SIP session settings

Setting

Description

Default  

Use Global SIP Station Session Settings

This option specifies whether stations inherit the values that are defined for the global SIP station.

This option is available for station configurations only.

Use SIP Session Timer and SIP Session Timeout

This setting indicates whether an OPTIONS messages is sent to the remote device when a SIP session times out. By default, a timeout occurs after 60 seconds. If the remote device does not respond to the OPTIONS message, the call is disconnected.

Selected

SIP Register Interval

This setting specifies the amount of time in days, hours, minutes or seconds.

1 day

Disconnect on Broken RTP

This setting determines if a VoIP call remains active after audio has been disrupted. Audio is considered disrupted if no RTP, RTCP and no comfort noise packet is received from the remote device. By default, this parameter is turned on (checked).

Selected

Media Timing

This setting specifies the timing on an INVITE request that contains a new media description in the SIP message body in the existing signaling session.

The available options are:

• Normal

• Delayed

Delayed

Media reINVITE Timing

This setting indicates the type of timing on a re-INVITE request that contains a new media description in the SIP message body in the existing signaling session.

The available options are:

• Normal

• Delayed

Delayed

Terminate Analysis on Connect

This setting indicates whether the call analysis procedure terminates when a SIP connection indication from the network is received.  

Example: CIC makes its PSTN call via SIP calls through a SIP/ISDN gateway. This particular SIP/ISDN gateway only sends a SIP connect message back to Interaction Center after the remote party answers the call. If call analysis is used, select this setting so that call analysis terminates when the SIP connect message is received.

Example: CIC makes its PSTN call via SIP calls through a SIP/analog gateway. This particular SIP/Analog gateway always sends a SIP connect message back to CIC prematurely, before the remote party answers the call. If call analysis is used, deselect this setting so that call analysis continues after the SIP connect message is received.

Tip: If the connection is to a station, the configuration of this option for the station determines the call analysis behavior.

Selected

Disable Media Server Passthru

This setting stops the media server from rewriting the SSRC header.

Not selected

Station Connections are Persistent

This setting determines whether a persistent voice connection to the CIC server is maintained. If so, then the audio path does not disconnect until the station initiates the disconnection.

If this setting is not selected, when CIC determines that the audio path to the station is no longer needed, CIC initiates the disconnection.

Recommended settings:

• Operators—Selected: If you want to handle more calls than the phone is capable of handling, select this setting. For example an operator wants to handle up to 20 simultaneous calls.

• Call Center Agents—Selected: If call center agents use an IP phone with a headset and also uses the CIC clients, select this setting.

Not selected

Connection Call Warm Down Time

This value represents the number of seconds a connection call should remain connected after the regular call is disconnected. Once this timeout is expired, the connection call will be disconnected.

Note: This option is not used for persistent connection calls.

5 seconds

Note: Decreasing the default value (5 seconds) can cause stability issues and is not recommended.

Call Appearances

Select the number of call appearances the phone can handle. CIC will send up to the configured number of calls to the phone. The default value for this option is 1.

Note: If Persistent is selected, the number of call appearances defaults to 1 and is grayed-out.  

Recommended settings:

• General: This value should be over 1 for experienced phone users only.

• Cisco: The Cisco IP phone 7960 can have up to 6 line appearances (each line appearance is equivalent to a station). Each line appearance has a unique SIP address. Line appearances are different than call appearances. Each line appearance handles 2 call appearances. Configure the phone to one line appearance and then this station configuration to 1 or 2 call appearances.

• Pingtel: Pingtel Expressa IP phone has one line appearance that handles 4 call appearances. Set station configuration to 1, 2, 3, or 4 call appearances.

This option does not apply to managed IP phones.

Enable AutoAnswer

If selected, this option sets the phone's "Enable Talk Event" attribute to yes. This attribute allows the phone to automatically receive phone calls.

If this option is not selected, the agent will not be able to pick up calls from the CIC clients or from the toast message.

Selected