[asterisk-users] Reinvite after DTMF?

[Yuan LIU]

>From: "Wilson Pickett" <spamsucks2005 at gmail.com>
>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.)

Yuan Liu