<div class="gmail_quote">On Sun, Nov 23, 2008 at 5:54 PM, Eric ManxPower Wieling <span dir="ltr"><<a href="mailto:eric@fnords.org">eric@fnords.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
The term you are looking for is "reinvite". Reinvites allow two devices<br>
to send audio directly between the two end points of the call. the<br>
SIGNALING stays on the servers, but the audio can be sent directly<br>
between the two end points.</blockquote><div><br></div><div>This still leaves the SIP signaling hairpin on Caller 2's system. </div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
nik600 wrote:<br>>> a) Caller 1 - Trunk A/B - Trunk B/C - Caller3<br>
>><br>
>> or<br>
>><br>
>> b) Caller 1 - Trunk A/C - Caller3<br>
>><br>
>> So:<br>
>><br>
>> is it possible to avoid the scenario a) ?</blockquote><div><br></div><div>Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's session with Caller 2 and send a new INVITE to Caller 3. So, this is how you do it from a SIP protocol perspective. I'm not sure to what extent Asterisk supports this capability.</div>
<div><br></div><div>--</div><div>Raj Jain</div></div>