I suspect i’m having an issue with the pjsip ps_endpoints options rewrite_contact and rtp_symmetric. Long story short, sip phone behind ipv4/ipv6 translation. SIP Contact headers contain the ipv6 address but the real source from asterisk point of view is the ipv4 address. The asterisk “pjsip show endpoint” shows proper ipv4 in the “Contact:”, and the SIP call setup works, ie. phone rings, and you can dial out. But once the audio starts, on the asterisk cli you see an error for trying to connect to the ipv6 address which is not possible as the asterisk is on ipv4.
WARNING acl.c: Cannot connect to 2605:b100:349:5cd1:0:4:375a:3e01: Cannot assign requested address