Click Fraud PORTech MV-378 - 8 channel GSM/VoIP Gateway
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PORTech MV-378L - 8 channel GSM/VoIP Gateway - 4G LTE

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02/05/2024 £1,268.80 exc. VAT
£1,522.56 inc. VAT
Availability: Usually ships in 1 to 2 weeks
Key Features
  • Standard SIP protocol
  • Send and receive SM/texts
  • Full web browser configuration

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PORTech MV-378L - 8 channel GSM/VoIP Gateway - 4G LTE

Please Note: This product may have a lead time of up to 1-2 weeks. Please call 0330 088 0195 and speak to our sales team who can confirm the current available stock for your requirement

The PORTech MV-378 is a VoIP GSM Gateway with 8 channels that provides call termination - VoIP to GSM  and call origination - GSM to VoIP. The MV-378 has full support for Asterisk and connects as a SIP trunk so you can take advantage of lower cost calls to mobiles using the SIM card. It supports 8 SIMs - therefore supports 8 GSM channels at a time.


  • VoIP (SIP) - GSM conversion
  • GSM - VoIP conversion
  • 8 Simultaneous GSM calls (with 8 SIMs inserted)
  • Capacity for 50 LAN to mobile route settings
  • Capacity for 50 mobile toLAN route settings
  • Voice response for setting and status (dial in from mobile)
  • Standard SIP protocol
  • Send and receive SM/texts
  • Full web browser configuration

It allows the users to make eight simultaneous calls from IP phones to GSM or GSM to IP phones. The PORTech MV-378 is a quad-band device, which should work in most countries.

The Portech MV-378 can take calls from the user - mobile or landline - and forward the call via the net to an IP PBX, VoIP gateway or VoIP service provider and onto SIP phones, analogue phones, PSTN or mobile.

Easy Two Stage Dialling
A user can dial the number of the SIM inside the gateway, the MV-378 then gives the user a dial tone and DTMF signalling is then used to link up the desired destination. The MV-378 then routes the call via VoIP or GSM depending on the configuration. Everything can be set up / configured via the web admin.

When an VoIP phone and the MV-378 both register to the SIP proxy server or Asterisk PBX, you can dial any destination number from the VoIP phone directly.


  • 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
  • GSM: Quad-band : 850/900/1800/1900 MHz
  • Codec: G.711 u-Law/a-Law, G.723.1 (5.3k), G.723.1 (6.3k), G.729A, G.729A/B, Voice Quality, VAD, CNG, AEC, LEC, Packet loss

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