Re: [asterisk-users] Streaming for ASR

2016-10-17 Thread Lefteris Zafiris 
On Mon, 17 Oct 2016 18:44:39 -0300
Joshua Colp <jc...@digium.com> wrote:
 
> >>
> >> The UnicastRTP channel driver allows you to send RTP to a specific
> >> target address with media. Combined with Chanspy (or Snoop channels in
> >> ARI) you can duplicate audio from a channel and send it off to where
> >> you want.  
> >
> > So originate a new channel, make one leg a UnicastRTP and the other a
> > chanspy to spy on the channel you're interested in transcribing?  
> 
> Theoretically, yes. Orchestrating stuff could get complicated but that 
> is a way to get media out.
> 

There is also EAGI, not very flexible but still an option.

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Re: [asterisk-users] AGI: How to break out of AGI when stream_file escape_digits are detected in middle of long sequence of files?

2016-10-11 Thread Lefteris Zafiris 
On Mon, 10 Oct 2016, at 22:47, Jonathan H wrote:
> For reasons best known to myself, I call a python agi (PYST2 - love
> it!) which streams a series of very short files in quick succession.
> 
> Like this:
> 
> escape_digits = str("0")
> agi.stream_file(promptFile,escape_digits)
> 
> and this is what I see on the AGI debug:
> 
> AGI Tx >> 200 result=0 endpos=6784
> AGI Rx << STREAM FILE
> /home/DefaultPrompts/en_GB/female/wx/low "0" 0
> --  Playing
> '/home/DefaultPrompts/en_GB/female/wx/low.alaw' (escape_digits=0)
> (sample_offset 0) (language 'en_GB')
> AGI Tx >> 200 result=48 endpos=1440
> 
> The FIRST line is a file that finished normally, with result=0, the
> LAST line is a file that was interrupted, and receives result=48 from
> asterisk.
> 
> Yes, zero stops the file playing, but it just goes onto the next file.
> 
> So, how do I get that variable? It doesn't become available in python,
> and trying to get it via either
> 
> agi.env[result']
> agi.get_variable('result')
> 
> just makes python choke and the AGI quits. That would be fine if it
> was a result of result being 48!
> 
> I feel like I'm going slightly bonkers here because it's something so
> obvious, and yet I've googled so hard over this!
> 
> Thanks.

Hello,

as the docs on stream_file() mention it returns the digit if one was
pressed.
So you might want to try something like:

pressed_digit = agi.stream_file(promptFile,escape_digits)

in you case the raw value 48 is converted to its ascii equivalent so
pressed_digit will have the value 0

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Re: [asterisk-users] Phone Number Validation

2016-03-29 Thread Lefteris Zafiris 
On Tue, 29 Mar 2016 09:53:15 +0100
Rizwan H Qureshi  wrote:

> Hi Everyone,
> I need to develop a service which tells me whether a given phone number is
> in service and is valid or not. It can be international number. This is
> basically to clean the list of leads we have. Is there any service which
> can give me the required information?
> 
> I currently have an international numbering plan database which only tells
> me if the given phone number is in valid format up to a certain area code.
> But I need to know whether it will ring or not. Any help will be
> appreciated.
> 
> Thanks
> 
> Best Ragards
> Rizwan H Qureshi
> 
> V: +44 (0) 7544180726
> linkedin.com/in/rhqureshi

This might come handy: https://github.com/googlei18n/libphonenumber

Regards,

Lefteris

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Re: [asterisk-users] Voice recognition IVR Is it possible?

2016-02-23 Thread Lefteris Zafiris 
On Tue, 23 Feb 2016 22:56:50 +0100
Frank  wrote:

> On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote:
> 
> > Google?...  
> 
> Yeah... searched "google voice recognition api asterisk", browsed though
> various results. Nothing helpful for a beginner, very confusing bla
> bla...
> 
> Thanks anyway for your help.
> 
> F.
> 
> 

Hello Frank,

google indeed makes it very hard to figure out how to enable the speech API and
get a key. I guess it is intentional since they still consider is a technology
preview. Anyway have a look here on how to get there:
https://github.com/zaf/asterisk-speech-recog/issues/9#issuecomment-44586754
Hope that helps,

Lefteris

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Re: [asterisk-users] Anyone doing speech to text?

2015-08-28 Thread Lefteris Zafiris 
On Fri, 28 Aug 2015 12:11:14 +0300
Amelye Chatila amec...@gmail.com wrote:

 I have a similar situation here, I want to include TTS in my asterisk IVR
 system. Could someone give suggestion(s) please, I prefer open-source
 thanks in advance!
 

Hello,

what follows is a mostly incomplete list of Text To Speech (TTS) and Speech To 
Text (STT)
solutions available for asterisk.

-Regarding the TTS free and open source available options: 

Asterisk comes with festival (http://www.cstr.ed.ac.uk/projects/festival/) 
support
(app_festival) already build in. Decent quality, supports mainly English. 

There is support for flite (http://www.festvox.org/flite/) available as a 3rd 
party
plugin : http://zaf.github.io/Asterisk-Flite/ Quality at par with festival, 
much easier
to setup and use, supports only English.

Also support for espeak (http://espeak.sourceforge.net/):
http://zaf.github.io/Asterisk-eSpeak/ Average quality, supports a wide range of 
languages.


-Free plugins/scripts that provide TTS from a remote not-so-free service:

GoogleTTS : http://zaf.github.io/asterisk-googletts/ Great quality, lots of 
languages,
free of charge but NOT suitable for any serious/commercial use. It is not a 
service
Google officially provides but just a hack that gets synthesized speech data 
from their
translate page. It's more suitable for testing/developing and home use.

MsTTS: http://zaf.github.io/asterisk-mstts/ using Microsoft's Translator voice
synthesis engine.

iSpeech: http://zaf.github.io/asterisk-ispeech/ using iSpeech API 
(http://www.ispeech.org)


-Other non free solutions:

Cepstral: http://www.cepstral.com/en/telephony/asterisk

Speech Technology Group: http://www.asteriskexchange.com/listings/1001


-Regarding the STT options:

Google Speech: http://zaf.github.io/asterisk-speech-recog/ the API is limited 
at the
moment in something like 50 requests/day and considered a technology preview.

iSpeech: http://zaf.github.io/asterisk-ispeech/

Lumevox: http://www.lumenvox.com/partners/digium/asterisk.aspx

Sphinx: http://cmusphinx.sourceforge.net/wiki/asteriskdetails

Vestec: http://www.asteriskexchange.com/listings/113


Regards,

Lefteris Zafiris

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[asterisk-users] agitator - FastAGI reverse proxy

2015-01-16 Thread Lefteris Zafiris 
Hello,

FastAGI doesn't get (or deserve) much love these days but a lot of people are 
still widely
using it. Here is a small project of mine, started trying to scratch my own 
itch, that
some might find useful.

https://github.com/zaf/agitator

It is a reverse proxy for the FastAGI protocol with some interesting features. 
Most
noticeably request based routing, HA/failover, load balancing and TLS 
encryption for
FastAGI sessions (something asterisk unfortunately is still lacking). It is 
written in
Go and it is quite fast and light on resources. First public release, feedback 
and patches
are more than welcome ;)

Regards,

Lefteris Zafiris

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Re: [asterisk-users] Asterisk-eSpeak and Asterisk 12

2014-08-28 Thread Lefteris Zafiris 
On Thu, 28 Aug 2014 17:22:54 +0200
Olivier oza.4...@gmail.com wrote:

 On a side note, with Asterisk 11, I'm getting this :
 
 gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
 -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o
 app_espeak.o app_espeak.c
 app_espeak.c: In function ‘espeak_exec’:
 app_espeak.c:219:13: error: dereferencing pointer to incomplete type
 app_espeak.c:221:47: error: dereferencing pointer to incomplete type
 
 (My plaftform is still Debian Wheezy).
 
 
 2014-08-28 16:50 GMT+02:00 Olivier oza.4...@gmail.com:
  Hi,
 
  I'm giving a look at [1] with this:
 
  cd /tmp
  git clone https://github.com/zaf/Asterisk-eSpeak.git
  cd Asterisk-eSpeak
  ln -s path-to-asterisk-folder/include/asterisk.h
  ln -s path-to-asterisk-folder/include/asterisk
  make
 
  I'm getting this:
  gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes
  -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o
  app_espeak.o app_espeak.c
  In file included from asterisk.h:21:0,
   from app_espeak.c:34:
  asterisk/autoconfig.h:7:32: fatal error: asterisk/buildopts.h: File Not
  found (above line translated)
 
  I can't find any buildopts.h anywhere in Asterisk 12 source files
  though it exists in Asterisk 11.
 
  Did I miss something ?
 
  Regards
 
  PS: If possible, I would prefer to keep asterisk external modules in
  seperate folder. Is there a smarted way to get (smater than the above)
  ?
 
 
  [1] http://zaf.github.io/Asterisk-eSpeak/
 

Hello,

please make sure that you are using the latest trunk code and not some older 
'stable' release.
You can get it from here: http://github.com/zaf/Asterisk-eSpeak/tarball/master

Regards,

Lefteris Zafiris

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[asterisk-users] Automatic Speech Recognition and Text To Speech using iSpeech

2013-05-21 Thread Lefteris Zafiris 
Hi,

a set of AGI scripts that provide ASR and TTS for asterisk using the
iSpeech API (http://www.ispeech.org/) are available on this page:

http://zaf.github.io/asterisk-ispeech/

This is the first public release, updates will soon follow.
Feel free to test and report.

Regards,

Lefteris Zafiris
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Re: [asterisk-users] Voicemail to text for Asterisk

2012-10-22 Thread Lefteris Zafiris 
On Mon, 22 Oct 2012 12:47:51 -0700
Carlos Alvarez car...@televolve.com wrote:
 
 In-house transcriptions are definitely out of the question, but any
 experience with outsourced solutions would be useful.  As far as I can tell
 the current service is automated, and as awful as Google Voice, yet they
 find it useful.  Their existing carrier uses Broadsoft and I'm not sure if
 they have that built in.
 

Voice recognition for asterisk based on Google speech API is already 
available[1],
the problem with this service is that it's limited to 20-30 seconds of speech 
data,
which isn't suitable for transcripting voicemails.
If you are able to find a reliable way of chopping speech samples in segments 
no bigger
than 20 seconds based on silence detection, so words wont be cut in half, you 
might come
up with something very similar to Google Voice transcription service.
But I would recommend against using this into production since google haven't 
yet
defined the terms of service for speech recognition, and its more or less a 
hack for
now.

[1] http://zaf.github.com/asterisk-speech-recog/


Lefteris Zafiris

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Re: [asterisk-users] html/js/flash/air SIP clients?

2012-08-02 Thread Lefteris Zafiris 
On Thu, 2 Aug 2012 10:27:59 +0800
Arstan Jusupov arst...@gmail.com wrote:

 Dear list,
 I am looking for an open source SIP client(or any SDK) that can work
 on a browser. It may be based html5, javascript, flash, adobe air. I
 have done some research myself and I would like to ask the community
 if they have any further hints for me. Real life experience would be
 awesome.


http://www.sipml5.org/

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Re: [asterisk-users] OT - mstts.agi - Where to find API key ?

2012-06-06 Thread Lefteris Zafiris 
On Wed, 6 Jun 2012 16:37:01 +0200
Olivier oza_4...@yahoo.fr wrote:

 I stricly followed instructions steps 1 and 2 and I'm very to report
 it works !

I m glad you got it working. Microsoft really tried it's best to make it
as complicated as possible.

 Thanks for your detailed answer.
 May I post here suggestions that may help others to use this script ?
 
That's what this list is all about.


Lefteris Zafiris

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Re: [asterisk-users] RPM availability -- [asterisk-announce] Asterisk 1.8.9.3 Now Available

2012-03-05 Thread Lefteris Zafiris 
On 03/05/2012 09:52 PM, Jason Parker wrote:
 On 03/05/2012 01:49 PM, Eric Germann wrote:
 Will a 1.8.10.0 build be imminent or should we go ahead and push this in to 
 production with testing?

 Thanks!

 EKG

 
 ~20 minutes
 

Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled
against 1.8.7:

asterisk18-addons-core-1.8.7.0-2_centos5
asterisk18-addons-mysql-1.8.7.0-2_centos5

Is this a problem with the repo? Are these packages
obsolete/unmaintained or have been replaced by others?




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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-12 Thread Lefteris Zafiris 
On 01/12/2012 05:50 PM, Danny Nicholas wrote:
 Two more offerings - #1 - add DTMF parameter so function can be stopped by
 pressing a digit or digits other than * or #  - #2 - add an option to
 silence the beep.  If you were using this in an IVR and wanted to say
 press 1 or say help for help,  silencing the beep before recording would
 (IMO) make the rendering sound more professional/less mechanical.

Both features added:

-
Usage
-
agi(speech-recog.agi,[lang],[timeout],[intkey],[NOBEEP])
Records from the current channel untill the timeout (set to 10 seconds
by default, -1 for no timeout) is reached or the interrupt key (# by
default) is pressed.
If NOBEEP is set, no beep sound is played back to the user to indicate
the start of the recording.

There is now also the option to enable SSL for encrypted communication
between your pbx and the google voice server.

Updated code can be found here:
https://github.com/zaf/asterisk-speech-recog/tarball/master


Lefteris Zafiris

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Re: [asterisk-users] 44Khz files in Asterisk 10

2012-01-09 Thread Lefteris Zafiris 
On Mon, 9 Jan 2012 13:59:07 -0600
Danny Nicholas da...@debsinc.com wrote:

 Hi gang,
 
 I'm thrilled to be able to use a better quality sound
 in Asterisk 10, but have to change my wav files to sln44 to get the
 benefit. Is there some conf setting I'm missing that would let me
 play a wav at 44 Khz instead of having to do this?
 
  
 
 Sox mon-0a.wav h-1a.wav -t raw -r 44100 jan01.sln44
 
  
 
 Not a biggie if no, since this is a decent work-around.

At the moment format_wav only supports playback of 8 and 16kHz wav
files, so for higher frequencies u have to use the raw slin format.

Just another note, bumbing up the frequency of the sound files doesn't
always means better sound quality. In most cases asterisk will have
to resample the file before playing it back to you which will degrade
the quality. In previous versions of asterisk codec_resample was using
libresample which has very poor quality and introduces clipping and
distortion in the resampled sound. Vesrion 10 has switched to speex
resampler which is fast but not top quality, and its mainly
tested and optimised with voice data so it wont be good to use for
anything else, eg MOH.

The optimal is to match the frequency of the codec you are
using, eg 8kHz or 16kHz for wideband codecs. And if you have to
resample use another resampler like sox (with dithering, lowpass etc)
instead of letting asterisk do this for you.


Lefteris Zafiris

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Re: [asterisk-users] 44Khz files in Asterisk 10

2012-01-09 Thread Lefteris Zafiris 
On Mon, 9 Jan 2012 14:40:47 -0600
Danny Nicholas da...@debsinc.com wrote:

 What do I need to set to play 16 Khz wav files?
 
Rename them to .wav16


Lefteris Zafiris


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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-07 Thread Lefteris Zafiris 
On 01/07/2012 09:34 AM, Bruce B wrote:
 Added two new features to the script: Timeout value and speechdata type.
 
 *exten = s,n,agi(speech-recog.agi,en-US,3000,phoneNumb)*
 - Will listen for 3 seconds and sanitize return as a single number without
 any spaces in between. This helps when one reads phone number in format
 415-554-2323 and google returns, 415 554 2323 as result which is not very
 usable.
 
 *exten = s,n,agi(speech-recog.agi,en-US,2,string)*
 - Will listen for 20 second and return result as provided by Google
 untouched.
 
 It would be great to see them in future versions as I seem to need them
 dearly in a real life scenario.
 
 Updated script attached.
 
 -Bruce

Thank you Bruce for the testing and the suggestions.
Both features added in the script. Timeout can now be set by the user,
also -1 means no timeout and the recording keeps going till # is pressed.
Space gets stripped between digits, this is now the default behavior and
there's no need to determine the 'speechdata' type.
The updated code can be found here:
https://github.com/zaf/asterisk-speech-recog/tarball/master

Next on my TODO list is to make use of the asterisk speech recognition
API (https://wiki.asterisk.org/wiki/display/AST/Speech+Recognition+API)
This will make the application actually usable for real case scenarios
and not a proof of concept as it is now.


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-06 Thread Lefteris Zafiris 
On Fri, 6 Jan 2012 20:46:14 -0500
Bruce B bruceb...@gmail.com wrote:

 Does sox have more features on a Debian system than RHEL? Is that why
 it won't work on RHEL?
 
RHEL's 5 version of sox is really old and outdated. The command syntax
and the switches are totally different compared to recent versions of
sox.
Anyway I'm not sure audio normalization and the rest we use sox for is
really needed. My tests so far didn't show any improvements in
detection rates. Keep in mind that all this is still WIP and the
option to use sox is more for testing than for serious use.


Lefteris Zafiris


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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
On 01/04/2012 07:51 AM, Bruce B wrote:
 And with recent version 14.3.2 I get:
 
 /usr/local/bin/sox FAIL formats: no handler for file extension `flac'
  -- speech-recog.agi: /usr/local/bin/sox failed: 512
 -- SIP/-002eAGI Script speech-recog.agi completed, returning 0
 
 Regards,
 
 
 On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote:
 
 Very interesting. I just tried to get it to work but it complains about
 sox. Probably you used a different version of sox?

 *PBX-*CLI /usr/bin/sox: invalid option -- -*
 */usr/bin/sox: invalid option -- n*
 */usr/bin/sox: invalid option -- o*
 */usr/bin/sox: -r must be given a positive integer*
 * -- speech-recog.agi: /usr/bin/sox failed: 512*

 I am using: *Package sox-12.18.1-1.el5_5.1.i386 *

 Thanks,



Note to self: Never release anything asterisk related without testing
on RHEL/Centos 5

Thank you for reporting this. I have replaced sox with flac and it seems
to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
You can get the updated code here:
https://github.com/zaf/asterisk-speech-recog/tarball/master


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote:
 this looks great - is there any chance of coverting the googletts.agi
 to use flac as well ?
 
 Julian
 

In googletts.agi we get the voice data from google in mp3 and we convert
it in a format that asterisk can read and playback (slin). If we store it
in flac asterisk wont be able to read it natively and we would have to
convert it each time we want to play it back to the user.

In the speech recognition script we have to convert the voice data in
flac before sending it to google because that's the accepted format.

Is there some particular reason you want the googletts.agi data in flac?


Lefteris Zafiris



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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote:
 the only reason is that I didn't want to have to install sox. Lazy.
 that's all ;) Just another piece of software to find and install
 
 running on amazon ec2, is the best thing to download the source and
 compile sox ?
 
 Thanks
 

It should be on your distro repos already.


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
On Wed, Jan 4, 2012 at 8:47 PM, Michelle Dupuis mdup...@ocg.ca wrote:
 Wow - nice!  A few quick questions:

 1.  How long can the recording be for translation?
At the moment the recording timeout is set at 15sec. I haven't tested
yet the max
length  of voice data ta google accepts (all this voice recognition
stuff is undocumented).
I have read that it is between 10-20 seconds but havent really went to
test this yet. On my todo list is
to add the option to cut the sound data in smaller chunks before
sending them to google and get rid of the
recording length limitations.

 2.  Any limitation on how much text the return (transcribed) variable can
 hold?
This better be answered by the astsrisk devs but empirically talking i
have loaded in dialplan variables really big
chunks of text (like the complete gpl license) without having any problems.

 3.  Any commercial / terms of use limitations?
This is a gray area at the moment. Voice recognition is undocumented
in google's API and i guess not
officially supported yet. I hope it gets covered by the general TOS of
google services:
 http://www.google.com/accounts/TOS


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
On Wed, Jan 4, 2012 at 8:27 PM,  isr...@gmail.com wrote:
 Does anyone know what languages are supported?

For sure english and spanish, since its undocumented i don't have a
complete list
yet.


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 

 Works beautifully. Amazing job Lefteris. Thanks.

 The best result I got in probability was 0.9725632 by saying, hello. I
 think there is some non-phonetic logic built-in as well. I tried, 1, 2 and
 I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
 0.97256315. Probably Google sees the pattern?!

 What are some of the other tricks (if any) or consideration that one should
 make while creating a strong speech recognition enabled IVR?

Google accepts sound files at any sampling rate (up to 44.1kHz) so if
you can use some wideband codec ( eg g722)
It can greatly improve the sound quality and the detection rates. For
now the script supports 8kHz and 16kHz sampling rates
for recording and it can be set by editing the scripts user defined
parameters ( the variable $samplerate).
Anything that improves the recording sound clarity will help, a good
phone, low background noise level etc.
I have also read that normalizing the recording and setting the gain
to -5 db improves detection rates. I m experimenting with this at the
moment and there will be some new code soon (as soon as i get sox
working in RHEL/Centos 5 :P ).


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
On Wed, 04 Jan 2012 14:48:22 -0500
sean darcy seandar...@gmail.com wrote:

 This is really spectacular. Thanks.
 
 I'm running Fedora 15, so I can use flac or sox. Any reason to prefer 
 one over the other?
 
 sean

We have to convert the voice data to flac format before sending them to
google, this can be done by both sox and flac encoder. For now the
script uses flac encoder for compatibility with older distros (mainly
RHEL 5). Sox is a bit more flexible and also gives you the option to
edit the sound data (normalizing, changing levels etc).


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

2012-01-04 Thread Lefteris Zafiris 
Fresh code is out! The use of sox can be now optionally enabled by the
user if the system has a recent version of the program (won't work in
RHEL/Centos 5)
This is done by editing the script and setting the variable 'use_sox'.
When sox is used the audio gets normalized, low frequency noise (100Hz)
is removed and also possible DC offset is corrected. Those are supposed
to improve the recognition results(?). The settings are still a bit
experimental, feel free to play with them and report what settings
improved your results.

get the new version here:
https://github.com/downloads/zaf/asterisk-speech-recog/asterisk-speech-recog-0.3.tar.gz


Lefteris Zafiris

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[asterisk-users] Speech recognition in asterisk using google voice API

2012-01-03 Thread Lefteris Zafiris 
Hello,
I have written an agi script that uses google voice API for voice
recognition.
The script records from the current channel untill the pound key (#) is
pressed or the timeout (15 seconds) is reached. The recording is send
over to google speech recognition service and the returned text string
is assigned to a channel variable.
More info and dialplan examples can be found in the README file:
https://raw.github.com/zaf/asterisk-speech-recog/master/README

The script is available here:
https://github.com/zaf/asterisk-speech-recog

The code is still young and not roughly tested so comments, suggestions
and bug reports are more than welcome.


Lefteris Zafiris

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[asterisk-users] AGI script that uses google's text to speech engine

2011-12-13 Thread Lefteris Zafiris 
Hello,

version 0.3 of the asterisk-googletts AGI script just got released,
most noticeable changes are:
The script can now be used to easily build IVRs.  
Fixed compatibility with asterisk 1.4 and older.
Fixed compatibility with older perl versions(5.8.8).
Better input handling.

The latest release, documentation and dialplan examples can be found
here: http://zaf.github.com/asterisk-googletts/

A big thank you to all the users that contributed with feedback,
bug reports and suggestions.


Lefteris Zafiris

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Re: [asterisk-users] AGI script that uses google's text to speech engine

2011-12-01 Thread Lefteris Zafiris 
On Thu, 1 Dec 2011 09:43:29 -0500
bakko asannu...@gmail.com wrote:

 Hello,
 
 when I use the Agi, sometimes not play the phrase:
 
 WARNING[30391]: file.c:650 ast_openstream_full: File ggl_U0sBo0 does
 not exist in any format
 
 Regards

Seems like the script failed to convert the mp3 data that gets from
google to raw slinear. In that case mpg123 or sox
failed to run. It would be very helpful if you could send the full
console output with verbosity set to 3. Please reply to my mail address
so we don't pollute the list.

Thanks for the feedback


Lefteris Zafiris
  


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Re: [asterisk-users] AGI script that uses google's text to speech engine

2011-12-01 Thread Lefteris Zafiris 
On Thu, 01 Dec 2011 17:23:59 +
Kingsley Tart kings...@skymarket.co.uk wrote:

 Hi. Aside from converting spaces to plus signs, you don't encode any
 special characters before putting them in the URL. It might be safer
 to run $line through some sort of encoding before calling Google with
 it, even if most special characters probably don't result in any
 sound. Google say and if you give it an ampersand, but unescaped
 you couldn't include that in the string.
 
 You may decide to have an option to locally cache pre-produced sound
 files in case that phrase is used again.
 
 Cheers,
 Kingsley.
 

Thanks for the suggestion. Ther's already some sort of input sanitation:
 $AGI{arg_1} =~ s/[\\\/|*~^\(\)\[\]\{\}\n\r]/ /g;
that strips most special characters but i guess it needs some more work.
As for the caching the script supports it already, its enabled by
default and controlled by these 2 variables in the script:
 $usecache   = 1;
 $cachedir   = /tmp;
Voice data gets stored in the cachedir for future use so we don't have
to fetch it from google each time.


Lefteris Zafiris

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Re: [asterisk-users] AGI script that uses google's text to speech engine

2011-12-01 Thread Lefteris Zafiris 
On Thu, 1 Dec 2011 11:35:21 -0600
Danny Nicholas da...@debsinc.com wrote:

 I personally don't like the use of mpg123 for playback - would prefer
 use of the internal Playback/background functions.  Still seems to
 be a nice effort though.

mpg123 used to convert the mp3 data that we get from google to wav. The
wav file is passed to sox that converts it to raw slinear and then its
played back by asterisk using the 'stream file' agi command.

I don't really like calling all these system commands but I thought it
would be better for the users to have the voice data in sln than mp3
since format_mp3 module isn't available in many installations.


Lefteris Zafiris 

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Re: [asterisk-users] AGI script that uses google's text to speech engine

2011-12-01 Thread Lefteris Zafiris 
On Thu, 1 Dec 2011 21:51:21 +0100
Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote:

 This is because you need to add /tmp to the STREAM command, ie:
 
 print STREAM FILE /tmp/$tmpname \$intkey\\n;
 
 $tmpname seems to not contain the path, so it will look in
 /var/lib/asterisk/sounds for the file...
 
 This at least made it work for me... (After fixing some other things
 to make it work with asterisk 1.2...)
 
 BR,
 Torbjörn Abrahamsson
 

$tmpname is supposed to include the full path together with the temp
dir since its created with the option 'TMPDIR = 1' during the call of
tempfile() and it does so in my system that runs perl 5.14.2. I guess
that might not be true for older versions of perl(?)
Can you please tell me what version of perl you are using?
The oldest perl I can get my hands on is 5.8.8 on RHEL 5.x machines.
I will try to test there and see whats going on.

Thanks for the feedback.


Lefteris Zafiris

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Re: [asterisk-users] AGI script that uses google's text to speech engine

2011-12-01 Thread Lefteris Zafiris 
On Thu, 1 Dec 2011 23:23:56 +0100
Torbjörn Abrahamsson torbjorn.abrahams...@gmail.com wrote:

 This was run on an Fedora 8 machine, with perl 5.8.8. I also found it
 odd that the path was not included...
 
 // T
 
It seems this is an issue with older versions of perl or at least with
5.8.8. Since this version is used in RHEL/CentOS 5.x that many people
run on their servers, this is a serious problem.

Changing the way tempfile() is called from: 
tempfile(ggl_XX, TMPDIR = 1, UNLINK = 1)
to:
tempfile(ggl_XX, DIR = $tmpdir, UNLINK = 1)
seems to address this issue.

An updated version including this fix can be obtained here:
http://github.com/zaf/asterisk-googletts/tarball/master


Lefteris Zafiris

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[asterisk-users] AGI script that uses google's text to speech engine

2011-11-30 Thread Lefteris Zafiris 
Hello,
I have written an AGI script for asterisk that uses google translate for
text to speech synthesis.
It supports a variety of different languages, local caching for the voice
data and wideband audio.
The voice in most languages is female and the quality of the synthesized
speech is very high.
More info about the script can be found here:
http://zaf.github.com/asterisk-googletts/
the first public release ca be obtained here:
https://github.com/downloads/zaf/asterisk-googletts/asterisk-googletts-0.2.tar.gz

To get a sample of the speech synthesis quality try this link:
http://translate.google.com/translate_tts?tl=enq=this+is+a+test+for+google+text+to+speech+engine

The code is still very young so suggestions, comments and bug reports are
more than welcome.

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Re: [asterisk-users] IAX MOS Score measuring solution

2011-08-25 Thread Lefteris Zafiris 
On Thu, 25 Aug 2011 19:56:51 +0300
Stelios Koroneos skoron...@digital-opsis.com wrote:

 Greetings !
 
 Has anyone used any solution for getting the MOS Score on IAX channels
 using codes like g729.
 I have found a few but all are measuring sip and/or a-ulaw.
 
 Regards
 
 Stelios

You can extract data like rtt, jitter and packet loss from the
dialplan with something like:
${CHANNEL(rtpqos|audio|all)}
Based on these u can calculate R and MOS using the formulas on this
page:
http://www.nessoft.com/kb/50


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[asterisk-users] Flite module for asterisk

2011-08-21 Thread Lefteris Zafiris 
Version 2.0 of app_flite just got released.

Flite For Asterisk provides the Flite dialplan application, which
allows you to use the Flite TTS Engine with Asterisk.
It supports 8kHz and 16kHz sample rates to provide the best
possible sound quality along with the use of wideband codecs. 
It works with asterisk 1.6 , 1.8 , 10.

http://zaf.github.com/Asterisk-Flite/


Lefteris Zafiris

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[asterisk-users] espeak module for asterisk

2011-08-21 Thread Lefteris Zafiris 
Version 2.0 of app_espeak just got released.

eSpeak For Asterisk provides the Espeak dialplan application,
which allows you to use the Espeak speech synthesizer with Asterisk.

It supports the following languages:
Afrikaans, Albanian, Armenian,Cantonese, Catalan, Croatian, Czech,
Danish, Dutch, English, Esperanto, Estonian, Finnish, French, Georgian,
German, Greek, Hindi, Hungarian, Icelandic, Indonesian, Italian,
Kannada, Kurdish, Latvian, Lojban, Macedonian, Malayalam, Mandarin,
Norwegian, Polish, Portuguese, Romanian, Russian, Serbian, Slovak,
Spanish, Swahili, Swedish, Tamil, Turkish, Vietnamese, Welsh.

It supports 8kHz and 16kHz sample rates to provide the best possible
sound quality along with the use of wideband codecs. Works with
asterisk 1.6 , 1.8 , 10.

http://zaf.github.com/Asterisk-eSpeak/


Lefteris Zafiris

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Re: [asterisk-users] use ILBC installed from asterisk yum repositories

2011-08-02 Thread Lefteris Zafiris 
On Tue, 2 Aug 2011 11:42:19 -0500
Bob Pierce westman...@gmail.com wrote:

 Is there a process for installing the ILBC codec under this
 environment, or will I have to un-install the RPMs and build Asterisk
 from source?

You can write a short makefile for just codec_ilbc module, build it and
install it on your running asterisk system. You will have to install the
asterisk18-devel package and get the asterisk source code either from
a tar or from the srpm. If you are familiar with the basics of writing
makefiles its pretty trivial to write one that builds codec_ilbc, I have
done this in numerous systems that use the digium rpms and it works
flawlessly. This method can also be used to build other modules that
are missing from the digium rpms.

---
Lefteris Zafiris

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Re: [asterisk-users] Asterisk 1.8 minimum modules/configuration

2011-06-07 Thread Lefteris Zafiris 
On 06/07/2011 08:04 PM, Chris Bagnall wrote:
 Greetings list,
 
 Has anyone compiled (or could point me at) a list of the minimum required 
 modules and conf files for a very basic 1.8 deployment?
 
Basic deployment is hard to specify, but in any case you can use the
following modules as a base to build your system. Its a set of modules
that provides very basic sip support for asterisk, and it can be
considered very close to absolute minimal. You will propably have to add
more modules for dialplan apps, channels, codes etc.

[modules]
autoload=no

load = res_musiconhold.so
load = res_smdi.so
load = res_rtp_asterisk.so
load = res_timing_timerfd.so
load = codec_ulaw.so
load = format_pcm.so
load = app_dial.so
load = pbx_config.so
load = chan_local.so
load = chan_sip.so


Lefteris Zafiris


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Re: [asterisk-users] Flite issue

2011-04-22 Thread Lefteris Zafiris 
On 04/22/2011 11:05 AM, virendra bhati wrote:
 Hi Asterisk guys,
 
 Flite is not working with asterisk 1.6.2.17.
 Flite is working with asterisk 1.4.
 Please help me how to use it with asterisk 1.6 ...
 Thanks in advance.


Thers an app_flite module compatible with asterisk 1.6.x and 1.8:

http://zaf.github.com/Asterisk-Flite/



Lefteris Zafiris


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Re: [asterisk-users] Flite issue

2011-04-22 Thread Lefteris Zafiris 
On 04/22/2011 02:09 PM, virendra bhati wrote:
 Hi,
 
 I already know about that link thanks for update me. But problem come when
 we start installation of that packages..
 
 [root@cent210 zaf-Asterisk-Flite-5af2b91]# make
 gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
 -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_flite.o
 app_flite.c
 app_flite.c:39:25: error: flite/flite.h: No such file or directory

You don’t have the flite headers installed in your system.
As the README file clearly says they are needed for app_flite in order
to compile and load. Install flite either from your distros repos
(judging by the hostname i assume you are running centos so you have to
search for it in 3rd party repos) or from source.



Lefteris Zafiris

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Re: [asterisk-users] Flite issue

2011-04-22 Thread Lefteris Zafiris 
On 04/22/2011 03:50 PM, virendra bhati wrote:
 Hi,
 
 from where I get Header file of flite. If you tell me step by step process
 then it will be easy to get use of such application for me.
 
 Yes I am using *CentOS release 5.6*.
 

You can find flite packages in the Extra Packages for Enterprise Linux
(EPEL) repo ( http://fedoraproject.org/wiki/EPEL ).
To install that repo read here:
http://fedoraproject.org/wiki/EPEL/FAQ#Using_EPEL
To install flite and its header files u just have to run:
yum install flite flite-devel

The other option (in case u dont want to use 2rd party repos) is to
download the source of flite, compile it and install it.



Lefteris Zafiris


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[asterisk-users] Wi-Fi sip phones with auto provisioning

2009-12-03 Thread Lefteris Zafiris 
Im looking for wifi sip phones that support auto provisioning and work
flawlessly with atserisk. Can anyone suggest me some models?

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Re: [asterisk-users] Wi-Fi sip phones with auto provisioning

2009-12-03 Thread Lefteris Zafiris 
Fred Posner wrote:
 On Dec 3, 2009, at 8:49 AM, Lefteris Zafiris wrote:
 
 Im looking for wifi sip phones that support auto provisioning and work
 flawlessly with atserisk. Can anyone suggest me some models?

 
 Don't know of any wifi phone that works flawlessly whatsoever. Best to 
 consider a DECT style phone.
 
 ---fred
 http://qxork.com
 

Ok i can live with wifi phones that work *almost* flawlessly with
asterisk :D
DECT is not an option for now.


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[asterisk-users] espeak app for asterisk 1.6

2009-09-01 Thread Lefteris Zafiris 
I have written a module for asterisk that uses the eSpeak
speech synthesizer (http://espeak.sourceforge.net/) to
render text to speech. The source is available here:
http://zaf.github.com/Asterisk-eSpeak/
It's similar to app_festival and app_flite.
It's only tested against asterisk 1.6.1 on x86 Linux but it must be
working for other 1.6 branches too. Comments, fixes and suggestion are
welcome.


===
 Espeak For Asterisk 1.6
===

This provides the Espeak dialplan application, which allows you to use
the Espeak speech synthesizer with Asterisk. This module invokes the Espeak TTS 
engine
locally, and uses it to render text to speech.


Requirements

Asterisk 1.6 header files
Espeak libraries and header files
**It is recommended to use espeak version 1.41.01 or newer.
Earlier version of epseak had an file descriptor leak that could 
cause asterisk to crash. If upgrading is not an option patch your 
current 
version of epseak with the espeak.patch provided here.
libsndfile libraries and header files
libresample libraries and header files



Installation

$ make
$ make install

To install the sample configuration file, issue the following command after
the 'make install' command:

$ make samples

-
Usage
-
Espeak(text[,intkeys,language]):  This will invoke the eSpeak TTS engine,
send a text string, get back the resulting waveform and play it to
the user, allowing any given interrupt keys to immediately terminate
and return.

  
Examples

dialplan sample code for your extensions.conf
  
;Espeak Demo
exten = 1234,1,Answer()
;;Play mesage using default language as set in 
espeak.conf
exten = 1234,n,Espeak(This is a simple espeak test in 
english.,any,)
;;Play message in Spanish
exten = 1234,n,Espeak(Esta es una simple prueba espeak en 
español.,any,es)
;;Play message in Greek
exten = 1234,n,Espeak(Αυτό είναι ένα απλό 
τέστ του espeak στα ελληνικά.,any,el)
;;Read a text file from disk (relative to the channel 
language)
;;and play it with espeak using the asterisk channel 
language.
exten = 1234,n,ReadFile(MYTEXT=/path/${LANGUAGE}/myfile,200)
exten = 1234,n,Espeak(${MYTEXY},any,${LANGUAGE})
exten = 1234,n,Hangup()

---
License
---
The Espeak module for asterisk is distributed under the GNU General Public 
License v2. See COPYING for details.
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Re: [asterisk-users] espeak app for asterisk 1.6

2009-09-01 Thread Lefteris Zafiris 
Steve Edwards wrote:
 On Tue, 1 Sep 2009, Lefteris Zafiris wrote:
 
 I have written a module for asterisk that uses the eSpeak speech 
 synthesizer (http://espeak.sourceforge.net/) to render text to speech. 
 The source is available here: http://zaf.github.com/Asterisk-eSpeak/
 
 I hope it sounds a whole lot better in practice than it does on their
 sample available at http://espeak.sourceforge.net/samples/raven.ogg
 
 Cepstral's Allison font is miles ahead.
 
Actually it sounds like that demo. You can improve it a bit by altering
the voice settings (speed pitch etc) in the config but it cannot
match Cepstrals solution. Flite (in which Cepstral is based) gives a
better sound but its limited in voice support (i think it supports only
English)

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Re: [asterisk-users] Flite module for asterisk 1.6.x

2009-08-31 Thread Lefteris Zafiris 
Klaus Darilion wrote:
 
 Lefteris Zafiris schrieb:
 I have written a simple application for asterisk 1.6 that uses the Flite
 tts engine to render text to speech.
 Source is available here: http://zaf.github.com/Asterisk-Flite/
 It works more or less like the festival app, can use cache etc.
 Its only tested against asterisk 1.6.1 on X86 linux but i guess it works
 for other 1.6 branches too. Comments, fixes and suggestion are welcome.
 
 Hi Zaf!
 
 I wonder what is the benefit of using Flite over Festival?
 
 thanks
 klaus
 

Flite is lightweight, simple and easy to install, thers no need for
configuration to get it running no deamon etc etc and can be ran even in
embedded systems. For more info check flite home page:
http://www.speech.cs.cmu.edu/flite/



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[asterisk-users] Flite module for asterisk 1.6.x

2009-08-29 Thread Lefteris Zafiris 
I have written a simple application for asterisk 1.6 that uses the Flite
tts engine to render text to speech.
Source is available here: http://zaf.github.com/Asterisk-Flite/
It works more or less like the festival app, can use cache etc.
Its only tested against asterisk 1.6.1 on X86 linux but i guess it works
for other 1.6 branches too. Comments, fixes and suggestion are welcome.

zaf


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