General telephony parameters

Parameter

Description

Default

Allow  Recording of External Transfers

Determines whether a call can be recorded  when an agent transfers it to an external number.

If you do not select this option, a call is not recorded if it is transferred to an external party and only external parties are involved. If you do select this option, the call recording continues after the transfer occurs.

Not selected

Append SIP Call-ID to CallLog

Determines whether the call log includes the SIP call ID.

Auto Disconnect Last Party

Determines whether the CIC system automatically disconnects the last party in a conference.

Selected

Block All External Transfers

Blocks all consult and blind transfers to external parties and remote numbers. This option works with the Disconnect Conferences with Only External Parties option to help prevent toll fraud.

Not selected

Board Event Window Limit

Specifies the maximum number of allowable digital board events.

Excessive board events can generate in-sync/out-of-sync errors from a bad board or a bad digital line. This option prevents bad equipment (such as T-1, ISDN, and other digital lines) from causing excessive errors and slowing down CIC.

Note: If board events are disabled on a board, the board remains registered as active.
However, events are disabled for the amount of time specified for the Event Recovery Time option.

If more than this number of digital events is counted during the Board Event Window Time, then digital events are disabled on the board.

The minimum number of events is 5.

20

Board Event Window Time (ms)

Specifies the number of consecutive milliseconds during which digital board events are counted. The minimum number is 10000 ms (10 seconds).

Note: If 20 events occur within 60 seconds, CIC disables the board that generated the events.

See also Board Event Window Limit.

60000 ms
(60 seconds)

Call Analysis Diagnostic Record

Enables diagnostic call recordings for call analysis.

Diagnostic records are 8k, mono, 8bit unsigned PCM.

Note: If Media Servers are not used for advanced operations, diagnostic recordings are
placed in the default Recordings directory
and have the following naming convention: DR_<callid>.pcm.

If Media Servers are used for advanced operations, diagnostic recordings  are stored on
each Media Server with the following naming convention:

%inin_trace_root%\<date>\diagnostics\<lastTwoDigitsOfCallId>\<callId>_<16RandomCharacters>_ca.wav

Diagnostic recordings stored on Media Servers are automatically deleted after seven days.

Not selected

Keyword Spotting Diagnostic Recording

Enables diagnostic recordings for keyword spotting.

Interaction Analyzer uses diagnostic recordings to help determine accuracy in keyword spotting. For more information, see Interaction Analyzer Technical Reference in the PureConnect Documentation Library on the CIC server.

Not selected

Fetch Diagnostic Recordings from Media Server

Retrieves all diagnostic recordings from the Media Server and from the [drive]:[IC installation folder]/logs/diagnostics.

On

Confirm  Station Connection

Sends an eCallEvent_StationConnectionConfirmation event. This event can be intercepted and handled  by the ConfirmStationConnection initiator and the StationConnectionConfirmation  tool step. This allows the user to intercept a remote client connection call and change the behavior via a handler.

Not selected

Diagnostic  Recording Extension Time (sec)

Sets the time (in seconds) that a diagnostic recording will be extended beyond the completion of the call analysis operation. Support Services may increase this period to troubleshoot call analysis result issues in order to thoroughly analyze the audio. The acceptable values are 3 through 30.

3 seconds

Disconnect Conferences with Only External Parties

Disconnects conferences in which the only members are external parties and remote numbers.

If all internal parties leave a conference, and only external parties remain on the call, the conference is disconnected. This option works with the Block all External Transfers parameter to help prevent toll fraud.

Not selected

Enable Call Analysis for Conference Add Party

Enables the ability to add a party to a conference immediately or to perform call analysis before adding the party to the conference.

Not selected

Enable Secure Input feature

Activates the Secure Input feature, which separates and encrypts data to protect it from theft or misuse.

When this parameter is elected, you can create and configure secure input forms in the Secure Input Forms container. You can then assign individual secure input forms to different workgroups in the Workgroup container.

Not selected

Extension Dialing Analysis Type

Used with extension dialing. When CIC uses extension dialing to place a call, and the number dialed includes  a /xxx where xxx is an IVR entry or extension number, CIC attempts to detect if the dialed number is answered by a human voice or an answering machine or an IVR script.

The acceptable values are:

• Voice: CIC waits for the dialed number to answer and then automatically sends the DTMF tones for the extension. This has the advantage that you can always be certain that the DTMF will be sent. The disadvantage is that when a human answers the call, the DTMF will play in that person's ear.

• Answering Machine: CIC attempts to recognize an IVR or automated voice before sending the DTMF tones. If it does not detect an automated voice, it assumes a human voice answered and does not send the DTMF extension digits. This kind of detection is less reliable than voice detection.

Voice

Global No Connection Timeout (min)

Sets a timer on all calls made on the CIC system. The CIC system disconnects a call if the call does not enter a connected state before the expiration of this timer.

30 minutes

Held Call Timeout (seconds)

Specifies the number of seconds a call can remain on hold. When the timer elapses, the call flow proceeds to a menu to present the caller with options. This menu is generated by the System_HeldInteractionTimer handler. You can modify the call flow behavior in the CustomHeldInteractionTimer handler.

The default value is 900 seconds (15 minutes).The minimum value is 2 seconds.

Note: If you change this parameter, you do not need to restart CIC in order for it to take effect.

900 seconds

Honor No Answer Timeout

Sets CIC to honor the no answer value presented to it by the caller, which is usually the client or handler tool  step.

Not selected.

Inband Dial Delay (ms)

Sets the time in milliseconds that the system waits before dialing in-band digits to the external party of a call. In-band dial digits are most commonly used for extension dialing and account code dialing.

500 milliseconds

Intercom Calls ASR Enabled

Allows a user to toggle permissions for allowing ASR enabled intercom calls on the system, as well as set a threshold for concurrent intercom ASR enabled sessions.

 The acceptable values are:

• On

• Off

• 0

On

Line Event Window Limit

Disables line events if a digital line starts generating abnormal numbers of events and degrades CIC performance.

Such events can generate in-sync/out-of-sync errors from a bad board or bad digital line. This safety mechanism prevents bad equipment (such as T-1, ISDN, and other digital lines) from causing excessive errors and dragging down CIC. The event and shutdown threshold is configurable with three parameters: Line Event Window Limit, Line Event Window Time, and Event Recovery Time.

Enter the maximum number of digital line events to detect within the Line Event Window Time window before disabling digital events on the excessive board. If a digital line exceeds this limit, CIC disables events from that line for the specified Event Recovery Time.

The minimum event limit is 5. The board remains registered as active, but events are disabled for the specified time.

Note: Not all lines implement this option.

20 events

Line Event Window Time

Specifies the number of consecutive milliseconds during which digital line events are counted. The minimum window is 10000 ms (10 seconds).

Notes: If 20 events occur within 60 seconds, CIC disables the line that generated the events.

Not all lines implement this options

60000 ms (60 seconds)

Maximum ASR Session Limit

Sets a maximum value of resources that are expected to be used concurrently at any given time in the  system for ASR sessions.

Specify any integer.

Note: CIC derives a proper value if you do not provide one.

0

Maximum Number of Calls per Station

Sets the maximum number of outgoing calls originating from a single station that are allowed to exist.

When an outgoing call is made, TsServer checks a map of Call IDs to Station IDs and, if the station has the specified number of outgoing calls still active, the new call attempt will fail. If not, an entry should be placed in the map. When the call disconnects, the entry is removed from the map.

5

Minimum ASR Session Limit

Sets a minimum value of resources that are expected to be available to the system for any ASR session  (for a minimum level of resources that should not be fixed to any line, and therefore available to all sessions).

The acceptable values are:

• 0: Indicates that no minimum level is required

• Any integer

0

Optimize Audio for Conferences

This option applies only to conferences. If you select this option, CIC automatically identifies the dominant speaker and mutes all of the other calls on the conference.

Conference calls hosted by Interaction Media Server include features such as dominant speaker detection with echo cancellation (muting errant noises from other callers), automatic level control (volume), support for Interactive Voice Response (IVR) input, and other optimizations.

Note: This parameter globally controls dominant speaker detection with echo cancellation for conference calls.

If you experience audio issues when this parameter is enabled, you can enter the ConferenceTypeDominantSpeakerDiagnosticRecording property through the Media Servers container in Interaction Administrator and set the value to true.  This property creates diagnostic recordings that you can send to a PureConnect Customer Care representative for analysis.

Selected

Perform Call Analysis

Contact your authorized PureConnect Customer Care representative for information on this setting.

For more information, see the Interaction Media Server Technical Reference document in the PureConnect Documentation Library.

Not selected

Play Digits Tone Specification

Sets the number duration of the on or off time between DTMF tones.

By default Telephony Services uses DTMF tones of 100ms in duration with 50ms off time between tones at vendor-specific levels. In some circumstances it may be necessary to adjust the duration on or off time so that the receiving system can correctly detect different tones.

This parameter applies to all DTMF played using the Play Digits tool step. The acceptable values are X:Y,  where X=on time (ms) and Y=off time (ms).

90:60

Recording  Silence Time

Sets the duration (milliseconds) of silence that is required to disconnect a call that is in voice mail.

This parameter is valid only for calls that are on the system, such as station-to-station calls, as opposed to physical external lines. External lines have the same configuration option, which is available on that line’s configuration page in Interaction Administrator.

0

Remote Station Timeout Override

This parameter should be modified only with the direction of a PureConnect Customer Care representative.

Any modifications to the value can have a significant impact on the accuracy of call progress analysis, and those changes will be noted in the appropriate log files. Modifying this parameter without explicit instruction from a PureConnect Customer Care representative could result in a billable support incident to restore functionality.

Not specified

Ringback Silence Detection Time

Dialogic and Aculab: Adjusts the maximum period of silence allowed before call analysis will detect no ringback. If unspecified, TS defaults to 11 seconds for connection calls (calls to a remote station), and 20 seconds for all other outbound calls. The minimum value allowed is 8 seconds.

Note: This setting has no effect if Media Servers are used for advanced operations.

Not specified

Ringback  File

Specifies the .wav file that is played to the caller during alerting to the destination user or station. .wav files are located in the  resource directory: Ringback.wav?playlocation=mediaserver

Caution: If you replace the RingBack.wav file in the Resources directory with your own customized file,  then this file will be overwritten when you update to a newer release. If you have replaced this file in the Resources directory, back up your customized file before you begin the update process. Then restore the file after the upgrade is complete. This applies to the files on IC servers and on Media Servers.

RingBack.wave

Telephony  Supported IP Versions

This parameter is available if the appropriate server parameter is selected.

Specifies the type of protocol address family that is supported by the CIC server.

For more information, see SIP Line Transport and SIP Station Transport.

Note: If this parameter is set to IPv4, then all lines and stations that specify the Address Family as IPv6, or the Media Address Family as IPv6, are unusable and shown as "Unusable" in the Status column.

If this parameter is set to IPv6, then all lines and stations that specify the Address Family as IPv4, or the Media Address Family as IPv4, are unusable and shown as "Unusable" in the Status column.

For more information, contact your authorized PureConnect Customer Care representative.

IPv4 and IPv6

Ts Consult  Transfer To Intercom Move All Queues Active

Specifies that when a call is consult transferred to an intercom call, the transferring call is placed on ALL queues of the replaced intercom call.

Selected

Waiting Call Indication

Enables call waiting. If this parameter is not selected, call waiting is globally disabled for all alert types.

Selected

Use Voice  Buffer Input

Specifies whether the voice buffer input is activated.

If you enable this parameter, it decreases the usage of voice and conference resources. If you do not enable this parameter, there is no timeout.

Selected

Timeout  (seconds)

Sets the number of seconds to wait before freeing up the resource after a play switches to the VoiceBufferInput role.

Acceptable values are 5-20000 seconds.

5

Warm Down Time (ms)

Specifies the duration in milliseconds that the audio remains in Interaction Center after a user plays a digit(s).

15 milliseconds

Zone Page Delay

Sets the maximum time in milliseconds that the CIC server should wait for the stations being paged to respond to a page request. Stations that take longer than this timeout to respond are be included in the page. If all stations respond before this timeout is reached, the page begins to load immediately.

1500 milliseconds