@voicenter-team/opensips-js 1.0.137 → 1.0.139
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/index.d.ts +39 -0
- package/dist/opensips-js.cjs.js +36 -36
- package/dist/opensips-js.es.js +645 -609
- package/dist/opensips-js.iife.js +23 -23
- package/dist/opensips-js.umd.js +23 -23
- package/package.json +1 -1
- package/src/types/listeners.d.ts +2 -0
package/dist/index.d.ts
CHANGED
@@ -52,6 +52,7 @@ declare class AudioModule {
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private isCallAddingInProgress;
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private muteWhenJoinEnabled;
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private isDNDEnabled;
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+
private isCallWaitingEnabled;
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private muted;
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private microphoneInputLevelValue;
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private speakerVolumeValue;
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@@ -92,6 +93,18 @@ declare class AudioModule {
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private set callAddingInProgress(value);
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get muteWhenJoin(): boolean;
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get isDND(): boolean;
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+
/**
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* Gets the current state of the call waiting feature.
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*
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* When call waiting is enabled (true), incoming calls will be allowed even when
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* other calls are active.
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*
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* When call waiting is disabled (false) and there are already active calls,
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* any new incoming calls will be automatically rejected with a "busy" status.
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*
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* @returns {boolean} True if call waiting is enabled, false if disabled
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*/
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get isCallWaiting(): boolean;
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get speakerVolume(): number;
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get microphoneInputLevel(): number;
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get getActiveCalls(): {
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@@ -165,12 +178,35 @@ declare class AudioModule {
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mergeCall(roomId: number): void;
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mergeCallByIds(firstCallId: string, secondCallId: string): void;
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setDND(value: boolean): void;
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+
/**
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* Sets the call waiting feature state.
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*
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* When call waiting is disabled (false) and there are already active calls,
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* any new incoming calls will be automatically rejected with a "busy" status.
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*
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* When call waiting is enabled (true), incoming calls will be allowed even when
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* other calls are active.
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*
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* This setting is used in the shouldTerminateNewSession method to determine whether
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* to automatically terminate new incoming sessions when the user is already on a call.
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*
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* @param {boolean} value - True to enable call waiting, false to disable
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*/
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setCallWaiting(value: boolean): void;
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private startCallTimer;
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setActiveRoom(roomId: number | undefined): Promise<void>;
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private getNewRoomId;
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private setupCall;
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private removeCall;
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private activeCallListRemove;
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+
/**
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* Determines whether a new incoming session should be automatically terminated
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* based on Do Not Disturb (DND) settings and Call Waiting settings.
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*
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* @param {RTCSessionEvent} event - The event containing the new RTC session
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* @returns {boolean} True if the session should be terminated automatically, false otherwise
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*/
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private shouldTerminateNewSession;
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private newRTCSessionCallback;
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setMuteWhenJoin(value: boolean): void;
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setMicrophoneSensitivity(value: number): void;
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@@ -285,6 +321,8 @@ declare type changeCallTimeListener = (event: { [key: string]: ITimeData }) => v
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declare type changeCallVolumeListener = (event: ChangeVolumeEventType) => void
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+
declare type changeIsCallWaitingListener = (value: boolean) => void
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+
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declare type changeIsDNDListener = (value: boolean) => void
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declare type changeIsMutedListener = (value: boolean) => void
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@@ -713,6 +751,7 @@ declare interface OpenSIPSEventMap extends UAEventMap {
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changeAvailableDeviceList: changeAvailableDeviceListListener
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changeMuteWhenJoin: changeMuteWhenJoinListener
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changeIsDND: changeIsDNDListener
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+
changeIsCallWaiting: changeIsCallWaitingListener
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changeIsMuted: changeIsMutedListener
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changeActiveStream: changeActiveStreamListener
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addRoom: addRoomListener
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