I have an asterisk 11.6, connected to a voip provider with a classic sip trunk. When I receive an incoming call on this trunk, I sometimes try to terminate it on an external pstn number, using this same trunk to dial out.
When I do this, calls are established correctly (invites, ack, ok), but immediately after the remote server reinvite both legs to have the media directly and take my box out of the rtp loop. Makes sense to me, except that the provider fails somewhere, and there is no audio on the call (the RTP flow doesnt appear at all on the asterisk box - and I am struggling to get an answer from the voip provider.
Is there a way to refuse (at initial call setup) the reinvite, ou reject it when it arrives ?
It looks like directmedia applies to asterisk initiating the reinvites, not handling them.
Any help will be much appreciated,