Show pageOld revisionsBacklinksExport to PDFBack to top This page is read only. You can view the source, but not change it. Ask your administrator if you think this is wrong. {{:wiki:logo-apple.png?100|}} ====== 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 ====== **02/01/2020** * 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 ====== **28/05/2018** * 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. installation_guide/changelog.txt Last modified: 2020/04/01 13:37by javier