at
> is a way to get media out.
>
There is also EAGI, not very flexible but still an option.
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Lefteris Zafiris
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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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Lefteris Zafiris
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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 whi
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 anyw
tails
Vestec: http://www.asteriskexchange.com/listings/113
Regards,
Lefteris Zafiris
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tches
are more than welcome ;)
Regards,
Lefteris Zafiris
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hing ?
> >
> > 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)
> > ?
> >
> >
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
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-
On Thu, 2 Aug 2012 10:27:59 +0800
Arstan Jusupov 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
> Thanks,
> Bruce
>
It is under the folder samples/wolfram/
https://github.com/zaf/asterisk-speech-recog/tree/master/samples/wolfram
Lefteris Zafiris
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> 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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Lefteris Zafiris
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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-addon
rypted 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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On Mon, 9 Jan 2012 14:40:47 -0600
"Danny Nicholas" wrote:
> What do I need to "set" to play 16 Khz wav files?
>
Rename them to .wav16
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Lefteris Zafiris
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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 l
eech-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
ent 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.
loads/zaf/asterisk-speech-recog/asterisk-speech-recog-0.3.tar.gz
--------
Lefteris Zafiris
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oogle, 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).
------
ate).
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 soo
On Wed, Jan 4, 2012 at 8:27 PM, 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
f 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
-
sox ?
>
> Thanks
>
It should be on your distro repos already.
Lefteris Zafiris
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rmat.
Is there some particular reason you want the googletts.agi data in flac?
--------
Lefteris Zafiris
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s,
>>
>>
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
, suggestions
and bug reports are more than welcome.
Lefteris Zafiris
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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
s 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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HEL 5.x machines.
I will try to test there and see whats going on.
Thanks for the feedback.
Lefteris Zafiris
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e the voice data in sln than mp3
since format_mp3 module isn't available in many installations.
Lefteris Zafiris
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cript 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
onvert 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 feed
://translate.google.com/translate_tts?tl=en&q=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.
------
Lefteris Zafiris
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:
http://zaf.github.com/Asterisk-eSpeak/
Asterisk flite module:
http://zaf.github.com/Asterisk-Flite/
Lefteris Zafiris
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u 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
--
l, 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/
----
Lefter
It works with asterisk 1.6 , 1.8 , 10.
http://zaf.github.com/Asterisk-Flite/
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tems 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.
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Lefteris Zafiris
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> 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 =>
e 2rd party repos) is to
download the source of flite, compile it and install it.
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Lefteris Zafiris
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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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risk 1.6.x and 1.8:
http://zaf.github.com/Asterisk-Flite/
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Lefteris Zafiris
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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 w
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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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/Aste
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 Lin
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 t
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 gue
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