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...rom the default. > > Thanks, > > Jason > Jason-- Assuming you follow the installation instructions, and install the Mbrola and other goodies for all the possible different voices, then you can, while running festival in a terminal window, run the following commands (in 195): (SayText "This is the default voice. It sounds like K.E.D. Diphone?") (tts "text_to_read" nil) (voice_cstr_us_awb_arctic_multisyn) (SayText "This is AWB arctic multisyn") (tts "text_to_read" nil) (voice_cstr_us_jmk_arctic_multisyn) (SayText "This is JMK arctic...

...having it say jokes at double speed, well, it reminds me of speed reading. There must be some lever or pulley or switch or something to modify the speed. AND Hoo boy, try putting this in your siteinit.scm file: (set! voice_default 'voice_cstr_us_awb_arctic_multisyn) and listen to this: (SayText "Hello there, kyootee pie.") (SayText "Don't you just love the sound of my voice.") (SayText "My wife, Sonya, Makes the best bread there ever was") Best synthetic voice I've ever heard. murf -- Steve Murphy Electronic Tools Company -------------- next par...

...] host=127.0.0.1 port=1314 usecache=no cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n Everything is running on the same box. I have rebooted... nothing is var log messages either. The local festival_client connects and I can put in (SayText "I can say numbers like") and it works great. The festival_server log show only this for the calls from asterick: client(11) Thu Aug 18 17:53:01 2005 : accepted from (my machine name here) client(11) Thu Aug 18 17:53:01 2005 : disconnected So it looks like it is connecting right. How...

Hi all, I need to simply use Asterisk to receive incoming calls in an IVR manner. It should authenticate users and read data from MySQL table that match their ID through Text-to-speech. I already have Asterisk 2.6 (Asterisk@home). I understand that I need to use Festival and AGI but do not know what to do exactly. Any help is appreciated. Thanks

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24

...n/festival --server server Wed Aug 4 17:49:04 2010 : Festival server started on port 1314 client(1) Wed Aug 4 17:50:12 2010 : accepted from localhost client(1) Wed Aug 4 17:50:12 2010 : disconnected I have to end the call after sometime. Festival works fine if I got into its console and type SayText(?hello john?) Please let me know how I can fix this. Thanks, Davinder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100804/0484e940/attachment.htm

..., it will consider that as silence also. My AEL dialplan associated with the calling is: > 100 => { > Answer(); > WaitForSilence(5000,2,60); > AGI(agi://127.0.0.1/playmessage,${CALLID}); > AGI(agi://127.0.0.1/saytext,"Goodbye."); > Hangup(); > } And the CLI just outputs: > == Using SIP RTP CoS mark 5 > > Channel SIP/twilio-0000006e was answered > -- Executing [100 at makeCall:1] Answer("SIP/twilio-0000006e", "") in > new s...

..., it will consider that as silence also. My AEL dialplan associated with the calling is: > 100 => { > Answer(); > WaitForSilence(5000,2,60); > AGI(agi://127.0.0.1/playmessage,${CALLID}); > AGI(agi://127.0.0.1/saytext,"Goodbye."); > Hangup(); > } And the CLI just outputs: > == Using SIP RTP CoS mark 5 > > Channel SIP/twilio-0000006e was answered > -- Executing [100 at makeCall:1] Answer("SIP/twilio-0000006e", "") in > new s...

...39; (escape_digits=) (sample_offset 0) -- Playing 'visibility' (escape_digits=) (sample_offset 0) -- <DAHDI/1-1> Playing 'digits/10.ulaw' (language 'en') -- Playing 'miles' (escape_digits=) (sample_offset 0) -- AGI Script Executing Application: (SayText) Options: (sunrise . 6 58 A M ) Once it played "wx/winds", it just read numbers and didn't play any of the other files. I've seen this before and it later goes away. But restarting Asterisk didn't do it. Any ideas what's going on?

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr