<br><br>
<div><span class="gmail_quote">On 10/16/06, <b class="gmail_sendername">Johansson Olle E</b> <<a href="mailto:olle@voop.com">olle@voop.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br>13 okt 2006 kl. 18.09 skrev Kevin P. Fleming:<br><br>BUT, that's exactly what we're doing in Asterisk
1.4 - all IFs and<br>BUTs regarded, if there's only two SIP endpoints in the<br>call, we will set up the call with RTP media directly between them<br>without a RE-invite.</blockquote>
<div> </div>
<div>Where can I find more information about this patch? </div>
<div> </div>
<div>Can it be disabled if you for some reason want to keep Asterisk in the media path?</div>
<div> </div><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">-- <br>Morten Isaksen<br><a href="http://www.misak.dk/blog/">http://www.misak.dk/blog/</a> </blockquote></div>