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Portech mV-374U: 4 Ports VoIP 3G Gateway Umts Gateway

Model Number: MV-374U
Country of Origin: Taiwan
Export Markets:
  • North America
  • South America
  • Eastern Europe
  • Southeast Asia
  • Africa
  • Oceania
  • Mid East
  • Eastern Asia
  • Western Europe
  • South Asia
  • Southern Europe
  • Northern Europe
  • Central America
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Portech mV-374U: 4 Ports VoIP 3G Gateway Umts Gateway

Product Specifications/Features
    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

    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.
    12.Provide CDR
    13.24 months warranty

    Specification
    1.Protocols:SIP (RFC2543,RFC3261)
    2.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
    3.Codec:G.711 u-Law,G.711 a-Law,G.729A,G.729A/B
    Voice Quality,VAD,CNG,AEC,LEC,Packet loss

    Frequecny:
    2G Quad Band:850/900/1800/1900MHZ
    3G/UMTS: for all world and Japan (SoftBank Mobile,Docomo)
    3G/UMTS Version: 3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850,2100 MHz

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