Hi, haven't check your trace, but a fast guess is that you do not fix the contact of the 200 OK re-INVITE is no fixed and carries back to asterisk a private contact.
So, do fix_nated_contact() for the replies coming from behind a NAT too.. Regards, Bogdan wrote: > Hi, > > My setup: > - 11.22.33.44 : openSIPS 1.6.3 > - 11.22.33.45 : one of the Asterisk 1.6.2.13 servers > - 88.77.66.55 : my public ip-address > - 192.168.1.10 : my local ip-address (NAT) > > All is working well except Session Timers where the Re-Invite > originates from Asterisk. > > I have a SIP trace ( http://pastebin.com/raw.php?i=NRDdaktn ) of a > call initiated by a softphone on my pc (192.168.1.10). > When Asterisk sends the Re-Invite (line 290) my softphone receives > this Re-Invite correctly. > The 100 Trying and 200 OK are also handled as it should. > But on line 455 you see openSIPS forwarding the ACK to 192.168.1.10 > instead of 88.77.66.55. > > Does anyone know why this isn't working? > Thanks in advance! > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users