[asterisk-users] Reinvite after DTMF?

[Yuan LIU]

>From: "Wilson Pickett" <spamsucks2005 at gmail.com>
>Date: Thu, 3 May 2007 09:19:25 +0200
>
>On 5/2/07, Yuan LIU <yliu11 at hotmail.com> wrote:
>> >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.)
>
>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?