I am running Asterisk 16.20.0 with FreePBX 18.104.22.168 on Google cloud platform.
I have a PJSIP trunk connected to my VSP and a remote PJSIP extension that has a Grandstream Budge Tone-201 handset.
Echo test works fine.
The problem I have is that an incoming call is answered ok by the extension but there is no RTP stream established. If I place the call on hold, the caller hears MoH and I can see that the RTP stream starts using the rtp set debug on command. When I take the caller back off hold, the call RTP stream continues and the audio is OK.
How do I get the RTP stream to establish itself on answer without having to place the caller on hold and retrieving the call?