From: "Wilson Pickett" <[EMAIL PROTECTED]>
Date: Thu, 3 May 2007 09:19:25 +0200
On 5/2/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
>From: "Wilson Pickett" <[EMAIL PROTECTED]>
>Date: Wed, 2 May 2007 15:30:21 +0200
>
>Is there a way to do the following scenario?
>
>1) my asterisk box receives an incoming call from a toll free number
>provider such as nufone, voicepulse, etc.
>2) It then dials a number via SIP and outputs a DTMF sequence.
At this point, I assume, the destination SIP has not been invited? The
purpose of the DTMF is either determine which SIP destination to invite or
to perform some other dial plan functions.
>ok, that part we do every day.
>
>3) After DTMF though, is it possible to get the two SIP channels
>(original SIP caller plus SIP called) hooked together and have my pbx
>no longer in the call at all?
>
>tia
If the above is true, then there shouldn't be a problem if all other
conditions for reinvite are satisfied, because Asterisk will only execute
Dial at this point, and that Dial could follow with reinvite. (I assume
that
the original SIP caller is in fact the toll free provider.)
So what is in the dialplan once the DTMF is sent? The two channels are
already bridged, how can asterisk then bow out? I don't see a way,
Maybe I missed something here. In my understanding, the only parties in the
call at DTMF stage are the originator and Asterisk. The destination is not
in the picture yet. Is this correct? What is the purpose of the said DTMF
sequence? Do you have a sample dial plan?
Yuan Liu
but
I thought I'd ask if someone else did?
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