PORTech MV-378: 8 Ports VoIP GSM Gateway

Quantity Order:

Origin:

Taiwan

Payment Method:

Telegraphic Transfer (T/T)

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.

MV-378 IP: 5060 port from Asterisk/ IP PBX
The call automatically switches from a busy line to available line.
* 5060 port can be changed
* just set one sip trunk in asterisk. Simultaneous 8 calls

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.

Major Function
1. VoIP( SIP) , GSM conversion.( MV-378)
2. VoIP( SIP) , CDMA conversion.( MV-378C) - CDMA 2000( 800/ 1900MHz)
3. VoIP( SIP) , UMTS conversion.( MV-378U) for all world and Japan ( SoftBank Mobile/ Docomo)
MV-378U: mobile to lan 2 stage dialing-free mode.
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.
12. All functions can be set on web.
Provide CDR

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