4 Channels VoIP GSM/WCDMA/UMTS Gateway
MV-374 is a 4 channels VoIP GSM/WCDMA/UMTS Gateway for call termination (VoIP to GSM/WCDMA/UMTS ) and origination ( GSM/WCDMA/UMTS to VoIP). It is SIP based and compatible with Asterisk,SIP Proxy Server,VoipBuster. It can enable to make 4 calls simultaneously from IP phones to GSM/WCDMA/UMTS networks and GSM/WCDMA/UMTS networks to IP phone.
- MV-374/MV-378: New version. Build in Dial peer Server,Stun Server and NAT Function.
- 5060 is behalf no
- It has 4/8 lines: 5064,5066,5068,5070,5072,5074,5076,5078
- The call automatically switches from a busy line to available line.
- So user just send call to 5060 port from Asterisk/IP PBX *5060 Port can be changed to any port user needs
New Developping: STDP Box is released in the market
*It's an external box with Dial Peer Server,Stun Server,NAT function all in one that can handle all MV-37X,total volume up to 8 ports each.
(E.g you can handle MV-370*2,MV-372*3 with one STDP Box)
Option SBK-32 :32 SIMs Remote SIM Bank and SIM Server
Connect with PORTech GSM Gateway via internet
SIM cards no longer need to be installed in GSM Gateway anymore;
You can deploy your GSM Gateway in different locations.
Centralize and supervise all SIMs in one place.
Option GSM Booster: BT-918/BT-921
It can improve the cellular signal, simple installation
Work with Dial peer Server (free)
- 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.
- Provide CDR
- Monitor the signal of all GSM ports
- VoIP(SIP),GSM conversion.
- VoIP(SIP),UMTS conversion for all world
- 50 sets of LAN --> MOBILE routes setting,50 sets of MOBILE --> LAN routes setting.
-Support one stage diaing
-Support free mode-two stage dialing and assigned mode-one stage dialing
- Voice response for setting and status(dial in from mobile).
- For call termination (VoIP to GSM/UMTS ) and origination ( GSM/UMTS to VoIP).
- Standard SIP(RFC2543,RFC3261) protocol,Communicates with other gateway or PC
- Receive SMS and Send SMS (2G/3G)
- Allows your program Send/receive SMS with AT Command
- Call Back feature
- All functions can be set on web.
- Provide CDR
- 1 year warranty
- Protocols:SIP (RFC2543,RFC3261)
- 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.729A,G.729A/B
- Voice Quality,VAD,CNG,AEC,LEC,Packet loss
- U Version:2G 850,900,1800,1900MHz,3G 850,2100 MHz
- A Version:2G 850,900,1800,1900MHz,3G 850,1900 MHz
- G Version:2G 850,900,1800,1900MHz,3G 800/850/900/1900/2100HMZ