[Asterisk-Users] How to monitor DTMF tones in a call?

Moises Silva moises.silva at gmail.com
Sat May 20 13:26:39 MST 2006


You can check that info in www.asterisk.org or voip-info.org

If you have problems applying the patch let me know, may be I can make
you a patch for the 1.2.7.1 specially.

Regards

On 5/19/06, Obelix <asterisklists at adontendev.net> wrote:
> Quoting Moises Silva <moises.silva at gmail.com>:
> Hi,
>
> I am ready to try out this patch, both PlayDTMF and SendDTMF and want to know
> which branch I should work from.
>
> I am not quite experienced with compiling from SVN directly and would like to
> know whether to download the latest 1.2.7.1 and apply the patch to it or use
> the latest from SVN.
>
> Can you give me a list of commands I should apply to SVN?
>
> /Obelix
>
>
> > I have uploaded a patch for some manager events that allow to know
> > when DTMF has been received or sent. Please take a look at this:
> >
> > http://bugs.digium.com/view.php?id=6082
> >
> > and if you can, test it and report feedback. Im having problems to
> > call the attention of bug marshalls for comitting this change. I think
> > this week i will enter to IRC in asterisk-dev to try to make that
> > bugmarshalls pay attention to it.
> >
> > Best Regards
> >
> > On 4/30/06, Obelix <asterisklists at adontendev.net> wrote:
> > >
> > > Is there a way to monitor the DTMF tones on a channel?
> > >
> > > I have a prepaid application working in asterisk. When the user dials a
> > call and
> > > wants to cancel the call before it is answered, there is now way to do it
> > > without hanging up and redialling the access number.
> > >
> > > Is there way to monitor a sequence of DTMF tones and cancel the call?
> > >
> > > If I use a SIP gateway or proxy rather than dial asterisk directly will
> > that be
> > > possible?
> > >
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