Product Details

MV-374

  • Model:MV-374

Product Introduction


       

 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,Trixbox,3CX,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.   
   
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

 

 Dial Peer Software   Dial Peer Guide  Catalog    User manual     Download SMS Program  
 MV-37X works with Elastix  
 
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.
12.
13. 

  All functions can be set on web. 
Provide CDR
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

 

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