Default Outside Call Sequence

Default = Normal. See Ring Tones.

This setting is only used with analog extensions. It sets the ringing pattern used for incoming external calls. For details of the ring types see System Settings > System > Telephony > Ring Tones.

This setting can be overridden by a user's Call Management > Users > Add/Edit Users > Telephony > Call Settings > Outside Call Sequence setting. Note that changing the pattern may cause fax and modem device extensions to not recognize and answer calls.

Default Inside Call Sequence

Default = Ring Type 1. See Ring Tones.

This setting is only used with analog extensions. It sets the ringing pattern used for incoming internal calls. For details of the ring types see System Settings > System > Telephony > Ring Tones. This setting can be overridden by a user's Call Management > Users > Add/Edit Users > Telephony > Call Settings > Inside Call Sequence setting.

Default Ring Back Sequence

Default = Ring Type 2. See Ring Tones.

This setting is only used with analog extensions. It sets the ringing pattern used for ringback calls such as hold return, park return, voicemail ringback, and Ring Back when Free. For details of the ring types see System Settings > System > Telephony > Ring Tones.

This setting can be overridden by a user's Call Management > Users > Add/Edit Users > Telephony > Call Settings > Ringback Call Sequence setting.

Restrict Analog Extension Ringer Voltage

Default = Off.

Supported on IP500 V2 systems only. If selected:

• The ring voltage on analog extension ports on the system is limited to a maximum of 40V Peak-Peak.

• The message waiting indication (MWI) settings for analog extension are limited to Line Reversal A, Line Reversal B or None.

• Any analog extension already set to another MWI setting is forced to Line Reversal A.

Companding Law

These settings should not normally be changed from their defaults. They should only be used where 4400 Series phones (ULAW) are installed on systems which have A-Law digital trunks.

A-Law or U-Law> PCM (Pulse Code Modulation) is a method for encoding voice as data. In telephony, two methods of PCM encoding are widely used, A-Law and U-Law (also called Mu-Law or µ-Law). Typically U-Law is used in North America and a few other locations while A-Law is used elsewhere. As well as setting the correct PCM encoding for the region, the A-Law or U-Law setting of a system when it is first started affects a wide range of regional defaults relating to line settings and other values.

For IP500 V2 systems, the encoding default is set by the type of Feature Key installed when the system is first started. The cards are either specifically A-Law or U-Law.

Dial Delay Time (secs)

Default = 4 (USA/Japan) or 1 (ROW). Range = 1 to 30 seconds.

This setting sets the time the system waits following a dialed digit before it starts looking for a short code match. In situations where there are potential short codes matches but not exact match, it also sets the delay following the dialing of a digit before dialing complete is assumed.

Dial Delay Count

Default = 0 digits (USA/Japan) or 4 digits (ROW). Range = 0 to 30 digits.

This setting sets the number of digits dialed after which the system starts looking for a short code match regardless of the Dial Delay Time.

Default No Answer Time (secs)

Default = 15 seconds. Range = 6 to 99999 seconds.

This setting controls the amount of time before an alerting call is considered as unanswered. How the call is treated when this time expires depends on the call type.

• For calls to a user:

• • the call follows the user's Forward on No Answer settings if enabled. If not set, the call goes to voicemail if available or else continues to ring.

  • This timer is also used to control the duration of call forwarding if the forward destination does not answer.

  • It also controls the duration of ringback call alerting.

  • For a user, this setting is overridden by the user's Call Management > Users > Add/Edit Users > Telephony > Call Settings > No Answer Time setting if different.

• For calls to hunt groups:

  • This setting controls the time before the call is presented to the next available hunt group member.

  • This setting is overridden by the group's Call Management > Group > Add/Edit Group > Group > Fallback > Group No Answer Time setting if different.

If the system includes users who are using Avaya Workplace Client on iOS devices, it is recommended to set the time to at least 20 seconds. You should do this for either the system default, or for the individual users and any hunt groups to which they belong.

Hold Timeout (secs)

Default = US: 120 seconds/ROW: 15 seconds. Range = 0 (Off) to 99999 seconds.

This setting controls how long calls remain on hold before recalling to the user who held the call. The user's wrap-up time is also added.

Note that the recall only occurs if the user has no other connected call. Recalled calls will continue ringing and do not follow forwards or go to voicemail.

Park Timeout (secs)

Default = 300 seconds. Range 0 (Off) to 99999 seconds.

This setting controls how long calls remain parked before recalling to the user who parked the call.

Note that the recall only occurs if the user has no other connected call. Recalled calls will continue ringing and do not follow forwards or go to voicemail.

Ring Delay

Default = 5 seconds. Range = 0 to 98 seconds.

This setting is used when any of the user's programmed appearance buttons is set to Delayed ringing. Calls received on that button will initially only alert visually. Audible alerting will only occur after the ring delay has expired.

This setting can be overridden by a ring delay set for an individual user (Call Management > Users > Add/Edit Users > Telephony > Multi-line Options > Ring Delay).

Call Priority Promotion Time (secs)

Default = Disabled. Range = Disabled, 10 to 999 seconds.

When calls are queued for a hunt group, higher priority calls are placed ahead of lower priority calls, with calls of the same priority sort by time in queue. External calls are assigned a priority (1-Low, 2-Medium or 3-High) by the Incoming Call Route that routed the call. Internal calls are assigned a priority of 1-Low. This option can be used to increase the priority of a call each time it has remained queued for longer than this value. The calls priority is increased by 1 each time until it reaches 3-High.

In situations where calls are queued, high priority calls are placed before calls of a lower priority. This has a number of effects:

• Mixing calls of different priority is not recommended for destinations where Voicemail Pro is being used to provided queue ETA and queue position messages to callers since those values will no longer be accurate when a higher priority call is placed into the queue. Note also that Voicemail Pro will not allow a value already announced to an existing caller to increase.

• If the addition of a higher priority call causes the queue length to exceed the hunt group's Queue Length Limit, the limit is temporarily raised by 1. This means that calls already queued are not rerouted by the addition of a higher priority call into the queue.

Default Currency

Default = Locale specific.

This setting is used with ISDN Advice of Charge (AOC) services. Note that changing the currency clears all call costs stored by the system except those already logged through SMDR. The currency is displayed in the system SMDR output.

Default Name Priority

Default = Favor Trunk.

For SIP trunks, the caller name displayed on an extension can either be that supplied by the trunk or one obtained by checking for a number match in the extension user's personal directory and the system directory. This setting determines which method is used by default. For each SIP line, this setting can be overridden by the line's own Name Priority setting if required. Select one of the following options:

• Favor Trunk: Display the name provided by the trunk. For example, the trunk may be configured to provide the calling number or the name of the caller. The system should display the caller information as it is provided by the trunk. If the trunk does not provide a name, the system uses the Favor Directory method.

• Favor Directory: Search for a number match in the extension user's personal directory and then in the system directory. The first match is used and overrides the name provided by the SIP line. If no match is found, the name provided by the line, if any, is used.

Media Connection Preservation

Default = Enabled.

When enabled, attempts to maintain established calls despite brief network failures. Call handling features are no longer available when a call is in a preserved state. When enabled, Media Connection Preservation applies to SCN links and Avaya H.323 phones that support connection preservation.

Phone Failback

Default = Automatic.

Phones are permitted to failover to the secondary gatekeeper when the IP Office Line link to the primary gatekeeper is down.

When set to Automatic, if a phone’s primary gatekeeper has been up for more than 10 minutes, the system causes the phone to failback if the phone is not in use. If the phone is in use, the system will reattempt failback 10 seconds after the phone ceases to be in use.

When set to Manual, phones remain in failover until manually restarted or re-registered, after which the phone attempts to fail back.

DSS Status

Default = Off

This setting affects Avaya display phones with programmable buttons. It controls whether pressing a DSS key set to another user who has a call ringing will display details of the caller. When off, no caller information is displayed.

Auto Hold

Default = On (Off for the United States locale).

Used for users with multiple appearance buttons. When on, if a user presses another appearance button during a call, their current call is placed on hold. When off, if a users presses another appearance button during a call, their current call is disconnected.

Show Account Code

Default = On This setting controls the display and listing of system account codes.

• When on: When entering account codes through a phone, the account code digits are shown while being dialed.

• When off: When entering account codes through a phone, the account code digits are replaced by s characters on the display.

Inhibit Off-Switch Forward/Transfer

Default = On

When enabled, this setting stops any user from transferring or forwarding calls externally.

Restrict Network Interconnect

Default = Off.  

When this option is enabled, each trunk is provided with a Network Type option that can be configured as either Public or Private. The system will not allow calls on a public trunk to be connected to a private trunk and vice versa, returning number unobtainable indication instead.

Due to the nature of this feature, its use is not recommended on systems also using any of the following other system features:  multi-site networks, VPNremote, application telecommuter mode.

Include location specific information

Default = Off.

When set to On, this setting is available in the trunk configuration settings when Network Type is set to Private.

Set to On if the PBX on the other end of the trunk is toll compliant.

Drop External Only Impromptu Conference

Default = On.

If selected, when the last remaining internal user in a conference exits the conference, the conference is ended, regardless of whether it contains any external callers.

If not selected, the conference is automatically ended when the last internal party or trunk that supports reliable disconnect exits the conference. The Inhibit Off-Switch Forward/Transfer option above is no longer applied to conference calls.

Visually Differentiate External Call

Default = Off.

This setting is applied to the lamp flashing rate used for bridged appearance and call coverage appearance buttons on 1400, 1600 and 9600 Series phones and on their button modules. When selected, external calls alerting on those buttons will use a slow flash (200ms on/50ms off). If not selected or if the call is internal, normal flashing (500ms on/500ms off) is used.

Unsupervised Analog Trunk Disconnect Handling

Default = Off.

When using analog trunks, various methods are used for trunk supervision. That is to detect when the far end of the trunk has disconnected and so disconnect the local end of the call. Depending on the locale, the system uses Disconnect Clear signaling and or Busy Tone Detection. This setting should only be enabled if it is know that the analog trunks do not provide disconnect clear signaling or reliable busy tone. For Server Edition this field is only available on Expansion System (V2) systems.

When enabled:

• Disconnect Clear signaling detection is disabled. Busy tone detection remains on.

• Unsupervised transfers and trunk-to-trunk transfers of analog trunk calls are not allowed. The Allow Analog Trunk to Trunk Connect setting on analog trunks (Line | Analog Options) is disabled.

• If Voicemail Pro is being used for external call transfers, Supervised Transfer actions should be used in call flows rather than Transfer actions.

• All systems in the network must have this setting set to match each other.

High Quality Conferencing

Default = On.

Supports the use of the G.722 codec. IP lines and extensions using G.722 are provided with wide band audio. If High Quality Conferencing is enabled, when several wide band audio devices are in the same conference, the system will ensure that the audio between them remains wide band, even if the conference also contains other lines and devices using narrow band audio (analog devices, digital devices and IP devices using codecs other than G.722).

Digital/Analogue Auto Create User

Default = On. (IP500 V2 only. Default = Off for Server Edition/On for others)

When enabled, an associated user is created for each digital/analogue extension created. Digital/analogue extension creation occurs on initial start up, reset of configuration, or addition of new digital/analogue expansion units or plug-in modules.

Directory Overrides Barring

Default = On.

When enabled, barred numbers are not barred if the dialed number is in the External Directory.

Advertize Callee State To Internal Callers

Default = Off.

When enabled, for internal calls, additional status information is communicated to the calling party.

Not supported for SIP endpoints except for J100 Series phones (not including the J129).

• When calling another internal phone and the called phone is set to Do Not Disturb or on another call, the calling phone displays “Do Not Disturb” or “On Another Call” rather than “Number Busy”.

• On 9400, 9500, 9600, and J100 Series phones, if a line appearance is programmed on a button on phone A and that line is in use on phone B, then phone A displays the name of the current user of the line and the line number.

• If a line appearance on a phone is in use elsewhere in the system and another extension unsuccessfully attempts to seize that line, the phone displays “In Use:<name>” where <name> is the name of the user currently using the line.

This configuration parameter sets the system wide default. Individual users can be configured for this feature using the setting Call Management > Users > Add/Edit Users > Telephony > Call Settings > Advertize Callee State To Internal Callers

Internal Ring on Transfer

Default = Off.

When enabled, the transfer enquiry calls ring with internal ring tone even if the call being transferred is external. If the user completes the transfers before being answered, the ring tone played to the target changes to the ring tone appropriate for the call being transferred.

This feature is supported on phone series: 1400, 1600, 9400, 9500, 9600, J100 and analog phones. This feature is not supported on SIP and H.323 DECT phones.

Enforcement

Default = On.

When on, a user PIN is required.

Minimum Length

Default = 6. Maximum 15 digits.

The number of users with login codes less than six digits is displayed below the field in red colored text.

Complexity

Default = On.

When on, the following complexity rules are enforced.

• The user extension number cannot be used.

• A PIN consisting of repeated digits is not allowed (111111).

• A PIN consisting of forward or backward sequence are not allowed. Examples: 123456, 654321.

Send RTCP to an RTCP Collector

When the check box is selected, system RTCP reporting is enabled. For IP Office Release 10.0 and higher, in addition to having the individual phones send RTCP call quality reports, the system can also send RTCP reports for calls.

Server Address

This Sets the address of the third-party QoS monitoring application to which the system sends RTCP reports.

UDP Port Number

The destination port. The default for this filed is 5005.

RTCP reporting interval (secs)

This setting sets the time interval at which the system sends RTCP reports.