similar to: AGI script that uses google's text to speech engine

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:

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 Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called "Microsoft Translator". This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good

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 <<

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.

Hello. I would like to run asterisk as an user other than root. I have seen some tutorials on the web, but I would like to know if there is some ?official? how-to for this. Is there? I looked at a thread on reviewboard regarding this (https://reviewboard.asterisk.org/r/654/). It was Paul Belangers work trying to make the installation process take care of this. But the conclusion seem to

Hello, We are setting up an asterisk system for voicemail with video possibilities. We are not using the voicemail app, but rather writing our own dialplan logic. The part of recording, and replaying, the voicemail works, and we receive both an h264 and an wav-file. What I now wonder is how to convert these into one file playable by a (standard) media player. I have not found any real good

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 Linux but it must be working for other 1.6 branches too. Comments, fixes and suggestion are

Hello! We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some problems when using realtime for peers. We connect the PBX to a sip peer at an ITSP, and when we try to dial the peer we get: Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing Dial("SIP/dev02-08c36f28", "SIP/3246 at 989800-out||W") in new stack Jan 23 09:02:07 DEBUG[2236]

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>

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 any further hints for me. Real life experience would be awesome. Thanks, Regards, Arstan Jusupov -------------- next part -------------- An HTML attachment was

Hello I have a problem where I would like to be able to send an arbitrary SIP domain when sending a call to a registered friend. By default the from domain is set to the IP of the Asterisk server, but I would like to set it to something else. The case is that when a call from a foreign domain comes in to the Asterisk, it will connect it to the callee (but with the domain changed). When

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Im looking for wifi sip phones that support auto provisioning and work flawlessly with atserisk. Can anyone suggest me some models?

Greetings list, Has anyone compiled (or could point me at) a list of the minimum required modules and conf files for a very basic 1.8 deployment? We have lots of 1.4 boxes in production, and I'm currently setting up a pair of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the existing 1.4 boxes. All the new installs need to do is receive calls via IAX and send them out via

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

The Error Which I face I have attached. I need a clarification of Why I face this error and how to overcome this. Anybody know Please help...... -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150309/51d674d5/attachment.html> -------------- next part -------------- A non-text attachment was