[asterisk-users] Packet2Packet bridging occurring when not wanted

[Cameron Jenkins]

Hi,

I have Asterisk set up on Fedora with a single SIP trunk, with a few
handsets configured. The Asterisk box has both public and private
addressing, so "canreinvite=no" is set on both the SIP trunk and handset
configurations so I can get around the nasty NAT issues.

One odd behaviour I am seeing is certain destinations are resulting in
different SIP codes being sent back to Asterisk, which then initiates
unwanted Packet2Packet bridging.

http://www.pastebin.ca/849961 <--- Working call. Line 280 shows a "SIP/2.0
183 Session Progress" and the RTP stream works as intended. I hung the call
up before being answered in case you were wondering where the answer part of
the debug occurs

http://www.pastebin.ca/849965 <--- Non-working call. Line 235 shows a
"SIP/2.0 180 Ringing", and for some odd reason, Packet2Packet bridging is
initiated despite "canreinvite=no" being set.

This in turn causes me some one way audio behaviour, and I've got no idea
how to fix it. Anyone able to offer any suggestions.

Regards,
Cameron Jenkins