Vitaliy, thank you for being a second set of eyes on this. This issue was my
fault completely--I had neglected to remove the fromdomain parameter on the
Asterisk side when I was testing something else, so the calls coming from
Asterisk were of course appearing to come from example.com which
According to your description BYE was sent using the information from
R-URI which had no 5080 port.
Asterisk should have added port 5080 to the outgoing Invite contact so
that it could be used for in-dialog routing.
Can you show a full trace with sip traffic between kamailio and
asterisk. To
I've been working on integration of Asterisk and Kamailio, currently on the
same host with different ports, and have come across a problem with calls that
originate from the Asterisk side (PSTN/DAHDI) and route through Kamailio to a
SIP UAC. In short, when the SIP UAC (10.1.1.9) sends the BYE,
