GSMs

ITS CGW GSM

Written by Friday, 23 November 2012 10:41
The CGW- VIP is an advanced Cellular-to-VoIP gateway. Unique by its pure-IP and fully managed 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 several connectivity options and many useful applications. It suits perfectly to offices with an IP-PBX as a cost saving Cellular-to-VoIP and VoIP-to-Cellular Gateway. And it can be used to establish a virtual private voice network between branches with toll-free calls.More Of CGW-VIP ApplicationsBusiness continuityLast mile access tool for underserved areasA virtual private voice network for branchesA costs saving gateway       Main FeaturesSupports VoIP, SIP, and GSM connectionsInteroperates with all standard IP-PBX/soft switch equipmentEmbedded web serverLocal and remote management via user-friendly web interfaceLocal and remote upgrades of configuration files and firmwareSupports registration of SIP extensionsEnables creation of a virtual office with remote extensionsBuilt-in firewall and NATBuilt-in Least-Cost-Routing (LCR)CLIP/CLIRInterfacesLAN / WAN 10/100 Base-T, RJ-45VoIPVoice signaling: SIPLocal SIP server/registerCodecs: G.711 PCM (μ/A-Law), G.729*Echo canceler: G.168-2002CellularGSM module: integrated quad band 850/900/1800/1900 MHzGSM module: 2Antenna: 2 external…

2N Helios Easy Gate

Written by Friday, 23 November 2012 10:37
2N® VoiceBlue NextTeach Your IP Exchange to Save MoneyThe 2N® VoiceBlue Next, a successor to the 2N VoiceBlue Lite gateway, belongs to the new generation of VoIP gateways. It brings you a significant reduction in the costs of calls to mobile networks and is compatible with a wide range of IP PBXs. Its simple and intuitive setting via a web interface enables you to quickly set up the VoIP interface as well as the gateway itself.                   The 2N® VoiceBlue is a GSM gateway designed with the goal to reduce costs as much as possible. This is ensured by an automated saving solution able to route the call according to time, pre-code or free minutes. The CallBack service is able to reduce roaming fees on trips abroad. If your company has other branch offices in several countries, you will greatly appreciate the possibility to call local mobile networks completely free of charge using VPN tariffs.Compared to our competitors’ products, the 2N® VoiceBlue offers you a unique advantage…

MV 374 - 378

Written by Friday, 23 November 2012 10:31
4 or 8 Channels VoIP GSM/CDMA/UMTS GatewayMV-374 - 378 is a 4 or 8 channels VoIP GSM/CDMA/UMTS Gateway for call termination (VoIP to GSM/CDMA/UMTS ) and origination ( GSM/CDMA/UMTS to VoIP). It is SIP based and compatible with Asterisk,Trixbox,3CX,SIP Proxy Server,VoipBuster.Option SBK-32 :32 SIMs Remote SIM Bank and SIM Server Connect with PORTech GSM Gateway via internet You can deploy your GSM Gateway in different locations. Centralize and supervise all SIMs in one place.  Work with Dial peer Server (free)1.Dial Peer Server can manage 128 GSM ports at same time   User just need to set one SIP trunk   Send call to dial peer IP:5060 port from Asterisk/IP PBX   The call automatically switches from a busy line to available line.2.Provide CDR3.Monitor the signal of all GSM ports     Major Function1.VoIP(SIP),GSM conversion.(MV-378)2.VoIP(SIP),CDMA conversion.(MV-378C) - CDMA 2000(800/1900MHz)VoIP(SIP),UMTS conversion.(MV-378U) for all world and Japan (SoftBank Mobile/Docomo)MV-378U: mobile to lan 2 stage dialing-free mode.3.When calling party call MV-378U sim card,the calling party will hear dial tone and enter any destination number.**How to differentiate mobile to lan-2 stage dialing is…

MV 370 - 372

Written by Friday, 23 November 2012 10:28
1 or 2 Ports VoIP GSM Gateway / VoIP CDMA Gateway/VoIP UMTS GatewayMV-370 -372  is a 1 Port VoIP GSM/CDMA/UMTS Gateway for call termination (VoIP to GSM/CDMA/UMTS ) and origination (GSM/CDMA/UMTS to VoIP). Support Asterisk,Trixbox,3CX,SIP Proxy Server,VoipBuster.                             Major Function    VoIP(SIP),GSM conversion.(MV-370)    VoIP(SIP),CDMA conversion.(MV-370C) - CDMA 2000(800/1900MHz)    VoIP(SIP),UMTS conversion.(MV-370U) for all world and Japan (SoftBank Mobile,Docomo)    MV-370U: mobile to lan 2 stage dialing-free mode. When calling party call MV-370U sim card,the calling party will hear dial tone and enter any destination number.    **How to differentiate mobile to lan-2 stage dialing is available?**    UMTS Mobile call UMTS Mobile: when the called party answer, the calling party press any DTMF.    If the called party hear DTMF Voice,this feature is available;contrariwise**    50 sets of LAN --> MOBILE routes setting,50 sets of MOBILE --> LAN routes setting.    -Support one stage diaing    *When lan phone and MV-370 both register SIP proxy Server or Asterisk or VoipBuster,you can dial any destination number from lan phone directly.    *Please note,SIP…