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,SIP Proxy Server,VoipBuster. It can enable to make 2 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS network to IP phone.
Catalog (230KB) ¡@ User manual (2.1MB) new!! (11/11/08)

<For sale abroad>
 
MV-372

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Update Firmware: How to update? ¡@Download AT Command

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<<Major Function>>Where to buy?
1. VoIP(SIP)¡BGSM conversion.(MV-372)
2.

VoIP(SIP)¡BCDMA conversion.(MV-372C) - CDMA 2000(800MHz)
3.

VoIP(SIP)¡BUMTS conversion.(MV-372U) for all world and Japan (SoftBank Mobile)
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¡A50 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¡ARFC3261) protocol¡Acommunicates 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.
 
 
<<Specification>> 
Protocols¡GSIP (RFC2543¡ARFC3261)
TCP/IP¡GIP/TCP/UDP/RTP/RTCP/¡BCMP/ARP/RARP/SNTP¡BDHCP/DNS Client¡BIEEE802.1P/Q¡BToS/DiffServ¡BNAT Traversal¡BSTUN¡BuPnP¡BIP Assignment¡BStatic IP¡BDHCP¡BPPPoE
Codec¡GG.711 u-Law¡BG.711 a-Law¡BG.723.1 (5.3k)¡BG.723.1 (6.3k)¡BG.729A¡BG.729A/B
Voice Quality¡BVAD¡BCNG¡BAEC¡BLEC¡BPacket loss

Frequency¡G
Dual BAND:900/1800 MHZ
Tri Band: 900/1800/1900MHZ
Tri Band: 850/1800/1900MHZ
Quad Band:900/1800/1900/850MHZ
3G/UMTS Version for all world and Japen (SoftBank Mobile)
3G:EDGE/GPRS 850, 900, 1800, 1900 MHz / HSDPA/UMTS 850, 1900, 2100 MHz
CDMA 2000(800MHZ)

 
<<How to stack up>
 
<<Application>>

 

<<Example of Application>>

1. How to connect with VoipBuster
Please register to voipbuster first.
(1) When you go oversea,you can receive local call without roaming fee.
When you go oversea,you insert your local sim in the MV-372
And set mobile to lan *,your overseas's phone number.
When you are in oversea,all incoming call of your local mobile will transfer to your mobile number in oversea auto. So you don't need to pay the roaming fee.
voipbuster's rate is very cheap.(landline is free. mobile rate is cheap) http://www.voipbuster.com
(2)When you out of office,you can call any international phone number via MV-372. (If the destination number is landline,you only need to pay the local gsm cost.)
set mobile to lan *,* in the MV-372
When you out of office,you can call the mobile of MV-372.
Then hear dial tone,you can dial any phone number.(even international call. landline is free. mobile rate is cheap)

2.When you call the ch.1 MV-372 gsm number,it will provide dial tone and you enter a destination number.
¡@ Then ch.2 MV-372 will dial this number and connect.
ch.1 MV-372: mobile to lan set route table *,*
ch.2 MV-372:lan to mobile set route table *,#
Additionally, two channels MV-372 both need to register proxy server or Asterisk.
And proxy server/asterisk set the route that the prefix of destination number dial out from ch.2 MV-372.
*The channel 2 MV-372's ip: the first ip + :5062 (e.g http://192.168.0.100:5062)

3. When you call the ch.1 MV-372 gsm number,ch.2 MV-372 will dial this specific number and connect
ch.1 MV-372: mobile to lan set route table *,specific destination number
ch.2 MV-372:lan to mobile set route table *,#
Additionally, two channels MV-372 both need to register proxy server.
And proxy server set the route that the prefix of destination number dial out from ch.2 MV-372.
When you call the ch.1 MV-372 gsm number,ch.2 MV-372 will dial this specific number and connect

 
<< Simple Steps>>

Step 1. Change the Network setting if you need (Network/network setting)
Step 2. Register SIP proxy Server or Asterisk or VoipBuster if you need (sip setting/service domain)
Step 3. Set Route ( request )

mobile to lan:
(1)*,* --->it is two stage dialing.
 when mobile call in,MV-372 will provide dial tone and you can enter ip or asterisk extension or phone number.
 * If you want to enter phone number,please note your asterisk need to have route of destination number.
(2)*, specific extension or IP or phone number
 when mobile call in,MV-372 will connect with this specific extension or IP or phone number auto
 ¡´ If you want to set specific phone number,please note your asterisk need to have route of destination number.
Lan to Mobile:
(1) *,* --->it is two stage dialing.
 when lan phone call in,MV-372 will provide dial tone and you can enter mobile number.
(2)*, specific mobile number
 when lan phone call in,MV-372 will connect with the specific mobile number auto.
(3)*,#--->It is 1 stage dialing
 When lan phone and MV-372 both register Asterisk,
you can dial any destination number from lan phone directly.
 * Please note:Asterisk need to set route of destination number that dial out from MV-372

*The channel 2 MV-372's ip: the first ip + :5062 (e.g http://192.168.0.100:5062)
*All changes both need to click "save and change"

 
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