On Mon, Nov 24, 2008 at 12:14 AM, Raj Jain <[EMAIL PROTECTED]> wrote:
>
> 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
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
>
Maybe because there is no such thing as a "SIP trunk", at least in the
Asterisk world. Most of us call them "peer" or "friend".
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 se
Maybe my question is not clear or is too stupid? (sorry)
Maybe this is already done in SIP trunking?
Or (worste case) is impossible to do that?
Thanks
On Fri, Nov 21, 2008 at 8:53 AM, nik600 <[EMAIL PROTECTED]> wrote:
> Hi to all.
>
> i-ve got a question:
>
> what happen when a call between 2 t
Hi to all.
i-ve got a question:
what happen when a call between 2 trunks is transferred to another trunk?
For example, suppose that i have 4 trunk A,B,C,D:
Caller 1 - Trunk A/B - Caller2
Then Caller 2 transfer to Caller 3 behind Trunk B/C
What happend?
a) Caller 1 - Trunk A/B - Trunk B/C - C
