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Discovery Telecom
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GOIP, Dinstar, Portech, Openvox, DTT
Tel./Fax: +44 20 37691919
E-mail: info@madcom.uk

Portech MV-372

Portech MV-372
Price 450 $
Other currency 421 ˆ

Main characteristics		  
Vendor Portech
Category Home office
Standart 2G/900/1800/850/1900 - VoIP/SIP
SMS(MMS) Yes
Antenna 1
Q'ty of channels 2
Q'ty SIM per channel 1
Connectors FME female, TNC female, SMA female
Interface USB, Ethernet, RJ-45
Management WEB, PC
Dimensions, HWD 170 õ 145 õ 41 mm
Weight 1 kg
Shipment Worldwide

The main functions and characteristics

2 Ports VoIP GSM Gateway MV-372 is a 2 channel VoIP GSM 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)

3. 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. 
    **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


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