mirror of
https://github.com/jambonz/jambonz-feature-server.git
synced 2025-12-20 16:50:39 +00:00
72345f83c1
* 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
45 lines
947 B
JavaScript
45 lines
947 B
JavaScript
const CallSession = require('./call-session');
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/**
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* @classdesc Subclass of CallSession. Represents a CallSession
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* that was initially a child call leg; i.e. established via a Dial verb.
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* Now it is all grown up and filling out its own CallSession. Yoo-hoo!
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* @extends CallSession
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*/
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class AdultingCallSession extends CallSession {
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constructor({logger, application, singleDialer, tasks, callInfo, accountInfo}) {
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super({
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logger,
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application,
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srf: singleDialer.dlg.srf,
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tasks,
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callInfo,
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accountInfo
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});
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this.sd = singleDialer;
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this.sd.dlg.on('destroy', () => {
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this.logger.info('AdultingCallSession: called party hung up');
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this._callReleased();
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});
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this.sd.emit('adulting');
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}
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get dlg() {
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return this.sd.dlg;
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}
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get ep() {
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return this.sd.ep;
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}
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get callSid() {
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return this.callInfo.callSid;
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}
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}
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module.exports = AdultingCallSession;
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