Informing Agents of Supervisor Presence

SIP Server now reliably informs agents of a supervisor's in-call presence in multi-site and complex single-site scenarios, when the Providing Call Participant Info feature is enabled. This feature applies to all types of call supervision—single-site, multi-site, and remote supervision. SIP Server reports the supervisor information in the EventUserEvent messages distributed to the logged-in agents. This information is primarily used by T-Library clients, such as Workspace Desktop, to display parties participating in the call.

The supervisor-related information is reported in the Extensions attribute of the relevant event using the following key-value pairs:

• LCTSupervisor<n>—An integer that represents the supervisor of the call, where n is an integer value starting from 0.
• LCTSupervisor<n>_location—A name of the switch to which this supervisor belongs.
• LCTSupervisor<n>_monitoredDN—An integer that represents the agent monitored by this supervisor.
• LCTSupervisor<n>_mode—Supervision mode.

A supervisor can switch between supervision modes and whenever there is change in supervision mode, SIP Server reports the change in EventPrivateInfo.

Using the EventUserEvent and EventPrivateInfo messages, Workspace Desktop could improve the customer experience by providing the accurate status of call supervision scenarios.

Note: Supervision mode is distributed only in the first EventUserEvent message generated immediately after a supervisor answers the call.

Sample Scenario

The following sample scenario describes the enhanced LCTParty interface with the supervision information:

1. Internal DNs 1001 and 1002 are provisioned on Switch A.
2. DN 1002 subscribes to monitor DN 1001 (mute mode, call scope).
3. Inbound call from DN 21001 on Switch B is routed to DN 1001.
4. Call supervision started.

SIP Server generates EventUserEvent—immediately after a supervisor answers the call—for DNs 1001@A and 1002@A with the following information:

EventUserEvent
  AttributeExtensions
    'LCTParty0' '21001'
    'LCTParty0_location' 'B'
    'LCTParty1' '1001'
    'LCTParty1_location' 'A'
    'LCTPartiesLength' 2
    'LCTSupervisor0' '1002'
    'LCTSupervisor0_location' 'A'
    'LCTSupervisor0_mode' 'mute'
    'LCTSupervisor0_monitoredDN' '1001'
    'LCTSupervisorLength' 1
  AttributeConnID 1

Feature Configuration

In the TServer section of the SIP Server Application, set the following configuration options:

• sip-enable-call-info—Set this option to true.
• sip-enable-call-info-extended—Set this option to true.

sip-enable-call-info

Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately

If set to true, SIP Server does the following:

• Distributes the information about call participants except their locations and the supervisor-related information (see the sip-enable-call-info-extended option) to logged-in agents by using the SIP NOTIFY method and EventUserEvent messages.
• Distributes an EventPrivateInfo(4024) message, with the MonitorMode key in AttributeExtensions, to a supervisor and agent DNs indicating that monitoring mode was changed.

If set to false, SIP Server does not distribute an EventPrivateInfo(4024) message when the monitoring mode changes.

sip-enable-call-info-extended

Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately

This option applies only when sip-enable-call-info is enabled. When this option is set to true, SIP Server generates the supervisor information (LCTSupervisor<n> key-value pairs) and the location of call participants (LCTParty<n>_location) in EventUserEvent.

Feature Limitations

• When multi-site supervision is established with the call scope and if a monitored agent leaves the call, the requests submitted by the supervisor to switch between supervision modes will be rejected by SIP Server.

• When multi-site supervision is established with the agent scope and if consultation call supervision is started, the supervisor will not be aware of the consultation call even though the supervisor will be able to hear audio from the consultation call.

Upgrade Notes

This feature is available starting with SIP Server version 8.1.101.74. If you run SIP Server version 8.1.101.59 and later, Genesys recommends the following upgrading procedure:

1. Configure the Application-level option sip-enable-call-info-extended to false in all backup instances of SIP Servers.
2. Upgrade the backup SIP Servers.
3. Configure sip-enable-call-info-extended to true in backup SIP Servers.
4. Promote backup SIP Servers to primary.
5. Repeat steps 1 to 3 on the new backup (formerly primary) SIP Servers.