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installation_guide:changelog [2017/07/31 21:31] – [14.0 (12/07/2017)] javierinstallation_guide:changelog [2020/04/01 13:37] (current) – [Voximal ChangeLog] javier
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 ====== Voximal ChangeLog ====== ====== Voximal ChangeLog ======
 +The old name of Voximal is VXI* : [[vxi_installation_guide:changelog|full VXI* ChangeLog]]
 +Voximal is the new generation of [[vxi_installation_guide:changelog#120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#130|branch 13.0]] 
 +====== 14.2 ======
 +  * 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 <input>.
 +  * 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/GoogleDialogflow connexions.
 +  * mod: Correction coredumps with command "clear cache" and files older than x days.
 +  * add: Add <script> extension to execute variable contents as an ECMA script.
 +  * 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:lang="context1:context2").
 +  * mod: Correction to send the JSGF grammars to the ASR engine with uniMRCP.
 +  * add: Add a new bargein mode called "early speech". Gets speech and interrupts prompts during the end of the prompts.
 +  * add: Average values for feedback and response time (between recognizes and prompts).
 +  * mod: Correction in the messaging waiting step (when the stream/recognize have a result).
 +  * add: Add match RegExp expressions in the grammars, prefix the item with the '@' character.
 +  * add: Add multilanguage support for Google Speech (with xml:lang="fr-FR:es-ES").
 +  * mod: Correction coredump in assignments with void EcmaScript objects.
 +  * mod: Correction to use the attribut timeout in the <prompt>.
 +  * add: Add <clear namelist=":"/> to clear event and prompt counters.
 +  * 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 "context:" on fields after a noinput/nomatch event.
 +  * 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 '1' (Nuance ASR).
 +  * 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/grammar parsing to find full words and not substrings.
 +  * 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, to bufferize the STT streaming before starting the session.
 +  * 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 "[]" break marks as "{}" for Dialogflow.
 +  * add: Add "unanswered" mode to execute a VoiceXML session without a real call (function "url").
 +  * mod: Improvments with the Dialogflow integration (set request payload, process result parameters, stability).
 +  * add: Add "{}" break marks in prompt texts to add prerecorded audio files or wait time in ms.
 +  * 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/DialogFlow)
 +  * 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/transcribe grammars in a single document.
 +  * 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 <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.
 +  * add: File streaming feature to get the speaking audio flow in the interpreter.
 +====== 14.1 ======
 +  * 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 PID file from the interpreter too.
 +  * mod: Correction Voximal cleanup at Asterisk exit.
 +  * add: /var/run/voximal/ 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 ======
-===== 14.0 (12/07/2017) ===== 
   * add: Correction for DTMF grammar with white spaces.   * 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 replace word/string in the STT results.
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   * add: Start the interpreter from the Asterisk module.   * add: Start the interpreter from the Asterisk module.
   * add: Remote License System integration.   * add: Remote License System integration.
-  * mod: Correction to support https: uri as Vxml() parameter. +  * mod: Correction to support https:// uri as Vxml() parameter. 
- +  
----- +
- +
-The old name of Voximal is VXI* : [[vxi_installation_guide:changelog|full change log]] +
- +
-Voximal is a fork of [[vxi_installation_guide:changelog#120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#130|branch 13.0]]  +
- +
- +
  • installation_guide/changelog.1501536699.txt.gz
  • Last modified: 2017/07/31 21:31
  • by javier