[asterisk-users] No audio on remote SIP calls

[Jonathan Moore]

Hi everyone.

We have an asterisk server in our main office and phones at each
remote site.  The remote offices are connected via a MPLS which, to my
knowledge has no natting going on.

The problem I have is that any call from a remote phone to a remote
phone (even on the same remote lan) results in no audio.  If I make a
call from the same LAN the asterisk server is on, to one of these
remote sites, I get perfect two way audio.  If I play a call from one
phone to another at a remote site, there is no audio, however, I do
hear messages (such as voicemail, things from Playback(), etc) that
originate on the asterisk server.

I've tried adjusting canreinvite= in sip.conf in hopes in might have
some effect, but so far nothing.

Suggestions on where else to look, or what the problem might be?

Which configs would be useful in troubleshooting?

Thanks.

-jonathan