[asterisk-users] Intercepting DTMF to initiate Voice Drop

[Raj Jain]

Transfer does not work for Voice Drop because it adds significant
delay from the time the transfer key is pressed and the transfer
extension is dialed to the playback.

Voice Drop has a critical requirement that the switchover from A's
speaking voice to announcement playback needs to be as seamless as
possible. The person picking up the message on the answering machine
must not be able to detect a gap between the two voices. That is why
this needs to be done in one shot.


On Jan 25, 2008 11:41 AM, Don Pobanz <dpobanz at hastingsutilities.com> wrote:
> From: Raj Jain - Friday, January 25, 2008 10:07 AM
> > I'm trying to implement a Voice Drop service within Asterisk
> > dial-plan. The service is supposed to work as following:
> >
> > 1. A initiates a call to B
> > 2. The call is answered by B's answering machine
> > 3. A hears the answering machine's greeting and the recording beep
> > 4. A speaks a few words into the recording to personalize the message
> > 5. A presses some DTMF keys (say, '##') to initiate Voice Drop
> > 6. PBX intercepts DTMF and starts playing a prerecorded
> > announcement to B
> > 7. A is released from the call as soon as the Voice Drop is initiated
> > 8. PBX releases the call to B at the end of the announcement
> >
> > Any thoughts, ideas?
>
> After talking with B, A could transfer the call to an extension such as
> 123 with a dial plan something like:
>
> Exten => 123,1,Playback(file)
> Exten => 123,n,Playback(file)
> Exten => 123,n,hangup
>
> A will need to be able to transfer outgoing calls ('T' option).
>
> Don Pobanz
>
>
>
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-- 
Raj Jain

mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org