date:20090302

Rajagopal, Sridhar (Sridhar) wrote:
 Hi all,

 I am planning to run freeswitch on powerpc MPC8358. Please let me know if any 
 changes needs to be done in the code

 Regards
 Sridhar
   
It may be easier to say what will currently stop Freeswitch working.

The lack of an MMU is a problem right now, so Blackfins are out, which 
is sad. Cores without hardware floating point may not perform all that 
well, but should work. Endianness should not be a problem. I think 
machines which choke on misaligned access are probably OK, too.

Checking that list, you should be OK on a PPC.

Steve


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Re: [Freeswitch-users] ESL Wrapper

pardon?
ESL is just a client library for event socket to make it easier to make
event socket apps.
ESL == Event Socket Library


On Mon, Mar 2, 2009 at 3:29 AM, Gopal krishnan gopal2krish...@gmail.comwrote:

 Hi,
   Actually what is the difference between ESL in FS 1.0.3 and event socket
 in FS 1.0.2. Is the FS 1.0.3 ESL superior?

 On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex rex.alex...@yahoo.com wrote:

 Hi All, I did what you have all suggested. Now its working perfectly.
 Thanks a lot for all your assistance. Rex.

 Raymond Chandler wrote:
  and it will probably be a good idea to do make phpmod-install so that the
 .so and .php files gets into the correct place to be included -Ray Mathieu
 Rene wrote:   You need your distro's php dev pakage.  On 26-Feb-09, at
 6:25 AM, Rex_Alex wrote:   Hi All, I tried svn up  ./bootstrap.sh 
 ./configure  make  install and did Mathieu's suggestion but getting
 error as below,  [r...@server esl]# make phpmod make
 MYLIB=../libesl.a  SOLINK=-shared -Xlinker -x 
 CFLAGS=-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE  -g
 -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall  -Werror
 -Wunused-variable -Wwrite-strings -Wstrict-prototypes 
 -Wmissing-prototypes 
 CXXFLAGS=-I/root/freeswitch-1.0.3/libs/esl/src/include  -DHAVE_EDITLINE
 -g -ggdb -I../../libs/libedit/src/ -fPIC  CXX_CFLAGS= -C php make[1]:
 php-config: Command not found make[1]:  Entering directory
 `/root/freeswitch-1.0.3/libs/esl/php' g++ 
 -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g  -ggdb
 -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o 
 esl_wrap.cpp:717:18: error: zend.h: No such file or directory 
 esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory 
 esl_wrap.cpp:719:17: error: php.h: No such file or directory 
 esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory 
 esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or 
 directory esl_wrap.cpp:767: error: âE_ERRORâ was not declared in this 
 scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of 
 âZEND_RSRC_DTOR_FUNCâ with no type esl_wrap.cpp:788: error: 
 âSWIG_landfillâ was not declared in this scope esl_wrap.cpp:788:  error:
 expected â,â or â;â before â{â token esl_wrap.cpp:793: error:  variable or
 field âSWIG_ZTS_SetPointerZvalâ declared void  esl_wrap.cpp:793: error:
 âzvalâ was not declared in this scope  esl_wrap.cpp:793: error: âzâ was
 not declared in this scope  esl_wrap.cpp:793: error: expected
 primary-expression before âvoidâ  esl_wrap.cpp:793: error: expected
 primary-expression before â*â token  esl_wrap.cpp:793: error: âtypeâ was
 not declared in this scope  esl_wrap.cpp:793: error: expected
 primary-expression before âintâ  esl_wrap.cpp:793: error: initializer
 expression list treated as  compound expression esl_wrap.cpp:793: error:
 expected â,â or â;â  before â{â token make[1]: *** [esl_wrap.o] Error 1
 make[1]: Leaving  directory `/root/freeswitch-1.0.3/libs/esl/php' make:
 *** [phpmod]  Error 2 [r...@server esl]# Please tell me where am i
 wrong? Thanks, Rex   mercutioviz wrote:  On Wed, Feb 25, 2009 at 11:34
 AM, Brian West wrote:  If he's on  1.0.3 I don't think it has php in it..
 Can't he do the whole  bootstrap process? svn up  ./bootstrap.sh 
 ./configure   make install And then do Mathieu's suggestion? -MC 
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Re: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number

put origination_caller_id_number in the dial string of any call and you can
set the caller id individually for that leg

{origination_caller_id_number=1234}any normal dial string


On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX prometheus...@gmx.net wrote:

 Hello,

 I have the following problem while providing callback (mod_eventsocket
 is used):
 1) I want to call a certain destination number A with a suppressed
 caller_id_number (this works fine with some vars in the origination string)
 2) The destination number A picks up the phone and enters a target
 number B by DTMF
 3) freeswitch then forwards the call to target number B by DTMF and I
 want to show the number A. I do this with uuid_setvar. The problem is
 that it still shows unknown.
 This is all with SIP.

 uuid_setvar however worked if I did not set the caller_id_number to
 unknown. Per default this is then 000 and can then be changed
 with uuid_setvar to the number of A.
 But if I set caller_id_number to unknown I can no longer change it to A.

 Any hint?

 Best regards
 Peter






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Re: [Freeswitch-users] Orginate: getting status of call fail

The best way would be to add a few custom variables and add a secondary
system that monitors the CDR data and uses the
custom variables to identify what you want to do with the failed calls.



On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton 
nik.middle...@noblesolutions.co.uk wrote:

  Hi Guys,



 I’ve been running a test script written in lua which now works very well
 thanks to Anthony’s fix to stream file.



 Right now I’m using an event socket to initiate the call and passing the
 name of the script along with originate thus:



 $dialstring = originate
 {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum
 'lua(helloworld.lua )';

 $result = $obj -bgapi_command($dialstring);



 The script gets fired (it would appear) on answer.  However, if the number
 is invalid , timed out or was busy, I’m not sure the script gets executed or
 am I wrong?



 I want to be able to fire an event back on what happed to the call in the
 event that it failed for whatever reason.



 I know I can simply call the originate and pass the number as an argument
 and execute the dial within the script but I’m led to believe that’s not
 very efficient, or am I completely wrong?



 Looking for the most FS friendly way here



 Regards,

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Re: [Freeswitch-users] Orginate: getting status of call fail

2009-03-02 Thread Nik Middleton

That's what I was wondering, however, won't the response to the bagi
(not the initial) give me the info on the call result?

 

Regards

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 02 March 2009 14:00
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orginate: getting status of call fail

 

The best way would be to add a few custom variables and add a secondary
system that monitors the CDR data and uses the 
custom variables to identify what you want to do with the failed calls.




On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:

Hi Guys,

 

I've been running a test script written in lua which now works very well
thanks to Anthony's fix to stream file.

 

Right now I'm using an event socket to initiate the call and passing the
name of the script along with originate thus:

 

$dialstring = originate
{ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/
Mygw/phonenum 'lua(helloworld.lua )';

$result = $obj -bgapi_command($dialstring);

 

The script gets fired (it would appear) on answer.  However, if the
number is invalid , timed out or was busy, I'm not sure the script gets
executed or am I wrong?

 

I want to be able to fire an event back on what happed to the call in
the event that it failed for whatever reason.

 

I know I can simply call the originate and pass the number as an
argument and execute the dial within the script but I'm led to believe
that's not very efficient, or am I completely wrong?

 

Looking for the most FS friendly way here

 

Regards,


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Re: [Freeswitch-users] Orginate: getting status of call fail

yes if you match the job uuid from bgapi to the SWITCH_EVENT_BACKGROUND_JOB
event, you would get the result in that event.


On Mon, Mar 2, 2009 at 8:49 AM, Nik Middleton 
nik.middle...@noblesolutions.co.uk wrote:

  That’s what I was wondering, however, won’t the response to the bagi (not
 the initial) give me the info on the call result?



 Regards




  --

 *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
 freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony
 Minessale
 *Sent:* 02 March 2009 14:00
 *To:* freeswitch-users@lists.freeswitch.org
 *Subject:* Re: [Freeswitch-users] Orginate: getting status of call fail



 The best way would be to add a few custom variables and add a secondary
 system that monitors the CDR data and uses the
 custom variables to identify what you want to do with the failed calls.


  On Sat, Feb 28, 2009 at 4:49 PM, Nik Middleton 
 nik.middle...@noblesolutions.co.uk wrote:

 Hi Guys,



 I’ve been running a test script written in lua which now works very well
 thanks to Anthony’s fix to stream file.



 Right now I’m using an event socket to initiate the call and passing the
 name of the script along with originate thus:



 $dialstring = originate
 {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/Mygw/phonenum
 'lua(helloworld.lua )';

 $result = $obj -bgapi_command($dialstring);



 The script gets fired (it would appear) on answer.  However, if the number
 is invalid , timed out or was busy, I’m not sure the script gets executed or
 am I wrong?



 I want to be able to fire an event back on what happed to the call in the
 event that it failed for whatever reason.



 I know I can simply call the originate and pass the number as an argument
 and execute the dial within the script but I’m led to believe that’s not
 very efficient, or am I completely wrong?



 Looking for the most FS friendly way here



 Regards,


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[Freeswitch-users] First time setting up FreeSwitch and SPA3102 / SPA3000

2009-03-02 Thread Aplayful Idiot

Hi.

I have little background in telephony and need to use a PBX but would like
to start first with a small test set-up.

I have a SPA3102 attached to the box running FS and to a ordinary phone
line.

I registered SPA in conf/directory/default/line1.xml and this works to a
point but I can't get caller id numbers from incoming calls. All FS sees is
line1 which is found in file line1.xml as variable
name=effective_caller_id_number value=line1/.

Looking back over the FS wiki, I'm now wondering if the SPA was registered
or set-up in FS correctly but reading the documentation is confusing me a
bit. Sometimes I think the analogue-phone-line-SPA-FS is like a softphone
which is registered to an extension numbered xml file in
conf/directory/default/ but then issues like not getting outside incoming
caller id's makes me think I've got this all wrong.

Can someone help me out with this?

Thanks.
api
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Re: [Freeswitch-users] Ghost Sessions in CLI after a longterm test

2009-03-02 Thread Michael Jerris

Could you please post this to jira along with a thread apply all bt of  
a core file taken from the process with the stuck sessions.

Mike

On Mar 2, 2009, at 2:06 AM, rod wrote:

 Hi All,

 I ran some longer tests with FS 1.0.3 acting as an SBC.
 The test machine has the following specs:
- Intel Quad Core Q9550
- 8GB RAM (far too much from what I saw)

 After 3 days running SIPP with 750 simultaneous calls (1500  
 channels) at
 20cps mean (50cps max) and call duration of 35s, I stopped SIPP.

 In the CLI, using status command I got this:

 freeswi...@internal status
 UP 0 years, 2 days, 22 hours, 48 minutes, 3 seconds, 859 milliseconds,
 607 microseconds
 15817560 session(s) since startup
 22 session(s) 0/500

 But when I use show channels or show calls, I see nothing. So I'm
 wondering where are these 22 sessions ?

 FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs.

 Successful call --  5271434
 Failed call --- 1554  (less than 0.03%)

 regards,
 rod.



 complete SIPP summary:

 -- Scenario Screen  [1-9]: Change
 Screen --
  Call-rate(length) Port   Total-time  Total-calls  Remote-host
 50.0(35000 ms)/1.000s   5060  254259.42 s  5273022
 10.10.10.254:5060(UDP)

  0 new calls during 0.856 s period  7 ms scheduler resolution
  0 calls (limit 750)Peak was 750 calls, after 15 s
  0 Running, 34 Paused, 0 Woken up
  15544 out-of-call msg (discarded)
  1 open sockets
  9213070274 Total echo RTP pckts 1st stream 0.000 last period RTP rate
 (kB/s)
  0 Total echo RTP pckts 2nd stream  0.000 last period RTP rate  
 (kB/s)

 Messages  Retrans   Timeout
 Unexpected-Msg
  INVITE -- 5273022   0 0
 100 -- 5273022   0   1554
 180 -- 0 0   0
 183 -- 0 0   0
 200 --  E-RTD1 5271434   0   0
 ACK -- 5271434   0
   Pause [35.0s] 5271434   0
 BYE -- 5271434   0 0
 200 -- 5271434   0   0

 -- Test Terminated
 


 - Statistics Screen --- [1-9]: Change
 Screen --
  Start Time | 2009-02-27
 09:11:31
  Last Reset Time| 2009-03-02
 07:49:10
  Current Time   | 2009-03-02
 07:49:11
 -+--- 
 +--
  Counter Name   | Periodic value| Cumulative value
 -+--- 
 +--
  Elapsed Time   | 00:00:00:857  |
 70:37:39:429
  Call Rate  |0.000 cps  |   20.739
 cps
 -+--- 
 +--
  Incoming call created  |0  |
 0
  OutGoing call created  |0  |
 5273022
  Total Call created |   |
 5273022
  Current Call   |   34
 |
 -+--- 
 +--
  Successful call|0  |
 5271434
  Failed call|0  |
 1554
 -+--- 
 +--
  Response Time 1| 00:00:00:000  |
 00:00:00:240
  Call Length| 38:32:13:386  |
 00:00:36:131
 -- Test Terminated
 


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Re: [Freeswitch-users] Sangoma USB FXO U100 + Asus notebook?

2009-03-02 Thread Michael Jerris

I think any issues we have are related to pri, the analog doesn't seem  
to generate any major bug reports.

Mike

On Mar 2, 2009, at 6:47 AM, Fred wrote:

 Thanks guys for the feedback. So, the OpenZap driver isn't ready for
 production yet?



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Re: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number

Hello Anthony,

I do this when I orginate the call. This way we suppress the cid when we
call party A and transfer A to an internal extension (our callback
application).
But now comes the part that does not work:
After A enters the target number B (via DTMF), we set the cid variables
via uuid_setvar and then transfer A via uuid_transfer to party B.
However uuid_setvar does not work in that case.

BUT: If we do the same scenario and do not suppress the cid in the
originate part, then uuid_setvar works correctly and sets the cid_number.

Best regards
Peter

Anthony Minessale schrieb:
 put origination_caller_id_number in the dial string of any call and
 you can set the caller id individually for that leg

 {origination_caller_id_number=1234}any normal dial string


 On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX prometheus...@gmx.net
 mailto:prometheus...@gmx.net wrote:

 Hello,

 I have the following problem while providing callback (mod_eventsocket
 is used):
 1) I want to call a certain destination number A with a suppressed
 caller_id_number (this works fine with some vars in the
 origination string)
 2) The destination number A picks up the phone and enters a target
 number B by DTMF
 3) freeswitch then forwards the call to target number B by DTMF and I
 want to show the number A. I do this with uuid_setvar. The problem is
 that it still shows unknown.
 This is all with SIP.

 uuid_setvar however worked if I did not set the caller_id_number to
 unknown. Per default this is then 000 and can then be
 changed
 with uuid_setvar to the number of A.
 But if I set caller_id_number to unknown I can no longer change it
 to A.

 Any hint?

 Best regards
 Peter






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Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

2009-03-02 Thread Andy Spitzer

Woof!

On Sun, 01 Mar 2009 21:28:18 -0500, Brian West br...@freeswitch.org wrote:

 NO.  You want something that people THINK exists and works well...
 Reliable human/voice detection doesn't exist in ANY form.

I beg to differ.  See http://www.freepatentsonline.com/5521967.html for one way 
to do it.  It works rather well and can quickly descriminate between voice and 
tone.  I've no idea who owns that patent now (not me, for sure).

There is a simpler, less reliable way of differentiating voice from tone, that 
as far as I know isn't patented.  If you compare the RMS power levels of 
sequential 40 mS periods, call progress tones will have very consistent power 
levels from sample to sample.  So if 5 or more 40 mS periods have about the 
same power measurement (within say, 2%), it's a tone.  Voice will have dramatic 
power level differences over that same period.  This works very well in today's 
telephony environment, where tones are computer generated.  In the old days 
when ringback tone was generated off the audio hum from the 20 Hz ring voltage 
generator...not so well.

--Woof!

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Re: [Freeswitch-users] Problems with record_stereo

2009-03-02 Thread Michael Collins

On Fri, Feb 27, 2009 at 6:07 AM, Helmut Kuper helmut.ku...@ewetel.de wrote:
 Hello,

 I play around with record_session and would like to have caller and
 callee separated on left and right channel. I found record_stereo is
 used for this. Unfortunately it doesn't work. A and B leg are still
 mixed. Additionally I found that B leg is significant louder than A leg,
 but both legs were local extensions.

Just to confirm - you are trying to record each leg of the call into a
separate file? In other words, one call creates two separate audio
recordings?

-MC

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Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

i think that's what mod_vmd does

On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer w...@nortel.com wrote:

Woof!

On Sun, 01 Mar 2009 21:28:18 -0500, Brian West br...@freeswitch.org
wrote:

NO. You want something that people THINK exists and works well...
Reliable human/voice detection doesn't exist in ANY form.

I beg to differ. See http://www.freepatentsonline.com/5521967.html for
one way to do it. It works rather well and can quickly descriminate between
voice and tone. I've no idea who owns that patent now (not me, for sure).

There is a simpler, less reliable way of differentiating voice from tone,
that as far as I know isn't patented. If you compare the RMS power levels
of sequential 40 mS periods, call progress tones will have very consistent
power levels from sample to sample. So if 5 or more 40 mS periods have
about the same power measurement (within say, 2%), it's a tone. Voice will
have dramatic power level differences over that same period. This works
very well in today's telephony environment, where tones are computer
generated. In the old days when ringback tone was generated off the audio
hum from the 20 Hz ring voltage generator...not so well.

--Woof!

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Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

2009-03-02 Thread Michael Collins

On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
 i think that's what mod_vmd does

I think that's right. It just does the opposite - instead of looking
for differing power levels it looks for the same power level. In other
words it tries to detect distinctly non-human sound. I'll bet you
could futz with that code and have it fire off events when it detects
what it believes is human speech.

-MC

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Re: [Freeswitch-users] pocketsphinx and event socket

Some more info:
the system I am working on is a copy (dd copy) of a system where the
pizza demo works on.
The only thing I changed was to update to the current freeswitch trunk
12293 (it was 10003 before).

Do I need to update the model? I did a make in the model directory, but
no change.

Best regards
Peter

Peter P GMX schrieb:
 Hello Brian,

 thanks for the info. I am a step further, but it cannot load the grammar
 files.
 I am sending through event_socket:

 SendMsg
 call-command: execute
 execute-app-name: detect_speech
 execute-app-arg: pocketsphinx yes no

 However I get the message (also when I am using Pizza demo):
 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
 sofia/internal/1...@sip2.server.com Command Execute
 detect_speech(pocketsphinx yes no)
 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
 pocketsphinx_asr_load_grammar() Can't open language model
 /usr/local/freeswitch/grammar/model/communicator.
 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
 switch_ivr_detect_speech() Error loading Grammar
 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
 pocketsphinx_asr_close() Port Closed.

 However the grammar files are there:
 r...@sip2:/usr/local/freeswitch/grammar/model/communicator#
 r...@sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
 total 12752
 drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
 drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
 -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
 -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
 -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
 -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
 -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
 -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
 -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
 -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances


 Any hint?

 Best regards
 Peter

 Brian West schrieb:
   
 You can accomplish this  here is an example using ESL in perl

 http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344

 /b

 On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:

   
 
 Or back to the basics: Is it possible to use pocketsphinx through  
 event
 socket?
 
   
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[Freeswitch-users] Howto config early dial

2009-03-02 Thread Sergio Alecha

In asterisk, with the parameter AMPBADNUMBER = FALSE it is possible to use
early dial Grandstream telephones. How do Freeswitch in?
thank you very much.
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Re: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number

Hello Anthony,

sorry for being tenacious but in some cases it works in a way we need it:
If I a am not suppressing the cid numer when calling A, the following
scenario works:

* A receives a Call (originate) with CID '00' (default from
  switch_caller.c)
* A dials some digits via DTMF, the app set the cid variables via
  uuid_setvar and uuid_transfers the call to B. B receives a call
  with the right cid set. 

Maybe I simply modify the default cid '00'  to a different value
in switch_caller.c? Is there a special reason why this is '00'?

I am using trunk version 12293.

Best regards
Peter

Anthony Minessale schrieb:
 origination_caller_id number is not ok as a variable unless its in {}
 as part of the dial string
 it's an exception that is parsed before the channel is even created.

 I think you are drawing the wrong conclusion about what works and
 doesn't work.
 If you can produce a dial string that contains
 {origination_caller_id_number=x} you will always be able to set it.

 I assume you are using a recent version of FS as we did have a small
 bug with this variable a few weeks ago.


 On Mon, Mar 2, 2009 at 10:19 AM, Peter P GMX prometheus...@gmx.net
 mailto:prometheus...@gmx.net wrote:

 Hello Anthony,

 I do this when I orginate the call. This way we suppress the cid
 when we
 call party A and transfer A to an internal extension (our callback
 application).
 But now comes the part that does not work:
 After A enters the target number B (via DTMF), we set the cid
 variables
 via uuid_setvar and then transfer A via uuid_transfer to party B.
 However uuid_setvar does not work in that case.

 BUT: If we do the same scenario and do not suppress the cid in the
 originate part, then uuid_setvar works correctly and sets the
 cid_number.

 Best regards
 Peter

 Anthony Minessale schrieb:
  put origination_caller_id_number in the dial string of any call and
  you can set the caller id individually for that leg
 
  {origination_caller_id_number=1234}any normal dial string
 
 
  On Sun, Mar 1, 2009 at 3:38 PM, Peter P GMX
 prometheus...@gmx.net mailto:prometheus...@gmx.net
  mailto:prometheus...@gmx.net mailto:prometheus...@gmx.net
 wrote:
 
  Hello,
 
  I have the following problem while providing callback
 (mod_eventsocket
  is used):
  1) I want to call a certain destination number A with a
 suppressed
  caller_id_number (this works fine with some vars in the
  origination string)
  2) The destination number A picks up the phone and enters a
 target
  number B by DTMF
  3) freeswitch then forwards the call to target number B by
 DTMF and I
  want to show the number A. I do this with uuid_setvar. The
 problem is
  that it still shows unknown.
  This is all with SIP.
 
  uuid_setvar however worked if I did not set the
 caller_id_number to
  unknown. Per default this is then 000 and can then be
  changed
  with uuid_setvar to the number of A.
  But if I set caller_id_number to unknown I can no longer
 change it
  to A.
 
  Any hint?
 
  Best regards
  Peter
 
 
 
 
 
 
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Re: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number

2009-03-02 Thread Michael Collins

On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX prometheus...@gmx.net wrote:
 Hello Anthony,

 sorry for being tenacious but in some cases it works in a way we need it:
 If I a am not suppressing the cid numer when calling A, the following
 scenario works:

    * A receives a Call (originate) with CID '00' (default from
      switch_caller.c)
    * A dials some digits via DTMF, the app set the cid variables via
      uuid_setvar and uuid_transfers the call to B. B receives a call
      with the right cid set.

 Maybe I simply modify the default cid '00'  to a different value
 in switch_caller.c? Is there a special reason why this is '00'?


Check vars.xml to confirm that you have actually set a default caller
ID. Most likely you'll still have the default caller id number set to
all zeroes, which is the default.

-MC

 I am using trunk version 12293.

 Best regards
 Peter


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Re: [Freeswitch-users] hide caller_id_number, transfer and then change caller_id_number

Since you did not describe the exact way you are doing it with enough detail
or any trace I can't begin to tell you
what your problem is.  you did not even mention what variable you are using
or show examples.

All I can do is tell you again that if you set the
origination_caller_id_number in the dial string it will
be the most likely to work for you.




On Mon, Mar 2, 2009 at 4:08 PM, Michael Collins m...@freeswitch.org wrote:

 On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX prometheus...@gmx.net wrote:
  Hello Anthony,
 
  sorry for being tenacious but in some cases it works in a way we need it:
  If I a am not suppressing the cid numer when calling A, the following
  scenario works:
 
 * A receives a Call (originate) with CID '00' (default from
   switch_caller.c)
 * A dials some digits via DTMF, the app set the cid variables via
   uuid_setvar and uuid_transfers the call to B. B receives a call
   with the right cid set.
 
  Maybe I simply modify the default cid '00'  to a different value
  in switch_caller.c? Is there a special reason why this is '00'?
 

 Check vars.xml to confirm that you have actually set a default caller
 ID. Most likely you'll still have the default caller id number set to
 all zeroes, which is the default.

 -MC

  I am using trunk version 12293.
 
  Best regards
  Peter
 

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Re: [Freeswitch-users] pocketsphinx and event socket

Thanks Addison.
The Pizza files are there (as mentionned is it a copy of an already
working system).
In fact freeswitch is complaning about
/usr/local/freeswitch/grammar/model/communicator which he cannot load

So somehow freeswitch is not willing to open the files, but I have no
clue why. So any hints are welcome.

Best regards
Peter


Addison Martin schrieb:
 Peter,

 You need the grammar files for the pizza demo:
 http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo
 has lonks to premade fles for everyhting to get the pizza demo working
 with pocketshinx.  Those to not come with the source code when you
 update from SVN.

 Nik



 On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX prometheus...@gmx.net wrote:
   
 Some more info:
 the system I am working on is a copy (dd copy) of a system where the
 pizza demo works on.
 The only thing I changed was to update to the current freeswitch trunk
 12293 (it was 10003 before).

 Do I need to update the model? I did a make in the model directory, but
 no change.

 Best regards
 Peter

 Peter P GMX schrieb:
 
 Hello Brian,

 thanks for the info. I am a step further, but it cannot load the grammar
 files.
 I am sending through event_socket:

 SendMsg
 call-command: execute
 execute-app-name: detect_speech
 execute-app-arg: pocketsphinx yes no

 However I get the message (also when I am using Pizza demo):
 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event()
 sofia/internal/1...@sip2.server.com Command Execute
 detect_speech(pocketsphinx yes no)
 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145
 pocketsphinx_asr_load_grammar() Can't open language model
 /usr/local/freeswitch/grammar/model/communicator.
 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041
 switch_ivr_detect_speech() Error loading Grammar
 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219
 pocketsphinx_asr_close() Port Closed.

 However the grammar files are there:
 r...@sip2:/usr/local/freeswitch/grammar/model/communicator#
 r...@sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al
 total 12752
 drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 .
 drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 ..
 -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING
 -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params
 -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef
 -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means
 -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict
 -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump
 -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices
 -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances


 Any hint?

 Best regards
 Peter

 Brian West schrieb:

   
 You can accomplish this  here is an example using ESL in perl

 http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344

 /b

 On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote:



 
 Or back to the basics: Is it possible to use pocketsphinx through
 event
 socket?


   
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Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

Andy Spitzer wrote:
 Woof!

 On Sun, 01 Mar 2009 21:28:18 -0500, Brian West br...@freeswitch.org wrote:

   
 NO.  You want something that people THINK exists and works well...
 Reliable human/voice detection doesn't exist in ANY form.
 

 I beg to differ.  See http://www.freepatentsonline.com/5521967.html for one 
 way to do it.  It works rather well and can quickly descriminate between 
 voice and tone.  I've no idea who owns that patent now (not me, for sure).
   
Since when did a patent mean a problem is solved? For things like speech 
recognition you can achieve pretty high accuracy in voice detection, but 
in that case you can delay the audio and make decisions that span the 
start of the speech burst. For most telephony purposes you need to make 
a decision on the very first frame of speech, as you can't afford to add 
latency. That turns it into a tough problem. Something like the VAD in 
G.729 is about the best people can currently do, but its far from perfect.
 There is a simpler, less reliable way of differentiating voice from tone, 
 that as far as I know isn't patented.  If you compare the RMS power levels of 
 sequential 40 mS periods, call progress tones will have very consistent power 
 levels from sample to sample.  So if 5 or more 40 mS periods have about the 
 same power measurement (within say, 2%), it's a tone.  Voice will have 
 dramatic power level differences over that same period.  This works very well 
 in today's telephony environment, where tones are computer generated.  In the 
 old days when ringback tone was generated off the audio hum from the 20 Hz 
 ring voltage generator...not so well.
   
That is *not* VAD. What you describe just says is its energy steady. I 
will trigger on music, background noise and maybe even some of the fast 
pulsed tone signals. A proper VAD won't.

Regards,
Steve


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Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

Hi,

mod_vmd is a bit more sophisticated than that. It looks for the signal 
being narrowband energy. However, mod_vmd isn't very reliable, as it 
takes a rather high SNR for its narrowband detector to work. So high 
that a lossy codec like G.711 can barely manage it.

Regards,
Steve

Anthony Minessale wrote:
 i think that's what mod_vmd does

 On Mon, Mar 2, 2009 at 11:16 AM, Andy Spitzer w...@nortel.com 
 mailto:w...@nortel.com wrote:

 Woof!

 On Sun, 01 Mar 2009 21:28:18 -0500, Brian West
 br...@freeswitch.org mailto:br...@freeswitch.org wrote:

  NO.  You want something that people THINK exists and works well...
  Reliable human/voice detection doesn't exist in ANY form.

 I beg to differ.  See
 http://www.freepatentsonline.com/5521967.html for one way to do
 it.  It works rather well and can quickly descriminate between
 voice and tone.  I've no idea who owns that patent now (not me,
 for sure).

 There is a simpler, less reliable way of differentiating voice
 from tone, that as far as I know isn't patented.  If you compare
 the RMS power levels of sequential 40 mS periods, call progress
 tones will have very consistent power levels from sample to
 sample.  So if 5 or more 40 mS periods have about the same power
 measurement (within say, 2%), it's a tone.  Voice will have
 dramatic power level differences over that same period.  This
 works very well in today's telephony environment, where tones are
 computer generated.  In the old days when ringback tone was
 generated off the audio hum from the 20 Hz ring voltage
 generator...not so well.

 --Woof!

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Re: [Freeswitch-users] Detecting the origin of voice activity using VAD