Call from SIP Trunk Disconnects in about 30s

Thanks for the insight! I read other topics that discuss this here and here along with your posts also discussing the blockers.

I did find PJSIP documentation on tel: URI at a few places [0] [1] (the last ticket is 12 years ago :eyes: )

At this point, I guess my best bet is to try out sip.conf instead. Please let me know if you have any other ideas, I do not think I will be able to convince my ISP to not use tel: URI