[asterisk-dev] Attended sip transfers - please test

[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 ?

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)    http://www.alphaque.com/
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