Good afternoon,
I have a setup with two Transports (Two Ethernet Cards) where the first is the PBX server and the second is my provider SIP trunk.
The first one is 172.xx.xx.0/24 where the PBX server is .254
The second transport is 198.xx.xx.xx
Enabled both under Advanced settings on Asterisk SIP settings where the second is pointed to the trunk transport under trunks.
All networks are listed under Local Networks and asterisk restarted.
The problem is as follows
When i make an outside call from any extension the call goes through fine and i hear audio incoming and outgoing
When i try to call an extension internally no audio goes through and crashes at 30sec (nat issue?)
sngrep shows that the invite is coming from 198.xxx which is wrong i pressume.
When i change the transport to 0.0.0.0/0 udp the audio starts working internally and sngrep shows 172.xx.xx.xx invite which is the correct one.
Funny thing is i have a similar setup on FreePBX16 but not causing me this issue with the extensions.
Is there any easy fix to fool the extensions to think that the invite is coming from 172.xx whilst keeping the 198.xx transport to register my trunk?\
Thank you
*edit keeping it at 0.0.0.0/0 udp seems to register my trunk also whilst working with the audio
is this correct to keep it?