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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.
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.
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
Major Function
1.VoIP(SIP),GSM conversion.(MV-378)
2.VoIP(SIP),CDMA conversion.(MV-378C) - CDMA 2000(800/1900MHz)
VoIP(SIP),UMTS conversion.(MV-378U) for all world and Japan (SoftBank Mobile/Docomo)
MV-378U: mobile to lan 2 stage dialing-free mode.
3.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.
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
Frequecny:
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, 1900,2100 MHz
CDMA 2000(800MHZ/1900MHZ)
**Please note**
1. Most CDMA -2000 operators don't offer Answer signal.
So VoIP to Mobile, MV-378 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
4. CDMA version doesn't support DTMF
Κατηγορίες
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