POR Tech MV 374

We bought a PORTech MV-374 four-port GSM-SIP gateway, which I hooked up to a Cisco Call Manager Express. Here's how!

  • Configure IP address.
  • Don't change the username! I did this and locked myself out. I haven't tried just changing the password as yet.

Normally the four channels work as four independent SIP ports (5060, 5062, 5064, 5066). This is sub-optimal if you want to hunt for a free outgoing line (for LAN to mobile) because the CME (apparently) can't hunt past busy SIP peers. Get around this by enabling forwarding.

  • Under Fwd Settings, enable Forward Enable for Mobile 1, 2.
  • Change the URL:Ports to wan.ip.address:5060, wan.ip.address:5062, wan.ip.address:5064
  • Submit, but don't reboot just yet. Change to Mobile 3, 4.
  • Under Fwd Settings, enable Forward Enable for Mobile 3, 4.
  • Change the URL:Ports to,, wan.ip.address:5060

The address is an internal one for the second module.

  • Configure SIP registration under Service Domain. Make mobile 1 and 2 active and set the display, user, and register names and password to something different for each mobile. I recommend using only numbers, eg make all four fields 8002 for mobile 1 (on my four digit extension system)
  • Set Proxy Server to CME's IP address

Note, there's a problem with mobile 3 and 4 registration as the SIP messages appear to come from I haven't worked out how to fix this yet. It may be possible to change the IP to something on the same subnet as the WAN interface (by logging in to the second module at http://wan.ip.address:8080) and then fiddling with the master/slave settings, but this looked awfully painful and the manual recommended not touching it. I can live with just two Mobile to LAN lines for now. In fact I tried doing this and it didn't work at all - it wasn't possible to bridge the second module's interface and avoid NAT. Furthermore, changing the IP doesn't seem to be supported and the technical support people also advised that I don't do it.

  • Under Mobile to LAN, set up routes as appropriate. Asterisks for both CID and URL mean that any incoming number can make any outgoing call. This is okay for testing.
  • On the CME, set up SIP and add in users for the two mobile channels
 voice register global
mode cme
  source-address xxx.xxx.xxx.xxx port 5060
  authenticate register
 voice register dn  1
  number 8002
  name 8002
  label GSM
 voice register dn  2
  number 8003
  name 8003
  label GSM
 voice register pool  1
  id mac 0003.7E00.38DB
  number 1 dn 1
  dtmf-relay rtp-nte
  username 8002 password 8002
  codec g711ulaw
 voice register pool  2
  id mac 0003.7E00.38DB
  number 1 dn 2
  dtmf-relay rtp-nte
  username 8003 password 8003
  codec g711ulaw

Now add in dial-peers for each number you want to route through the GSM gateway

 dial-peer voice 2 voip
  destination-pattern 04........
  session protocol sipv2
  session target ipv4:wan.ip.address
  dtmf-relay rtp-nte
  codec g711ulaw
  no vad

If you're doing this with a group plan that has cheap/free calls between numbers on the same account, just set up an identical dial-peer (with a unique sequence number) for each number (eg, destination-pattern 0499999999).