installation_guide:changelog

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installation_guide:changelog [2017/08/01 13:26] – [Voximal ChangeLog] javierinstallation_guide:changelog [2018/04/08 23:07] – borja
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 The old name of Voximal is VXI* : [[vxi_installation_guide:changelog|full ChangeLog]] The old name of Voximal is VXI* : [[vxi_installation_guide:changelog|full ChangeLog]]
  
-Voximal is a fork of [[vxi_installation_guide:changelog#120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#130|branch 13.0]] +Voximal is the new generation of [[vxi_installation_guide:changelog#120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#130|branch 13.0]] 
  
  
 +<code>
 +14.2 (07/04/2018)
 +------------------
  
-===== 14.0 ===== +add: Add Google Text To Speech support. 
-**12/07/2017** +add: Add Google Speech Streaming features to improve the results. 
-  * add: Correction for DTMF grammar with white spaces. +add: Add transferaudio support. 
-  * add: Add 'hidden' grammars to set replace word/string in the STT results. +mod: Corrections in STT streaming (async thread mode). 
-  * add: Add 'hidden' grammars to set "Phrases" to Google STT. +mod: Correction to detect the pause/stop with Google Speech Streaming. 
-  * add: Silences record with STT, with speechrecordsilence option. +add: Add codecs ULAW and OGG for Google Speech Streaming. 
-  * mod: Changes for Google Speech API V1 (not beta). +add: Add a HTTP client for the <data> tag (to keep SSL connections with chatbots APIs). 
-  * add: Add the property recognizemodel for Watson STT. +mod: Corrections with STT streaming and bargein. 
-  * add: Escape the " by \" in the JSON string contents. +mod: Improvements in the STT streaming integration. 
-  * mod: Correction freePBX module for the option dialformat. +mod: Support maxspeechtimeout with the STT streaming. 
-  * mod: Correction to support uniMRCP configuration. +mod: Correction to allows flexible URI in the attribute dest with the <transfer>. 
-  * mod: Correction crashes with JSON/TEXT <data> requests. +add: Add bargein support with the Speech API Streaming. 
-  * mod: Correction to support HTTPS server reset connection (Keep-Alive with timeouts). +mod: Correction to remove spaces from the digits/number grammar with STT. 
-  * mod: Correction DEV logs for STT. +add: Add Google Speech API with streaming. 
-  * mod: Correction with the speech beeps. +add: File streaming feature to get the speaking audio flow in the interpreter.
-  * add: Integration of the TTS Amazon/Polly with CLI commands. +
-  * mod: Correction HTTPS read timeouts (when SSL datas pendings). +
-  * mod: Escape the HTTP parameters characters. +
-  * mod: Corrections in the chunk HTTP download. +
-  * mod: Use the directories files and streams for the log contents. +
-  * add: Add parameter lang for the builtin grammar text (text?lang=x). +
-  * mod: Correction of an issue with the MRCPsynth extra parameters. +
-  * add: Add a mark for the VM and specific Voximal installs. +
-  * add: Use the sensibility end completetimeout to adjust the speech recording for STT. +
-  * add: Add JSON support for <data> +
-  * mod: Correction of a memoryleak with a debug trace in the <assign>. +
-  * mod: Add error messages relative to write disk errors and MSQ read errors. +
-  * mod: Set the PlayListSize to 1 to avoid MSQ lock when the MSQ size is to small. +
-  * add: Support STT with menus/options/grammars using (interpreter filters results). +
-  * add: Add speechprovider parameter for the accouts in the FreePBX module. +
-  * add: Integration of the STT IBM/Watson Cloud API (bluemix). +
-  * add: Integration of the STT Microsoft Cloud API. +
-  * add: Integration of ASR/STT with HTTP interface. +
-  * mod: Correction in MD5 functions for cache managment. +
-  * add: Integration of the TTS IBM/Watson Cloud API (bluemix). +
-  * add: Added speechverbio to support Verbio bultins grammars. +
-  * add: Integration of the TTS iSpeech Cloud API. +
-  * mod: Update for the new TTS Microsoft/Bing Cloud API. +
-  * mod: Disable the default POST/100-continue and add the property fetchcontinuetimeout. +
-  * mod: Disable the unload grammar execution by default. +
-  * add: Add wav16 and sln16 support. +
-  * mod: Allows to use one free port with an invalid key. +
-  * mod: Correction to avoid sending grammar actions without finishing the playlist queue. +
-  * add: Asterisk 14 support. +
-  * mod: Disable the POST continue for the TTS requests by default. +
-  * mod: Correction to fully support the POST continue to pass HTTP1.0 proxies. +
-  * add: Integration of the TTS Microsoft Bing Voice Output API. +
-  * add: Integration of the VoiceRSS Cloud Text-to-Speech API. +
-  * mod: Enable to start without configuration file, with defaults parameters. +
-  * mod: Change log directory to /var/log/voximal. +
-  * mod: Change cache directory to /var/cache/voximal. +
-  * add: Support of NLSML answers from Telisma ASR engine. +
-  * add: Option unimrcp to start unimrcpserver, as voximald. +
-  * add: Option cacheclear to clear the cache directories at startup. +
-  * add: Integration of the CereProc Cloud Text-to-Speech API. +
-  * add: Support ogg format (Vorbis OGG 8kHz). +
-  * add: Add max retries to avoid to disable the license immediately. +
-  * mod: Correction in the number and accurency DMTF builtin grammars. +
-  * mod: Correction to not inspect the tags with DTMF grammars. +
-  * add: Support sln format (PCM 16bit 8kHz Raw). +
-  * add: Add a parameter to use CALLERID with originate. +
-  * mod: Correction to parse the cookies parameter 'secure' and 'httponly'. +
-  * mod: Correction for MRCPsynth using without cache. +
-  * mod: Correction to allow VoiceXML execution after throwing the event disconnect. +
-  * add: Added mrcpsynthparams for accounts too. +
-  * add: Integration of the Voxygen Cloud hosted. +
-  * add: Clean text results from Loquendo ASR (speechclean parameter removes spaces and CR). +
-  * mod: Correction to catch and process the error.grammar events. +
-  * mod: Correction to support alternate prompt using <value>. +
-  * add: Refund to use functions, names and directories based on voximal. +
-  * add: Integration of the SpeechAAS TTS hosted. +
-  * add: Json HTTP/POST request support (with enctype="application/json") +
-  * mod: Set the Speech-Language property of the uniMRCP (for builtins grammars). +
-  * add: Auto-extend threads in the interpreter if needed. +
-  * add: Start the interpreter from the Asterisk module. +
-  * add: Remote License System integration. +
-  * mod: Correction to support https: uri as Vxml() parameter.+
  
-----+ 
 +14.1 (06/10/2017) 
 +------------------ 
 + 
 +add: Add the server name to the frame Hello with SSL. 
 +add: Add speechrecordsilence parameter for the Speech recording. 
 +mod: Correction regression with JSGF grammars with mode=dtmf. 
 +mod: Add the utterance in the nomatch for the SpeechToText. 
 +mod: Stop the process with signal TERM and after KILL. 
 +add: Add option -pid to create the voximald.pid PID file from the interpreter too. 
 +mod: Correction Voximal cleanup at Asterisk exit. 
 +add: /var/run/voximal/voximal.pid file. 
 +mod: Correction of JSON/Ecma conversions. 
 +mod: Correction of the GUI/General parameters. 
 +mod: Correction in the install script. 
 +add: Support for Asterisk 15. 
 + 
 + 
 +14.0 (12/07/2017) 
 +------------------ 
 + 
 +add: Correction for DTMF grammar with white spaces. 
 +add: Add 'hidden' grammars to set replace word/string in the STT results. 
 +add: Add 'hidden' grammars to set "Phrases" to Google STT. 
 +add: Silences record with STT, with speechrecordsilence option. 
 +mod: Changes for Google Speech API V1 (not beta). 
 +add: Add the property recognizemodel for Watson STT. 
 +add: Escape the " by \" in the JSON string contents. 
 +mod: Correction freePBX module for the option dialformat. 
 +mod: Correction to support uniMRCP configuration. 
 +mod: Correction crashes with JSON/TEXT <data> requests. 
 +mod: Correction to support HTTPS server reset connection (Keep-Alive with timeouts). 
 +mod: Correction DEV logs for STT. 
 +mod: Correction with the speech beeps. 
 +add: Integration of the TTS Amazon/Polly with CLI commands. 
 +mod: Correction HTTPS read timeouts (when SSL datas pendings). 
 +mod: Escape the HTTP parameters characters. 
 +mod: Corrections in the chunk HTTP download. 
 +mod: Use the directories files and streams for the log contents. 
 +add: Add parameter lang for the builtin grammar text (text?lang=x). 
 +mod: Correction of an issue with the MRCPsynth extra parameters. 
 +add: Add a mark for the VM and specific Voximal installs. 
 +add: Use the sensibility end completetimeout to adjust the speech recording for STT. 
 +add: Add JSON support for <data> 
 +mod: Correction of a memoryleak with a debug trace in the <assign>. 
 +mod: Add error messages relative to write disk errors and MSQ read errors. 
 +mod: Set the PlayListSize to 1 to avoid MSQ lock when the MSQ size is to small. 
 +add: Support STT with menus/options/grammars using (interpreter filters results). 
 +add: Add speechprovider parameter for the accouts in the FreePBX module. 
 +add: Integration of the STT IBM/Watson Cloud API (bluemix). 
 +add: Integration of the STT Microsoft Cloud API. 
 +add: Integration of ASR/STT with HTTP interface. 
 +mod: Correction in MD5 functions for cache managment. 
 +add: Integration of the TTS IBM/Watson Cloud API (bluemix). 
 +add: Added speechverbio to support Verbio bultins grammars. 
 +add: Integration of the TTS iSpeech Cloud API. 
 +mod: Update for the new TTS Microsoft/Bing Cloud API. 
 +mod: Disable the default POST/100-continue and add the property fetchcontinuetimeout. 
 +mod: Disable the unload grammar execution by default. 
 +add: Add wav16 and sln16 support. 
 +mod: Allows to use one free port with an invalid key. 
 +mod: Correction to avoid sending grammar actions without finishing the playlist queue. 
 +add: Asterisk 14 support. 
 +mod: Disable the POST continue for the TTS requests by default. 
 +mod: Correction to fully support the POST continue to pass HTTP1.0 proxies. 
 +add: Integration of the TTS Microsoft Bing Voice Output API. 
 +add: Integration of the VoiceRSS Cloud Text-to-Speech API. 
 +mod: Enable to start without configuration file, with defaults parameters. 
 +mod: Change log directory to /var/log/voximal. 
 +mod: Change cache directory to /var/cache/voximal. 
 +add: Support of NLSML answers from Telisma ASR engine. 
 +add: Option unimrcp to start unimrcpserver, as voximald. 
 +add: Option cacheclear to clear the cache directories at startup. 
 +add: Integration of the CereProc Cloud Text-to-Speech API. 
 +add: Support ogg format (Vorbis OGG 8kHz). 
 +add: Add max retries to avoid to disable the license immediately. 
 +mod: Correction in the number and accurency DMTF builtin grammars. 
 +mod: Correction to not inspect the tags with DTMF grammars. 
 +add: Support sln format (PCM 16bit 8kHz Raw). 
 +add: Add a parameter to use CALLERID with originate. 
 +mod: Correction to parse the cookies parameter 'secure' and 'httponly'. 
 +mod: Correction for MRCPsynth using without cache. 
 +mod: Correction to allow VoiceXML execution after throwing the event disconnect. 
 +add: Added mrcpsynthparams for accounts too. 
 +add: Integration of the Voxygen Cloud hosted. 
 +add: Clean text results from Loquendo ASR (speechclean parameter removes spaces and CR). 
 +mod: Correction to catch and process the error.grammar events. 
 +mod: Correction to support alternate prompt using <value>. 
 +add: Refund to use functions, names and directories based on voximal. 
 +add: Integration of the SpeechAAS TTS hosted. 
 +add: Json HTTP/POST request support (with enctype="application/json") 
 +mod: Set the Speech-Language property of the uniMRCP (for builtins grammars). 
 +add: Auto-extend threads in the interpreter if needed. 
 +add: Start the interpreter from the Asterisk module. 
 +add: Remote License System integration. 
 +mod: Correction to support https:// uri as Vxml() parameter. 
 +</code>
  
  
  • installation_guide/changelog.txt
  • Last modified: 2020/04/01 13:37
  • by javier