This shows you the differences between two versions of the page.

Link to this comparison view

Both sides previous revision Previous revision
Next revision
Previous revision
installation_guide:changelog [2017/07/31 21:43] javierinstallation_guide:changelog [2020/04/01 13:37] (current) – [Voximal ChangeLog] javier
Line 1: Line 1:
 ====== Voximal ChangeLog ====== ====== Voximal ChangeLog ======
 +The old name of Voximal is VXI* : [[vxi_installation_guide:changelog|full VXI* ChangeLog]]
-The old name of Voximal is VXI* [[vxi_installation_guide:changelog|full change log]]+Voximal is the new generation of [[vxi_installation_guide:changelog#120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#130|branch 13.0]] 
-Voximal is a fork of [[vxi_installation_guide:changelog#120|VXI* 12.0]] , and integrates [[vxi_installation_guide:changelog#130|branch 13.0]]  
 +====== 14.2 ======
-===== 14.0 =====+  * 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 ======
 **12/07/2017** **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.
Line 83: Line 189:
   * 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. 
- +  
----- +
- +
  • installation_guide/changelog.1501537425.txt.gz
  • Last modified: 2017/07/31 21:43
  • by javier