similar to: Asterisk IVR functionality

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

Sheesh. I only joined here a few days ago and already there's a flame war. Look, to remove your name from the list is easy. It tells you where to go to manage your subscription down there at the bottom. If you want another mailing list, why not go to yahoo!! or topica and set one up, or set one up yourself. It ain't rocket science with mailman. Even an idiot like me has managed it.

I just wanted to try out Festival, but I can't get it patched. I'm thinking that there is something missing from the steps listed at http://www.marko.net/asterisk/archives/0209/0389.html. >>tar xvzf festival-1.4.2-release.tar.gz >>patch -p0 </usr/src/asterisk-ng/festival-1.4.2-diff >> (or wherever the patch is located) When I run the patch command, I get the

Until last week my wine runs a Microcontroler Programmer under serial port very well, But in this week have problems to work wine for serial ports on my new Mandrake 8.1. Well, in Mdk 8.1 there is no ttyS0 devices but links to /dev/tts/0 and /dev/tts/1 I realy don't remember how was until last week, but I think about users and perms to devices. I try to chmod a+rw for /dev/tts/0 and tts/1

Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i

Hello Group, I have been going through all the chit-chat about TTS and the various engines available to integrate with Asterisk incl. flite/festival, espeak, Nuance etc but I am wondering if anyone's tried any or all of these to compile on a Sparc based Solaris platform? If not, then what is the best way for me to accomplish a production environment TTS service when most of my servers or the

Can anyone suggest a shell script in linux bash for encodimg raw audio output from text to speech app into an ogg file. I am running festival speech. The command line for basic tts is festival> bin/festival --tts myfile.txt Thanks -- Raena Lea-Shannon --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this

Good afternoon. I try to put nut on gentoo. Has executed installation by a standard command emerge nut. All has passed successfully. I use UPS POWERCOM BNT-1500AP. It is connected to serial port 1, i.e. for my system it/dev/tts/0. In/etc/nut/ups.conf [powercom] driver = powercom port = /dev/tts/0 desc = "PowerCom BNT-1500AP on IP-PBX" modelname = "BNT-1500AP" linevoltage = 230

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

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've succeeded running older version ov TextAloud on my machine (to some extend). It means that now I can type my registration code in the application and continue with running TextAloud. Later at some moment it ties to use TTS engines and then it shows the message that TTS engine isn't installed. I've installed AT&T Natural voices in TextAloud directory and have instructed

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:

All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards,

Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/56254a0e/attachment.htm

(forgive the brief interruption to -users with a mostly -dev issue, just wanted to publicize this on behalf of the larger community) If there are any ambitious coders out there (not too many shekels yet, but I expect some folks may pony-up) please see: www.voip-info.org/?page=Asterisk+Bounty+Email+TTS We are at $150 & counting. Maybe lobby your exec's for $50 to contribute to this,

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

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse

Hi, For an organization welcoming turists (in France), I would be curious to learn about successful use (with Asterisk) of Text-To-Speech in spanish (and english). I took a look at Cepstral's web site and saw there 2 "Americas Spanish" voices (along a bunch of english voices). 1. In this context, according to your experience, is it acceptable to use an "Americas Spanish"

Can someone point me to a page about writing a text file to call an external number and play a TTS with cepstral? I know it includes the creation of a .call file but beyond that im a bit lost.