[quote=“chris.mylonas”]what is the output of “sip show peers” from the asterisk CLI.
Sounds like the phones are using the public IP address of the asterisk server to register. Can you confirm they are registering to the 10.x.x.x address of asterisk.[/quote]
Not sure i fully understand your question. Asterisk is on a public address, not on 10.x.x.x
The SIP devices are on my LAN (behind a NAT that currently holds a dynamic IP being 184.108.40.206) and hold private IP 10.0.0.10 and 10.0.0.11
This is the result of "sip show peers"
11/11 220.127.116.11 D N A 5441 OK (161 ms)
12/12 18.104.22.168 D N A 5441 OK (151 ms)
I just have tried putting canreinvite=nonat and it working now. I get a direct audio flow without going through the internet.
But it seems that this method would fail if two extensions - each behind a separate NAT and thus on two different private LANs - call each other.
How can I force the canreinvite only when the two extensions are on the same private LAN ?