TTS and existing audio file

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TTS and existing audio file

Postby roll72 » Thu Feb 14, 2013 7:44 am

Hello,

I got Vicibox VERSION: 2.2.1-237 BUILD: 100510-2015. I'm trying to use the TTS with some TTS and ready wav file as described in http://www.vicidial.org/images/TTS_prompt_settings.jpg, I saw we can use reday wav or gsm file using "audio src" syntax, but could we use TTS without ceptral installed and then with only reday audio src files ?

Do someone got some documentation on it (I read the 2.4 manager manual but not found anything)?

thanks in advance
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Re: TTS and existing audio file

Postby williamconley » Thu Feb 14, 2013 9:15 am

You did not list your Vicibox version. The version you listed is for Vicidial (which is good), but the installer (Vicibox) version is also quite handy to list in your specs. As is your asterisk version.

Have you tried your solution to see if it works?

I doubt that it will, because I believe the commands in question are executed by Cepstral ... which obviously won't work if Cepstral is missing.
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Re: TTS and existing audio file

Postby roll72 » Fri Feb 15, 2013 4:21 am

Hello,

I saw the new cepstral do not support vicidial as previously (I found a wiki on that).
I was wondering to modify the script who run Cepstral voice generation, and replace it with a command like 'sox' which could make the concanetation of multiple audio file in one.

what do you think of that ?
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Re: TTS and existing audio file

Postby mflorell » Fri Feb 15, 2013 6:45 am

The new version of Cepstral works just fine with Vicidial, as long as you buy it with the batch option.
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Re: TTS and existing audio file

Postby williamconley » Sat Feb 16, 2013 3:25 pm

sox does not help because it will merely convert/combine/split/modify existing audio files. if you have an audio file generated by cepstral ... asterisk will just play it. so sox is not needed. but you must purchase the version of cepstral that will write the file to disk to use this method (and any necessary conversion already happens with the existing cepstral/vicidial stock methodology). if you purchase a cepstral license that writes directly into an asterisk channel, however, sox cannot help as there is no file to convert.
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Re: TTS and existing audio file

Postby roll72 » Mon Feb 18, 2013 3:50 am

What is the version of the cepstral to choose ? do I need to purchase the "save to file" option, or it is done automatically by the cache process in the vicidial script ?
My need : some of the message will be the same for many leads, so I don't need it to be regenarate each time.

thanks
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Re: TTS and existing audio file

Postby williamconley » Mon Feb 18, 2013 3:23 pm

For version: You need to purchase the save to file option and the version that will fit the Asterisk version in your system. Of course, you could try the new asterisk 1.8 version of Vicidial, but that configuration is still in Beta.
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Re: TTS and existing audio file

Postby ctc_olsen » Sun Apr 21, 2013 1:13 pm

mflorell wrote:The new version of Cepstral works just fine with Vicidial, as long as you buy it with the batch option.


Hi. What is meant by "batch option"? We purchased Callie and to ports. I was looking for a confirmation elsewhere. Will Cepstral 6 work with Vicidial with Asterisk version 1.4?
VERSION: 2.4-309a BUILD: 110430-1642 (Upgrade from CE 2.0,ISO) | Asterisk 1.4.27.1-1 | VmWare vCenter Server Ver 4.1.0| No additional software | No Digium/Sangoma Hardware
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Re: TTS and existing audio file

Postby williamconley » Sun Apr 21, 2013 2:43 pm

Cepstral 6 is not directly compatible with asterisk 1.4.

if you want cepstral 6, you will need to purchase a license that will allow you to write the audio to disk.
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Re: TTS and existing audio file

Postby elmer » Fri Mar 07, 2014 11:13 pm

WARNING[7125]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Mar 8 11:48:32] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 8 11:48:32] -- Executing [916267985848@default:3] Hangup("Local/916267985848@default-feff,2", "") in new stack
[Mar 8 11:48:32] == Spawn extension (default, 916267985848, 3) exited non-zero on 'Local/916267985848@default-feff,2'
[Mar 8 11:48:32] -- Executing [h@default:1] DeadAGI("Local/916267985848@default-feff,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODE-------") in new stack
[Mar 8 11:48:32] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 8 11:48:32] -- Executing [917075452561@default:2] Dial("Local/917075452561@default-aa87,2", "SIP/17075452561@APN||TtoR") in new stack
[Mar 8 11:48:32] WARNING[7128]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Mar 8 11:48:32] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 8 11:48:32] -- Executing [917075452561@default:3] Hangup("Local/917075452561@default-aa87,2", "") in new stack
[Mar 8 11:48:32] == Spawn extension (default, 917075452561, 3) exited non-zero on 'Local/917075452561@default-aa87,2'
[Mar 8 11:48:32] -- Executing [h@default:1] DeadAGI("Local/917075452561@default-aa87,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODE-------") in new stack
[Mar 8 11:48:32] == Parsing '/etc/asterisk/manager.conf': [Mar 8 11:48:32] Found
[Mar 8 11:48:32] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 8 11:48:32] -- Executing [914062914969@default:1] AGI("Local/914062914969@default-0c47,2", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 8 11:48:32] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 8 11:48:32] -- Executing [914062914969@default:2] Dial("Local/914062914969@default-0c47,2", "SIP/14062914969@APN||TtoR") in new stack
[Mar 8 11:48:32] WARNING[7133]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Mar 8 11:48:32] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 8 11:48:32] -- Executing [914062914969@default:3] Hangup("Local/914062914969@default-0c47,2", "") in new stack
[Mar 8 11:48:32] == Spawn extension (default, 914062914969, 3) exited non-zero on 'Local/914062914969@default-0c47,2'
[Mar 8 11:48:32] -- Executing [h@default:1] DeadAGI("Local/914062914969@default-0c47,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODE-------") in new stack
[Mar 8 11:48:32] == Parsing '/etc/asterisk/manager.conf': [Mar 8 11:48:32] Found
[Mar 8 11:48:32] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 8 11:48:32] -- Executing [919095873108@default:1] AGI("Local/919095873108@default-83a5,2", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 8 11:48:32] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 8 11:48:32] -- Executing [919095873108@default:2] Dial("Local/919095873108@default-83a5,2", "SIP/19095873108@APN||TtoR") in new stack
[Mar 8 11:48:32] WARNING[7137]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Mar 8 11:48:32] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 8 11:48:32] -- Executing [919095873108@default:3] Hangup("Local/919095873108@default-83a5,2", "") in new stack
[Mar 8 11:48:32] == Spawn extension (default, 919095873108, 3) exited non-zero on 'Local/919095873108@default-83a5,2'
[Mar 8 11:48:32] -- Executing [h@default:1] DeadAGI("Local/919095873108@default-83a5,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODE-------") in new stack



help me for this error
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