Using Asterisk 16.6.2 with PJSIP.
I have an issue where when I “forward” a call from my ITSP back to my ITSP there is no audio.
I’ve tried using Progress() and Answer() before dialing the ITSP without much luck.
What’s weird is … when I Playback() and audio file right before Dial()'ing the ITSP the prompt is heard and the two-way audio between the parties works just fine. This is where I’m getting confused … what would Playback() do to the channel?
Also, I’m using MixMonitor() here as well and although it reports in the CLI that the recording has started, the file is 0 bytes unless I do the above.
I assumed that Progress would be where to go to get the RTP packets flowing but no luck.
Not NAT related.
Any advice appreciated