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

2016-02-22 Thread Frank
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? --

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

2016-02-22 Thread Laszlo
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

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

2016-02-22 Thread Frank
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. " :-(

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

2016-02-22 Thread Duncan
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

[asterisk-users] Troubles with MessageSend command

2016-02-22 Thread Julien Sansonnens
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

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

2016-02-22 Thread Daniel Heckl
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

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

2016-02-22 Thread Daniel Chavez
Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin directory. Where do I go from here? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

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

2016-02-22 Thread Daniel Heckl
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? >

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

2016-02-22 Thread John Kiniston
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

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

2016-02-22 Thread 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? --

Re: [asterisk-users] Grandstream Early Dial

2016-02-22 Thread Jean-Denis Girard
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 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

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

2016-02-22 Thread Daniel Heckl
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? > > -- >

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

2016-02-22 Thread Daniel Chavez
Thanks for the link. Are there no free alternatives for speech recognition? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

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

2016-02-22 Thread John Kiniston
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

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

2016-02-22 Thread Daniel Chavez
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,

Re: [asterisk-users] Windstream SIP Trunk settings

2016-02-22 Thread Mark Wiater
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,

Re: [asterisk-users] Windstream SIP Trunk settings

2016-02-22 Thread Frank
On Mon, 2016-02-22 at 08:20 -0500, James Cass wrote: > register string: :@:5060 Try: register => 5551231234:sec...@sipdomain.com/5551231234 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

[asterisk-users] Windstream SIP Trunk settings

2016-02-22 Thread James Cass
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.