[Freeswitch-users] Audio only in one direction when calling FS from Skype

2009-08-15 Thread Scott Torr 
Freeswitch-users,

I'm very new to Freeswitch and have installed the following in VMware
Server 2.0.1


ubuntu-8.04.3-server-i386.iso (udate/upgrade)
skype-debian_2.0.0.72-1_i386.deb
FreeSWITCH Version 1.0.trunk (14492)
mod_skypiax


Using the FS wiki to install/setup and making small changes to default
XML configs as required.


(phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
PSTN)


The MS-Skype client is on the same local network as FS.


I can make calls from the phone to the Skype Client and this works OK.
Audio path OK: phone--FS--Skype


But ,if I call from the MS-Skype client to the phone I hear no audio
from the MS-Skype client.  

Audio path: phone--FS--Skype

Likewise a call from a Skype 'online number' can hear for example the
default 5000 ivr but the DTMF tones from the PSTN phone are not detected
by FS when action application=start_dtmf / is used.



Any suggestions or pointers in the right direction would be much
appreciated. 

Most likely I have overlook something very obvious to others.


Thanks in advance,
Scott Torr 






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Re: [Freeswitch-users] Audio only in one direction when calling FS from Skype

2009-08-16 Thread Scott Torr 

I have used a MS-Skype client on the local area network to receive Skype
calls and do not have any audio continuity problems, and would conclude
that there is not an issue with the firewall if this is working
correctly.

To the best of my knowledge the external ip addresses and local networks
are configured properly in FS.


When I use the (linux Skpye client)=(mod_skypiax)=(FS) to recieve
Skype calls however I cannot hear the incoming audio.

This problem only occurs with an incoming Skype call (ie cannot hear
their audio, they can hear mine).

If I make an outgoing call from FS to Skype there are no audio problem.

So this is an asymmetrical problem.

I have used wireshark and can see UDP packets going in both directions
on the local area network.
So the incoming audio data is hitting the (linux Skype client).


How can I investigate where the data is being dropped?

Is it between:

(linux Skype client)--(mod_skypiax)
or
(mod_skypiax)--(FS)
 

I need some advice on what debug commands to use to see where the audio
is being dropped or some clues as to how to pinpoint where the problem
is occurring.


For example is (mod_skypiax) getting the audio data from (linux Skype
client)?

How do investigate that?
How do I see that?
Are there channel IO byte counters?

The machine is using snd_dummy, has no vmware sound card.

It is strange the problem is only in one direction?



Thanks,
Scott Torr



On Sat, 15 Aug 2009 17:20 -0400, Michael Jerris m...@jerris.com
wrote:
 Make sure your external ip addresses and local networks are configured  
 properly.
 
 Mike
 
 On Aug 15, 2009, at 1:31 PM, Scott Torr wrote:
 
  Freeswitch-users,
 
  I'm very new to Freeswitch and have installed the following in VMware
  Server 2.0.1
 
 
  ubuntu-8.04.3-server-i386.iso (udate/upgrade)
  skype-debian_2.0.0.72-1_i386.deb
  FreeSWITCH Version 1.0.trunk (14492)
  mod_skypiax
 
 
  Using the FS wiki to install/setup and making small changes to default
  XML configs as required.
 
 
  (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
  PSTN)
 
 
  The MS-Skype client is on the same local network as FS.
 
 
  I can make calls from the phone to the Skype Client and this works OK.
  Audio path OK: phone--FS--Skype
 
 
  But ,if I call from the MS-Skype client to the phone I hear no audio
  from the MS-Skype client.
 
  Audio path: phone--FS--Skype
 
  Likewise a call from a Skype 'online number' can hear for example the
  default 5000 ivr but the DTMF tones from the PSTN phone are not  
  detected
  by FS when action application=start_dtmf / is used.
 
 
 
  Any suggestions or pointers in the right direction would be much
  appreciated.
 
  Most likely I have overlook something very obvious to others.
 
 
  Thanks in advance,
  Scott Torr
 
 
 
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[Freeswitch-users] Screaming monkeys on ext 5000

2009-08-23 Thread Scott Torr 
Just a quick note, and I'm sure why, but screaming monkeys does not play
on the the default installation.

I have not looked into why, but thought I would just quickly let you
know.

Perhaps I have not done something?

regards,
sbt 

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

2009-12-22 Thread Scott Torr 
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?

2009-12-23 Thread Scott Torr 
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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 -- 
 Sincerely,
 
 Giovanni Maruzzelli
 Cell : +39-347-2665618
 
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Re: [Freeswitch-users] Skypiax: not able to detect Inband dtmf tones from pstn call?

2009-12-24 Thread Scott Torr 
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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 -- 
 Anthony Minessale II
 
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 pstn:+19193869900

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