do as anthm say :-) On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale <anthony.minessale@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@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 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618