@wazo/sdk 0.37.30 → 0.37.31
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
- package/dist/wazo-sdk.js +4 -4
- package/esm/domain/CallSession.js +1 -1
- package/esm/domain/Phone/CTIPhone.js +1 -1
- package/esm/domain/Phone/WebRTCPhone.js +10 -2
- package/esm/domain/__tests__/CallSession.test.js +1 -1
- package/esm/domain/__tests__/IndirectTransfer.test.js +1 -1
- package/esm/domain/__tests__/Room.test.js +1 -1
- package/esm/index.js +10 -2
- package/esm/simple/Auth.js +10 -2
- package/esm/simple/Configuration.js +10 -2
- package/esm/simple/Phone.js +10 -2
- package/esm/simple/index.js +10 -2
- package/esm/simple/room/LocalParticipant.js +10 -2
- package/esm/simple/room/Participant.js +10 -2
- package/esm/simple/room/RemoteParticipant.js +10 -2
- package/esm/simple/room/Room.js +10 -2
- package/esm/simple/room/SipRoom.js +10 -2
- package/esm/simple/room/Stream.js +10 -2
- package/esm/simple/room/__tests__/RemoteParticipant.test.js +10 -2
- package/esm/simple/utils.js +10 -2
- package/lib/domain/CallSession.js +1 -1
- package/lib/domain/Phone/CTIPhone.js +1 -1
- package/lib/domain/Phone/WebRTCPhone.js +10 -2
- package/lib/domain/__tests__/CallSession.test.js +1 -1
- package/lib/domain/__tests__/IndirectTransfer.test.js +1 -1
- package/lib/domain/__tests__/Room.test.js +1 -1
- package/lib/index.js +10 -2
- package/lib/simple/Auth.js +10 -2
- package/lib/simple/Configuration.js +10 -2
- package/lib/simple/Phone.js +10 -2
- package/lib/simple/index.js +10 -2
- package/lib/simple/room/LocalParticipant.js +10 -2
- package/lib/simple/room/Participant.js +10 -2
- package/lib/simple/room/RemoteParticipant.js +10 -2
- package/lib/simple/room/Room.js +10 -2
- package/lib/simple/room/SipRoom.js +10 -2
- package/lib/simple/room/Stream.js +10 -2
- package/lib/simple/room/__tests__/RemoteParticipant.test.js +10 -2
- package/lib/simple/utils.js +10 -2
- package/package.json +1 -1
- package/src/domain/CallSession.js +1 -1
- package/src/domain/Phone/WebRTCPhone.js +9 -1
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@@ -368,7 +368,7 @@ class CallSession {
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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@@ -2042,7 +2042,7 @@ class CallSession {
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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@@ -390,7 +390,7 @@ class CallSession {
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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@@ -1504,6 +1504,7 @@ class WebRTCPhone extends Emitter {
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}
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
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return callSession;
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@@ -1720,6 +1721,7 @@ class WebRTCPhone extends Emitter {
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this.shouldSendReinvite = false;
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.acceptedSessions[callSession.getId()] = true;
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this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
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@@ -1819,7 +1821,7 @@ class WebRTCPhone extends Emitter {
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// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
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if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
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logger.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
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this.holdSipSession(this.currentSipSession,
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this.holdSipSession(this.currentSipSession, this.currentCallSession);
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}
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const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
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}
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const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
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@@ -2309,6 +2312,8 @@ class WebRTCPhone extends Emitter {
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}
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
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});
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logger.info('WebRTC call rejected', session.id);
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const callSession = this._createCallSession(session);
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callSession.endTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
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});
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@@ -2576,6 +2583,7 @@ class WebRTCPhone extends Emitter {
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displayName: identity ? identity.displayName || number : number,
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startTime: fromSession ? fromSession.startTime : new Date(),
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creationTime: fromSession ? fromSession.creationTime : null,
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answerTime: fromSession ? fromSession.answerTime : null,
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? new Date() : null;
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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package/esm/index.js
CHANGED
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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}
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
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return callSession;
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this.shouldSendReinvite = false;
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.acceptedSessions[callSession.getId()] = true;
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this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
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// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
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if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
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logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
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this.holdSipSession(this.currentSipSession,
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this.holdSipSession(this.currentSipSession, this.currentCallSession);
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}
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const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
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const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
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}
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
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});
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logger$7.info('WebRTC call rejected', session.id);
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const callSession = this._createCallSession(session);
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callSession.endTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
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displayName: identity ? identity.displayName || number : number,
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startTime: fromSession ? fromSession.startTime : new Date(),
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creationTime: fromSession ? fromSession.creationTime : null,
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answerTime: fromSession ? fromSession.answerTime : null,
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endTime: fromSession ? fromSession.endTime : null,
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answered: state === SessionState.Established,
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paused: this.client.isCallHeld(sipSession),
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package/esm/simple/Auth.js
CHANGED
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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}
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get answered() {
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callSession.answerTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
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this.shouldSendReinvite = false;
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this._updateCallSession(callSession);
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this.acceptedSessions[callSession.getId()] = true;
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this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
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// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
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if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
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logger$8.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
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this.holdSipSession(this.currentSipSession,
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this.holdSipSession(this.currentSipSession, this.currentCallSession);
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const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
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const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
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callSession.creationTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
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logger$8.info('WebRTC call rejected', session.id);
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const callSession = this._createCallSession(session);
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callSession.endTime = new Date();
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this._updateCallSession(callSession);
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this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
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startTime: fromSession ? fromSession.startTime : new Date(),
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creationTime: fromSession ? fromSession.creationTime : null,
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answerTime: fromSession ? fromSession.answerTime : null,
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// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
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set answered(value ) {
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this.answerTime = value ? this.answerTime || new Date() : null;
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}
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get answered() {
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callSession.answerTime = new Date();
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|
+
this._updateCallSession(callSession);
|
|
9096
9097
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9097
9098
|
|
|
9098
9099
|
return callSession;
|
|
@@ -9309,6 +9310,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9309
9310
|
|
|
9310
9311
|
this.shouldSendReinvite = false;
|
|
9311
9312
|
callSession.answerTime = new Date();
|
|
9313
|
+
this._updateCallSession(callSession);
|
|
9312
9314
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9313
9315
|
|
|
9314
9316
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9408,7 +9410,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9408
9410
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9409
9411
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9410
9412
|
logger$8.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9411
|
-
this.holdSipSession(this.currentSipSession,
|
|
9413
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9412
9414
|
}
|
|
9413
9415
|
|
|
9414
9416
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9619,6 +9621,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9619
9621
|
}
|
|
9620
9622
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9621
9623
|
callSession.creationTime = new Date();
|
|
9624
|
+
this._updateCallSession(callSession);
|
|
9622
9625
|
|
|
9623
9626
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9624
9627
|
|
|
@@ -9898,6 +9901,8 @@ class WebRTCPhone extends Emitter {
|
|
|
9898
9901
|
}
|
|
9899
9902
|
|
|
9900
9903
|
callSession.creationTime = new Date();
|
|
9904
|
+
this._updateCallSession(callSession);
|
|
9905
|
+
|
|
9901
9906
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
9902
9907
|
});
|
|
9903
9908
|
|
|
@@ -9941,6 +9946,8 @@ class WebRTCPhone extends Emitter {
|
|
|
9941
9946
|
logger$8.info('WebRTC call rejected', session.id);
|
|
9942
9947
|
const callSession = this._createCallSession(session);
|
|
9943
9948
|
callSession.endTime = new Date();
|
|
9949
|
+
this._updateCallSession(callSession);
|
|
9950
|
+
|
|
9944
9951
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
9945
9952
|
});
|
|
9946
9953
|
|
|
@@ -10165,6 +10172,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10165
10172
|
displayName: identity ? identity.displayName || number : number,
|
|
10166
10173
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10167
10174
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10175
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10168
10176
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10169
10177
|
answered: state === SessionState.Established,
|
|
10170
10178
|
paused: this.client.isCallHeld(sipSession),
|
package/esm/simple/Phone.js
CHANGED
|
@@ -584,7 +584,7 @@ class CallSession {
|
|
|
584
584
|
|
|
585
585
|
// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
|
|
586
586
|
set answered(value ) {
|
|
587
|
-
this.answerTime = value ? new Date() : null;
|
|
587
|
+
this.answerTime = value ? this.answerTime || new Date() : null;
|
|
588
588
|
}
|
|
589
589
|
|
|
590
590
|
get answered() {
|
|
@@ -9093,6 +9093,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9093
9093
|
}
|
|
9094
9094
|
|
|
9095
9095
|
callSession.answerTime = new Date();
|
|
9096
|
+
this._updateCallSession(callSession);
|
|
9096
9097
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9097
9098
|
|
|
9098
9099
|
return callSession;
|
|
@@ -9309,6 +9310,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9309
9310
|
|
|
9310
9311
|
this.shouldSendReinvite = false;
|
|
9311
9312
|
callSession.answerTime = new Date();
|
|
9313
|
+
this._updateCallSession(callSession);
|
|
9312
9314
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9313
9315
|
|
|
9314
9316
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9408,7 +9410,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9408
9410
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9409
9411
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9410
9412
|
logger$8.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9411
|
-
this.holdSipSession(this.currentSipSession,
|
|
9413
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9412
9414
|
}
|
|
9413
9415
|
|
|
9414
9416
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9619,6 +9621,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9619
9621
|
}
|
|
9620
9622
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9621
9623
|
callSession.creationTime = new Date();
|
|
9624
|
+
this._updateCallSession(callSession);
|
|
9622
9625
|
|
|
9623
9626
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9624
9627
|
|
|
@@ -9898,6 +9901,8 @@ class WebRTCPhone extends Emitter {
|
|
|
9898
9901
|
}
|
|
9899
9902
|
|
|
9900
9903
|
callSession.creationTime = new Date();
|
|
9904
|
+
this._updateCallSession(callSession);
|
|
9905
|
+
|
|
9901
9906
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
9902
9907
|
});
|
|
9903
9908
|
|
|
@@ -9941,6 +9946,8 @@ class WebRTCPhone extends Emitter {
|
|
|
9941
9946
|
logger$8.info('WebRTC call rejected', session.id);
|
|
9942
9947
|
const callSession = this._createCallSession(session);
|
|
9943
9948
|
callSession.endTime = new Date();
|
|
9949
|
+
this._updateCallSession(callSession);
|
|
9950
|
+
|
|
9944
9951
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
9945
9952
|
});
|
|
9946
9953
|
|
|
@@ -10165,6 +10172,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10165
10172
|
displayName: identity ? identity.displayName || number : number,
|
|
10166
10173
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10167
10174
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10175
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10168
10176
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10169
10177
|
answered: state === SessionState.Established,
|
|
10170
10178
|
paused: this.client.isCallHeld(sipSession),
|
package/esm/simple/index.js
CHANGED
|
@@ -3584,7 +3584,7 @@ class CallSession {
|
|
|
3584
3584
|
|
|
3585
3585
|
// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
|
|
3586
3586
|
set answered(value ) {
|
|
3587
|
-
this.answerTime = value ? new Date() : null;
|
|
3587
|
+
this.answerTime = value ? this.answerTime || new Date() : null;
|
|
3588
3588
|
}
|
|
3589
3589
|
|
|
3590
3590
|
get answered() {
|
|
@@ -9357,6 +9357,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9357
9357
|
}
|
|
9358
9358
|
|
|
9359
9359
|
callSession.answerTime = new Date();
|
|
9360
|
+
this._updateCallSession(callSession);
|
|
9360
9361
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9361
9362
|
|
|
9362
9363
|
return callSession;
|
|
@@ -9573,6 +9574,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9573
9574
|
|
|
9574
9575
|
this.shouldSendReinvite = false;
|
|
9575
9576
|
callSession.answerTime = new Date();
|
|
9577
|
+
this._updateCallSession(callSession);
|
|
9576
9578
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9577
9579
|
|
|
9578
9580
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9672,7 +9674,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9672
9674
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9673
9675
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9674
9676
|
logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9675
|
-
this.holdSipSession(this.currentSipSession,
|
|
9677
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9676
9678
|
}
|
|
9677
9679
|
|
|
9678
9680
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9883,6 +9885,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9883
9885
|
}
|
|
9884
9886
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9885
9887
|
callSession.creationTime = new Date();
|
|
9888
|
+
this._updateCallSession(callSession);
|
|
9886
9889
|
|
|
9887
9890
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9888
9891
|
|
|
@@ -10162,6 +10165,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10162
10165
|
}
|
|
10163
10166
|
|
|
10164
10167
|
callSession.creationTime = new Date();
|
|
10168
|
+
this._updateCallSession(callSession);
|
|
10169
|
+
|
|
10165
10170
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
10166
10171
|
});
|
|
10167
10172
|
|
|
@@ -10205,6 +10210,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10205
10210
|
logger$7.info('WebRTC call rejected', session.id);
|
|
10206
10211
|
const callSession = this._createCallSession(session);
|
|
10207
10212
|
callSession.endTime = new Date();
|
|
10213
|
+
this._updateCallSession(callSession);
|
|
10214
|
+
|
|
10208
10215
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
10209
10216
|
});
|
|
10210
10217
|
|
|
@@ -10429,6 +10436,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10429
10436
|
displayName: identity ? identity.displayName || number : number,
|
|
10430
10437
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10431
10438
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10439
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10432
10440
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10433
10441
|
answered: state === SessionState.Established,
|
|
10434
10442
|
paused: this.client.isCallHeld(sipSession),
|
|
@@ -3610,7 +3610,7 @@ class CallSession {
|
|
|
3610
3610
|
|
|
3611
3611
|
// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
|
|
3612
3612
|
set answered(value ) {
|
|
3613
|
-
this.answerTime = value ? new Date() : null;
|
|
3613
|
+
this.answerTime = value ? this.answerTime || new Date() : null;
|
|
3614
3614
|
}
|
|
3615
3615
|
|
|
3616
3616
|
get answered() {
|
|
@@ -9357,6 +9357,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9357
9357
|
}
|
|
9358
9358
|
|
|
9359
9359
|
callSession.answerTime = new Date();
|
|
9360
|
+
this._updateCallSession(callSession);
|
|
9360
9361
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9361
9362
|
|
|
9362
9363
|
return callSession;
|
|
@@ -9573,6 +9574,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9573
9574
|
|
|
9574
9575
|
this.shouldSendReinvite = false;
|
|
9575
9576
|
callSession.answerTime = new Date();
|
|
9577
|
+
this._updateCallSession(callSession);
|
|
9576
9578
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9577
9579
|
|
|
9578
9580
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9672,7 +9674,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9672
9674
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9673
9675
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9674
9676
|
logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9675
|
-
this.holdSipSession(this.currentSipSession,
|
|
9677
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9676
9678
|
}
|
|
9677
9679
|
|
|
9678
9680
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9883,6 +9885,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9883
9885
|
}
|
|
9884
9886
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9885
9887
|
callSession.creationTime = new Date();
|
|
9888
|
+
this._updateCallSession(callSession);
|
|
9886
9889
|
|
|
9887
9890
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9888
9891
|
|
|
@@ -10162,6 +10165,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10162
10165
|
}
|
|
10163
10166
|
|
|
10164
10167
|
callSession.creationTime = new Date();
|
|
10168
|
+
this._updateCallSession(callSession);
|
|
10169
|
+
|
|
10165
10170
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
10166
10171
|
});
|
|
10167
10172
|
|
|
@@ -10205,6 +10210,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10205
10210
|
logger$7.info('WebRTC call rejected', session.id);
|
|
10206
10211
|
const callSession = this._createCallSession(session);
|
|
10207
10212
|
callSession.endTime = new Date();
|
|
10213
|
+
this._updateCallSession(callSession);
|
|
10214
|
+
|
|
10208
10215
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
10209
10216
|
});
|
|
10210
10217
|
|
|
@@ -10429,6 +10436,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10429
10436
|
displayName: identity ? identity.displayName || number : number,
|
|
10430
10437
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10431
10438
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10439
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10432
10440
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10433
10441
|
answered: state === SessionState.Established,
|
|
10434
10442
|
paused: this.client.isCallHeld(sipSession),
|
|
@@ -3610,7 +3610,7 @@ class CallSession {
|
|
|
3610
3610
|
|
|
3611
3611
|
// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
|
|
3612
3612
|
set answered(value ) {
|
|
3613
|
-
this.answerTime = value ? new Date() : null;
|
|
3613
|
+
this.answerTime = value ? this.answerTime || new Date() : null;
|
|
3614
3614
|
}
|
|
3615
3615
|
|
|
3616
3616
|
get answered() {
|
|
@@ -9357,6 +9357,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9357
9357
|
}
|
|
9358
9358
|
|
|
9359
9359
|
callSession.answerTime = new Date();
|
|
9360
|
+
this._updateCallSession(callSession);
|
|
9360
9361
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9361
9362
|
|
|
9362
9363
|
return callSession;
|
|
@@ -9573,6 +9574,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9573
9574
|
|
|
9574
9575
|
this.shouldSendReinvite = false;
|
|
9575
9576
|
callSession.answerTime = new Date();
|
|
9577
|
+
this._updateCallSession(callSession);
|
|
9576
9578
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9577
9579
|
|
|
9578
9580
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9672,7 +9674,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9672
9674
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9673
9675
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9674
9676
|
logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9675
|
-
this.holdSipSession(this.currentSipSession,
|
|
9677
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9676
9678
|
}
|
|
9677
9679
|
|
|
9678
9680
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9883,6 +9885,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9883
9885
|
}
|
|
9884
9886
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9885
9887
|
callSession.creationTime = new Date();
|
|
9888
|
+
this._updateCallSession(callSession);
|
|
9886
9889
|
|
|
9887
9890
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9888
9891
|
|
|
@@ -10162,6 +10165,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10162
10165
|
}
|
|
10163
10166
|
|
|
10164
10167
|
callSession.creationTime = new Date();
|
|
10168
|
+
this._updateCallSession(callSession);
|
|
10169
|
+
|
|
10165
10170
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
10166
10171
|
});
|
|
10167
10172
|
|
|
@@ -10205,6 +10210,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10205
10210
|
logger$7.info('WebRTC call rejected', session.id);
|
|
10206
10211
|
const callSession = this._createCallSession(session);
|
|
10207
10212
|
callSession.endTime = new Date();
|
|
10213
|
+
this._updateCallSession(callSession);
|
|
10214
|
+
|
|
10208
10215
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
10209
10216
|
});
|
|
10210
10217
|
|
|
@@ -10429,6 +10436,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10429
10436
|
displayName: identity ? identity.displayName || number : number,
|
|
10430
10437
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10431
10438
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10439
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10432
10440
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10433
10441
|
answered: state === SessionState.Established,
|
|
10434
10442
|
paused: this.client.isCallHeld(sipSession),
|
|
@@ -3584,7 +3584,7 @@ class CallSession {
|
|
|
3584
3584
|
|
|
3585
3585
|
// Retro-compatibility: `answered` was a boolean before. We can reproduce the behaviour with a getter/setter
|
|
3586
3586
|
set answered(value ) {
|
|
3587
|
-
this.answerTime = value ? new Date() : null;
|
|
3587
|
+
this.answerTime = value ? this.answerTime || new Date() : null;
|
|
3588
3588
|
}
|
|
3589
3589
|
|
|
3590
3590
|
get answered() {
|
|
@@ -9357,6 +9357,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9357
9357
|
}
|
|
9358
9358
|
|
|
9359
9359
|
callSession.answerTime = new Date();
|
|
9360
|
+
this._updateCallSession(callSession);
|
|
9360
9361
|
this.eventEmitter.emit(ON_CALL_ACCEPTED, callSession, cameraEnabled);
|
|
9361
9362
|
|
|
9362
9363
|
return callSession;
|
|
@@ -9573,6 +9574,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9573
9574
|
|
|
9574
9575
|
this.shouldSendReinvite = false;
|
|
9575
9576
|
callSession.answerTime = new Date();
|
|
9577
|
+
this._updateCallSession(callSession);
|
|
9576
9578
|
this.acceptedSessions[callSession.getId()] = true;
|
|
9577
9579
|
|
|
9578
9580
|
this.eventEmitter.emit(ON_CALL_ANSWERED, callSession);
|
|
@@ -9672,7 +9674,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9672
9674
|
// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
|
|
9673
9675
|
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
|
|
9674
9676
|
logger$7.info('WebRTC hold call after resume', { id: this.currentSipSession.id });
|
|
9675
|
-
this.holdSipSession(this.currentSipSession,
|
|
9677
|
+
this.holdSipSession(this.currentSipSession, this.currentCallSession);
|
|
9676
9678
|
}
|
|
9677
9679
|
|
|
9678
9680
|
const promise = this.unholdSipSession(sipSession, callSession, withEvent, isConference);
|
|
@@ -9883,6 +9885,7 @@ class WebRTCPhone extends Emitter {
|
|
|
9883
9885
|
}
|
|
9884
9886
|
const callSession = this._createOutgoingCallSession(sipSession, cameraEnabled || false);
|
|
9885
9887
|
callSession.creationTime = new Date();
|
|
9888
|
+
this._updateCallSession(callSession);
|
|
9886
9889
|
|
|
9887
9890
|
this.eventEmitter.emit(ON_PLAY_PROGRESS_SOUND, this.audioOutputDeviceId, this.audioOutputVolume);
|
|
9888
9891
|
|
|
@@ -10162,6 +10165,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10162
10165
|
}
|
|
10163
10166
|
|
|
10164
10167
|
callSession.creationTime = new Date();
|
|
10168
|
+
this._updateCallSession(callSession);
|
|
10169
|
+
|
|
10165
10170
|
this.eventEmitter.emit(ON_CALL_INCOMING, callSession, wantsToDoVideo);
|
|
10166
10171
|
});
|
|
10167
10172
|
|
|
@@ -10205,6 +10210,8 @@ class WebRTCPhone extends Emitter {
|
|
|
10205
10210
|
logger$7.info('WebRTC call rejected', session.id);
|
|
10206
10211
|
const callSession = this._createCallSession(session);
|
|
10207
10212
|
callSession.endTime = new Date();
|
|
10213
|
+
this._updateCallSession(callSession);
|
|
10214
|
+
|
|
10208
10215
|
this.eventEmitter.emit(ON_CALL_REJECTED, callSession);
|
|
10209
10216
|
});
|
|
10210
10217
|
|
|
@@ -10429,6 +10436,7 @@ class WebRTCPhone extends Emitter {
|
|
|
10429
10436
|
displayName: identity ? identity.displayName || number : number,
|
|
10430
10437
|
startTime: fromSession ? fromSession.startTime : new Date(),
|
|
10431
10438
|
creationTime: fromSession ? fromSession.creationTime : null,
|
|
10439
|
+
answerTime: fromSession ? fromSession.answerTime : null,
|
|
10432
10440
|
endTime: fromSession ? fromSession.endTime : null,
|
|
10433
10441
|
answered: state === SessionState.Established,
|
|
10434
10442
|
paused: this.client.isCallHeld(sipSession),
|