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

[Giovanni Maruzzelli]

Ooops, Had not seen you got it in the dialplan...

try to move it after the "answer" and test again.

Other than this, only thing that comes in my mind is that the
conversion from the pstn to sip (skype partner that gives pstn access)
to skype is ruining the dtmfs beyond recognition... but you said that
at spectral analisys they're fine...

So, I have no idea.

-giovanni

On Wed, Dec 23, 2009 at 7:08 PM, Scott Torr
<scott.torr.fs at letterboxes.org> wrote:
> 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
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:+19193869900
>> >
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>> >
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618