Hi Anthony,
Yes,
The "start_dtmf" application is in the dialplan.
One question I still have is will the Goertzel algorithm in
libteletone_detect.c be able to detect and decode the DTMF tones once
they have past through the PSTN and Skype network traversing various
codecs?
1) They sound audible and clear.
2) A spectrum graph clearly shows the two frequencies.
How bad does the signal need to degrade before the DTMF tones cannot be
detected?
Can you suggest a way to play recordings through the "start_dtmf"
application.
This way I can test various wave forms.
** BUG **
Why does samples=0?
One thing I have noted is that when "start_ivr_async.c" calls:
teletone_dtmf_detect(&pvt->dtmf_detect, frame->data, frame->samples);
for a skypiax call the samples=0
for a SIP call the samples=160
I hope this may help track down the problem.
Perhaps in time with better understanding of the internal workings of fs
and may be able to post solutions rather than problems?
regards,
Scott Torr
On Tue, 22 Dec 2009 09:21 -0600, "Anthony Minessale"
<anthony.miness...@gmail.com> wrote:
> add "start_dtmf" app to your dialplan before bridge to start the inband
> dtmf
> detector.
>
>
> On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr
> <scott.torr...@letterboxes.org>wrote:
>
> > ubuntu-8.04.3-server-amd64.iso (update/upgrade)
> > FreeSWITCH Version 1.0.trunk (15787)
> > skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
> > mod_skypiax
> >
> > (POTS)-->(PSTN)-->(skypeIN)-->(skype_client)-->(skypiax)-->(fs)
> >
> > <extension name="Indial_to_fs_via_skypeIN">
> > <condition field="destination_number" expression="^501$">
> > <action application="start_dtmf" />
> > <action application="record_session"
> >
> >
> > data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
> > <action application="playback" data="/root/Hello_16000.wav" />
> > </condition>
> > </extension>
> >
> >
> > fs>console loglevel 7
> >
> >
> > If I dial 501 from from a sip phone using "inband" dtmf I can see the
> > dtmf tones being detected and decoded by fs in the debug log.
> >
> >
> > If however I use a pstn phone and dial my skypeIN telephone number the
> > call comes into fs via skypiax but when I generate dtmf tones on the
> > phone they are not detected or decoded by fs.
> >
> > If I take the record_session file and spectrum analyze the recorded
> > tones appear to be within spec.
> >
> >
> > Can anybody suggest why this is not working for me?
> >
> >
> > Is the correct sample rate being used in libteletone_detect.c?
> > Does the Goertzel algorithm work for other sample rates other than
> > 8000hz?
> >
> >
> > I'm not sure why I can not get this to work?
> >
> >
> >
> > regards,
> > Scott Torr
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
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