VoIP Mailing List Archives

[rj2807 at gmail.com]

Hi,

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
To acheive this I need to intercept DTMF in the middle of a call and
initiate an action based on that. I couldn't find an option in the
Dial() application to break out of it on receipt of a particular DTMF
sequence. Does the Dial() application support such a capability?

I've tried the 'G' option in the Dial() application but that splits
the call as soon as it is answered, whereas, I need to split the call
after it is established based on a DTMF stimulus. Are there any other
ways of accomplishing this goal?

Any thoughts, ideas?

Thank you,

Raj Jain

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

[dpobanz at hastingsuti...]

From: Raj Jain - Friday, January 25, 2008 10:07 AM

[rj2807 at gmail.com]

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:

[atis at iq-labs.net]

On 1/25/08, Raj Jain <rj2807 at gmail.com> wrote: