Gateway Voip SIP-GSM Portech MV372 (2 channel GSM)

Portech MV372

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Gateway Voip SIP-GSM Portech MV372 (2 channel GSM)

MV-372 is a 2 channel 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 2 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS networks to IPphone.

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Gateway Voip SIP-GSM Portech MV372 (2 channel GSM)

2 Ports VoIP GSM Gateway / VoIP CDMA Gateway/VoIP UMTS Gateway

MV-372 is a 2 channel 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 2 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS networks to IP phone.

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.

Major Function

1. VoIP(SIP),GSM conversion.(MV-372)

2. VoIP(SIP),CDMA conversion.(MV-372C) - CDMA 2000(800/1900MHz) VoIP(SIP),UMTS conversion.(MV-372U) for all world and Japan (SoftBank Mobile,Docomo) MV-372U: mobile to lan 2 stage dialing-free mode. When calling party call MV-372U sim card,the calling party will hear dial tone and enter any destination number.

3. **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-372 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 all AT Command

10. Call Back feature

11. All functions can be set on web.

12. Provide CDR

13. 24 months 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
  • Frequency: Quad Band:850/900/1800/1900MHZ  3G/UMTS Version for all world and Japan (SoftBank Mobile/Docomo) 3G:EDGE/GPRS / HSDPA/UMTS CDMA 2000(800/1900MHZ) 

**Please note** 1. Most CDMA -2000 operators don't offer Answer signal. So VoIP to Mobile, MV-372 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