[asterisk-users] sip trunking and call transfer

[Raj Jain]

On Sun, Nov 23, 2008 at 5:54 PM, Eric ManxPower Wieling <eric at fnords.org>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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