On Sun, Nov 23, 2008 at 5:54 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]>wrote:
> The term you are looking for is "reinvite". Reinvites allow two devices > to send audio directly between the two end points of the call. the > SIGNALING stays on the servers, but the audio can be sent directly > between the two end points. This still leaves the SIP signaling hairpin on Caller 2's system. > nik600 wrote: > >> a) Caller 1 - Trunk A/B - Trunk B/C - Caller3 > >> > >> or > >> > >> b) Caller 1 - Trunk A/C - Caller3 > >> > >> So: > >> > >> is it possible to avoid the scenario a) ? 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. -- Raj Jain
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