similar to: speech-recog.agi

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:

Hi Marcelo, Thanks for replying, I do not know what this branch is. Could you please let me know. Also, I enabled google cloud speech API only from the console. Do I need more API enabled? On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhterres at gmail.com> wrote: > Did you already tried the cloud_api branch? > > Regards, > > Marcelo H. Terres <mhterres at

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! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

Hi Jonathan Thanks ! That would indeed be wonderful, at this point I really do not care whether I need to use Python or Lua or JS. I was following http://zaf.github.io/asterisk-speech-recog/ but hit a road end with (for the lack of sane word ) copulating Google's Key On Wed, Jul 19, 2017 at 2:28 PM, Jonathan H <lardconcepts at gmail.com> wrote: > Yes! But I can only tell you if

Thanks for the link. Are there no free alternatives for speech recognition?

Hi Daniel On Mon, 2016-02-22 at 19:40 +0100, Daniel Heckl wrote: > try this http://zaf.github.io/asterisk-speech-recog/. > I have tested it myself, it works very well. I wanted to try it, but I obtain the following error message: "speech-recog.agi,en-US: API key is missing. Aborting. " :-( What am I missing? Any hint? Frank

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: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

Hi, I'm trying to integrate Google cloud speech recognition v2 in it. I can get the audio recorded, have created Service key and API key but whenever I try to access it, I just get 403 access denied. I am at my wits end here. Has anybody tried it ? were you successful ? Could you please guide me how to do it ? I'll be grateful to you if this works ! -- Warm Regds. MathuRahul

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

On Tue, 2016-02-23 at 17:06 +0000, 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.

On 7/19/17 2:37 AM, Marcelo Terres wrote: > This is the pjsip library. > > Is it possible to you to update pjsip for the latest version to test > if it solves the problem? > > On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cursor at telecomab.mx > <mailto:cursor at telecomab.mx>> wrote: > > I am getting frequent segfaults on a new Asterisk

I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 00007fb2d535723f sp 00007fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+180000] Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 00007f4afea0c23f sp 00007f4a7f7e35c0 error 4 in

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-00000055] bridge.c: Bridge 14410400-5e04-4358-af0c-45fd71f6f5cd: switching from simple_bridge technology to native_rtp     Most calls just do:

If SIP goes to the same provider then yes. Still I would check a packet capture for better understanding. BTW, did you try iax debug? ??, 20 ???. 2017 ?. ? 19:46, Carlos Chavez <cursor at telecomab.mx>: > On 4/20/17 12:45 AM, Kseniya Blashchuk wrote: > > Can it happen that the routes lead the traffic through another interface? > Did you try a packet capture with tcpdump? Do the

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

Here is a weird problem that I have had several reports on lately. Customers using our 1.4 based product placing calls to large companies (two examples - AT&T and ComEd, numbers for which I will provide below) that have auto-attendants based on voice recog get disconnected - by the remote end - when the "transfer to an agent" occurs. The message they hear just before disconnect is

Greetings Friends, I would be grateful if you can help me undestand how to make my R code more efficiently. I have read in R intoductory tutorial that a for loop is not used so ofter (and is not maybe not that efficient) compared to other languages. So I am trying to build understanding how to get the equivalent of a for loop using more R-oriented thinking. If I got it right one way to do that

All; I have a customer who is looking for a good speech to text solution, either open source or reasonably priced commercial product, I'm open to suggestions. Thanks; John V -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150826/64d6c224/attachment.html>

On 7/20/17 8:47 AM, Marcelo Terres wrote: > Which version of Asterisk are you using? Are you compiling it with the > bundle pjproject ? > > --with-pjproject-bundled > > Regards, > > Marcelo H. Terres <mhterres at gmail.com <mailto:mhterres at gmail.com>> > IM: mhterres at jabber.mundoopensource.com.br > <mailto:mhterres at

I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4. I tried using the Freepbx 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting random segfaults: [175711.476685] asterisk[2942]: segfault at 188 ip 00007fc6c41abffc sp 00007fc608575890 error 4 in libasteriskpj.so.2[7fc6c4144000+14c000] I then proceeded