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installation_guide:changelog [2017/07/31 21:40]
javier
installation_guide:changelog [2018/05/29 11:51] (current)
borja
Line 1: Line 1:
 ====== Voximal ChangeLog ====== ====== Voximal ChangeLog ======
  
 +The old name of Voximal is VXI* : [[vxi_installation_guide:​changelog|full ChangeLog]]
  
-===== 14.0 ===== +Voximal is the new generation ​of [[vxi_installation_guide:changelog#​120|VXI12.0]] , and integrates [[vxi_installation_guide:changelog#​130|branch 13.0]] 
-**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). +
-  * addIntegration 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 filewith 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. +
-  * modCorrection 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.+
  
----- 
  
-The old name of Voximal is VXI* : [[vxi_installation_guide:​changelog|full change log]]+<​code>​ 
 +14.2 (28/​05/​2018) 
 +------------------
  
-Voximal is a fork of [[vxi_installation_guide:changelog#​120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#​130|branch 13.0]] +modChange the default mark with the parameter id if set. 
 +mod: Change the Cereproc Cloud integration (SpeakExtended support). 
 +add: Add parameter speed (and promptspeed) to change the TTS voice rate with Cereproc and Watson. 
 +add: Integration of GoogleSpeech V1.1with the enhanced model. 
 +add: Add Google Text To Speech support. 
 +add: Add Google Speech Streaming features to improve the results. 
 +add: Add transferaudio support. 
 +mod: Corrections in STT streaming (async thread mode). 
 +mod: Correction to detect the pause/stop with Google Speech Streaming. 
 +add: Add codecs ULAW and OGG for Google Speech Streaming. 
 +add: Add a HTTP client for the <​data>​ tag (to keep SSL connections with chatbots APIs). 
 +mod: Corrections with STT streaming and bargein. 
 +mod: Improvements in the STT streaming integration. 
 +mod: Support maxspeechtimeout with the STT streaming. 
 +mod: Correction to allows flexible URI in the attribute dest with the <​transfer>​. 
 +add: Add bargein support with the Speech API Streaming. 
 +mod: Correction to remove spaces from the digits/​number grammar with STT. 
 +add: Add Google Speech API with streaming. 
 +addFile streaming feature to get the speaking audio flow in the interpreter.
  
 +
 +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>​