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 192.168.33.102:5060, 192.168.33.102:5062, wan.ip.address:5060
The 192.168.33.102 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 192.168.33.102. 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 no-reg label GSM ! voice register dn 2 number 8003 name 8003 no-reg 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).
