[Freeswitch-users] Skypiax: not able to detect "Inband" dtmf tones from pstn call?

[Scott Torr]

You will need to elaborate a bit more?

Not sure where you want me to move the <action application="start_dtmf"
/> statement to?

Also,
In what way is a sip call handled differently to a skypiax call?
Why would the sip call detect and decode properly?

<extension name="Indial_to_fs_via_skypeIN">
 <condition field="destination_number" expression="^501$">
   <action application="start_dtmf" />
   <action application="answer" />
   <action application="record_session"
   data="/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
   <action application="playback" data="/root/Hello_16000.wav" />
 </condition>
</extension>


regards,
Scott Torr


On Tue, 22 Dec 2009 16:26 +0100, "Giovanni Maruzzelli"
<gmaruzz at celliax.org> wrote:
> do as anthm say :-)
> 
> On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale
> <anthony.minessale at 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.fs at 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
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
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> >
> >
> >
> > --
> > Anthony Minessale II
> >
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> >
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> >
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> >
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> 
> 
> 
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
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