Voximal ChangeLog

The old name of Voximal is VXI* : full ChangeLog

Voximal is the new generation of VXI* 12.0 , and integrates branch 13.0

14.2 (28/05/2018)

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 (06/10/2017)

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.