Posted: Fri Mar 28, 2003 12:53 pm Post subject: [Asterisk-Dev] chan_sip.c - fix for Asterisk not on port 506
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There's problems with the SIP channel if Asterisk is running on other than
port 5060 - Asterisk was not including its port number in From and Contact
headers.
Attached is a fix.
I've also been hunting down *s problems with being a client to Free World
Dialup - which boils down to lack of proper handling of
Contact: and Record-Route: in the 200 OK reply.
But in the process I did notice another minor thing: RFC3261 says that
the ACK to a 200 OK should have a new branch in the Via header. See the
example in section 24.2 and also 8.1.1.7. This should probably be fixed
(or, perhaps, the branch= should be left off entirely?)
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