[asterisk-dev] Attended sip transfers - please test

[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 ?

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