[asterisk-users] sip trunking and call transfer
Eric "ManxPower" Wieling
eric at fnords.org
Sun Nov 23 16:54:01 CST 2008
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 servers, but the audio can be sent directly
between the two end points.
NAT, transcoding, and the T and t options to dial (as well as other
things) will prevent reinvies from happening.
> 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?
> On Fri, Nov 21, 2008 at 8:53 AM, nik600 <nik600 at gmail.com> wrote:
>> 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 - Caller3
>> b) Caller 1 - Trunk A/C - Caller3
>> is it possible to avoid the scenario a) ?
>> Thanks to all
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