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 catch sip traffic on all interfaces use "-i any" option for
tcpdump or "-d any" for ngrep.
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, loose_route()
is returning -1 and the BYE is routed back to Kamailio (10.1.1.1:5060) instead
of Asterisk (10.1.1.1:5080). I am using the stock WITHINDLG route
configuration.
RR module settings are as follows:
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 1)
The BYE from the SIP UAC contains the following Route header which only
contains the contents of Kamailio's Record-Route header. I have attached the
full sip trace for review as well.
Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
What would be the best method to resolve this issue in either Asterisk or
Kamailio? Should I manually add a Record-Route header for the Asterisk
host:port to Kamailio config? Is there something to be done in Asterisk?
Thanks. -A
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