Commit 4d54c46

<a name="6.4.0"></a>

- New parameters for fetching recordings (start time and end time)

- Audio Streaming feature

<version>6.4.0</version>

* `target/freeclimb-java-client-6.4.0.jar`

defaultClient.setAccountId("AC0123456789abcdefABCDEF0123456789abcdef00");

version = '6.4.0'

googleJavaFormat('1.24.0').aosp().reflowLongStrings()

version := "6.4.0",

The `AudioStream` command transfers control of the call to a gRPC session. Upon completion of the gRPC session, if the actionUrl is specified, control can be returned to percl usage or the call will simply be hung up if the actionUrl is not specified.

Name | Type | Description | Notes

------------ | ------------- | ------------- | -------------

**location** | **URI** | The gRPC server location that will receive the grpc stream as a uri and must be port 80 or 443. |

**actionUrl** | **URI** | A request is made to this URL when the gRPC session is concluded. The PerCL script returned in response to the actionUrl will be executed on the call. | [optional]

**contentType** | **String** | The type and sample rate of the audio being received over the channel must match the environmental sample rate. | [optional]

**metaData** | **List&lt;String&gt;** | An arbitrary array of strings passed through FC to the GRPC server can be used to pass state or other information about the call. | [optional]

**privacyMode** | **Boolean** | Enables audio redaction with full call recording while gRPC session is running and blocks logging of any DTMFs received by FreeClimb. | [optional]

A gRPC session has ended for this Call its actionUrl is being invoked. A PerCL response is expected — unless the URL is invoked due to the participant hanging up.

Name | Type | Description | Notes

------------ | ------------- | ------------- | -------------

**requestType** | **String** | Context or reason why this request is being made. Will be audioStream - gRPC session for this Call has ended and its actionUrl is being invoked. | [optional]

**callId** | **String** | Unique ID for this Call, generated by FreeClimb. This is the call leg which ended its gRPC session. | [optional]

**accountId** | **String** | Account ID associated with your account. | [optional]

**from** | **String** | Phone number of the party that initiated the Call (in E.164 format). | [optional]

**to** | **String** | Phone number provisioned to you and to which this Call is directed (in E.164 format). | [optional]

**callStatus** | **CallStatus** | | [optional]

**direction** | **CallDirection** | | [optional]

**conferenceId** | **String** | This is only populated if request pertains to a Conference. Otherwise, it is set to null. | [optional]

**queueId** | **String** | This is only populated if the request pertains to a Queue. Otherwise, it is set to null. | [optional]

**alias** | **String** | A nicely-formatted version of the phone number. | [optional]

**dateCreatedISO** | **OffsetDateTime** | The date that this resource was created in ISO 8601 format (e.g., 2022-01-01T00:00:00.000Z). | [optional]

**dateUpdatedISO** | **OffsetDateTime** | The date that this resource was last updated in ISO 8601 format (e.g., 2022-01-01T00:00:00.000Z). | [optional]

**startTimeISO** | **OffsetDateTime** | Start time of the Call in ISO 8601 format (e.g., 2022-01-01T00:00:00.000Z). Empty if the Call has not yet been dialed. | [optional]

**connectTimeISO** | **OffsetDateTime** | Time the Call was answered in ISO 8601 format (e.g., 2022-01-01T00:00:00.000Z). Empty if the Call has not yet been dialed. | [optional]

**endTimeISO** | **OffsetDateTime** | End time of the Call in ISO 8601 format (e.g., 2022-01-01T00:00:00.000Z). Empty if the Call did not complete successfully. | [optional]

**audioStreamDuration** | **Integer** | Length of time that the Call used the audio stream in seconds. This value is empty or zero when the Call did not use the audio stream. | [optional]

**callerName** | **String** | The caller ID name (CNAM) for this Call. Empty if unavailable. | [optional]

**webRTC** | **Boolean** | Indicates whether this Call was initiated via WebRTC. | [optional]

**subresourceUris** | [**CallResultAllOfSubresourceUris**](CallResultAllOfSubresourceUris.md) | | [optional]

The list of subresources for this Call. These include things like logs and recordings associated with the Call.

Name | Type | Description | Notes

------------ | ------------- | ------------- | -------------

**logs** | **String** | The URI for the logs associated with this Call. | [optional]

**recordings** | **String** | The URI for the recordings associated with this Call. | [optional]