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installation_guide:changelog [2017/07/31 21:37] – javier | installation_guide:changelog [2018/04/08 23:07] – borja | ||
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====== Voximal ChangeLog ====== | ====== Voximal ChangeLog ====== | ||
- | ===== 14.0 ===== | + | The old name of Voximal is VXI* : [[vxi_installation_guide:changelog|full ChangeLog]] |
- | **12/ | + | |
- | * add: Correction for DTMF grammar with white spaces. | + | |
- | * add: Add ' | + | |
- | * add: Add ' | + | |
- | * 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 < | + | |
- | * 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 | + | |
- | * 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? | + | |
- | * mod: Correction of an issue with the MRCPsynth extra parameters. | + | |
- | * add: Add a mark for the VM and specific | + | |
- | * add: Use the sensibility end completetimeout to adjust the speech recording for STT. | + | |
- | * add: Add JSON support for < | + | |
- | * mod: Correction of a memoryleak with a debug trace in the < | + | |
- | * 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/ | + | |
- | * 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/ | + | |
- | * mod: Disable the default POST/ | + | |
- | * 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 / | + | |
- | * mod: Change cache directory to / | + | |
- | * add: Support of NLSML answers from Telisma ASR engine. | + | |
- | * add: Option unimrcp to start unimrcpserver, | + | |
- | * 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 ' | + | |
- | * 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 < | + | |
- | * 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=" | + | |
- | * 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. | + | |
- | | + | |
- | ---- | + | Voximal is the new generation of [[vxi_installation_guide: |
- | The old name of Voximal is VXI* : [[vxi_installation_guide: | ||
- | Voximal is a fork of [[vxi_installation_guide: | + | < |
+ | 14.2 (07/ | ||
+ | ------------------ | ||
+ | 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 < | ||
+ | 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 < | ||
+ | add: Add bargein support with the Speech API Streaming. | ||
+ | mod: Correction to remove spaces from the digits/ | ||
+ | add: Add Google Speech API with streaming. | ||
+ | add: File streaming feature to get the speaking audio flow in the interpreter. | ||
+ | |||
+ | |||
+ | 14.1 (06/ | ||
+ | ------------------ | ||
+ | |||
+ | 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: / | ||
+ | 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/ | ||
+ | ------------------ | ||
+ | |||
+ | add: Correction for DTMF grammar with white spaces. | ||
+ | add: Add ' | ||
+ | add: Add ' | ||
+ | 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 < | ||
+ | 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/ | ||
+ | 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? | ||
+ | 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 < | ||
+ | mod: Correction of a memoryleak with a debug trace in the < | ||
+ | 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/ | ||
+ | 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/ | ||
+ | mod: Disable the default POST/ | ||
+ | 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 / | ||
+ | mod: Change cache directory to / | ||
+ | add: Support of NLSML answers from Telisma ASR engine. | ||
+ | add: Option unimrcp to start unimrcpserver, | ||
+ | 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 ' | ||
+ | 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 < | ||
+ | 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=" | ||
+ | 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. | ||
+ | </ | ||