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installation_guide:changelog [2017/08/01 13:26]
javier [Voximal ChangeLog]
installation_guide:changelog [2018/05/29 11:51] (current)
borja
Line 3: Line 3:
 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 (28/​05/​2018)
 +------------------
  
-===== 14.0 ===== +modChange the default mark with the parameter id if set. 
-**12/​07/​2017** +mod: Change the Cereproc Cloud integration ​(SpeakExtended support). 
-  * addCorrection for DTMF grammar ​with white spaces. +add: Add parameter speed (and promptspeed) to change ​the TTS voice rate with Cereproc and Watson. 
-  * add: Add '​hidden'​ grammars to set replace word/string in the STT results. +add: Integration of GoogleSpeech V1.1, with the enhanced model
-  * add: Add '​hidden'​ grammars to set "​Phrases"​ to Google STT. +addAdd Google Text To Speech ​support. 
-  * add: Silences record with STT, with speechrecordsilence option+addAdd Google Speech Streaming features ​to improve ​the results
-  ​* ​mod: Changes for Google Speech API V1 (not beta). +add: Add transferaudio support
-  ​* ​add: Add the property recognizemodel for Watson ​STT+mod: Corrections in STT streaming ​(async thread mode). 
-  ​* ​add: Escape the " by \" in the JSON string contents. +mod: Correction ​to detect the pause/​stop ​with Google Speech Streaming
-  * mod: Correction freePBX module for the option dialformat+add: Add codecs ULAW and OGG for Google Speech Streaming
-  * modCorrection to support ​uniMRCP configuration+add: Add a HTTP client ​for the <​data> ​tag (to keep SSL connections ​with chatbots APIs)
-  * modCorrection crashes with JSON/TEXT <​data>​ requests. +mod: Corrections with STT streaming ​and bargein
-  * mod: Correction ​to support HTTPS server reset connection (Keep-Alive with timeouts). +mod: Improvements in the STT streaming integration
-  * mod: Correction DEV logs for STT. +mod: Support ​maxspeechtimeout ​with the STT streaming
-  * mod: Correction with the speech beeps+mod: Correction ​to allows flexible URI in the attribute dest with the <​transfer>​
-  ​* ​add: Integration of the TTS Amazon/​Polly with CLI commands+add: Add bargein ​support with the Speech API Streaming
-  * mod: Correction HTTPS read timeouts (when SSL datas pendings). +mod: Correction ​to remove spaces ​from the digits/number ​grammar ​with STT
-  * mod: Escape the HTTP parameters characters. +add: Add Google Speech API with streaming
-  * mod: Corrections in the chunk HTTP download. +add: File streaming feature ​to get the speaking audio flow in the interpreter.
-  * 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>​