Re: [asterisk-users] speech-recog.agi

[Jonathan H]

Argh! Of course, you are correct! I had not noticed that either - why
does he hide the working version right at the end?!

Anyway, that is your immediate requirement resolved, but I will still
post python version for anyone who wishes for it, soon!



On 19 October 2017 at 23:32, Carlos Chavez  wrote:
> On 10/19/17 3:53 PM, Jonathan H wrote:
>
>> That's because it uses a deprecated API and endpoint.
>>
>> However, funny you should ask this, because I've just finished
>> updating my Google TTS routine to take advantage of the new
>> streamlined API.
>>
>> If you can wait a couple of days, I've stick it up on the repo -
>> BUT... it's going to require python3.5+, the way I do it...
>>
>> Would that work for you?
>>
> Thank you.  I just realized that I just needed to RTFM.  There is a new
> version of the speech-recog agi that uses the Cloud Speech API.  It is
> listed on the web site almost at the end. That one works perfectly.  Of
> course alternatives are always welcome.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)8116-9161
>
>
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Re: [asterisk-users] speech-recog.agi

[Carlos Chavez]

On 10/19/17 3:53 PM, Jonathan H wrote:


That's because it uses a deprecated API and endpoint.

However, funny you should ask this, because I've just finished
updating my Google TTS routine to take advantage of the new
streamlined API.

If you can wait a couple of days, I've stick it up on the repo -
BUT... it's going to require python3.5+, the way I do it...

Would that work for you?

Thank you.  I just realized that I just needed to RTFM.  There is a 
new version of the speech-recog agi that uses the Cloud Speech API.  It 
is listed on the web site almost at the end. That one works perfectly.  
Of course alternatives are always welcome.


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Re: [asterisk-users] speech-recog.agi

[Mike Diehl]

If you'll release it for python, I'll take a stab at porting it to perl.

Mike


On October 19, 2017 4:53:52 PM EDT, Jonathan H  wrote:
>That's because it uses a deprecated API and endpoint.
>
>However, funny you should ask this, because I've just finished
>updating my Google TTS routine to take advantage of the new
>streamlined API.
>
>If you can wait a couple of days, I've stick it up on the repo -
>BUT... it's going to require python3.5+, the way I do it...
>
>Would that work for you?
>
>On 19 October 2017 at 18:41, Carlos Chavez  wrote:
>> I want to try using google for speech recognition in Asterisk and
>I
>> found a ready made AGI:
>>
>> http://zaf.github.io/asterisk-speech-recog/
>>
>> I have followed all the steps listed in the web site but I keep
>getting
>> this error:
>>
>> AGI Tx >> 200 result=99981 (timeout)
>endpos=22720
>> AGI Rx << VERBOSE "Unable to get recognition
>data." 3
>>
>> I made sure all the dependencies are met and that my API key for
>Google
>> Cloud Speech is correct (cut and paste).  Any pointers to get this to
>work
>> or any other quick waysto start using Google for speech recognition
>in
>> Asterisk?  Thanks.
>>
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> +52 (55)8116-9161
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>Check out the new Asterisk community forum at:
>https://community.asterisk.org/
>
>New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] speech-recog.agi

[Jonathan H]

That's because it uses a deprecated API and endpoint.

However, funny you should ask this, because I've just finished
updating my Google TTS routine to take advantage of the new
streamlined API.

If you can wait a couple of days, I've stick it up on the repo -
BUT... it's going to require python3.5+, the way I do it...

Would that work for you?

On 19 October 2017 at 18:41, Carlos Chavez  wrote:
> I want to try using google for speech recognition in Asterisk and I
> found a ready made AGI:
>
> http://zaf.github.io/asterisk-speech-recog/
>
> I have followed all the steps listed in the web site but I keep getting
> this error:
>
> AGI Tx >> 200 result=99981 (timeout) endpos=22720
> AGI Rx << VERBOSE "Unable to get recognition data." 3
>
> I made sure all the dependencies are met and that my API key for Google
> Cloud Speech is correct (cut and paste).  Any pointers to get this to work
> or any other quick waysto start using Google for speech recognition in
> Asterisk?  Thanks.
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)8116-9161
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>  https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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[asterisk-users] speech-recog.agi

[Carlos Chavez]

I want to try using google for speech recognition in Asterisk and I 
found a ready made AGI:


http://zaf.github.io/asterisk-speech-recog/

I have followed all the steps listed in the web site but I keep 
getting this error:


AGI Tx >> 200 result=99981 (timeout) endpos=22720
AGI Rx << VERBOSE "Unable to get recognition data." 3

I made sure all the dependencies are met and that my API key for 
Google Cloud Speech is correct (cut and paste).  Any pointers to get 
this to work or any other quick waysto start using Google for speech 
recognition in Asterisk?  Thanks.



--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161


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[asterisk-users] [CFP] FOSDEM 2018, RTC devroom, speakers, volunteers neeeded

[FOSDEM RTC Team]

FOSDEM is one of the world's premier meetings of free software developers,
with over five thousand people attending each year.  FOSDEM 2018
takes place 3-4 February 2018 in Brussels, Belgium.  https://fosdem.org

This email contains information about:
- Real-Time Communications dev-room and lounge,
- speaking opportunities,
- volunteering in the dev-room and lounge,
- related events around FOSDEM, including the XMPP summit,
- social events (the legendary FOSDEM Beer Night and Saturday night dinners
  provide endless networking opportunities),
- the Planet aggregation sites for RTC blogs

Call for participation - Real Time Communications (RTC)
===

The Real-Time dev-room and Real-Time lounge is about all things involving
real-time communication, including: XMPP, SIP, WebRTC, telephony,
mobile VoIP, codecs, peer-to-peer, privacy and encryption.  The dev-room
is a successor to the previous XMPP and telephony dev-rooms.
We are looking for speakers for the dev-room and volunteers and
participants for the tables in the Real-Time lounge.

The dev-room is only on Sunday, 4th of February 2018.  The lounge will
be present for both days.

To discuss the dev-room and lounge, please join the FSFE-sponsored
Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

To be kept aware of major developments in Free RTC, without being on the
discussion list, please join the Free-RTC Announce list:
http://lists.freertc.org/mailman/listinfo/announce

Speaking opportunities
--

Note: if you used FOSDEM Pentabarf before, please use the same account/username

Real-Time Communications dev-room: deadline 23:59 UTC on 30th of November.
Please use the Pentabarf system to submit a talk proposal for the
dev-room.  On the "General" tab, please look for the "Track" option and
choose "Real Time Communications devroom".
https://penta.fosdem.org/submission/FOSDEM18/

Other dev-rooms and lightning talks: some speakers may find their topic is
in the scope of more than one dev-room.  It is encouraged to apply to more
than one dev-room and also consider proposing a lightning talk, but please
be kind enough to tell us if you do this by filling out the notes in the form.
You can find the full list of dev-rooms at
   https://www.fosdem.org/2018/schedule/tracks/
and apply for a lightning talk at https://fosdem.org/submit

Main track: the deadline for main track presentations is 23:59 UTC
3rd of November.  Leading developers in the Real-Time Communications
field are encouraged to consider submitting a presentation to
the main track at https://fosdem.org/submit

First-time speaking?


FOSDEM dev-rooms are a welcoming environment for people who have never
given a talk before.  Please feel free to contact the dev-room administrators
personally if you would like to ask any questions about it.

Submission guidelines
-

The Pentabarf system will ask for many of the essential details.  Please
remember to re-use your account from previous years if you have one.

In the "Submission notes", please tell us about:
- the purpose of your talk
- any other talk applications (dev-rooms, lightning talks, main track)
- availability constraints and special needs

You can use HTML and links in your bio, abstract and description.

If you maintain a blog, please consider providing us with the
URL of a feed with posts tagged for your RTC-related work.

We will be looking for relevance to the conference and dev-room themes,
presentations aimed at developers of free and open source software about
RTC-related topics.

Please feel free to suggest a duration between 20 minutes and 55 minutes
but note that the final decision on talk durations will be made by the
dev-room administrators based on the number of received proposals.
As the two previous dev-rooms have been combined into one, we may decide to
give shorter slots than in previous years so that more speakers can
participate.

Please note FOSDEM aims to record and live-stream all talks.
The CC-BY license is used.

Volunteers needed
=

To make the dev-room and lounge run successfully, we are looking for
volunteers:

- FOSDEM provides video recording equipment and live streaming,
  volunteers are needed to assist in this
- organizing one or more restaurant bookings (dependending upon number of
  participants) for the evening of Saturday, 3rd of February
- participation in the Real-Time lounge
- helping attract sponsorship funds for the dev-room to pay for the
  Saturday night dinner and any other expenses
- circulating this Call for Participation to other mailing lists

Related events - XMPP and RTC summits
=

The XMPP Standards Foundation (XSF) has traditionally held a summit
in the days before FOSDEM.  There is discussion about a similar
summit taking place on the 2nd of February 2018
http://wiki.xmpp.org/web/Summit_22 - please join the mailing
li

Re: [asterisk-users] Dahdi get latest

[Tzafrir Cohen]

On Wed, Oct 18, 2017 at 01:35:34PM -1000, Jean-Denis Girard wrote:
> Le 18/10/2017 à 02:11, Jerry Geis a écrit :
> > I am trying to use dahdi complete 2.11.1 with a 4.13 kernel. - NOT
> > working for know reasons.
> > I tried applying two patches but still get compile errors. AHHH!
> > 
> > How do I just use git to get the latest with the fixes 
> > 
> > This command did not work - I still get the errors.
> > git clone git://git.asterisk.org/dahdi/linux
> >  dahdi-linux
> 
> Hi Jerry,
> 
> Maybe you missed this patch:
> https://issues.asterisk.org/jira/browse/DAHLIN-356
> 
> Or you can try my fork:
> git clone -b next https://github.com/sysnux/dahdi-linux.git

Thanks for the fix!

As mentioned in that bug report, that patch missed a few minor things.
See the -v3 patch there.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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