Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
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.miness...@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...@letterboxes.orgwrote:
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
fsconsole 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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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
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
gmar...@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...@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
fsconsole 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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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
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...@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
gmar...@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...@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}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
Ooops, Had not seen you got it in the dialplan...
try to move it after the answer and test again.
Other than this, only thing that comes in my mind is that the
conversion from the pstn to sip (skype partner that gives pstn access)
to skype is ruining the dtmfs beyond recognition... but you said that
at spectral analisys they're fine...
So, I have no idea.
-giovanni
On Wed, Dec 23, 2009 at 7:08 PM, Scott Torr
scott.torr...@letterboxes.org wrote:
You will need to elaborate a bit more?
Not sure where you want me to move the action application=start_dtmf
/ statement to?
Also,
In what way is a sip call handled differently to a skypiax call?
Why would the sip call detect and decode properly?
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=answer /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
regards,
Scott Torr
On Tue, 22 Dec 2009 16:26 +0100, Giovanni Maruzzelli
gmar...@celliax.org wrote:
do as anthm say :-)
On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
add start_dtmf app to your dialplan before bridge to start the inband
dtmf
detector.
On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr scott.torr...@letterboxes.org
wrote:
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com
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IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
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pstn:+19193869900
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[Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
action application=start_dtmf /
action application=record_session
data=/root/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav/
action application=playback data=/root/Hello_16000.wav /
/condition
/extension
fsconsole loglevel 7
If I dial 501 from from a sip phone using inband dtmf I can see the
dtmf tones being detected and decoded by fs in the debug log.
If however I use a pstn phone and dial my skypeIN telephone number the
call comes into fs via skypiax but when I generate dtmf tones on the
phone they are not detected or decoded by fs.
If I take the record_session file and spectrum analyze the recorded
tones appear to be within spec.
Can anybody suggest why this is not working for me?
Is the correct sample rate being used in libteletone_detect.c?
Does the Goertzel algorithm work for other sample rates other than
8000hz?
I'm not sure why I can not get this to work?
regards,
Scott Torr
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
add start_dtmf app to your dialplan before bridge to start the inband dtmf
detector.
On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr
scott.torr...@letterboxes.orgwrote:
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
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
fsconsole 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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--
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:+19193869900
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
do as anthm say :-)
On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale
anthony.miness...@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...@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
fsconsole 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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--
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:+19193869900
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?
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...@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
fsconsole 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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