[asterisk-users] Call flash transfer

[Al lists]

Doesnt reinvite still keeps asterisk in SIP path?
lt looks like all it does is making both peers send rtp to each other.

what i'm trying to do is, a sip peer sends invite to asterisk, asterisk
calls another phone and in case of no answer, "tells peer" to send this call
to another number outside and get off completly,
why i'm trying to do this, first i have limited number of calls i can put on
the incoming peer, with this i can tell the peer to call someone else and
not using my trunk limited number of calls, also final destination number
will see the original caller id instead of overwritten caller id of
asterisk.
hope it makes sense



On Sat, Mar 12, 2011 at 5:47 PM, C F <shmaltz at gmail.com> wrote:

> Reinvite its called
>
> On Sat, Mar 12, 2011 at 1:22 PM, Al lists <asteriskal at gmail.com> wrote:
> > is there a way to have asterisk to flash transfer the call and not being
> in
> > sip path anymore?
> > for example, a sip trunk send a call to asterisk, asterisk rings a
> handset,
> > then sends/flash hooks the call to a cell phone through sip trunk, and
> not
> > being in path anymore?
> >
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