MV-370 is a VoIP GSM/CDMA/UMTS Gateway for call termination (VoIP to GSM/CDMA/UMTS ) and origination (GSM/CDMA/UMTS to VoIP). Support Asterisk,Trixbox,SIP Proxy Server,VoipBuster.

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.

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,SIP Proxy Server,VoipBuster. It can enable to make 4 calls simultaneously from IP phones to GSM/CDMA/UMTS networks andGSM/CDMA/UMTS network to IP phone.

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,SIP Proxy Server,VoipBuster. It can enable to make 8 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS network to IP phone

DMT-V: 8/16/24/31 ports VoIP GSM Gateway (support SIP and H.323)
~ with CDR,LCR,Remote Monitor
~ GSM ,CDMA can be combined in one DMT-V

DMT-V accepts incoming call from VoIP and choose one GSM channel to dial out according to the prefix of the destination mobile number. In this way, we can have least cost routing (LCR). DMT can provide Call Detail Record (CDR) for traffic and accounting management. User can remote monitor traffic via VNC

DMT accepts incoming call from E1/T1 PRI of PSTN or PBX and choose one GSM channel to dial out according to the prefix of the destination mobile number. In this way, we can have least cost routing. DMT can provide Call Detail Record (CDR) for traffic and accounting management.
*Besides PRI to GSM,you maybe also apply GSM to PRI (two ways).
GSM Frequecny:
Quad Band:900/1800/1900/850MHZ

~Application 1: E1*2 in , GSM*31 out
When you do call terminal, DMT-2 can accept incoming call from 2 E1/T1 PRI and choose one GSM channel to dial out according to the prefix of the destination mobile number

~Application 2: E1*1 in, GSM*31 and E1*1 out.
DMT-2 is connected between PSTN and PBX using the E1/PRI interface. It routes the mobile calls from PBX through mobile channels instead of going directly to the PSTN, hence reduces higher rate charge for fix to mobile calls. If the call is not a mobile call, it just goes directly to the PSTN. On the other side, the calls come from PSTN will be passed to the PBX. Thus the DMT-2 can perform the LCR(Least Cost Route) function. (More detail as follows)

DuMV@PCI: 2 ports GSM/VoIP PCI Card is Full function (both GSM and VoIP) on board , SIP based and compatible with Asterisk. It can enable to make 2 calls simultaneously from IP phones to GSM/CDMA/UMTS networks and GSM/CDMA/UMTS networks to IP phone.

Beside Voice,2 ports GSM/VoIP PCI Card support SMS sending and SMS receiving on web directly . Also allows your program Send/receive SMS with all AT Command .

Worldwide use ( 2G-850/900/1800/1900 MHz ,3G compatible with all world and Japan,CDMA 2000)