[asterisk-dev] Attended sip transfers - please test

[Olle E Johansson]

4 mar 2006 kl. 10.29 skrev Dinesh Nair:

>
>
> On 03/04/06 15:57 Olle E Johansson said the following:
>> 4 mar 2006 kl. 05.46 skrev Dinesh Nair:
>>>
>>> On 03/02/06 23:22 Olle E Johansson said the following:
>>>
>>>> "siptransfer" branch. Check it out from
>>>> http://svn.digium.com/svn/asterisk/team/oej/siptransfer
>>>
>>>
>>> is there any diffs we can use to apply to either trunk or 1.2.4  
>>> to  test this ? downloading the entire subversion repo just to  
>>> test  this is a little difficult.
>>> would it be safe to assume that only a few files were changed   
>>> (chan_sip.c obviously) to handle this ?
>>>
>>>
>> It is not a small patch and would be hard to backport at this stage.
>
> would a drop in replacement of chan_sip.c work then, or have there  
> been changes made further
> down the asterisk stack which would make this just as difficult ?
>
I don't think so. I have to spend my time fixing this new feature for  
the development version instead of
backporting to a release version that will never include this code.  
Sorry, gotta focus.

/O