> ## Documentation Index > Fetch the complete documentation index at: https://docs.voxeme.com/llms.txt > Use this file to discover all available pages before exploring further. # Update Call - Live Call Control > This api allows you to manipulate a call that is in progress -- i.e., "Live Call Control". The actions available are: - hangup a call - mute/unmute a party on a call - pause/resume the real-time audio stream provided via the [listen](https://www.jambonz.org/docs/webhooks/listen/) command - transfer one or both parties on a call to new application(s). Note that you can provide a different application for each party (e.g. queue the called party while the calling party dials a new number) - play a whisper prompt to a party. - place or remove a party that is in conference on hold. - mute/unmute all non-moderators in a conference. Generally, a single API call can only do one of the above operations. The JSON payload may contain the properties described in the table below. | property | description | when can this be used | ------------- |-------------| ---------------------| | call_hook | a new application to start executing on the call for the party associated with the call_sid | at any time a call is active | | child_call_hook | a new application to start executing on the call for the party associated with the child_call_sid | at any time a call is active and in a [dial](https://www.jambonz.org/docs/webhooks/dial/) verb| | call_status | Change the status of the call. Possible values are 'completed' or 'no-answer' (the former terminates an answered call, the latter a call that is ringing) | at any time a call is in-progress or ringing | | conf_hold_status |Possible values are 'hold' or 'unhold'. When playing a party on hold, a wait_hook can optionally be provided to play music or a message to the party while on hold.| Only valid when the party is in a conference. | | conf_mute_status |Possible values are 'mute' or 'unmute'. Note that this action applies to all non-morderators in the conference, not to a single member| Only valid when the party is in a conference. | | listen_status | Change the status of a listen stream. Possible values are 'pause' or 'resume'. Pausing a stream maintains the websocket connection but will discontinue sending audio over the connection. Resuming will start sending audio again. This may be useful, for example, when a caller is providing confidential information that you do not want to appear in a recording. | only when a listen command is active on the call (may be nested in an active dial command) | | mute_status | Mute or unmute a call that is currently in a Dial verb. Possible values are 'mute' or 'unmute'. Either party in a call may be modified, depending on the call_sid provided in the path of the request-uri. | only when a dial command is currently active on a call | | whisper | Play a mid-call whisper prompt to one of the parties on a call in progress. The whisper prompt is provided in a play or say verb, as shown in the examples below. The whisper may be played to either party on the call, depending on the call_sid provided in the path of the request-uri. The other party is briefly placed on hold while the prompt is played, and then reconnected to the other party afterwards.| only when a dial command is currently active on the call | | wait_hook | A relative or absolute url that can be supplied when placing a party that is in conference on hold.| Only when putting a party on conference hold | The call_status, listen_status, and mute_status properties are mutually exclusive -- only one may be provided in the same request. The whisper property may be provided alone, or together with a listen_status or mute_status property, in which case the whisper prompt is played after the listen_status or mute_status operation is complete. Finally, if call_hook is provided, then call_status_hook may also optionally be included. This is used to specify a new callback to receive call status events. Success is indicated by a 204 response ## OpenAPI ````yaml put /v1/Accounts/{account_sid}/Calls/{call_sid} openapi: 3.0.3 info: title: Voxeme API description: >- The Voxeme account-level REST API allows a user to create and manage calls, applications, speech credentials, carriers, and other resources that are associated with their account. ## Authentication The API uses Bearer authentication; i.e., a bearer token that must be provided in the Authorization header, e.g. `Authorization: Bearer 'your-api-token'` You can generate an api token for your account in the hosted portal. ## HTTP response codes - POST requests will return a 201 status on success and a JSON payload that includes a `sid` property referencing the unique identifier of the resource that has been created. - UPDATE and DELETE requests will return a 204 status on success with no body - GET requests will return a 200 response with a JSON payload ## API version The current version of the api is v1. version: 1.0.0 contact: {} servers: - url: '' security: [] tags: - name: v1 description: The current API version is v1. - name: Alerts description: >- Alerts are generated when some element of your configuration is not working properly; for instance, if you have configured a webhook that is unreachable, or you try to use a speech service without having configured speech credentials. - name: Applications description: >- Applications represent the call handling rules to be applied to voice calls either made or received by the platform. Applications can be created, queried, updated, and destroyed via the API. Applications have the following properties: |property|description| |--------|-----------| |account_sid|identifies the Account this application belongs to| |application_sid|unique identifier for the application| |name|user-specified name of application| |call_hook|web callback to invoke for new calls| |call_status_hook|web callback to invoke for call status notifications| |speech_recognizer_vendor|vendor to use for speech recognition. Default: google| |speech_recognizer_language|default language to use for speech recognition, if not specified in the 'gather' or 'transcribe' verbs. Default: en-US| |speech_synthesis_vendor|vendor to use for text to speech. Default: google| |speech_synthesis_voice|default voice to use for text to speech, if not specified in the 'say' verb. Default: en-US-Wavenet-C| - name: Calls description: >- Calls that are active or recently ended (within 10 minutes) can be retrieved and managed via the api. > Note: this api should not be used to retrieve call detail records. For that, please use the /RecentCalls api - name: RecentCalls description: >- RecentCalls represent call detail records (CDRs) that are kept in the database. Currently, 30 days of CDRs are retained in the time-series database. - name: SpeechCredentials description: >- SpeechCredentials represent AWS or Google tokens that are associated with your account for use in speech synthesis and recognition operations. > Note: you are currently restricted to a max of two SpeechCredentials - one for AWS and one for Google - name: VoipCarriers description: >- VoipCarriers represent SIP trunking providers that are associated with your account for use in placing and receiving PSTN calls. - name: SipGateways description: >- A SipGateway represents a single IP address or range of IPs that are associated with a VoipCarrier (e.g. the VoipCarrier places and receives calls from these IPs). When you create a VoipProvider, you need to also add the SipGateways for that provider in order for incoming calls to be properly classified and for outbound calls to be properly routed. paths: /v1/Accounts/{account_sid}/Calls/{call_sid}: parameters: - name: account_sid in: path required: true schema: type: string example: 5f8e8f60-4771-44cb-92a6-94ea66df0450 - name: call_sid in: path required: true schema: type: string example: a4941f23-66b8-4b52-b495-94192adee7ec put: tags: - v1 - Calls summary: Update Call - Live Call Control description: >- This api allows you to manipulate a call that is in progress -- i.e., "Live Call Control". The actions available are: - hangup a call - mute/unmute a party on a call - pause/resume the real-time audio stream provided via the [listen](https://www.jambonz.org/docs/webhooks/listen/) command - transfer one or both parties on a call to new application(s). Note that you can provide a different application for each party (e.g. queue the called party while the calling party dials a new number) - play a whisper prompt to a party. - place or remove a party that is in conference on hold. - mute/unmute all non-moderators in a conference. Generally, a single API call can only do one of the above operations. The JSON payload may contain the properties described in the table below. | property | description | when can this be used | ------------- |-------------| ---------------------| | call_hook | a new application to start executing on the call for the party associated with the call_sid | at any time a call is active | | child_call_hook | a new application to start executing on the call for the party associated with the child_call_sid | at any time a call is active and in a [dial](https://www.jambonz.org/docs/webhooks/dial/) verb| | call_status | Change the status of the call. Possible values are 'completed' or 'no-answer' (the former terminates an answered call, the latter a call that is ringing) | at any time a call is in-progress or ringing | | conf_hold_status |Possible values are 'hold' or 'unhold'. When playing a party on hold, a wait_hook can optionally be provided to play music or a message to the party while on hold.| Only valid when the party is in a conference. | | conf_mute_status |Possible values are 'mute' or 'unmute'. Note that this action applies to all non-morderators in the conference, not to a single member| Only valid when the party is in a conference. | | listen_status | Change the status of a listen stream. Possible values are 'pause' or 'resume'. Pausing a stream maintains the websocket connection but will discontinue sending audio over the connection. Resuming will start sending audio again. This may be useful, for example, when a caller is providing confidential information that you do not want to appear in a recording. | only when a listen command is active on the call (may be nested in an active dial command) | | mute_status | Mute or unmute a call that is currently in a Dial verb. Possible values are 'mute' or 'unmute'. Either party in a call may be modified, depending on the call_sid provided in the path of the request-uri. | only when a dial command is currently active on a call | | whisper | Play a mid-call whisper prompt to one of the parties on a call in progress. The whisper prompt is provided in a play or say verb, as shown in the examples below. The whisper may be played to either party on the call, depending on the call_sid provided in the path of the request-uri. The other party is briefly placed on hold while the prompt is played, and then reconnected to the other party afterwards.| only when a dial command is currently active on the call | | wait_hook | A relative or absolute url that can be supplied when placing a party that is in conference on hold.| Only when putting a party on conference hold | The call_status, listen_status, and mute_status properties are mutually exclusive -- only one may be provided in the same request. The whisper property may be provided alone, or together with a listen_status or mute_status property, in which case the whisper prompt is played after the listen_status or mute_status operation is complete. Finally, if call_hook is provided, then call_status_hook may also optionally be included. This is used to specify a new callback to receive call status events. Success is indicated by a 204 response operationId: updateCallLiveCallControl requestBody: content: application/json: schema: type: object properties: record: type: object properties: action: type: string example: stopCallRecording recordingID: type: string example: my-recording-id siprecServerURL: type: string example: sip:srs@52.52.132.163 example: record: action: stopCallRecording recordingID: my-recording-id siprecServerURL: sip:srs@52.52.132.163 responses: '202': description: Transfer party to new application headers: Access-Control-Allow-Origin: schema: type: string example: '*' Connection: schema: type: string example: keep-alive Content-Length: schema: type: string example: '8' Date: schema: type: string example: Sun, 30 May 2021 12:38:54 GMT Server: schema: type: string example: nginx/1.14.2 etag: schema: type: string example: W/"8-YaBXLEiT7zQxEyDYTILfiL6oPhE" x-powered-by: schema: type: string example: Express content: text/plain: examples: Transfer party to new application: value: Accepted ````