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

[Giovanni Maruzzelli]

Scott,
do as tony wrote,
=====
add "start_dtmf" app to your dialplan before bridge to start the
inband dtmf detector.
=====

-giovanni

On Wed, Dec 23, 2009 at 7:00 PM, Scott Torr
<scott.torr.fs at letterboxes.org> wrote:
> Yes,
> I noticed the Jira for the situation where the where the fs controlled
> skype client generates both an In Band audible DTMF tone and an API
> signal causing potential confusion for devices down the line. If only
> the skype client had an option not the generate the tone in the first
> place that would be good, but then I guess they (skype) think the client
> would only be an end device ;-)
>
> However that is not where I'm having a problem, as I'm purely dealing
> with 'In band' DTMF tones.
>
> The question I had on my mind was did the Skype codec faithfully
> transport the DTMF tones across the network?
>
> http://fs.torr.letterboxes.org/dtmf_compare.html
>
> >From these comparisons I would have to say that there in no major
> filtering or distortion of the DTMF tones when transmitted across the
> Skype network.
>
> So I would have to say that "you can receive calls from skypeIN with
> inband dtmfs".
>
>
> If someone has a different conclusion please let me know.
>
> regards,
> Scott Torr
>
>
> On Tue, 22 Dec 2009 16:25 +0100, "Giovanni Maruzzelli"
> <gmaruzz at celliax.org> wrote:
>> It is probably because mod_skypiax does not analize incoming audio
>> looking for dtmf, because the "normal" call from a Skype client peer
>> sends *both* inband and out of band (signaling) dtmf.
>>
>> So, I choose to only detect out of band (signaling) dtmfs, and ignore
>> possible inband dtmfs (in the audio flow), so to have the most
>> reliable source (signaling) and spare cpu (not analizing the incoming
>> audio).
>>
>> Never tought you can receive calls from skypeIN with inband dtmfs...
>>
>> Open a Jira for this, I'll think about.
>>
>> Also, let me know your toughts...
>>
>> -giovanni
>>
>>
>>
>>
>> On Tue, Dec 22, 2009 at 3:57 PM, 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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>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
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>
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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618