Per-Registration Call Parameters

Polycom phones support an optional per-registration feature that enables automatic call placement when the phone is off-hook.

The phones also support a per-registration configuration that determines which events cause the missed-calls counter to increment. You can enable/disable missed call tracking on a per-line basis.

In the following table, x is the registration number.

To view the list of maximum registrations for each phone model see Flexible Call Appearances.

Table 1. Per-Registration Call Parameters

Template

Parameter

Permitted Values

Change Causes

Restart or Reboot

sip-interop.cfg

call.advancedMissedCalls.addToReceivedList

Applies to calls on that are answered remotely.

0 (default) - Calls answered from the remote phone are not added to the local receive call list.

1 - Calls answered from the remote phone are added to the local

receive call list.

No

sip-interop.cfg

call.advancedMissedCalls.enabled

Use this parameter to improve call handling.

1 (default) - Shared lines can correctly count missed calls.

0 - Shared lines may not correctly count missed calls.

No

sip-interop.cfg

call.advancedMissedCalls.reasonCodes

Enter a comma-separated list of reason code indexes interpreted to mean that a call should not be considered as a missed call.

200 (default)

No

reg-advanced.cfg

call.autoAnswer.micMute

1 (default) - The microphone is initially muted after a call is auto-answered.

0 - The microphone is active immediately after a call is auto-answered.

No

reg-advanced.cfg

call.autoAnswer.ringClass

The ring class to use when a call is to be automatically answered using the auto-answer feature. If set to a ring class with a type other than answer or ring-answer , the setting are overridden such that a ringtone of visual (no ringer) applies.

ringAutoAnswer (default)

No

reg-advanced.cfg

call.autoAnswer.ringTone

Intercom (default) – Auto answer plays the intercom tone.

doubleBeep – Auto answer plays the double-beep tone.

No

reg-advanced.cfg

call.autoAnswer.SIP

This parameter cannot be used with VVX 101, 150, or 201 phones.

0 (default) - Disable auto-answer for SIP calls.

1 - Enable auto-answer for SIP calls.

No

reg-advanced.cfg call.autoAnswer.ringTone Sets the auto-answer tone on the phone.

intercom (default) – While auto answering a call, phone plays an intercom tone.

doubleBeep – Phone plays the double beep tone.

No

featurescfg

call.autoAnswerMenu.enable

1 (default) - The autoanswer menu displays and is available to the user.

0 - The autoanswer menu is disabled and is not available to the user.

No

sip-interop.cfg

call.BlindTransferSpecialInterop

0 (default) - Do not wait for an acknowledgment from the transferee before ending the call.

1 - Wait for an acknowledgment from the transferee before ending the call.

No

sip-interop.cfg

call.dialtoneTimeOut

The time is seconds that a dial tone plays before a call is dropped.

60 (default)

0 - The call is not dropped.

Yes

sip-interop.cfg

call.internationalDialing.enabled

Use this parameter to enable or disable the key tap timer that converts a double tap of the asterisk “*” symbol to the “+” symbol used to indicate an international call.

1 (default) - A quick double tap of “*” converts immediately to “+”. To enter a double asterisk “**”, tap “*” once and wait for the key tap timer to expire to enter a second “*”.

0 - You cannot dial”+” and you must enter the international exit code of the country you are calling from to make international calls.

This parameter applies to all numeric dial pads on the phone including for example, the contact directory.

Yes

sip-interop.cfg, site.cfg

call.internationalPrefix.key

0 (default)

1

No

sip-interop.cfg

call.localConferenceEnabled

1 (default) - The feature to join a conference during an active call is enabled and you can establish conferences on the phone.

0 - The feature to join a conference during an active call is disabled. When you try to join the Conference, an ‘Unavailable' message displays.

Yes

sip-interop.cfg

call.offeringTimeOut

Specify a time in seconds that an incoming call rings before the call is dropped.

60 (default)

0 - No limit.

Note that the call diversion, no answer feature takes precedence over this feature when enabled.

Yes

sip-interop.cfg

call.playLocalRingBackBeforeEarlyMediaArrival

Determines whether the phone plays a local ring-back after receiving a first provisional response from the far end.

1 (default) - The phone plays a local ringback after receiving the first provisional response from the far end. If early media is received later, the phone stops the local ringback and plays the early media.

0 - No local ringback plays, and the phone plays only the early media received.

No

site.cfg

call.playLocalRingBackBeforeEarlyMediaArrival

0 (default) - URL mode is used for URL calls.

1 - Number mode is used for URL calls.

No

sip-interop.cfg

call.ringBackTimeOut

Specify a time in seconds to allow an outgoing call to remain in the ringback state before dropping the call.

60 (default)

0 - No limit.

Yes

sip-interop.cfg

call.showDialpadOnProceeding

0 (default) – The phone does not show the dialpad button while a placed call is outgoing.

1 – The phone displays the dialpad button while a placed call is outgoing.

No

sip-interop.cfg, site.cfg

call.stickyAutoLineSeize

0 - Dialing through the call list uses the line index for the previous call. Dialing through the contact directory uses a random line index.

1 - The phone uses sticky line seize behavior. This helps with features that need a second call object to work with. The phone attempts to initiate a new outgoing call on the same SIP line that is currently in focus on the LCD. Dialing through the call list when there is no active call uses the line index for the previous call. Dialing through the call list when there is an active call uses the current active call line index. Dialing through the contact directory uses the current active call line index.

Yes

sip-interop.cfg, site.cfg

call.stickyAutoLineSeize.onHookDialing

0 (default)

If call.stickyAutoLineSeize is set to 1, this parameter has no effect. The regular stickyAutoLineSeize behavior is followed.

If call.stickyAutoLineSeize is set to 0 and this parameter is set to 1, this overrides the stickyAutoLineSeize behavior for hot dial only. (Any new call scenario seizes the next available line.)

If call.stickyAutoLineSeize is set to 0 and this parameter is set to 0, there is no difference between hot dial and new call scenarios.

A hot dial occurs on the line which is currently in the call appearance. Any new call scenario seizes the next available line.

Yes

sip-interop.cfg call.switchToLocalRingbackWithoutRTP

Determines whether local ringback plays in the event that early media stops.

0 (default) – No ringback plays when early media stops.

1 – The local ringback plays if no early media is received.

No

site.cfg

call.teluri.showPrompt

1 (default) - Phone displays a pop-up box to either call or cancel the number when tel URI is executed.

0 - Phone does not display the pop-up box.

No

sip-interop.cfg

call.urlModeDialing

0 (default) - Disable URL dialing.

1 - Enable URL dialing.

Yes