[asterisk-dev] Attended sip transfers - please test

[Olle E Johansson]

Friends,

Today I've released the first version of the code for the major  
rewrite of SIP REFER support in Asterisk.
This is code that I spent many, many hours working on last summer,  
and Terry Wilson
spent many, many hours testing, coming with suggestions, being  
frustrated with and
on me and finally drinking beer with me to celebrate when it worked.
(We had to be quick, since bugs and crashes where frequent)

One version of this code is in production, but for general use it  
still needs some polishing and
finishing up. Thanks to support from Voop A/S, Foniris Telecom and  
others I can now start to
do that. The bulk of the work was funded by Nuvio, Inc.

This patch is inspired from and in a small part based on Anthms  
earlier patch in the bug
tracker.

Some of the changes:

- We do not always send 200 OK to REFERs
- Transfers are handled by the dial plan
- We set channel variables so you can handle transfers in the dial plan
- Some transfer security implemented, more to come
- Support for INVITE with Replaces
- SIP text messaging on call parking

I will open a new bug report to handle the testing of this code. It  
is now available in the
"siptransfer" branch. Check it out from
http://svn.digium.com/svn/asterisk/team/oej/siptransfer

There are tons of comments within chan_sip that tells you more about  
this work.

At some point, I will integrate this into the test-this-branch branch  
as well, but not yet.

Regards,
/Olle


* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden