----- Original Message -----
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
