4 Channels VoIP GSM/CDMA/UMTS Gateway
MV-374 is a 4 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,SIP Proxy Server,VoipBuster. It can enable to make 4 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/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)
Major Function
1. VoIP(SIP),GSM conversion.(MV-374)
2. VoIP(SIP),CDMA conversion.(MV-374C) - CDMA 2000(800/1900MHz)
3. VoIP(SIP),UMTS conversion.(MV-374U) for all world and Japan (SoftBank Mobile/Docomo)
MV-374U: mobile to lan 2 stage dialing-free mode.
When calling party call MV-374U 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**
4. 50 sets of LAN --> MOBILE routes setting,50 sets of MOBILE --> LAN routes setting.
-Support one stage diaing
*When lan phone and MV-374 both register SIP proxy Server or Asterisk or VoipBuster, you can dial any destination number from lan phone directly.
*Please note,SIP proxy Server,Asterisk need to have the route of destination number. VoipBuster need to have credit.
-Support free mode-two stage dialing and assigned mode-one stage dialing
5. Voice response for setting and status(dial in from mobile).
6. For call termination (VoIP to GSM/CDMA/UMTS ) and origination ( GSM/CDMA/UMTS to VoIP).
7. Standard SIP(RFC2543,RFC3261) protocol,Communicates with other gateway or PC
8. Receive SMS and Send SMS (CDMA version,sms feature is unavailable)
9. Allows your program Send/receive SMS with AT Command
10. Call Back feature
11. All functions can be set on web.
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
Frequency:
Quad Band:900/1800/1900/850MHZ
3G/UMTS Version for all world and Japen (SoftBank Mobile/Docomo)
3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz
CDMA 2000(800/1900MHZ)
**Please note**
Most CDMA operators don't offer Polarity reversing . So VoIP to Mobile, MV-374 will connect soon. CDMA operators will start billing soon. It doesn't wait mobile side answer.
CDMA Version doesn't support SMS Feature and 180/183 unavailable
