Re: [SR-Users] Routing PSTN calls from Asterisk through Kamailio to UAC & loose_route

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