Skypiax: not able to detect "Inband" dtmf tones from pstn call?
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
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
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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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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
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@celliax.org> wrote:
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
_______________________________________________ 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
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@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@celliax.org> wrote:
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
_______________________________________________ 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
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-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
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.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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
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@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
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
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@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@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
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ 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
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@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@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@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
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ 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
_______________________________________________ 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
participants (3)
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Anthony Minessale
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Giovanni Maruzzelli
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Scott Torr