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
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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 internally resolves to 10.1.1.1, the same address as Kamailio, but without the port.
Thanks again. -A
On Thursday, March 26, 2015 05:55:16 PM Vitaliy Aleksandrov wrote:
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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Anthony Messina -
Vitaliy Aleksandrov