similar to: Regarding Text To Speech conversion

Thank You . But now i get solved with that error since I had some mistakes in installing googletts.agi Now when calling from my softphone i have written dialplan with an AGI script to convert from text to speech. It get executed without error but there is no sound getting played. My output, == Using SIP RTP CoS mark 5 -- Executing [1310 at Client-dial-Menu:1]

On Monday 09 Mar 2015, janani m wrote: > 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...... That's a very common error and what it means is, the AGI script "/var/lib/asterisk/agi-bin/googletts.agi" either has an incorrect #! line, or needs chmod +x run on it. What do you

On Tuesday 10 Mar 2015, janani m wrote: > Thank You . > > But now i get solved with that error since I had some mistakes in > installing googletts.agi > > Now when calling from my softphone i have written dialplan with an AGI > script to convert from text to speech. > > It get executed without error but there is no sound getting played. > > My output, [stuff

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

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:

I've got an agi that recognises speech (via Google) and another that turns text into speech (tts) (via Microsoft Translate). Both are web APIs, both called via seperate python AGIs. I've googled and I'm probably missing something pretty newbie 101 here, but is there any way, or fiddle, that I can play some audio to let the caller know that their weather forecast is being fetched,

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>

I had been using google tts, but it started requiring a captcha for my browser, and via linux I can't access http://translate.google.com/translate_tts?q=test (redirects to captcha) as so, its not reliable On 27 August 2015 at 17:16, Carlos Chavez <cursor at telecomabmex.com> wrote: > On 8/26/15 1:15 PM, Tech Support wrote: > > All; > > I have a customer who is

I can't seem to find a definitive answer on this, and I really don't want to risk breaking a production server to find out; so I am going to try asking this here, and maybe anyone else in the same situation searching the archives sometime in future will find the answer I get. Can you use variables in the target of a GoTo() statement? What I am specifically thinking of is this;

Hi list, n00b question, but I can't figure it out: [callthrough] exten => _+X.,1,NoOp(nothing here) #include "blockedall.conf" exten => _+X.,n(hangup),Hangup exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" = "anonymous"]?nocli:cli) ... more stuff that is handling the call ... I'm putting CLIs that I don't want to be able to call my

All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and

B.H. Hello, all :-) We have a cluster of Asterisk (v. 11.9) servers that host IVR applications. The servers work behind SIP proxy (kamailio) for load balancing. All servers are in 2 processor configuration, 8-10 cores per CPU. When a particular server gets about 500 concurrent calls, the sound quality begins to degrade, the sound plays slowly and with clicks. As far as i understand, it's

Hi I want to can we use asterisk as TTS server. Which can support mrcpv2 and ssml. Im looking for tts server with above requirement will asterisk 1.8 is useful for me. Any configuration available. Any opensource tts available. Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Reed, I've been using Speex to transmit TTS for years. It works very well with no tweaking. I use Microsoft TTS ("Microsoft Mike") with Speex at 16kHz wideband and VBR quality 6. Sometimes I forget that the sound is even coming from another computer and being compressed+decompressed. If anything, TTS seems easier for Speex to deal with than real voice. But I don't

Hello All, I was wondering if anyone's tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then what's their experience? Please Note, this does NOT need to be integrated with Asterisk ala MRCP or some module/plugin etc. I just wanted to know if someone's used it and and what their experience has been in

I have done research and reading around text to speech, and was wanting to get an updated query of where everyone is with this. I have installed festival 1.4 on CentOS 3.5 ( system was installed using Asterisk at Home ISO ). I also changed the directory php application to use the festival.pl to read the names of those who have not installed a greeting. It works well enough, though the only voices

On Mar 13, 2013 10:16 PM, "Amit Salunkhe" <amitsalunkhe21 at gmail.com> wrote: > Hi > > I want to know asterisk 1.8 as text to speech server. > > If we can use as TTS server then it support SSML. > > Any sample configuration available for this requirement. Plz help me with > support asterisk as tts server. > > Amit-- > -------------- next part

Hello I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ?