMTS (850// MHz)
Supports H.323, SIP
Support for failover between VoIP and GSM network
One 10/100 Ethernet port;
Support for Master/Slave architecture
Integrated SMS gateway
CallBack request by SMS and by HTTP
IVR (Interactive Voice Response)
LCR &#; List Cost Routing
CDR &#; Call Detail Records
Support for Fax server and T.38 fax protocol (expandable to 500 users);
Support for Mobility service (expandable to 500 users);
GSM gateway delivered with 2 free demo users (for Fax/SMS and mobility features testing)

Further information about QuesCom GSM gateways

QuesCom SMS Manager

QuesCom allows you to receive SMS on each GSM channels and to send SMS by any user from his personal computer. The new version of QuesCom GSM gateway supports:

to SMS format: [ protected]
Flash SMS supported
Send SMS to multiple users via Outlook

The QuesCom SMS Manager is a software for sending and/or receiving SMS, and allows to do:

SMS Campaigns
Campaign reports
Third-Party Application & Database Integration.

2-channel version:

TOPEX MobiLink IP is a GSM/UMTS small capacity gateway with VoIP interfaces. Its main functionality is to interconnect IP PBX or hibrid PBX with mobile networks. With TOPEX MobiLink IP you make significant savings on calls from IP to cellular networks and backwards.

FEATURES:

1 or 2 cellular modules Available in multiple configurations
Least Cost Routing by: prefix
Works with any IP-PBX or Softswitch based on SIP protocols (e.g. Topex multiSwitch, Topex IP-PBX, Asterisk&#;, Cisco Call-Manager&#;,etc),hybrid PBX with VoIP trunks
DISA capability
Advanced Incoming and outgoing call routing
SIP protocols supported
Advanced SMS capabilities
Send SMS from Web
SMS to and to SMS
Easy to use web based interface

Contact sales for special offer

4-channel version:

Topex VoiBridge is a GSM/UMTS small capacity gateway with VoIP or PRI interfaces. Its main functionality is to interconnect VoIP or PSTN networks with mobile networks.
With Topex VoiBridge you make significant savings on calls from IP or PSTN to cellular networks and backwards.

FEATURES:

Capacity:

up to 4 GSM/ UMTS channels
up to 4 SIM cards ( 1 SIM per channel )
2E1 intefaces &#; in & out (for PRI variant)
H323 and SIP supported
Advanced incoming call routing based on Caller- ID
Advanced call-routing of outgoing calls
SIM Server Ready &#; compatibilty with TOPEX SIM Manager
Advanced security with integrated firewall

BENEFITS:

Supports up to 4 cellular operators
No need to add any other gateway
Simple integration with VoIP and PSTN networks
Least Cost Routing
Fast return of investment

Contact sales for special offer

32 or 64 channels version:

Topex multiAccess is an advanced GSM gateway; its main functionality is to interconnect VoIP& PSTN networks with all kind of mobile networks (GSM, UMTS, CDMA).

FEATURES:

Fixed mobile convergence
VoIP &#; GSM gateway
Trunk gateway for NGN
Enterprise hybrid telephony solution
IP telephony for Triple Play Solutions (voice, video and data)
VoIP termination

CAPACITY:

VoIP Card
H323 and SIP &#; supported
up to 60 voice channels
2 E1 trunks
60 simultaneous calls
ISDN, SS7, R2, R1.5 &#; signaling protocols supported
up to 60 GSM/UMTS/CDMA channels

CONFIGURATION:

Up to 32 cellular channels &#; 2 GSM/UMTS channels per board
Up to 64 cellular channels &#; 4 GSM/UMTS channels per board

BENEFITS:

Conversion of PSTN calls into IP or mobile calls
Least Cost Routing
Multiple voice codecs

Contact sales for special offer

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 .

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: $. Additional E1 cards can be installed for E1 connectivity and/or pass through &#; interception of GSM calls)

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 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 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 ) , SDP (RFC ), Symmetric RTP, STUN (RFC), ENUM (RFC ), RTP Payload for DTMF Digits (RFC), 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

Contact: [ protected]

Synway

Synway SMG series GSM/CDMA VoIP Gateway is a multi-functional product used to efficiently realize the smooth transition between mobile and VoIP network. Different from SMG/ series, SMG adds more software features and adopts new hardware structure design. With new pop-up SIM card slot, users are able to change SIM card easily without disassembling equipment. Also, it is fully compatible with Synway SIMCloud/SIMbank solution.

Support 2~32 GSM/CDMA/WCDMA/Mobile Protocols
Superior Voice Quality by Designated DSP Chipsets
Support SMS and A Range of Mobile Applications

Benefits

Compact, convenient VoIP connectivity
High survivability&availability
Multi-function to ensure network security
Easy to install, configure, and maintain

Features

Rich and powerful SIP normalization for various needs and minimum communication costs
Advanced hardware and software architecture to guarantee high performance
Interoperable with softswitch, PBX, and IP-PBX
User-friendly GUI for better control and management

Function Description

GSM: Quad-band 850/900//MHz and customizable
CDMA: 800MHz
PSTN and VoIP Hotline
Up to 32 Concurrent Calls
Flexible Dial Rules and Manipulation Rules
SMS Sending and Receiving
USSD, Open API for SMS/USSD
BCCH
Carrier Selection
PIN Code Management
Remote SIM Card Management(SIMCloud)

Typical Application

Home page: http://www.synway.net
: [ protected]
Skype ID: x.alex_huang.x
: +86 571

SunComm

SunComm is a Taiwanese manufacturer of VoIP <-> GSM equipment.
Their product contains a decent GSM module (Siemens, SimCom etc) and an external antenna with magnetic feet.

Their current product line:

2 Channel SC-385 with 2SIM, 1LAN 1WAN , Quad Band ,SMS
4 Channels SC-495 with 4SIM,1WAN , Quad Band ,SMS
8 Channels SC-895 with 8SIM,1WAN, Quad Band ,SMS
SC-RS Remote SIM Switch with 32 SIM (to be used as remote SIM server)

SunComm VoIP GSM Gateway SC-495 with 4SIM, 1 WAN

(NAT & Auto hunting support)

Key Facts

For GSM &#; VoIP connection, Quad band 900 850 MHz; SIP protocol, Asterisk, VoIPBuster compatible, support NAT & Auto Hunting; CDMA VoIP Terminal: SC-495cdma
Packing Detail

Major Function:

4 x VoIP (SIP)- GSM conversion
50 sets of LAN -> MOBILE routes setting;
50 sets of MOBILE -> LAN routes setting
Voice response for setting & status (dial in from mobile)
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP (RFC, RFC) protocol,communicates with other gateway or PC
To be set on web

Specification:

SIP protocol: (RFC, RFC)
TCP/IP: IP/TCP/UDP/RTP/RTCP/&#;CMP/ARP/RARP/SNTP DHCP/DNS Client&#;IEEE802.1P/Q&#;ToS/DiffServ&#;NAT Traversal&#;STUN&#;uPnP&#;IP Assignment&#;Static IP&#;DHCP&#;PPPoE
Codec: G.711 u-Law&#;G.711 a-Law&#;G.723.1 (5.3k) G.723.1 (6.3k)&#;G.729A&#;G.729A/B&#;Voice Quality VAD&#;CNG&#;AEC&#;LEC&#;Packet loss
GSM Frequency: Dual band 900/MHz,
Tri band 900//MHz
Quad band 900///850MHz

SunComm VoIP GSM Gateway SC-895 with 8SIM, 1WAN (NAT & Auto hunting support)

Key Facts

For 8 GSM &#; VoIP connection; Quad band 900 850 MHz; SIP protocol, Asterisk compatilbe; support NAT & Auto Hungting; CDMA VoIP Terminal: SC-895cdma
Packing Detail

Major Function:

VoIP (SIP)- GSM conversion
50 sets of LAN -> MOBILE routes setting;
50 sets of MOBILE -> LAN routes setting
Voice response for setting & status (dial in from mobile)
For call termination (VoIP to GSM) and origination (GSM to VoIP)
Standard SIP (RFC, RFC) protocol,communicates with other gateway or PC
To be set on web

Specification:

SIP protocol: (RFC, RFC)
TCP/IP: IP/TCP/UDP/RTP/RTCP/&#;CMP/ARP/RARP/SNTP DHCP/DNS Client&#;IEEE802.1P/Q&#;ToS/DiffServ&#;NAT Traversal&#;STUN&#;uPnP&#;IP Assignment&#;Static IP&#;DHCP&#;PPPoE
Codec: G.711 u-Law&#;G.711 a-Law&#;G.723.1 (5.3k) G.723.1 (6.3k)&#;G.729A&#;G.729A/B&#;Voice Quality VAD&#;CNG&#;AEC&#;LEC&#;Packet loss
GSM Frequency: Dual band 900/MHz,
Tri band 900//MHz
Quad band 900///850MHz

Joan Hu
Suncomm Technology Co.,Ltd

Website : www.suncommtech.com.tw
: [ protected]
MSN: [ protected], [ protected]
Skype: liaison3.suncomm.com.tw
Taiwan :886-2-
Taiwan Fax:886-2-

Introducing GoIP, a SIP-GSM Gateway for Asterisk

How far we have come! The original Asterisk® claim to fame was its ability to interface with proprietary systems and legacy telephony hardware, the glue that literally kept companies stuck to their overpriced PBXs. And, just as wired systems began to lose their edge, along came the Bell Sisters to introduce cellular communications with billing that began when the started ringing and an end to toll-free calling and extra fees for text messaging on top of exorbitantly priced data service. The piece that traditionally has been missing from Asterisk deployments has been interconnectivity with cellular data services. Well, that was then, and this is now. Meet the GoIP GSM Gateway in one, four, eight, and 16-channel flavors to meet your every need. Our focus today will be the one-channel GoIP device, but the larger units work almost identically so, once you&#;ve mastered the device, it&#;s not rocket science to move to the 4-channel or 8-channel device (or even larger) if the extra GSM ports better meet your office&#;s requirements.

Let&#;s begin with the basics. What does it do? What does it cost? Why do I need it? How steep is the learning curve?

What Does It Do? In a nutshell, GoIP is a SIP-talkin&#; GSM gateway that sits on the same network as your Asterisk server. Once you configure a trunk and a few special Asterisk settings to support SMS messaging, you&#;ll have another full-featured provider for your PBX, only this one happens to be GSM cellular-based. The good news is GoIP brings to your PBX most of the same feature set that is available using your favorite GSM cellphone except now every extension on your PBX in a Flash&#; server can share the cellular connection both for calls and messaging. That means inbound and outbound cell calls as well as inbound and outbound SMS messaging for every extension on your PBX.

With today&#;s Nerd Vittles additions, here&#;s the new feature set using a GoIP device from any extension on your PBX:

Make outbound calls through the GoIP cellular trunk from any PBX extension
Receive incoming cellular calls and redirect them to any number on your PBX
Dictate text by and deliver SMS messages to any SMS-capable device
Use a browser to create and deliver outbound SMS messages to any SMS device
Receive incoming SMS messages and forward the messages to any address
Receive incoming SMS messages and forward the messages to any SMS number
Send an SMS message with a password and receive a callback with DISA dialtone

What Does It Cost? As much as we love Amazon for its referral revenue support of our blog and open source projects, we couldn&#;t find a single-channel GoIP offering at a reasonable price. The Amazon links provided above for the larger units are competitive (about $100 per port). For the single-channel model, eBay® is your friend. You&#;ll find multiple providers in the $150 price range. All of the units we&#;ve found ship from China. We used this provider who got the GoIP device to us exactly 14 days after we ordered it. Ours shipped with the latest firmware, but firmware updates are available here. AliExpress also sells the devices for about the same price. We&#;ve had good luck with them in the past.

The other expense with the GoIP devices is cellular service. For each channel, you&#;ll need a GSM SIM card just like what your GSM, AT&T, or T-Mobile cell uses. The good news is there are lots of other choices now. See WalMart for some options. Another option for low frequency use would be T-Mobile&#;s pay-by-the day plans. The $1 (unlimited SMS messaging) or $2/day (unlimited calls and unlimited SMS messaging) plans are almost perfect since you don&#;t need data. Just be sure to choose a GSM carrier, AT&T or T-Mobile in the U.S. market. Both are supported by StraightTalk. Our favorite remains the (almost) unlimited calling, text, and data $45 plan from StraightTalk. With their AT&T-compatible SIM (don&#;t buy it in a StraightTalk-locked !), it&#;s a simple matter of moving the SIM card from your cellphone to the GoIP&#;s GSM slot (connectors facing down). The GoIP unit can spoof an IMEI for picky providers.

Why Do I Need It? The two major advantages of adding a cellular trunk to your PBX are redundancy and portability. Except in the Hurricane Katrina situation, chances are that your Internet service provider and your cellular provider won&#;t both be dead in the water at the same time. The good news is that even with a hurricane, you can pack up your PBX in a Flash server or Raspberry Pi together with your GoIP device and move to higher ground. As fast as you can say "George Bush is a compassionate conservative," you&#;ll be back in business.

And then there are the mobile users such as construction site workers, mobile firefighters deployed to a site far from home and other first responders, or even the nomads that manage conventions in a different town every week. Think AstriCon! Rather than relying on crappy hotel WiFi service or paying an arm and a leg for installation of cable or DSL Internet service which often isn&#;t available anyway, now you have the flexibility to deploy a full-featured PBX at almost any temporary site with nothing more than a $30 Wi-Fi firewall/router, a PBX in a Flash Server or Raspberry Pi, and a GSM SIP trunk courtesy of GoIP. The only other ingredient you need is a little electricity. That could be a wall outlet, or a generator, or an inexpensive AC inverter for your vehicle. Did we mention it&#;ll work identically on the next site without spending an extra nickel. Hardware cost for the Mobile Communications Center (as shown below): about $250.

Last but not least are all of the organizations that could benefit from an SMS-based emergency messaging service. A dollar a day is a small price to pay to deploy a service that can alert the public, employees, or parents and students of emergency situations. Before you read about the next mass shooting or midnight tornado, give it some thought. We&#;ve already introduced SMS Blaster to make the job easy. Or you can roll your own by building a simple text file in /tmp/callees.txt with a 10-digit callee&#;s number on each line. Then add the following snippet to your Asterisk dialplan code and put your emergency message in line 2. You&#;ve just replaced a $100 a month message blasting service with a totally portable, self-managed solution. And you&#;ll recover your hardware costs in less than three months.

[goip-sms-blaster] exten => s,1,Answer exten => s,n,Set(SMSMSG="Here is where your emergency message goes.") exten => s,n,ReadFile(callees=/tmp/callees.txt) exten => s,n,Set(callees=${URIENCODE(${callees})}) exten => s,n,Set(callees=${REPLACE(callees,%0A,-)}) exten => s,n,Set(SMSNUM=${callees:0:10}) exten => s,n,While($[${LEN(${SMSNUM})}>9]) exten => s,n,NoOp(Here's where we send SMS message to: ${SMSNUM}) exten => s,n,Set(SMSOUT=${SMSNUM}%0A${SMSMSG}) exten => s,n,Set(SMSOUTRAW=${URIDECODE(${SMSOUT})}) exten => s,n,Set(MESSAGE(body)=${SMSOUTRAW}) exten => s,n,MessageSend(sip:goip_1) exten => s,n,Set(callees=${callees:13}) exten => s,n,Set(SMSNUM=${callees:0:10}) exten => s,n,Set(SMSNUM=${REPLACE(SMSNUM,-,0)}) exten => s,n,EndWhile() exten => s,n,Hangup()

How Steep Is the Learning Curve? Lucky for you, you&#;re not going to have to worry about the learning curve. After all, that&#;s why you come to Nerd Vittles, isn&#;t it? We&#;ve spent the better part of a week getting the GoIP to sit up and bark. If you&#;re a slow typist, it might take you 10 minutes to get everything set up and functional once you have your GoIP device and SIM card in hand. When we&#;re finished, you&#;ll have an easy way to make and receive calls through your GoIP device using any extension on your PBX. And you&#;ll have a simple utility to send and receive SMS messages. In fact, you&#;ll be able to dictate your SMS messages from any connected to your PBX and send them out to any number supported by SMS including the millions of Google Voice numbers. Last but not least, we&#;ll provide a utility to send password-protected SMS messages to GoIP and receive a return call with DISA dial tone to make outbound calls using any available trunks on your PBX.

A Word About Security. We&#;re a little paranoid when it comes to security so bear with us. Without impugning anyone&#;s integrity, suffice it to say this device is manufactured in China. Although the device reportedly runs Linux, none of its other firmware is open source, at least not that we could find. There also are three back doors into the system which can be triggered by SMS commands to the device itself. These are well documented in the GoIP User&#;s Manual. Whether there are other backdoors or whether the device "phones home" are questions we have neither the time nor the money to explore. Unless you do, you are well advised to treat the device in the same way you would treat a new employee on their first day at work. Don&#;t put the device on a private LAN in which other computers or devices on the LAN are not protected. Don&#;t use a SIM card with an automatic renewal feature or with authority to post charges against your credit or debit card. Change your Admin password to the device immediately. Don&#;t use a password you use elsewhere! Anyone can reset the device to factory defaults by knowing the default credentials and sending RESET admin in an SMS message to the device. We love the device, but be careful.

Initial Setup of the GoIP Device

To begin, you&#;ll need cellphone coverage in the place where you intend to connect your GoIP device. Verify this while the SIM card you plan to use is still installed in a working cellphone. Make a call and send an SMS message to verify that the site is appropriate. Next, verify that you have a place to connect your GoIP device to your LAN in the same location. Both of these are important first steps, or you&#;ll be wasting your time continuing on. Once the connectivity issues are out of the way, turn off your cell , remove the GSM SIM card, and insert it into the GoIP device with the connectors pointing downward. You should hear a click when the SIM card is properly seated. Now connect the device behind a hardware-based firewall/router that provides DHCP service. Plug an Ethernet cable into the LAN port of the GoIP device and connect it to your network. Finally, using the power adapter provided, apply power to the device. Watch the blinking lights. While booting the RUN light will flash on and off every 100 milliseconds. Once the RUN and CHANNEL lights flash GREEN once per second, you&#;re in business. Now use another cellphone to send a text message with the word INFO to the number associated with the SIM card you plugged into the GoIP Device. You should receive a return message telling you the DHCP LAN address associated with the GoIP CHANNEL port where you plugged in the SIM card. Write it down! We&#;re not going to use the PC port so you can ignore its IP address for now.

Asterisk Prerequisites for Today&#;s GoIP Project

We&#;ll be using PIAF-Green with Asterisk 11 and FreePBX 2.11 today so you&#;ll have to read between the lines if you&#;re using a prehistoric release or a non-FreePBX system. We&#;re also assuming you&#;ve installed Incredible PBX&#; 11 which provides the necessary components to get Google&#;s text-to-speech and speech-to-text features working. If you&#;d prefer to roll your own, then start by installing Lefteris Zafiris&#; GoogleTTS and Speech Recognition components for Asterisk. For PBX in a Flash users that aren&#;t using Incredible PBX, you can follow this tutorial to install all of the necessary components in one click.

Initial Setup of FreePBX for the GoIP Device

We&#;ve found that it&#;s easier to configure the FreePBX® side to support the GoIP, and then configure the GoIP unit. There are seven simple steps. If you don&#;t want SMS DISA callback support in your setup, skip the last two steps.

Add GoIP SIP Trunk
Add Custom SIP Settings
Add GoIP Outbound Route
Add GoIP Custom Destination
Add GoIP Misc Application
Add GoIP DISA Context
Add GoIP DISA Misc Application

1. Start by adding a new SIP Trunk to support the GoIP device. Be sure to match the device names we&#;ve shown exactly, or nothing will work. Our special thanks to samyantoun for his initial work on this. Replace 192.168.0.107 with the IP address of your GoIP. Replace 77 with whatever dialing prefix you want to use to make calls through the GoIP trunk. And add the number associated with your GoIP in the Outbound CallerID field. If you&#;re using the GoIP device behind a hardware-based firewall with no Internet port exposure, then you can leave password as the secret. Otherwise, you would want something very secure!

2. Add a couple of custom SIP entries at the bottom of Asterisk SIP Settings to support SMS messaging with Asterisk. Set accept_outofcall_messages=yes and outofcall_message_context=sms_message. Then Submit Changes.

3. Add an Outbound Route to make calls using your GoIP device using the dial prefix you chose for the trunk:

4. Next we need to add a FreePBX Custom Destination to support the Nerd Vittles speech-to-text module which we&#;ll be using to dictate and send SMS messages using any on your PBX. Under Admin -> Custom Destination, add an entry that looks like this:

5. Then we need to associate an extension number with the custom destination we just added. We&#;ve chosen which spells GoIP. Choose Applications -> Misc Application and enter the following:

6. DISA is an Asterisk function that lets someone call into your PBX and obtain dial tone to place an outbound call using the available trunks on your PBX. In the case of the GoIP device, this gets a little fancier. We&#;ll actually be sending an SMS message with a custom password to the GoIP device, and it will in turn call the SMS sender&#;s number and provide DISA dialtone after the user enters a special DISA PIN. Make the PIN and password very secure. We&#;ll get to the password in a minute. On the FreePBX side, add a DISA context in FreePBX under Applications -> DISA that looks something like the following with a secure PIN (not the one in the example):

7. In order to use DISA with GoIP, we&#;ll need an extension associated with the DISA function. We add this number using FreePBX Misc Application. You can use any available extension number you like. Just remember what you chose when we configure the GoIP side to support SMS DISA access. Here&#;s what we use:

Configuration of the GoIP Device

All of the GoIP device configuration is handled using a browser pointed to the internal IP address of the GoIP. If you haven&#;t already done so, send an SMS message with the word INFO to the number associated with your GoIP device. You will get a return message with the private IP address of the unit. Using a browser, point it to the IP address and login with username admin and password admin. It&#;s probably a good idea to reset your unit to factory defaults before beginning the setup just to make sure you&#;re starting with a clean slate. Send an SMS message to the device with the words RESET admin to initialize the hardware.

As we&#;ve mentioned, sending the admin password to the device with the RESET keyword forces a total reset of the device so you obviously want to change this admin password immediately unless you want to risk a total stranger sending a reset command to your device. Do it now under Tools -> Change Password -> Administration Level. It&#;s probably a good idea to change the other passwords as well.

Next, click Configurations. This is the screen on which you set everything. The Preference pane has the country-specific settings for both the network and your cellphone carrier so set them carefully. The IMEI will default to the actual IMEI of your unit. If your cellphone carrier requires registration of a specific IMEI before your SIM card will work, then you can spoof the IMEI using the IMEI of the cell that was previously used with this SIM card. For the East Coast of the United States, our setup looks like this:

If you&#;re using DHCP for the GoIP, the Network Configuration pane shouldn&#;t require any changes. We do recommend that you lock the DHCP address to the GoIP in your router so that it doesn&#;t inadvertently change down the road. You will note that a PPTP VPN tunnel for the device is supported although we haven&#;t yet played with it.

The Call Settings pane has all of your SIP settings for the GoIP. These have got to be right or nothing will work. Our setup (that works) is shown below. Start by clicking on each of the Settings and Preferences links to open up the sub-menus. Both 192.168.0.180 entries should be replaced with the IP address of your Asterisk server. The Number and Authentication ID both need to be goip_1 as shown. The password is password unless you changed your secret in the FreePBX trunk setup. DTMF Signaling should be changed to Outband and DTMF Type should be RFC. Ours still doesn&#;t work reliably, but that may be the lousy cellphone signal in our office. We recommend ULAW and ALAW exclusively for the Audio Codecs. You don&#;t want the overhead of codec translation particularly if you&#;re using a Raspberry Pi. On a normal server, G.729 would obviously reduce the bandwidth of GoIP voice calls. Get it working first and then experiment! The RTP port range should be - to match your Asterisk default setup.

The Call Divert pane is where we configure all of the Nerd Vittles magic. Forward Number(PSTN To VoIP) should be the number on your PBX to which you want inbound GoIP calls forwarded when someone calls the cellphone number associated with your GoIP device. This could be an extension, ring group, IVR, or even the DISA number we set up above. Just be sure you have a verrrrrry secure DISA PIN if you go this route! It&#;s your bill. The SMS Mode must be changed to Relay, and SMS Forward SIP Number must be s to work with the Nerd Vittles apps.

Once you have all of your settings entered, click the Save Settings link under Configurations. The unit will reload its SIP setup. It usually takes about 30 seconds. We recommend you now test the setup to make sure you can make a call to the GoIP number and have it forwarded to an extension on your Asterisk server. Then use an extension on your PBX to place an outbound call using the GoIP dial prefix you assigned above. If either call fails, check your settings for typos in both the FreePBX and GoIP configurations.

Adding the Nerd Vittles Apps to Support the GoIP Device

Now for the fun stuff. We&#;ve built a little shell script that sets up all of the Nerd Vittles applications we outlined above. It&#;s licensed as GPL2 code so you are more than welcome to make any changes or additions which you believe would be useful. We hope you&#;ll share them with the rest of us. The script puts everything in the proper place on Incredible PBX systems to support SMS messaging with Asterisk. You&#;ll be prompted for the following information:

address to which to forward incoming SMS messages
SMS number to which to forward incoming SMS messages
Very secure password to trigger PBX callbacks
Extension number to ring on callbacks

1. When incoming SMS messages are received by the GoIP unit, Asterisk will forward them to this address.

2. When incoming SMS messages are received by the GoIP unit, Asterisk will forward them to this SMS number. You can disable either the forwarding address or the forwarding SMS number (not both!) by editing the [sms_message] context in extensions_custom.conf and commenting out either of these lines with a semicolon:

exten => s,n,system(echo "SMS Message From ${SMSDID}: ${SMSMSG}"...

exten => s,n,MessageSend(sip:goip_1)

3. This password is what must be sent as an SMS message to the GoIP device to trigger a return call from Asterisk. Do NOT include any spaces in the password and make it very secure!

4. This is the extension number that will be used to place the return call from Asterisk. For DISA service, it would be in today&#;s setup. It could also be a regular extension on your PBX if you simply want to trigger a return call from your home or office extension when you send this password via SMS to the GoIP device. Note that the home or office extension must answer the call before the return call will be placed to your SMS device or .

Installation. To install the components (a one-minute job!), log into your server as root and issue the following commands:

cd /root rm GoIP-install.sh wget http://incrediblepbx.com/GoIP-install.sh chmod +x GoIP-install.sh ./GoIP-install.sh

If you ever need to make changes to your setup, just run the script again and answer the prompts.

Kicking the Tires. To make sure everything is working, try sending an SMS message to the GoIP with your secret password from #3 above. You should get a return call within 30 seconds. Next, from an extension on your PBX, dial . Dictate a brief message and then enter a number for delivery of the message via GoIP to some SMS device (not your GoIP unit!). Finally, send a "Hello World" SMS message to your GoIP device. It should be forwarded to both your address (#1) and SMS number (#2) within a few seconds. Enjoy!

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Originally published: Monday, September 30,

We are pleased to once again be able to offer Nerd Vittles&#; readers a 20% discount on registration to attend this year&#;s AC13NERD.

We are pleased to once again be able to offer Nerd Vittles&#; readers a 20% discount on registration to attend this year&#;s 10th Anniversary AstriCon in Atlanta. And, if you hurry, you also can take advantage of the early bird registration discount. Here&#;s the Nerd Vittles Discount Code:

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We&#;ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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