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

[Obelix]

Quoting Moises Silva <moises.silva at gmail.com>:

I downloaded and compiled this trunk version - Asterisk SVN-trunk-r28970. The
DTMF events show up in the logging system after I configured logger.conf to
output them, but they are not showing up in the Events.

On checking the SVN for the 6082 patch I saw a branch ../team/jcollie/bug6082.

I don't know what revision that branch is based on. Will compiling that branch
give me the facility?

I am not that familiar with the SVN workings, but if you give the instructions
to follow and may be a revision number or some other parameters to work with I
will be able to do the rest myself.


> 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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