[asterisk-dev] SIP transfer committed

[Olle E Johansson]

Friends,

I finally committed the last piece of the new SIP transfer support  
code. This greatly enhances the support of SIP
transfers - or at least is meaning to. The code has been tested on  
1.2 for almost a year in production, but the
trunk version is a port from this. A port in many cases means that  
one introduces new bugs.

This code will be tested heavily this week. If you test it and find  
new bugs, please make sure you report
them on the bug tracker so we can fix this quickly.

This patch affected both the handling of incoming INVITEs and REFERs.  
We now have support for INVITE/Replaces
in relation to REFER. We will have to test what needs to be done to  
support INVITE/Replaces in relation to call pickup,
that is an area I have not been focusing on yet. As kpfleming says: I  
demand that is may or may not work :-)

Thanks to Nuvio, Voop and Foniris for sponsoring the work with SIP  
transfers. It's been a long journey through
masquerading, channel locks and a lot of evil bugs.

/O