Please show a piece of opensips.cfg where you calling record_route() and SIP debug of such call
-----Original Message----- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Jesse Cloutier Sent: Wednesday, July 20, 2011 7:08 PM To: users@lists.opensips.org Subject: [OpenSIPS-Users] record_route and dialog module confusion Hello, I am a little confused as to how our opensips proxy is fitting into our topology. I have opensips setup as a proxy for dynamic routing and I believe I have it set up for state full routing. Our asterisk server calls the opensips proxy which calls one of our providers based on the dr routing. What confuses me is that all transactions after the initial invite go directly between our provider and the asterisk server. Bypassing the opensips proxy. I am calling record_route() on the calls so shouldnt all the transactions go through the proxy as well? I have also setup the dialog module and my dialogs never get destroyed because opensips never gets the "bye". Is calling record_route enough? Thanks, Jesse Cloutier Network Administrator _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
