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

[Moises Silva]

Im unable to give you SVN instructions since Im not an SVN
experiencied user, I only issue the commands in asterisk.org to get
the latest sources to generate the patches I want to upload to
bugs.digium.com

I think the best you could do is to adapt the patches to the asterisk
version you are using. Or remind me to do it for you in the weekend ;)


Regards

On 5/21/06, Obelix <asterisklists at adontendev.net> wrote:
> 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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