[asterisk-dev] CANCEL/BYE problems ? Please test!

[Olle E Johansson]

With the assistance of Leif Madsen (blitzrage) I've located a serious  
problem with CANCEL's in combination
with early media (183 Session Progress).
This caused the outbound call leg to just die, with no proper hangup.  
In some cases, a phone could ring
forever.

If you have this problem, please test one of the following branches:

- invitestate	http://svn.digium.com/svn/asterisk/team/oej/invitestate  
- for 1.2
- invitestate-1.4	http://svn.digium.com/svn/asterisk/team/oej/ 
invitestate-1.4 - for 1.4

and svn trunk for the development branch.

If we can get positive feedback on these changes quickly, I will  
merge the code into 1.2 and 1.4.
This will mean that we will release a new version of 1.2 soon, due to  
many bug fixes in chan_sip.

I've also fixed some issues with T.38 being hangup prematurely due to  
rtp timeouts. Hopefully
this no longer happens.

Thanks for testing!

/Olle