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8 Channels VoIP GSM/CDMA/UMTS Gateway
MV-378 is a 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. It can enable to make 8 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS networks to IP phone. MV-378 IP:5060 port from Asterisk/IP PBX
The call automatically switches from a busy line to available line.
*5060 port can be changed
*just set one sip trunk in asterisk. Simultaneous 8 calls
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.
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 Dial Peer Software Catalog User manual How to update  Download High Quality Picture
Download AT Command Download SMS Program  Q&A | | | | Major Function | | | 1. | | VoIP(SIP),GSM conversion.(MV-378) | | | 2. | | VoIP(SIP),CDMA conversion.(MV-378C) - CDMA 2000(800/1900MHz) | | | 3. | | VoIP(SIP),UMTS conversion.(MV-378U) for all world and Japan (SoftBank Mobile/Docomo)
MV-378U: mobile to lan 2 stage dialing-free mode.
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 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-378 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:850/900/1800/1900MHZ
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**
1. Most CDMA -2000 operators don't offer Answer signal.
So VoIP to Mobile, MV-370 will connect soon.
CDMA -2000 operators will start billing soon. It doesn't wait mobile side answer
2. CDMA Version doesn't support SMS Feature and 180/183 unavailable
3. CDMA version doesn't have remote SIM feature |
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