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https://github.com/jambonz/jambonz-feature-server.git
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Feature/minimal media anchoring (#36)
* initial WIP to remove freeswitch from media path when not recording or transcribing dial calls * implement release-media and anchor-media operations * mute/unmute now handled by rtpengine * Dial: dtmf detection now based on SIP INFO events from sbcs and rtpengine * add reason to gather action, bugfixes for transcribe and say
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@@ -21,9 +21,15 @@ class TaskSay extends Task {
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const {updateSpeechCredentialLastUsed} = require('../utils/db-utils')(this.logger, srf);
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const {writeAlerts, AlertType, stats} = srf.locals;
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const {synthAudio} = srf.locals.dbHelpers;
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const vendor = this.synthesizer.vendor || cs.speechSynthesisVendor;
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const language = this.synthesizer.language || cs.speechSynthesisLanguage;
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const voice = this.synthesizer.voice || cs.speechSynthesisVoice;
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const vendor = ('default' === this.synthesizer.vendor || !this.synthesizer.vendor) ?
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cs.speechSynthesisVendor :
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this.synthesizer.vendor;
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const language = ('default' === this.synthesizer.language || !this.synthesizer.language) ?
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cs.speechSynthesisLanguage :
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this.synthesizer.language;
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const voice = ('default' === this.synthesizer.voice || !this.synthesizer.voice) ?
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cs.speechSynthesisVoice :
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this.synthesizer.voice;
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const salt = cs.callSid;
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const credentials = cs.getSpeechCredentials(vendor, 'tts');
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