One of our users is able to place calls but has no audio.
Here’s the sip.conf entry + call log with verbose 3, SIP debugging, and RTP debugging turned on:
I can see that RTP traffic is being sent to the wrong address - this even though “nat=yes” is set. None of our other users has this problem, so what could it be?
When I run “sip show peers,” the user’s public IP address is shown, so SIP, at least, is working.