Settings for the BYOC Genesys Cloud CX data source

The Genesys Cloud CX API Adapter (formerly Genesys PureCloud) connects to a region-specific Genesys Cloud CX API to receive call events and data.

Field	Description
Name	Data source name. Only alphanumeric characters (no quotation marks and no back or forward slashes) are supported in the data source name (required).
Organization	Specify the organization to which you want to associated the data source. In a multi-tenant enabled environment, if the selected organization is associated to a tenant, the data source is also associated to that same tenant. In this scenario, when you click Save to save the data source, the tenant name of the organization that is associated to the data source is displayed in the screen heading. The data source can be associated to a particular tenant or have the Shared status. A data source associated to a particular tenant processes data only for that tenant. A data source that has the Shared status processes data for all tenants in the system.
Description	Description of the Data Source.
Time Zone	Select the correct time zone. Use the same time zone in Enterprise Manager and Recorder Manager Preferences.
Minimum Session Length	To prevent the retention of calls that are so short they are unlikely to contain usable data, specify a Minimum Session Length (seconds)—calls that are shorter than this will be deleted automatically. If set to 0 (zero), this feature is disabled and no calls will be deleted based on this setting. The maximum value is 3600 (or one hour). This setting applies to active CTI Sessions or VOX Sessions. Inactive CTI Sessions can be retained using the Session Auditing Policy.
Long Call Duration (minutes)	Duration of a call, in minutes. This number indicates the maximum duration of a call that the Integration Service tracks. All calls over this period are ended.
Contact Policy Type	This setting allows you to set the call stitching method. Follow the call—When follow the call is enabled, there will be one contact that includes all audio from the beginning of the call to the end. This is the default. Backoffice - Contact per call—Enables the “Back office” style of stitching, which creates sessions/interactions based on CTI calls. If one employee is on two calls at the same time (for example, a customer call and a consultation call), the system creates two sessions/interactions. This is used more often in trading environments rather than contact center environments.
Session Auditing Policy	Defines the type of session/interaction that should be marked and kept in the system. “Disabled” (the default) only marks sessions/interactions with some kind of content. Two additional options will create a basic entry in the database for calls that occurred but were not recorded: “Missed Recordings” will mark calls that should have been recorded, but were not, while “Full Switch” will mark all sessions/interactions for which we receive CTI without recording (for example, calls that were blocked, or interception calls that were met with a busy tone or unanswered ringtone). You may then search for these types of interactions in the Portal. Recorded employee segments marked via auditing will appear in playback once all Interactions in the related Contact are closed, after a delay of up to five minutes.
Recorder Allocation based on Audio Location	Use this setting in a SIP-based VoIP delivery environment to control how Recorders are selected to record calls, and ensure that each call is recorded by a Recorder local to the media gateway at which the call arrives. This setting is designed primarily for environments where recording occurs at multiple sites and each site has its own gateway. Note that this setting must be used on conjunction with either a Gateway Side Correlation Pool member group or an IP Extension Pool member group configured for the data source. Within these member groups, you must specify the IP addresses or host names of the phones associated with the member group in the IP Network Region configuration. The Recorder Allocation Based on Audio Location setting instructs the system to examine the IP address in a SIP message and compare this address to the addresses listed in the IP Network Region settings of the Gateway Side Correlation Pool or IP Extension Pool member group associated with the data source. If an address in the SIP message matches one of those specified in the member group IP Network Region settings, the system will route the call to a Recorder associated with that member group. The options for this setting are: Inactive—The system attempts to record the call, but does not use an address found in a SIP message to route the call to a particular Recorder. This is the default setting. From Signaling—The system examines the SIP header section of the SIP Invite. The system compares the address found in a SIP header field (such as From, Contact, Via, or Socket) to the addresses configured in the member group IP Network Region settings. If there is a match, the system routes the call to a Recorder associated with that member group. If the system does not find a match, the Recording Resource Allocation Behavior setting will determine how the call is recorded. From Media—The system examines the SDP message attached to a SIP Invite. The system compares the IP addresses found in the SDP message to those configured in the member group IP Network Region settings. If there is a match, it routes the call to a Recorder associated with that member group. If the system does not find a match, the Recording Resource Allocation Behavior setting determines how the call is recorded. This option does not work if SIP operates in Delayed-Offer mode. Note that this setting works in combination with the Recording Resource Allocation Behavior setting to determine recording behavior. If the Recording Resource Allocation Behavior setting specifies: IgnoreLine—The system acts as if the Inactive setting is selected for this setting. If either the From Signaling or From Media is selected, those settings are ignored. In this case, any Recorder associated with the data source can record the call. LineFirst—The system makes two attempts to record the call using the address obtained from the SIP message. If the system cannot successfully route the call to a Recorder using the SIP message address, any Recorder associated with the data source can record the call. LineExclusive—If the system cannot successfully route the call to a Recorder using the address from the SIP message, the call is not recorded. In this case, the system raises an alarm.
Default Employee
In the Default Employee section, specify the employee to be associated to a recording that has no employee associated to it. There are situations where recordings will not have employees associated to them. Examples include: IVR recordings where there is no employee or phone device Back office environments where phones are shared and not associated to a specific employee Assigning a Default Employee for Interactions to a data source provides a way to provide replay access to these recordings. When a Default Employee for Interactions is configured, any recordings made for the data source that have no explicit user associated are associated to the Default Employee for Interactions. First, select the Organization to which the Default Employee for Interactions belongs, and then select one employee as the Default Employee for Interactions. You cannot select an employee who has a configured end date as the Default Employee for Interactions. Also, you should not select an employee who will soon move or transfer to a different organization as the Default Employee for Interactions. Once an employee is selected as the Default Employee for Interactions, an error message displays on the data source screen if the employee has been deleted, terminated, or changed to a different organization since the last time the data source was saved.
Device IP Configuration	Select one of the following Device IP Configuration options and enter the corresponding IP Address: PBX Side - Near End—The PBX Side - Near End is the server used to send/receive the control messages to/from the extensions. If you have the same extensions in multiple data sources, you must specify an IP Address or Host Name for the signaling interface server. Otherwise, the setting is not mandatory, but recommended. The Recorder uses the source or destination IP addresses or host names of the signaling messages to identify the particular data source with which a call is associated. The recorder does not use any data inside the signaling itself to make this determination. For this reason, the source/destination of the IP packets presented to the Recorder must be different for each duplicate extension. Make sure that your network is set up in such a way as to allow for this. Example A proxy server in front of separate PBXs using duplicate extensions can cause IP packets to appear as though they are using the same IP address (even though the SIP signaling would indicate otherwise). This will prevent the duplicate extensions from being recorded correctly. PSTN Side - Far End—If you are using SIP Trunk Recording, set PSTN Side - Far End as the Server Type. This setting is only available for Avaya and Generic switch types, and if you have created a Gateway Side Correlation Pool Member Group, this setting is mandatory. For details on this, see the “SIP Trunk Recording” section in the Recorder Configuration and Administration Guide, or the related online help.
Time Zone Settings	Under TimeZone Settings, select a Local Time Tagging Mode from the drop-down list. Select DataSource to base tagging on the Time Zone specified in the data source. Select Organization to base tagging on that of the organization. This is useful in scenarios where employees are working in different regions, allowing you to unify tagging across multiple time zones. Please note that if you select Organization, it is still necessary to specify the correct time zone for the data source, above, as this is used in fallback selection for monitored extensions without an associated Employee or Profile (that is, extensions that don't belong to an employee). If you select Organization, it is still necessary to specify the correct time zone for the data source, above, as this is used in fallback selection for monitored extensions without an associated Employee or Profile (that is, extensions that don't belong to an Organization).
AudioHook Settings	These settings are only used with a Genesys Cloud CX with AudioHook integration. Organization ID - This field is required. Specify the Genesys cloud Organization ID that is used by the authentication process. The maximum number of characters allowed is 255. API Key - This field is required. Specify the API Key that is configured as part of the Genesys AudioHook configuration. This key is used by the authentication process. This is the same API Key you entered when creating an AudioHook integration in the Genesys portal. The maximum number of characters allowed is 255. Client Secret - This field is optional. Specify the Client Secret (or password) that is configured as part of the Genesys AudioHook configuration. The client secret is used as part of the authentication process to generate a request signature that can be validated on each request. The maximum number of characters allowed is 255.
Use Pause/Resume events from Genesys Client	Configure Verint's response to pause and resume events from Genesys Cloud CX: Under Genesys Platform API Settings, select an option from the Use Pause/Resume Events from Genesys Client list: No: Ignore pause and resume events. Yes for Audio: Pause and resume audio only. Yes for Audio and Screen: Pause and resume audio and agent screen.
Associated Integration Service Installations	Associated Integration Service Installations expands allowing you to select the server that is providing Integration Services for the recorder you are configuring this data source for.
Advanced Settings	Use the Key and Value fields to enter any proprietary pairs that are in use in your system. Only use Advanced Settings in consultation with Verint Field Engineers.