16 okt 2006 kl. 13.52 skrev Morten Isaksen:
On 10/16/06, Johansson Olle E <[EMAIL PROTECTED]> wrote:
13 okt 2006 kl. 18.09 skrev Kevin P. Fleming:
BUT, that's exactly what we're doing in Asterisk 1.4 - all IFs and
BUTs regarded, if there's only two SIP endpoints in the
call, we will set up the call with RTP media directly between them
without a RE-invite.
Where can I find more information about this patch?
Hmmm. Not very well documented. Mark created it in Pisa earlier this
year.
Can it be disabled if you for some reason want to keep Asterisk in
the media path?
Canreinvite = no - like always.
/O
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