VoIP GSM Gateways Fixed Cellular Terminal FCT

NEW PRODUCT**PLEASE NOTE THIS PRODUCT HAS A DELIVERY LEAD TIME OF 7-10 Days**
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. 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.

With 1SIM, 1LAN, SMS for GSM-VoIP connection, GSM termination & origination, SMS sending & receiving, Dual band, Tri Band 900 1800 1900MHz, Multi band 850 1900 1800MHz, SIP Asterisk compatible; SC-375C: CDMA; SC-385C: CDMA 2SIM
VOIP (SIP) - GSM conversion | 50 sets of LAN GSM MOBILE routes setting | 50 sets of GSM MOBILE LAN routes setting. | Voice response for setting and status (dial in from mobile). | For call termination (VOIP to GSM) and origination (GSM to VOIP). | Standard SIP (RFC2543, RFC3261) protocol, communicates with other gateway or PC. | All functions can be set on web. | Protocols: SIP(RFC2543, RFC3261) | 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.711a-Law G.723.1(5.3k)G.723.1(6.3k) G.729A G.729A/B | Voice Quality VAD CNG AEC LEC Packet loss | Dual band: EGSM 900/DCS 1800 MHZ Speech Service with EFR (Enhance Full Rate)/FR (Full Rate)/HR(Half Rate) Codec. Tri band: 900/1800/1900MHz, Multi band: 850/1800/1900MHz | SC-375sms: SMS sending & receiving
With 1SIM, 1LAN, SMS for GSM-VoIP connection, GSM termination & origination, SMS sending & receiving, Dual band, Tri Band 900 1800 1900MHz, Multi band 850 1900 1800MHz, SIP Asterisk compatible; SC-375C: CDMA; SC-385C: CDMA 2SIM
VOIP (SIP) - GSM conversion | 50 sets of LAN GSM MOBILE routes setting | 50 sets of GSM MOBILE LAN routes setting. | Voice response for setting and status (dial in from mobile). | For call termination (VOIP to GSM) and origination (GSM to VOIP). | Standard SIP (RFC2543, RFC3261) protocol, communicates with other gateway or PC. | All functions can be set on web. | Protocols: SIP(RFC2543, RFC3261) | 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.711a-Law G.723.1(5.3k)G.723.1(6.3k) G.729A G.729A/B | Voice Quality VAD CNG AEC LEC Packet loss | Dual band: EGSM 900/DCS 1800 MHZ Speech Service with EFR (Enhance Full Rate)/FR (Full Rate)/HR(Half Rate) Codec. Tri band: 900/1800/1900MHz, Multi band: 850/1800/1900MHz | SC-375sms: SMS sending & receiving

The PORTech MV-370 is a single channel VoIP GSM Gateway which allows for call termination (VoIP to GSM) and call origination (GSM to VoIP.) The gateway fully supports Asterisk, it connects as a SIP trunk and allows you take advantage of low cost calls to mobile phones via the SIM card. The MV-370 supports 1 SIM card so supports one GSM channel at a time. The PORTech MV-370 is a quad-band device, which should work in most territories worldwide.
The MV-370 can receive calls from the user (via mobile or landline) and then forward the call via the Internet to an IP PBX, VoIP gateway or ITSP (such as ourselves) and then onto SIP phones, analog phones, PSTN or a mobile phone.
Simple 2-stage dialling:- The user dials the number of the SIM thats inside the MV-370, the gateway then presents the user with a dial tone, and DTMF signalling is then used to pick up the desired destination. The MV-370 then routes the call via GSM or VoIP depending on the settings that have been configured. All settings can be configured via a Web interface.
When an IP phone and the MV-370 both register to the SIP proxy Server or Asterisk server, you can dial any destination number from IP phone directly.
Key Features:
•VoIP (SIP) - GSM conversion
•GSM - VoIP conversion
•Space for 50 mobile toLAN route settings
•Space for 50 LAN to mobile route settings
•Voice response for setting and status (dial in from mobile)
•Standard SIP protocol
•Full web browser configuration
•Send and receive SMS
Specification
•Protocols: SIP (RFC2543, RFC3261)
•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/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: Quad-band: GSM 850/900/1800/1900 MHz,
•Speech Service with EFR (Enhance Full Rate)/FR (Full Rate)/HR (Half Rate) Codec.
With 1SIM, 1LAN, SMS for GSM-VoIP connection, GSM termination & origination, SMS sending & receiving, Dual band, Tri Band 900 1800 1900MHz, Multi band 850 1900 1800MHz, SIP Asterisk compatible; SC-375C: CDMA; SC-385C: CDMA 2SIM
VOIP (SIP) - GSM conversion | 50 sets of LAN GSM MOBILE routes setting | 50 sets of GSM MOBILE LAN routes setting. | Voice response for setting and status (dial in from mobile). | For call termination (VOIP to GSM) and origination (GSM to VOIP). | Standard SIP (RFC2543, RFC3261) protocol, communicates with other gateway or PC. | All functions can be set on web. | Protocols: SIP(RFC2543, RFC3261) | 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.711a-Law G.723.1(5.3k)G.723.1(6.3k) G.729A G.729A/B | Voice Quality VAD CNG AEC LEC Packet loss | Dual band: EGSM 900/DCS 1800 MHZ Speech Service with EFR (Enhance Full Rate)/FR (Full Rate)/HR(Half Rate) Codec. Tri band: 900/1800/1900MHz, Multi band: 850/1800/1900MHz | SC-375sms: SMS sending & receiving

SC-385 is a 2-channel VoIP GSM Terminal for call termination (VoIP to GSM) and origination (GSM to VoIP). It is SIP based, compatible with Asterisk. It will allow you to make 2 calls simultaneously from VoIP to GSM networks or from GSM network to IP VoIP devices. If the SC-385 is registered to an IP PBX or Wifi IP PBX as Extension Numbers, users can extend IP Phone application to GSM Mobile Phone for VoIP GSM connection.

Main Features:
VoIP (SIP) GSM conversion. (SC-385)
50 sets of LAN -> MOBILE routes setting,
50 sets of MOBILE -> LAN routes setting.
Voice response for setting and status (dial in from mobile).
For call termination (VoIP to GSM) and origination (GSM to VoIP).
Standard SIP (RFC2543, RFC3261) protocol, communicates with other gateway or PC.
All functions can be set on web.

Specification:
Protocols: SIP (RFC2543, RFC3261) TCP/IP: IP/TCP/UDP/RTP/RTCP/CMP/ARP/RARP/SMTP 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 (SC-385): Quad band: 900/1800/850/1900MHzWith 1SIM, 1LAN, SMS for GSM-VoIP connection, GSM termination & origination, SMS sending & receiving, Dual band, Tri Band 900 1800 1900MHz, Multi band 850 1900 1800MHz, SIP Asterisk compatible; SC-375C: CDMA; SC-385C: CDMA 2SIM
VOIP (SIP) - GSM conversion | 50 sets of LAN GSM MOBILE routes setting | 50 sets of GSM MOBILE LAN routes setting. | Voice response for setting and status (dial in from mobile). | For call termination (VOIP to GSM) and origination (GSM to VOIP). | Standard SIP (RFC2543, RFC3261) protocol, communicates with other gateway or PC. | All functions can be set on web. | Protocols: SIP(RFC2543, RFC3261) | 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.711a-Law G.723.1(5.3k)G.723.1(6.3k) G.729A G.729A/B | Voice Quality VAD CNG AEC LEC Packet loss | Dual band: EGSM 900/DCS 1800 MHZ Speech Service with EFR (Enhance Full Rate)/FR (Full Rate)/HR(Half Rate) Codec. Tri band: 900/1800/1900MHz, Multi band: 850/1800/1900MHz | SC-375sms: SMS sending & receiving

2 SIM Cards
• SIP Connectivity
• LCR Functionality
• STUN Server Support
• Minute Management
• Auto Clip Routing
• Optional SMS server (XAPI SMS)The Smartest Way from IP to GSM Networks

2N VoiceBlue Lite - the first professionally built VoIP GSM Gateway! It is an ideal complementary product to any SIP-based IP PBX. It allows the user to get from the IP to a GSM network and vice versa very cheaply. The Ethernet (RJ 45) connection is a significant advantage in most cases.

2N VoiceBlue Lite offers VoIP services (internet telephony), all functions and advantages of digital GSM gateways and many other features. Thanks to its efficient LCR, the VoiceBlue Lite VoIP to GSM gateway (Fixed Cellular Terminal) always chooses the cheapest route to any GSM network used. Gaining a 100% control of all outgoing GSM calls is one of the main advantages of our VoIP GSM SIP Gateway compared with other, "home-made" solutions. With the help of voice prompts and efficient CLIP routing, 2N VoiceBlue Lite always routes GSM calls to the right IP phone.
By connecting 2N VoiceBlue Lite to your corporate SMS server (VoIP company), you can use SMS as an additional communication tool for your clients. SMS delivery reports are obtained automatically.

The 2N VoiceBlue Lite Fixed Cellular Terminal has been successfully tested and proved as compatible with following systems:

Cisco Call Manager
Alcatel Omni PCx Enterprise systems
Siemens Hi Path 3 000, 3 500, 3 800 PBX system
Asterisk PBX system

OnDo IP PBX
Televentage equipment

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

The PORTech MV-372 is a dual channel VoIP GSM Gateway which allows for call termination (VoIP to GSM) and call origination (GSM to VoIP.) The gateway fully supports Asterisk, it connects as a SIP trunk and allows you take advantage of low cost calls to mobile phones via the SIM card. The MV-372 supports 2 SIM cards so supports two GSM channels at a time. It allows the users to make two simultaneous calls from IP phones to GSM or GSM to IP phones. The PORTech MV-372 is a quad-band device, which should work in most territories worldwide. The MV-372 can receive calls from the user (via mobile or landline) and then forward the call via the Internet to an IP PBX, VoIP gateway or ITSP (such as ourselves) and then onto SIP phones, analog phones, PSTN or a mobile phone. Simple 2-stage dialling:- The user dials the number of the SIM thats inside the MV-372, the gateway then presents the user with a dial tone, and DTMF signalling is then used to pick up the desired destination. The MV-372 then routes the call via GSM or VoIP depending on the settings that have been configured. All settings can be configured via a Web interface. When an IP phone and the MV-372 both register to the SIP proxy Server or Asterisk server, you can dial any destination number from IP phone directly.

Key Features:
•VoIP (SIP) - GSM conversion
•GSM - VoIP conversion
•2 Simultaneous GSM calls (with 2 SIMs inserted)
•Space for 50 mobile toLAN route settings
•Space for 50 LAN to mobile route settings
•Voice response for setting and status (dial in from mobile)
•Standard SIP protocol
•Full web browser configuration
•Send and receive SMS
•Quad-band

Specification
•Protocols: SIP (RFC2543, RFC3261)
•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/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: Quad-band: 850/900/1800/1900 MHz

4 SIM Cards
• SIP Connectivity
• LCR Functionality
• STUN Server Support
• Minute Management
• Auto Clip Routing
• Optional SMS server (XAPI SMS)The Smartest Way from IP to GSM Networks

2N VoiceBlue Lite - the first professionally built VoIP GSM Gateway! It is an ideal complementary product to any SIP-based IP PBX. It allows the user to get from the IP to a GSM network and vice versa very cheaply. The Ethernet (RJ 45) connection is a significant advantage in most cases.

2N VoiceBlue Lite offers VoIP services (internet telephony), all functions and advantages of digital GSM gateways and many other features. Thanks to its efficient LCR, the VoiceBlue Lite VoIP to GSM gateway (Fixed Cellular Terminal) always chooses the cheapest route to any GSM network used. Gaining a 100% control of all outgoing GSM calls is one of the main advantages of our VoIP GSM SIP Gateway compared with other, "home-made" solutions. With the help of voice prompts and efficient CLIP routing, 2N VoiceBlue Lite always routes GSM calls to the right IP phone.
By connecting 2N VoiceBlue Lite to your corporate SMS server (VoIP company), you can use SMS as an additional communication tool for your clients. SMS delivery reports are obtained automatically.

The 2N VoiceBlue Lite Fixed Cellular Terminal has been successfully tested and proved as compatible with following systems:

Cisco Call Manager
Alcatel Omni PCx Enterprise systems
Siemens Hi Path 3 000, 3 500, 3 800 PBX system
Asterisk PBX system

OnDo IP PBX
Televentage equipment

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

. 2 SIM Cards
. H323 & SIP Connectivity
. LCR Functionality
. STUN Server Support
. Minute Management
. Auto Clip Routing
. Optional E]Mail/SMS Server
. Optional SIP Proxy (Soft PBX)
. Optional Internet GatewayThe Most Effective Way from IP to GSM


2N VoiceBlue Enterprise is a VoIP-to-GSM Gateway supporting SIP and SIP Proxy. It is a complementary product to any SIP based IP PBX and can also be used as a substitute to any SIP based IP PBX because the SIP proxy server can be included.

The 2N VoiceBlue Enterprise fixed cellular terminal can be used not only as a gateway to GSM but also as a SIP based IP PBX.
As VoiceBlue Enterprise uses the Siemens MC55 module, it is capable of data transfer (GPRS technology is implemented) and therefore can serve as a gateway from your LAN to the Internet. A firewall is embedded to provide maximum security.


All these features can be active and physically used at the same time. The whole system can be supplied as a "puzzle", which means that the customer chooses only the features he wants to have in the gateway. The customer shall pay no money for the features that he doesn't need.


With the latest version of web based SMS Server you will gain the advantage of a full review of your messages, whether they were delivered and exactly when.

The 2N VoiceBlue Enterprise Fixed Cellular Terminal has been successfully tested and proved as compatible with following systems:
Cisco Call Manager
Alcatel Omni PCx Enterprise systems
Siemens Hi Path 3 000, 3 500, 3 800 PBX system
Asterisk PBX system

OnDo IP PBX
Televentage equipment
Features:
SIP/SIP proxy
Up to 4 GSM channels
Standard SIP client embedded in one hardware unit
LCR according to free minutes and GSM providers
SIP proxy - REGISTRAR for IP phones included
Intelligent incoming call routing
DISA for incomming calls from GSM
Top voice quality (EFR super sound)

SMS sending and receiving (WEB interface)
Simple web based configuration
CDR buffer for up to 500,000 records
LOG and statistics saving
Integrated antenna splitter
Worldwide use (all frequencies are supported)

. 4 SIM Cards
. H323 & SIP Connectivity
. LCR Functionality
. STUN Server Support
. Minute Management
. Auto Clip Routing
. Optional E]Mail/SMS Server
. Optional SIP Proxy (Soft PBX)
. Optional Internet GatewayThe Most Effective Way from IP to GSM


2N VoiceBlue Enterprise is a VoIP-to-GSM Gateway supporting SIP and SIP Proxy. It is a complementary product to any SIP based IP PBX and can also be used as a substitute to any SIP based IP PBX because the SIP proxy server can be included.

The 2N VoiceBlue Enterprise fixed cellular terminal can be used not only as a gateway to GSM but also as a SIP based IP PBX.
As VoiceBlue Enterprise uses the Siemens MC55 module, it is capable of data transfer (GPRS technology is implemented) and therefore can serve as a gateway from your LAN to the Internet. A firewall is embedded to provide maximum security.


All these features can be active and physically used at the same time. The whole system can be supplied as a "puzzle", which means that the customer chooses only the features he wants to have in the gateway. The customer shall pay no money for the features that he doesn't need.


With the latest version of web based SMS Server you will gain the advantage of a full review of your messages, whether they were delivered and exactly when.

The 2N VoiceBlue Enterprise Fixed Cellular Terminal has been successfully tested and proved as compatible with following systems:
Cisco Call Manager
Alcatel Omni PCx Enterprise systems
Siemens Hi Path 3 000, 3 500, 3 800 PBX system
Asterisk PBX system

OnDo IP PBX
Televentage equipment
Features:
SIP/SIP proxy
Up to 4 GSM channels
Standard SIP client embedded in one hardware unit
LCR according to free minutes and GSM providers
SIP proxy - REGISTRAR for IP phones included
Intelligent incoming call routing
DISA for incomming calls from GSM
Top voice quality (EFR super sound)

SMS sending and receiving (WEB interface)
Simple web based configuration
CDR buffer for up to 500,000 records
LOG and statistics saving
Integrated antenna splitter
Worldwide use (all frequencies are supported)