While I am unable to debug for the moment (the system is live on chan_sip), I will be able to test/debug in a couple of weeks when I am adding a new gigaset handset.
basically, with chan_pjsip, any outbound call out the SIP trunk did not ‘ring’ (no call progress tone on the originating Gigaset handset). When the remote party answered, the call connected and there was audio.
I changed to chan_sip and it was fine.
I could see the 183 session progress etc but for whatever reason it just didn’t work.
I tried pjsip’s inband progress option and that made no difference.
While I am sure it’s probably a bug in the handset’s implementation, I just wanted to bring it up now in case anybody who knows the code might have a lightbulb moment in their head for why this might have happened.