[Asterisk-Users] current code release & chan_sip problem/question rport

[Todd Graham]

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

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