Portech MV-374G-3G GSM/UMTS/VoIP Gateway, 4x SIM, 1x LAN (Global Version: 3G 2G+3G 800/850/900/1900/2100MHZ)

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)

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 CDR
3. Monitor the signal of all GSM ports

Major Function

1. VoIP(SIP),GSM conversion.
2. VoIP(SIP),UMTS conversion for all world
3. 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
4. Voice response for setting and status(dial in from mobile).
5. For call termination (VoIP to GSM/UMTS ) and origination ( GSM/UMTS to VoIP).
6. Standard SIP(RFC2543,RFC3261) protocol,Communicates with other gateway or PC
7. Receive SMS and Send SMS (2G/3G) 
8. Allows your program Send/receive SMS with AT Command
9. Call Back feature
10. All functions can be set on web.
11. Provide CDR
12. 1 year warranty

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,G.711 a-Law,G.729A,G.729A/B
• Voice Quality,VAD,CNG,AEC,LEC,Packet loss

3G Frequecny:

• 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