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Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

...ble to access the ARI REST interface of the Asterisk server quite fine on the host machine. I suspect it has to do with RTP not being set up, but all the codec support is there. Here's a log for the SIP request from 192.168.1.50: <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 ---> INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 146.115.163.234:64009 ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Max-Forwards: 70 Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP> To: <sip:6002 at 192.168.1.139;transport=UDP> From:...