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installation_guide:changelog [2018/05/29 11:51] – borja | installation_guide:changelog [2020/04/01 13:35] – [{{:wiki:logo-apple.png?400|}}Voximal ChangeLog] javier | ||
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====== Voximal ChangeLog ====== | ====== Voximal ChangeLog ====== | ||
- | + | {{: | |
- | The old name of Voximal is VXI* : [[vxi_installation_guide: | + | The old name of Voximal is VXI* : [[vxi_installation_guide: |
Voximal is the new generation of [[vxi_installation_guide: | Voximal is the new generation of [[vxi_installation_guide: | ||
- | < | ||
- | 14.2 (28/ | ||
- | ------------------ | ||
- | |||
- | mod: Change 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 V1.1, with 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 < | ||
- | 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. | + | ====== 14.2 ====== |
- | add: Add speechrecordsilence parameter for the Speech recording. | + | **02/01/2020** |
- | 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/ | + | |
- | mod: Correction of JSON/Ecma conversions. | + | |
- | mod: Correction of the GUI/General parameters. | + | |
- | mod: Correction in the install script. | + | |
- | add: Support for Asterisk 15. | + | |
+ | * mod: Improve the AAI support. (Can be linked to User-To-User SIP header). | ||
+ | * mod: Improve the fetchaudio feature. | ||
+ | * mod: Add Google Assistant Dialogflow payload support. | ||
+ | * mod: Correction to ignore XML tags in the NLSML < | ||
+ | * mod: Important correction in the earlyspeech feature. | ||
+ | * add: Add ConfBridge conference application support (Transfer with conference: prefix). | ||
+ | * add: Add a property to manage the Google Credentials files name and allows serveral GoogleSpeech/ | ||
+ | * mod: Correction coredumps with command "clear cache" and files older than x days. | ||
+ | * add: Add < | ||
+ | * mod: Correction to force the score with Google Speech when empty final result. | ||
+ | * mod: Modification to send speechprovider with load grammar URI. | ||
+ | * mod: Correction aoround cut/break features. | ||
+ | * add: Add property to define the TTS text/ssml encoding for MRCP contents. | ||
+ | * mod: Add parameter a speechunanswered to keep grammars errors during unanswered processing. | ||
+ | * mod: Correction for MRCP (bad streaming state check). | ||
+ | * mod: Correction to extend the buffer to receive long ASR/STT results. | ||
+ | * mod: Correction crash with grammars and unaswered mode. | ||
+ | * mod: Update to the last openssl, grpc libraries (SSL/ALPN property). | ||
+ | * mod: Corrections for HTTP speech processing. | ||
+ | * add: Variable SPYGROUP for ChanSpy using. | ||
+ | * add: Support payload from Dialogflow webhooks (expect_user_response). | ||
+ | * add: Add multicontext support for Google Dialogflow (with xml: | ||
+ | * mod: Correction to send the JSGF grammars to the ASR engine with uniMRCP. | ||
+ | * add: Add a new bargein mode called "early speech" | ||
+ | * add: Average values for feedback and response time (between recognizes and prompts). | ||
+ | * mod: Correction in the messaging waiting step (when the stream/ | ||
+ | * add: Add match RegExp expressions in the grammars, prefix the item with the ' | ||
+ | * add: Add multilanguage support for Google Speech (with xml: | ||
+ | * mod: Correction coredump in assignments with void EcmaScript objects. | ||
+ | * mod: Correction to use the attribut timeout in the < | ||
+ | * add: Add <clear namelist=":"/> | ||
+ | * mod: Correction of the default memory audio directory (transfer announcement and karaoke features). | ||
+ | * add: Add parameters audiotruncate and audiopollard to cut TTS prompts. | ||
+ | * add: Add property encodingtype to set or modify the default POST encodingtype. | ||
+ | * mod: Do not clean text contents with the TTV (language=text or video). | ||
+ | * add: Add the parameters maxLogFileSize and maxContentDirSize. | ||
+ | * mod: Correction to keep the set " | ||
+ | * mod: Corrections around Google RPC timings and answers. | ||
+ | * mod: Correction to support prompt properties with the Watson API. | ||
+ | * mod: Correction to support DTMF results with score equal to ' | ||
+ | * mod: Correction to forward DTMF events to the speech API (ASR) if DTMF grammar is used. | ||
+ | * mod: Dynamic grammar now support empty srcexpr with throwing an error event. | ||
+ | * mod: Change the word/ | ||
+ | * add: Add the parameter Speech Provider in the General section in the FreePBX module. | ||
+ | * add: Add Dialogflow shadow variables (hangup, intent, name, property...). | ||
+ | * add: Add Parameters for the interpreter properties (section [interpreter]). | ||
+ | * add: Add parameter minimalspeech, | ||
+ | * mod: Correction for the TTS with MRCP configuration. | ||
+ | * add: Add property promptlang (and xml:lang) to force the languauge for all prompts. | ||
+ | * add: Add the option useredirect for SIP redirected calls (with OVH french VOIP provider). | ||
+ | * mod: Disable the option speechhotwordscore by default. | ||
+ | * add: Add " | ||
+ | * add: Add " | ||
+ | * mod: Improvments with the Dialogflow integration (set request payload, process result parameters, stability). | ||
+ | * add: Add " | ||
+ | * mod: Correction bug with the cutprompt feature. | ||
+ | * add: Add extraheaders property. | ||
+ | * add: Add parameters to generate silence to the audio streaming (with RTP silence). | ||
+ | * add: Add parameters to record audio streaming (GoogleSpeech/ | ||
+ | * mod: Remove too verbose traces during HTTP downloads. | ||
+ | * mod: Continue Playing the same audiotransfer during several sequential transfers. | ||
+ | * add: Add Yandex Synthesis support. | ||
+ | * add: Add Dialogflow integration with RPC (text/event and streaming modes). | ||
+ | * add: Add multiple text/ | ||
+ | * mod: Default STT streaming set to ulaw. | ||
+ | * add: Add the minspeech property to start the STT only if we reach a minimal speech duration. | ||
+ | * mod: Change 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 V1.1, with 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 < | ||
+ | * 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. | ||
- | 14.0 (12/07/2017) | + | ====== |
- | ------------------ | + | **28/05/2018** |
- | add: Correction for DTMF grammar with white spaces. | + | * add: Add the server name to the frame Hello with SSL. |
- | 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 | + | |
- | 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 | + | |
- | 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 | + | |
- | add: Add JSON support for < | + | |
- | mod: Correction | + | |
- | 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 | + | |
- | 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 | + | |
- | 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 | + | |
- | add: Asterisk 14 support. | + | |
- | mod: Disable | + | |
- | mod: Correction | + | |
- | 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, | + | |
- | 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 | + | |
- | mod: Correction | + | |
- | 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 | + | |
- | add: Clean text results from Loquendo ASR (speechclean parameter removes spaces and CR). | + | |
- | mod: Correction | + | |
- | 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 | + | |
- | add: Remote License System integration. | + | |
- | mod: Correction to support https:// uri as Vxml() parameter. | + | |
- | </ | + | |
+ | ====== 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. | ||
+ | |