One way audio (again…)

Hey I guys,

I’m using freepbx 13 but I think I have a general problem:

I have a pjsip-trunk to german Telekom and I have several local sip extensions, all connected by pjsip-channels.
What works:

  • I can make internal calls without any problem
  • I can make incoming calls directly to voicebox or echo-service, without any problem

As soon as I try to make an inbound or outbound call to / from a phone, I only have one-way audio (outgoing audio is working).

As I only use local phones, I don’t see any requirement for nat settings, right?
I also disabled canreinvite (directmedia) as I thought this could make some trouble.

What I don’t understand:
When I’m able to talk from outside to the echo-service than there shouldn’t be a problem for “trunk <-> freepbx”.
When I’m able to make internal calls there should be no problem for “phone <-> freepbx”.
So why can there be a problem with “trunk <-> freepbx <-> phone” when canreinvite is disabled?

THX FOR YOUR HELP IN ADVANCE!
CU
mts

Without seeing logs including the SIP traffic (sip set debug on), console output, and RTP debug (rtp set debug on) nothing immediately springs to mind.