One way audio (again...)

Hey I guys,

here is my 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?


P.S. freepbx 13