On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote:
>
> Requirements
>
> ...
> Speech API key from Google
Yes... OK... but... where and how can I obtain this API Key?
Where and how do I install it into my Asterisk box?
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On Tue, Feb 23, 2016 at 12:39 AM, Frank wrote:
> 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
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. "
:-(
CMU Sphinx is really good if you know what sentences you want to recognise
I am not sure how well it works with random stuff but if you have a list
of common phrasings then you can do really well (having used it
recently) - although I would say its much better at recognising North
American
Hi list,
I am trying to enable SIP SIMPLE communication in my test environment
(Asterisk 13.6.0)
I have two problems:
1. Using messagesend(), I don't want my users to be able to change their
own callerid name. I want the name that appears in the ${MESSAGE(from)} to
be set by config file, and
Read README, check the requirements and get the google speech api key.
Then add a custom destination in FreePBX and edit your extensions_custom.conf.
> Am 22.02.2016 um 21:03 schrieb Daniel Chavez :
>
> Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin
Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory.
Where do I go from here?
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I use FreePBX as well. There is no module for speech recognition. You have too
create a custom destination.
> Am 22.02.2016 um 20:53 schrieb Daniel Chavez :
>
> Thanks, this looks promising. I was wondering if there's an easier way to get
> this to work inside FreePBX?
>
I think I saw an old page on the voip-info wiki on how to use CMU Sphinx
with asterisk.
http://www.voip-info.org/wiki/view/Sphinx
IMHO It's not going to be anywhere as good as a commercial product without
a lot of work.
On Mon, Feb 22, 2016 at 11:34 AM, Daniel Chavez
Thanks, this looks promising. I was wondering if there's an easier way to get
this to work inside FreePBX?
I have all of the dependencies installed for it, but now I want to know if
there's a mod I can use in FreePBX to get it setup?
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Le 19/02/2016 12:24, Bryant Zimmerman a écrit :
> Jean
>
> If you moved the exten => _. Lines to the bottom of the context then
> you should like be able to get away from having to have two separate
> contexts. I use that method quiet often, but
Daniel,
try this http://zaf.github.io/asterisk-speech-recog/.
I have tested it myself, it works very well.
Daniel
> Am 22.02.2016 um 19:34 schrieb Daniel Chavez :
>
> Thanks for the link.
> Are there no free alternatives for speech recognition?
>
> --
>
Thanks for the link.
Are there no free alternatives for speech recognition?
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I saw Lumenvox offering Speech Recognition for asterisk at a past Astricon.
http://www.lumenvox.com/partners/digium/Asterisk.aspx
On Mon, Feb 22, 2016 at 11:00 AM, Daniel Chavez
wrote:
> Hello list,
> I was wondering if it were possible for asterisk to do a voice
Hello list,
I was wondering if it were possible for asterisk to do a voice recognition type
IVR?
Like you know how most banks and stuff do, where they ask you to say your
selection or key it in?
If this is possible, how can I set this up? I'm using FreePBX 2.11 on Linux,
CentOS 6.7 32-bit,
In my case, username is the BTN. I also set the fromdomain to be the sbc
that I'm registering with. Externip might help also?
[paetec]
host=10.250.0.5
username=btn
fromdomain=10.250.0.5
dtmfmode=rfc2833
externip=10.255.0.2
I've used these settings on both registering and non-registering trunks,
On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote:
> register string: :@:5060
Try:
register => 5551231234:sec...@sipdomain.com/5551231234
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New to
Does anyone on this list use Windstream as a SIP trunk provider?
If so, would you mind sharing your peer settings?
I'm using asterisk 13.7.2 and can't seem to get the inbound working
correctly (using registration). Outbound is fine, but they are seeing an
authentication error on their end.
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