Todd Graham
2004-Jun-18 21:05 UTC
[Asterisk-Users] current code release & chan_sip problem/question rport
Updated to the latest code release of * today. After compiling and reinstalling the SIP dialout connections through our media gateway stopped working. Finally tracked down the issue. In chan_sip.c in transmit_invite there was ;rport added to the INVITE via line of the msg: snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x;rport", inet_ntoa(p->ourip), ourport, p->branch); The old code did not have that ;rport, it ends with the branch . Can anyone explain what that does? I have taken it out, recompiled and can now make outbound calls again. Todd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040618/b44f83fa/attachment.htm
Brian K. West
2004-Jun-18 21:09 UTC
[Asterisk-Users] current code release & chan_sip problem/question rport
Modified Files: chan_sip.c Log Message: Enable support for RFC3581 (bug #1862) bkw ----- Original Message ----- From: Todd Graham To: asterisk-users@lists.digium.com Sent: Friday, June 18, 2004 11:05 PM Subject: [Asterisk-Users] current code release & chan_sip problem/question rport Updated to the latest code release of * today. After compiling and reinstalling the SIP dialout connections through our media gateway stopped working. Finally tracked down the issue. In chan_sip.c in transmit_invite there was ;rport added to the INVITE via line of the msg: snprintf(p->via, sizeof(p->via), "SIP/2.0/UDP %s:%d;branch=z9hG4bK%08x;rport", inet_ntoa(p->ourip), ourport, p->branch); The old code did not have that ;rport, it ends with the branch . Can anyone explain what that does? I have taken it out, recompiled and can now make outbound calls again. Todd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040618/0881b984/attachment.htm
Todd Graham
2004-Jun-18 21:17 UTC
[Asterisk-Users] current code release & chan_sip problem/question rport
Thanks Brian. Makes sense now. I guess I need to forward this extension info on to our media gateway vendor to get them to include in their SIP stack. Todd ----- Original Message ----- From: Brian K. West To: asterisk-users@lists.digium.com Sent: Friday, June 18, 2004 9:09 PM Subject: Re: [Asterisk-Users] current code release & chan_sip problem/question rport Modified Files: chan_sip.c Log Message: Enable support for RFC3581 (bug #1862) bkw