Hi,
I got a problem with asterisk 1.8.9.2. The same scenario is working fine
in 1.8.8.2.
Asterisk calls a SIP phone via a proxy, proxy phone and asterisk are on
the same LAN, no NAT.
Asterisk sends the INVITE to the proxy, the proxy sends INVITE to the
phone. The phone sends 180 RINGING back to the proxy. The proxy sends
180 RINGING to asterisk. So far so good. If the calling side decides to
cancel the call, asterisk sends the CANCEL directly to the phone. The
phone doesn't find the call and answers 404. In asterisk 1.8.8.2
asterisk sends the CANCEL to the proxy, which sends the CANCEL to the
phone and all ist fine.
I think, the new behavior comes from the lines
parse_ok_contact(p, req);
if (!reinvite) {
build_route(p, req, 1);
}
which are inserted in the handling of provisional SIP response.
Am I doing something wrong or is this a bug?
Thanks,
Karsten
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