Multiple Line Registrations Parameters

Each registration can be mapped to one or more line keys, however, a line key can be used for only one registration.

The maximum number of call appearances you can set varies by phone model.

Table 1. Multiple Registrations Parameters

Template

Parameter

Permitted Values

Change Causes Restart or Reboot

reg-advanced.cfg

reg.x.acd-agent-available

0 (default) - The ACD feature is disabled for registration.

1 - If both ACD login/logout and agent available are set to 1 for registration x, the ACD feature is enabled for that registration.

No

reg-advanced.cfg reg-advanced.cfg

reg.x.acd-login-logout reg.x.acd-agent-available

0 (default) - The ACD feature is disabled for registration.

1 - If both ACD login/logout and agent available are set to 1 for registration x, the ACD feature is enabled for that registration.

No

reg-advanced.cfg reg-advanced.cfg

reg.x.acd-login-logout reg.x.acd-agent-available

0 (default) - The ACD feature is disabled for registration.

1 - If both ACD login/logout and agent available are set to 1 for registration x, the ACD feature is enabled for that registration.

No

reg-basic.cfg

reg.x.address

The user part (for example, 1002) or the user and the host part (for example, 1002@polycom.com ) of the registration SIP URI or the H.323 ID/extension.

Null (default)

string address

No

reg-advanced.cfg

reg.x.advancedConference.maxParticipants

Sets the maximum number of participants allowed in a push to conference for advanced conference calls. The number of participants configured must match the number of participants allowed on the ALU CTS.

3 (default)

0 - 25

No

reg-advanced.cfg

reg.x.advancedConference.pushToConference

0 (default) - Disable push-to-conference functionality.

1 - Enable push-to-conference functionality.

No

reg-advanced.cfg

reg.x.advancedConference.subscribeForConfEvents

1 (default) - Conference participants to receive notifications for conference events is enabled.

0 - Conference participants to receive notifications for conference events is disabled.

No

reg-advanced.cfg

reg.x.advancedConference.subscribeForConfEventsOnCCPE

1 (default) - Enable the conference host to receive notifications for conference events.

0 - Disable the conference host to receive notifications for conference events.

No

reg-advanced.cfg

reg.x.auth.domain

The domain of the authorization server that is used to check the user names and passwords.

Null (default)string

No

reg-advanced.cfg

reg.x.auth.optimizedInFailover

The destination of the first new SIP request when failover occurs.

0 (default) - The SIP request is sent to the server with the highest priority in the server list.

1 - The SIP request is sent to the server which sent the proxy authentication request.

No

reg-basic.cfg

reg.x.auth.password

The password to be used for authentication challenges for this registration.

Null (default)

string - It overrides the password entered into the Authentication submenu on the Settings menu of the phone.

No

reg-advanced.cfg

reg.x.auth.useLoginCredentials

0 - (default) The Login credentials are not used for authentication to the server on registration x.

1 - The login credentials are used for authentication to the server.

No

reg-basic.cfg

reg.x.auth.userId

User ID to be used for authentication challenges for this registration.

Null (default)

string - If the User ID is non-Null, it overrides the user parameter entered into the Authentication submenu on the Settings menu of the phone.

No

reg-advanced.cfg

reg.x.bargeInEnabled

0 (default) - barge-in is disabled for line x.

1 - barge-in is enabled (remote users of shared call appearances can interrupt or barge in to active calls).

No

reg.x.bridgeInEnabled

0 (default) - Bridge In feature is disabled.

1 - Bridge In feature is enabled.

No

features.cfg

reg.x.broadsoft.userId

Enter the BroadSoft user ID to authenticate with the BroadSoft XSP service interface.

Null (default)

string

No

features.cfg

reg.x.broadsoft.useXspCredentials

If this parameter is disabled, the phones use standard SIP credentials to authenticate.

1 (default) - Use this value, if phone lines are registered with a server running BroadWorks R19 or earlier.

0 - Set to 0, if phone lines are registered with a server running BroadWorks R19 SP1 or later.

No

features.cfg

reg.x.broadsoft.xsp.password

Enter the password associated with the BroadSoft user account for the line. Required only when reg.x.broadsoft.useXspCredentials=1 .

Null (default)

string

No

reg-advanced.cfg

reg.x.callsPerLineKey

Set the maximum number of concurrent calls for a single registration x. This parameter applies to all line keys using registration x. If registration x is a shared line, an active call counts as a call appearance on all phones sharing that registration.

This per-registration parameter overrides call.callsPerLineKey .

24 (default)

1-24

VVX 101, 201

8 (default)

1 - 8

No

reg-advanced.cfg

reg.x.displayName

The display name used in SIP signaling and/or the H.323 alias used as the default caller ID.

Null (default)

UTF-8 encoded string

No

features.cfg

reg.x.enablePvtHoldSoftKey

This parameter applies only to shared lines.

0 (default) - To disable user on a shared line to hold calls privately.

1 - To enable users on a shared line to hold calls privately.

No

features.cfg

reg.x.enablePvtHoldSoftKey

This parameter applies only to shared lines.

0 (default) - To disable user on a shared line to hold calls privately.

1 - To enable users on a shared line to hold calls privately.

No

reg-advanced.cfg

reg.x.enhancedCallPark.enabled

0 (default) - To disable the BroadWorks Enhanced Call Park feature.

1 - To enable the BroadWorks Enhanced Call Park feature.

No

reg.x.filterReflectedBlaDialogs

1 (default) - bridged line appearance NOTIFY messages are ignored.

0 - bridged line appearance NOTIFY messages is not ignored

No

reg-advanced.cfg

reg.x.fwd.busy.contact

The forward-to contact for calls forwarded due to busy status.

Null (default) - The contact specified by divert.x.contact is used.

string - The contact specified by divert.x.contact is not used

No

reg-advanced.cfg

reg.x.fwd.busy.contact

The forward-to contact for calls forwarded due to busy status.

Null (default) - The contact specified by divert.x.contact is used.

string - The contact specified by divert.x.contact is not used

No

reg-advanced.cfg

reg.x.fwd.busy.status

0 (default) - Incoming calls that receive a busy signal is not forwarded

1 - Busy calls are forwarded to the contact specified by reg.x.fwd.busy.contact .

No

reg-advanced.cfg

reg.x.fwd.busy.status

0 (default) - Incoming calls that receive a busy signal is not forwarded

1 - Busy calls are forwarded to the contact specified by reg.x.fwd.busy.contact .

No

reg-advanced.cfg

reg.x.fwd.noanswer.contact

Null (default) - The forward-to contact specified by divert.x.contact is used.

string - The forward to contact used for calls forwarded due to no answer.

No

reg-advanced.cfg

reg.x.fwd.noanswer.contact

Null (default) - The forward-to contact specified by divert.x.contact is used.

string - The forward to contact used for calls forwarded due to no answer.

No

reg-advanced.cfg

reg.x.fwd.noanswer.ringCount

The number of seconds the phone should ring for before the call is forwarded because of no answer. The maximum value accepted by some call servers is 20.

0 - (default)

1 to 65535

No

reg-advanced.cfg

reg.x.fwd.noanswer.ringCount

The number of seconds the phone should ring for before the call is forwarded because of no answer. The maximum value accepted by some call servers is 20.

0 - (default)

1 to 65535

No

reg-advanced.cfg

reg.x.fwd.noanswer.status

0 (default) - The calls are not forwarded if there is no answer.

1 - The calls are forwarded to the contact specified by reg.x.noanswer.contact after ringing for the length of time specified by reg.x.fwd.noanswer.ringCount .

No

reg-advanced.cfg

reg.x.fwd.noanswer.status

0 (default) - The calls are not forwarded if there is no answer.

1 - The calls are forwarded to the contact specified by reg.x.noanswer.contact after ringing for the length of time specified by reg.x.fwd.noanswer.ringCount .

No

sip-interop.cfg

reg.x.gruu

1 - The phone sends sip.instance in the REGISTER request.

0 (default) - The phone does not send sip.instance in the REGISTER request.

No

debug.cfg

reg.x.gruu

Specify if the phone sends sip.instance in the REGISTER request.

0 (default)

1

No

reg-advanced.cfg

reg.x.header.pearlymedia.support

0 (Default) - The p-early-media header is not supported on the specified line registration.

1 - The p-early-media header is supported by the specified line registration.

No

reg-basic.cfg

reg.X.insertOBPAddressInRoute

1 (Default) - The outbound proxy address is added as the topmost route header.

0 - The outbound proxy address is not added to the route header.

No

reg-basic.cfg

reg.x.label

The text label that displays next to the line key for registration x.

The maximum number of characters for this parameter value is 256; however, the maximum number of characters that a phone can display on its user interface varies by phone model and by the width of the characters you use. Parameter values that exceed the phone's maximum display length are truncated by ellipses (…). The rules for parameter up.cfgLabelElide determine how the label is truncated.

Null (default) - the label is determined as follows:
  • If reg.1.useteluriAsLineLabel=1 , then the tel URI/phone number/address displays as the label.
  • If reg.1.useteluriAsLineLabel=0 , then the value for reg.x.displayName , if available, displays as the label. If reg.x.displayName is unavailable, the user part of reg.x.address is used.

UTF-8 encoded string

No

site.cfg

reg.x.line.y.label

Configure a unique line label for a shared line that has multiple line key appearances. This parameter takes effect when u p.cfgUniqueLineLabel=1 . If reg.x.linekeys=1 , this parameter does not have any effect.

x = the registration index number starting from 1.

y = the line index from 1 to the value set by reg.x.linekeys . Specifying a string sets the label used for the line key registration on phones with multiple line keys.

If no parameter value is set for reg.x.line.y.label , the phone automatically numbers multiple lines by prepending “<y>_” where <y> is the line index from 1 to the value set by reg.x.linekeys .

  • The following examples show labels for line 1 on a phone with user registration 1234, where reg.x.linekeys=2 :
    • If no label is configured for registration, the labels are “1_1234” and “2_1234”.
    • If reg.1.line.1.label=Polycom and reg.1.line.2.label=VVX , the labels display as ‘Polycom' and ‘VVX'.

No

site.cfg

reg.x.line.y.label

Configure a unique line label for a shared line that has multiple line key appearances. This parameter takes effect when u p.cfgUniqueLineLabel=1 . If reg.x.linekeys=1 , this parameter does not have any effect.

x = the registration index number starting from 1.

y = the line index from 1 to the value set by reg.x.linekeys . Specifying a string sets the label used for the line key registration on phones with multiple line keys.

If no parameter value is set for reg.x.line.y.label , the phone automatically numbers multiple lines by prepending “<y>_” where <y> is the line index from 1 to the value set by reg.x.linekeys .

  • The following examples show labels for line 1 on a phone with user registration 1234, where reg.x.linekeys=2 :
    • If no label is configured for registration, the labels are “1_1234” and “2_1234”.
    • If reg.1.line.1.label=Polycom and reg.1.line.2.label=VVX , the labels display as ‘Polycom' and ‘VVX'.

No

reg-basic.cfg

reg.x.lineAddress

The line extension for a shared line. This parameter applies to private lines and BroadSoft call park and retrieve. If there is no extension provided for this parameter, the call park notification is ignored for the shared line.

Null (default)

String

No

reg-advanced.cfg

reg.x.lineKeys

Specify the number of line keys to use for a single registration. The maximum number of line keys you can use per registration depends on your phone model.

1 (default)

1 to max

No

reg-advanced.cfg

reg.x.lineKeys

Specify the number of line keys to use for a single registration. The maximum number of line keys you can use per registration depends on your phone model.

1 (default)

1 to max

No

lync.cfg

reg.x.lisdisclaimer

This parameter sets the value of the location policy disclaimer. For example, the disclaimer may be “Warning: If you do not provide a location, emergency services may be delayed in reaching your location should you need to call for help.”

Null (default)

string, 0 to 256 characters

No

reg-advanced.cfg

reg.x.musicOnHold.uri

A URI that provides the media stream to play for the remote party on hold.

Null (default) - This parameter does not overrides voIpProt.SIP.musicOnHold.uri .

a SIP URI - This parameter overrides voIpProt.SIP.musicOnHold.uri .

No

reg-advanced.cfg

reg.x.offerFullCodecListUponResume

1 (default) - The phone sends full audio and video capabilities after resuming a held call irrespective of the audio and video capabilities negotiated at the initial call answer.

0 - The phone does not send full audio and video capabilities after resuming a held call.

No

reg-basic.cfg

reg.x.outboundProxy.address

The IP address or hostname of the SIP server to which the phone sends all requests.

Null (default)

IP address or hostname

No

sip-interop.cfg

reg.x.outboundProxy.failOver.failBack.mode

The mode for failover failback (overrides reg.x.server.y.failOver.failBack.mode ).

duration - (default) The phone tries the primary server again after the time specified by reg.x.outboundProxy.failOver.failBack.timeout expires.

newRequests - All new requests are forwarded first to the primary server regardless of the last used server.

DNSTTL - The phone tries the primary server again after a timeout equal to the DNS TTL configured for the server that the phone is registered to.

No

reg-advanced.cfg

reg.x.outboundProxy.failOver.failBack.timeout

3600 (default) -The time to wait (in seconds) before failback occurs (overrides reg.x.server.y.failOver.failBack.timeout ).

0, 60 to 65535 - The phone does not fail back until a failover event occurs with the current server.

No

reg-advanced.cfg

reg.x.outboundProxy.failOver.failRegistrationOn

1 (default) - The reRegisterOn parameter is enabled, the phone silently invalidates an existing registration.

0 - The reRegisterOn parameter is enabled, existing registrations remain active.

No

reg-advanced.cfg

reg.x.outboundProxy.failOver.onlySignalWithRegistered

1 (default) - The reRegisterOn and failRegistrationOn parameters are enabled, no signaling is accepted from or sent to a server that has failed until failback is attempted or failover occurs.

0 - The reRegisterOn and failRegistrationOn parameters are enabled, signaling is accepted from and sent to a server that has failed.

No

reg-advanced.cfg

reg.x.outboundProxy.failOver.reRegisterOn

This parameters overrides reg.x.server.y.failOver.reRegisterOn .

0 (default) - The phone won't attempt to register with the secondary server.

1 - The phone attempts to register with (or via, for the outbound proxy scenario), the secondary server.

No

reg-advanced.cfg

reg.x.outboundProxy.port

The port of the SIP server to which the phone sends all requests.

0 - (default)

1 to 65535

No

reg-advanced.cfg

reg.x.outboundProxy.transport

The transport method the phone uses to communicate with the SIP server.

DNSnaptr (default)

DNSnaptr, TCPpreferred, UDPOnly, TLS, TCPOnly

No

features.cfg

reg.x.path

0 (Default) - The path extension header field in the Register request message is not supported for the specific line registration.

1 - The phone supports and provides the path extension header field in the Register request message for the specific line registration.

No

sip-interop.cfg

reg.x.protocol.H323

You can use this parameter for the VVX 500/501, 600/601, and 1500.

0 (default) - H.323 signaling is not enabled for registration x.

1 - H.323 signaling is enabled for registration x.

No

sip-interop.cfg

reg.x.protocol.H323

You can use this parameter for the VVX 500/501, 600/601, and 1500.

0 (default) - H.323 signaling is not enabled for registration x.

1 - H.323 signaling is enabled for registration x.

No

sip-interop.cfg

reg.x.protocol.SIP

You can use this parameter for the VVX 500/501, 600/601, and 1500.

1 (default) - SIP signaling is enabled for this registration.

0 - SIP signaling is not enabled for this registration.

No

sip-interop.cfg

reg.x.proxyRequire

Null (default) - No Proxy-Require is sent.

string - Needs to be entered in the Proxy-Require header.

No

features.cfg

reg.x.regevent

0 (default) - The phone is not subscribed to registration state change notifications for the specific phone line.

1 - The phone is subscribed to registration state change notifications for the specific phone line.

This parameter overrides the global parameter voIpProt.SIP.regevent.

No

reg-advanced.cfg

reg.x.rejectNDUBInvite

Specify whether or not the phone accepts a call for a particular registration in case of a Network Determined User Busy (NDUB) event advertised by the SIP server.

0 (Default) - If an NDUB event occurs, the phone does not reject the call.

1 - If an NDUB event occurs, the phone rejects the call with a 603 Decline response code.

No

reg-advanced.cfg

reg.x.ringType

The ringer to be used for calls received by this registration. The default is the first non-silent ringer.

If you use the configuration parameters ringer13 and ringer14 on a single registered line, the phone plays SystemRing.wav.

default (default)

ringer1 to ringer24

No

reg-advanced.cfg

reg.x.ringType

The ringer to be used for calls received by this registration.

ringer2 (default) - Is the first non-silent ringer.

ringer1 to ringer24 - To play ringer on a single registered line.

No

site.cfg

reg.x.server.H323.y.address

Address of the H.323 gatekeeper.

Null (default)

IP address or hostname

No

site.cfg

reg.x.server.H323.y.address

Address of the H.323 gatekeeper.

Null (default)

IP address or hostname

No

site.cfg

reg.x.server.H323.y.address

Address of the H.323 gatekeeper.

Null (default)

IP address or hostname

No

site.cfg

reg.x.server.H323.y.expires

Desired registration period.

3600

positive integer

No

site.cfg

reg.x.server.H323.y.expires

Desired registration period.

3600

positive integer

No

site.cfg

reg.x.server.H323.y.expires

Desired registration period.

3600

positive integer

No

site.cfg

reg.x.server.H323.y.port

Port to be used for H.323 signaling. If set to Null, 1719 (H.323 RAS signaling) is used.

0 (default)

0 to 65535

No

site.cfg

reg.x.server.H323.y.port

Port to be used for H.323 signaling. If set to Null, 1719 (H.323 RAS signaling) is used.

0 (default)

0 to 65535

No

site.cfg

reg.x.server.H323.y.port

Port to be used for H.323 signaling. If set to Null, 1719 (H.323 RAS signaling) is used.

0 (default)

0 to 65535

No

site.cfg

reg.x.server.y.address

If this parameter is set, it takes precedence even if the DHCP server is available.

Null (default) - SIP server does not accepts registrations.

IP address or hostname - SIP server that accepts registrations. If not Null, all of the parameters in this table override the parameters specified in voIpProt.server.*

No

reg-advanced

reg.x.server.y.expires

The phone's requested registration period in seconds.

The period negotiated with the server may be different. The phone attempts to re-register at the beginning of the overlap period.

3600 - (default)

positive integer, minimum 10

No

reg-advanced

reg.x.server.y.expires.lineSeize

Requested line-seize subscription period.

30 - (default)

0 to 65535

No

reg-advanced

reg.x.server.y.expires.overlap

The number of seconds before the expiration time returned by server x at which the phone should try to re-register.

The phone tries to re-register at half the expiration time returned by the server if the server value is less than the configured overlap value.

60 (default)

5 to 65535

No

site.cfg

reg.x.server.y.failOver.failBack.mode

duration (default) - The phone tries the primary server again after the time specified by reg.x.server.y.failOver.failBack.timeout .

newRequests - All new requests are forwarded first to the primary server regardless of the last used server.

DNSTTL - The phone tries the primary server again after a timeout equal to the DNS TTL configured for the server that the phone is registered to.

registration - The phone tries the primary server again when the registration renewal signaling begins.

This parameter overrides voIpProt.server.x.failOver.failBack.mode)

No

site.cfg

reg.x.server.y.failOver.failBack.timeout

3600 (default) - The time to wait (in seconds) before failback occurs.

0 - The phone does not fail back until a failover event occurs with the current server.

60 to 65535 - If set to Duration, the phone waits this long after connecting to the current working server before selecting the primary server again.

No

site.cfg

reg.x.server.y.failOver.failRegistrationOn

1 (default) - The reRegisterOn parameter is enabled, the phone silently invalidates an existing registration (if it exists) at the point of failing over.

0 - The reRegisterOn parameter is disabled, existing registrations remain active.

No

site.cfg

reg.x.server.y.failOver.onlySignalWithRegistered

1 (default) - Set to this value and reRegisterOn and failRegistrationOn parameters are enabled, no signaling is accepted from or sent to a server that has failed until failback is attempted or failover occurs. If the phone attempts to send signaling associated with an existing call via an unregistered server (for example, to resume or hold a call), the call ends. No SIP messages are sent to the unregistered server.

0 - Set to this value and reRegisterOn and failRegistrationOn parameters are enabled, signaling is accepted from and sent to a server that has failed (even though failback hasn't been attempted or failover hasn't occurred).

No

site.cfg

reg.x.server.y.failOver.reRegisterOn

0 (default) - The phone does not attempt to register with the secondary server, since the phone assumes that the primary and secondary servers share registration information.

1 - The phone attempts to register with (or via, for the outbound proxy scenario), the secondary server. If the registration succeeds (a 200 OK response with valid expires), signaling proceeds with the secondary server.

This parameter overrides voIpProt.server.x.failOver.reRegisterOn .

No

site.cfg

reg.x.server.y.port

Null (default) - The port of the SIP server does not specifies registrations.

0 - The port used depends on reg.x.server.y.transport .

1 to 65535 - The port of the SIP server that specifies registrations.

No

site.cfg

reg.x.server.y.register

1 (default) - Calls can not be routed to an outbound proxy without registration.

0 - Calls can be routed to an outbound proxy without registration.

See voIpProt.server.x.register for more information, see SIP Server Fallback Enhancements on Polycom Phones - Technical Bulletin 5844 on Polycom Engineering Advisories and Technical Notifications.

No

sip-interop.cfg

reg.x.server.y.registerRetry.baseTimeOut

For registered line x, set y to the maximum time period the phone waits before trying to re-register with the server.Used in conjunction with reg.x.server.y.registerRetry.maxTimeOut to determine how long to wait.

60 (default)

10 - 120 seconds

No

sip-interop.cfg

reg.x.server.y.registerRetry.maxTimeout

For registered line x, set y to the maximum time period the phone waits before trying to re-register with the server. Use in conjunction with r eg.x.server.y.registerRetry.baseTimeOut to determine how long to wait. The algorithm is defined in RFC 5626.

180 - (default)

60 - 1800 seconds

No

reg-advanced.cfg

reg.x.server.y.retryMaxCount

The number of retries attempted before moving to the next available server.

3 - (default)

0 to 20 - 3 is used when the value is set to 0.

No

reg-advanced.cfg

reg.x.server.y.retryTimeOut

0 (default) - Use standard RFC 3261 signaling retry behavior.

0 to 65535 - The amount of time (in milliseconds) to wait between retries.

No

reg-advanced.cfg

reg.x.server.y.specialInterop

Specify the server-specific feature set for the line registration.

Standard (Default)

VVX 101:

Standard

GENBAND

ALU-CTS

DT

VVX 201:

Standard,

GENBAND

ALU-CTS

ocs2007r2

lync2010

All other phones:

Standard

GENBAND

ALU-CTS

ocs2007r2

lync2010

lcs2005

reg-advanced.cfg

reg.x.server.y.subscribe.expires

The phone's requested subscription period in seconds after which the phone attempts to resubscribe at the beginning of the overlap period.

3600 seconds - (default)

10 - 2147483647 (seconds)

You can use this parameter in conjunction with reg.x.server.y.subscribe.expires.overlap .

No

reg-advanced.cfg

reg.x.server.y.subscribe.expires

The phone's requested subscription period in seconds after which the phone attempts to resubscribe at the beginning of the overlap period.

3600 seconds - (default)

10 - 2147483647 (seconds)

You can use this parameter in conjunction with reg.x.server.y.subscribe.expires.overlap .

No

reg-advanced.cfg

reg.x.server.y.subscribe.expires.overlap

The number of seconds before the expiration time returned by server x after which the phone attempts to resubscribe. If the server value is less than the configured overlap value, the phone tries to resubscribe at half the expiration time returned by the server.

60 seconds (default)

5 - 65535 seconds

No

reg-advanced.cfg

reg.x.server.y.subscribe.expires.overlap

The number of seconds before the expiration time returned by server x after which the phone attempts to resubscribe. If the server value is less than the configured overlap value, the phone tries to resubscribe at half the expiration time returned by the server.

60 seconds (default)

5 - 65535 seconds

No

site.cfg

reg.x.server.y.transport

The transport method the phone uses to communicate with the SIP server.

DNSnaptr (default) - If reg.x.server.y.address is a hostname and reg.x.server.y.port is 0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If reg.x.server.y.address is an IP address, or a port is given, then UDP is used.

TCPpreferred - TCP is the preferred transport; UDP is used if TCP fails.

UDPOnly - Only UDP is used.

TLS - If TLS fails, transport fails. Leave port field empty (defaults to 5061 ) or set to 5061 .

TCPOnly - Only TCP is used.

No

site.cfg

reg.x.server.y.useOutboundProxy

1 (default) - Enables to use the outbound proxy specified in reg.x.outboundProxy.address for server x.

0 - Disable to use the outbound proxy specified in reg.x.outboundProxy.address for server x.

No

reg-advanced.cfg

reg.x.serverFeatureControl.callRecording

1 (default) - BroadSoft BroadWorks v20 call recording feature for individual phone lines is enabled.

0 - BroadSoft BroadWorks v20 call recording feature for individual phone lines is disabled.

No

reg-advanced.cfg

reg.x.serverFeatureControl.callRecording

1 (default) - BroadSoft BroadWorks v20 call recording feature for individual phone lines is enabled.

0 - BroadSoft BroadWorks v20 call recording feature for individual phone lines is disabled.

No

reg-advanced.cfg

reg.x.serverFeatureControl.cf

This parameter overrides voIpProt.SIP.serverFeatureControl.cf .

0 (default) - The server-based call forwarding is disabled.

1 - server based call forwarding is enabled.

Yes

reg-advanced.cfg

reg.x.serverFeatureControl.cf

This parameter overrides voIpProt.SIP.serverFeatureControl.cf .

0 (default) - The server-based call forwarding is disabled.

1 - server based call forwarding is enabled.

Yes

reg-advanced.cfg

reg.x.serverFeatureControl.dnd

This parameter overrides voIpProt.SIP.serverFeatureControl.dnd.

0 (default) - server-based do-not-disturb (DND) is disabled.

1 - server-based DND is enabled and the call server has control of DND.

Yes

sip-interop.cfg

reg.x.serverFeatureControl.localProcessing.cf

This parameter overrides voIpProt.SIP.serverFeatureControl.localProcessing.cf .

0 (default) - If reg.x.serverFeatureControl.cf is set to 1 the phone does not perform local Call Forward behavior.

1 - The phone performs local Call Forward behavior on all calls received.

No

sip-interop.cfg

reg.x.serverFeatureControl.localProcessing.cf

This parameter overrides voIpProt.SIP.serverFeatureControl.localProcessing.cf .

0 (default) - If reg.x.serverFeatureControl.cf is set to 1 the phone does not perform local Call Forward behavior.

1 - The phone performs local Call Forward behavior on all calls received.

No

sip-interop.cfg

reg.x.serverFeatureControl.localProcessing.dnd

This parameter overrides voIpProt.SIP.serverFeatureControl.localProcessing.dnd .

0 (default) - If reg.x.serverFeatureControl.dnd is set to 1, the phone does not perform local DND call behavior.

1 - The phone performs local DND call behavior on all calls received.

No

reg-advanced.cfg

reg.x.serverFeatureControl.securityClassification

0 (default) - The visual security classification feature for a specific phone line is disabled.

1 - The visual security classification feature for a specific phone line is enabled.

No

reg-advanced.cfg

reg.x.serverFeatureControl.securityClassification

0 (default) - The visual security classification feature for a specific phone line is disabled.

1 - The visual security classification feature for a specific phone line is enabled.

No

reg-advanced.cfg

reg.x.serverFeatureControl.signalingMethod

Controls the method used to perform call forwarding requests to the server.

serviceMsForwardContact (default)

string

No

sip-interop.cfg

reg.x.srtp.enable

1 (default) - The registration accepts SRTP offers.

0 - The registration always declines SRTP offers.

Yes

sip-interop.cfg

reg.x.srtp.offer

This parameter applies to the registration initiating (offering) a phone call.

0 (default) - No secure media stream is included in SDP of a SIP INVITE.

1 - The registration includes a secure media stream description along with the usual non-secure media description in the SDP of a SIP INVITE.

Yes

sip-interop.cfg

reg.x.srtp.require

0 (default) - Secure media streams are not required.

1 - The registration is only allowed to use secure media streams.

Yes

sip-interop.cfg

reg.x.srtp.simplifiedBestEffort

This parameter overrides sec.srtp.simplifiedBestEffort .

1 (default) - Negotiation of SRTP compliant with Microsoft Session Description Protocol Version 2.0 Extensions is supported.

0 - No SRTP is supported.

No

sip-interop.cfg

reg.x.strictLineSeize

0 (default) - Dial prompt is provided immediately without waiting for a successful OK from the call server.

1 - The phone is forced to wait for 200 OK on registration x when receiving a TRYING notify.

This parameter overrides voIpProt.SIP.strictLineSeize for registration x.

No

sip-interop.cfg

reg.x.tcpFastFailover

0 (default) - A full 32 second RFC compliant timeout is used.

1 - failover occurs based on the values of reg.x.server.y.retryMaxCount and voIpProt.server.x.retryTimeOut .

No

reg-advanced.cfg

reg.x.terminationType

Determines the type of termination that is used for the line where the line can be managed automatically on the VVX, the wireless handset, or on both. X = each registration index.

NULL (default)

VVX, DECT, or VVX-DECT

No

reg-advanced.cfg

reg.x.thirdPartyName

Null (default) - In all other cases.

string address -This field must match the reg.x.address value of the registration which makes up the part of a bridged line appearance (BLA).

No

reg-advanced.cfg

reg.x.thirdPartyName

Null (default) - In all other cases.

string address -This field must match the reg.x.address value of the registration which makes up the part of a bridged line appearance (BLA).

No

reg-advanced.cfg

reg.x.type

private (default) - Use standard call signaling.

shared - Use augment call signaling with call state subscriptions and notifications and use access control for outgoing calls.

No

reg-advanced.cfg

reg.x.type

private (default) - Use standard call signaling.

shared - Use augment call signaling with call state subscriptions and notifications and use access control for outgoing calls.

No

reg-advanced.cfg

reg.x.useCompleteUriForRetrieve

This parameters overrides voipPort.SIP.useCompleteUriForRetrieve .

1 (default) - The target URI in BLF signaling uses the complete address as provided in the XML dialog document.

0 - Only the user portion of the XML dialog document is used and the current registrar's domain is appended to create the full target URI.

No

sip-basic.cfg

voipProt.server.x.address

The IP address or hostname and port of a SIP server that accepts registrations. Multiple servers can be listed starting with x=1 to 4 for fault tolerance.

Null (default), IP address, or hostname

No

sip-interop.cfg

voIpProt.server.x.expires

The phone's requested registration period in seconds.

3600 (default)

positive integer, minimum 10

The period negotiated with the server may be different. The phone attempts to re-register at the beginning of the overlap period. For example, if expires="300" and overlap="5", the phone re-registers after 295 seconds (300-5).

No

sip-interop.cfg

voIpProt.server.x.expires

The phone's requested registration period in seconds. Note: The period negotiated with the server may be different. The phone will attempt to re-register at the beginning of the overlap period.

3600 (default)

positive integer, minimum 10

No

sip-interop.cfg

voIpProt.server.x.expires.lineSeize

Requested line-seize subscription period.

30 (default)

positive integer, minimum 10

No

sip-interop.cfg

voIpProt.server.x.expires.lineSeize

Requested line-seize subscription period.

30 (default)

positive integer, minimum 0 was 10

No

sip-interop.cfg

voIpProt.server.x.expires.overlap

The number of seconds before the expiration time returned by server x at which the phone should try to re-register. If the server value is less than the configured overlap value, the phone tries to re-register at half the expiration time returned by the server.

60 (default)

5 to 65536

No

sip-interop.cfg

voIpProt.server.x.expires.overlap

The number of seconds before the expiration time returned by server x at which the phone should try to re-register. The phone will try to re-register at half the expiration time returned by the server if the server value is less than the configured overlap value.

60 (default)

5 to 65535

No

sip-interop.cfg

voIpProt.server.x.failOver.failBack.mode

Specify the failover failback mode.

duration (default) - The phone tries the primary server again after the time specified by voIpProt.server.x.failOver.failBack.timeout

newRequests - All new requests are forwarded first to the primary server regardless of the last used server.

DNSTTL - The phone tries the primary server again after a timeout equal to the DNS TTL configured for the server that the phone is registered to.

registration - The phone tries the primary server again when the registration renewal signaling begins.

No

sip-interop.cfg

voIpProt.server.x.failOver.failBack.timeout

If voIpProt.server.x.failOver.failBack.mode is set to duration, this is the time in seconds after failing over to the current working server before the primary server is again selected as the first server to forward new requests to. Values between 1 and 59 result in a timeout of 60 and 0 means do not fail-back until a fail-over event occurs with the current server.

3600 (default)

0, 60 to 65535

No

sip-interop.cfg

voIpProt.server.x.failOver.failRegistrationOn

1 (default) - When set to 1, and the reRegisterOn parameter is enabled, the phone silently invalidates an existing registration (if it exists), at the point of failing over.

0 - When set to 0, and the reRegisterOn parameter is enabled, existing registrations remain active. This means that the phone attempts failback without first attempting to register with the primary server to determine if it has recovered.

No

sip-interop.cfg

voIpProt.server.x.failOver.onlySignalWithRegistered

1 (default) - When set to 1, and the reRegisterOn and failRegistrationOn parameters are enabled, no signaling is accepted from or sent to a server that has failed until failback is attempted or failover occurs. If the phone attempts to send signaling associated with an existing call via an unregistered server (for example, to resume or hold a call), the call ends. No SIP messages are sent to the unregistered server.

0 - When set to 0, and the reRegisterOn and failRegistrationOn parameters are enabled, signaling is accepted from and sent to a server that has failed (even though failback hasn't been attempted or failover hasn't occurred).

No

sip-interop.cfg

voIpProt.server.x.failOver.reRegisterOn

0 (default) - When set to 0, the phone won't attempt to register with the second.

1 - When set to 1, the phone attempts to register with (or by, for the outbound proxy scenario), the secondary server. If the registration succeeds (a 200 OK response with valid expires), signaling proceeds with the secondary server.

No

sip-basic.cfg

voIpProt.server.x.port

The port of the server that specifies registrations.

0 (default) - If 0, the port used depends on voIpProt.server.x.transport.

1 to 65535

No

voIpProt.server.x.protocol.SIP

1 (default) - Server is a SIP proxy/registrar

0 - If set to 0, and the server is confirmed to be a SIP server, then the value is assumed to be 1.

No

sip-interop.cfg

voIpProt.server.x.register

1 (default) - Calls can not be routed to an outbound proxy without registration.

0 - Calls can be routed to an outbound proxy without registration.

For more information, see Technical Bulletin 5844: SIP Server Fallback Enhancements on Polycom Phones.

No

sip-interop.cfg

voIpProt.server.x.registerRetry.baseTimeOut

The base time period to wait before a registration retry. Used in conjunction with voIpProt.server.x.registerRetry.maxTimeOut to determine how long to wait. The algorithm is defined in RFC 5626.

If both parameters voIpProt.server.x.registerRetry.baseTimeOut and reg.x.server.y.registerRetry.baseTimeOut are set, the value of reg.x.server.y.registerRetry.baseTimeOut takes precedence.

60 - (default)

10 - 120

No

sip-interop.cfg

voIpProt.server.x.registerRetry.maxTimeOut

The maximum time period to wait before a registration retry. Used in conjunction with voIpProt.server.x.registerRetry.maxTimeOut to determine how long to wait. The algorithm is defined in RFC 5626.

If both parameters voIpProt.server.x.registerRetry.maxTimeOut and reg.x.server.y.registerRetry.maxTimeOut are set, the value of reg.x.server.y.registerRetry.maxTimeOut takes precedence.

60 - (default)

10 - 1800

No

sip-interop.cfg

voIpProt.server.x.retryMaxCount

The number of retries that will be attempted before moving to the next available server.

3 (default)

0 to 20 - If set to 0, 3 is used.

No

sip-interop.cfg

voIpProt.server.x.retryTimeOut

0 (default) - Use standard RFC 3261 signaling retry behavior.

0 to 65535 - The amount of time (in milliseconds) to wait between retries.

No

sip-interop.cfg

voIpProt.server.x.specialInterop

Enables server-specific features for all registrations.

Standard (default)

VVX 101 = Standard, GENBAND, GENBAND-A2, ALU-CTS, DT

VVX 201 = Standard, GENBAND, GENBAND-A2, ALU-CTS, ocs2007r2, lync2010

All other phones = Standard, GENBAND, GENBAND-A2, ALU-CTS, DT, ocs2007r2, lync2010, lcs2005

No

sip-interop.cfg

voIpProt.server.x.subscribe.expires

The phone's requested subscription period in seconds after which the phone attempts to resubscribe at the beginning of the overlap period.

3600 - (default)

10 - 2147483647

No

sip-interop.cfg

voIpProt.server.x.subscribe.expires.overlap

The number of seconds before the expiration time returned by server x after which the phone attempts to resubscribe. If the server value is less than the configured overlap value, the phone tries to resubscribe at half the expiration time returned by the server.

60 - (default)

5 - 65535 seconds

No

sip-interop.cfg

voIpProt.server.x.transport

The transport method the phone uses to communicate with the SIP server.

Null or DNSnaptr (default) - If voIpProt.server.x.address is a hostname and voIpProt.server.x.port is 0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If voIpProt.server.x.address is an IP address, or a port is given, then UDP is used.

TCPpreferred - TCP is the preferred transport; UDP is used if TCP fails.

UDPOnly - Only UDP will be used.

TLS - If TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.

TCPOnly - Only TCP will be used.

No

sip-interop.cfg

voIpProt.server.x.useOutboundProxy

1 (default) - Enables to use the outbound proxy specified in voIpProt.SIP.outboundProxy.address for server x.

0 - Enables not to use the outbound proxy specified in voIpProt.SIP.outboundProxy.address for server x.

No

sip-interop.cfg

voIpProt.SIP.acd.signalingMethod

0 (default) - The ‘SIP-B' signaling is supported. (This is the older ACD functionality.)

1 - The feature synchronization signaling is supported. (This is the new ACD functionality.)

Yes

sip-interop.cfg

voIpProt.SIP.acd.signalingMethod

0 (default) - The ‘SIP-B' signaling is supported. (This is the older ACD functionality.)

1 - The feature synchronization signaling is supported. (This is the new ACD functionality.)

Yes

sip-interop.cfg

voIpProt.SIP.alertInfo.x.class

Alert-Info fields from INVITE requests are compared as many of these parameters as are specified (x=1, 2, ..., N) and if a match is found, the behavior described in the corresponding ring class is applied.

default (default)

See the list of ring classes in Ringtone Parameters.

No

sip-interop.cfg

voIpProt.SIP.alertInfo.x.class

Alert-Info fields from INVITE requests are compared as many of these parameters as are specified (x=1, 2, ..., N) and if a match is found, the behavior described in the corresponding ring class is applied.

default (default)

No

sip-interop.cfg

voIpProt.SIP.alertInfo.x.class

Alert-Info fields from INVITE requests are compared as many of these parameters as are specified (x=1, 2, ..., N) and if a match is found, the behavior described in the corresponding ring class is applied.

default (default)

See the list of ring classes in Ringtone Parameters.

No

sip-interop.cfg

voIpProt.SIP.alertInfo.x.value

Specify a ringtone for single registered line using a string to match the Alert-Info header in the incoming INVITE.

NULL (default)

No

sip-interop.cfg

voIpProt.SIP.alertInfo.x.value

Specify a ringtone for single registered line using a string to match the Alert-Info header in the incoming INVITE.

NULL (default)

No

sip-interop.cfg

voIpProt.SIP.alertInfo.x.value

Specify a ringtone for single registered line using a string to match the Alert-Info header in the incoming INVITE.

NULL (default)

No

sip-interop.cfg

voIpProt.SIP.allowTransferOnProceeding

1 (default) - Transfer during the proceeding state of a consultation call is enabled.

0 - Transfer during the proceeding state of a consultation call is enabled

2 - Phones will accept an INVITE with replaces for a dialog in early state. This is needed when using transfer on proceeding with a proxying call server such as openSIPS, reSIProcate or SipXecs.

No

sip-interop.cfg

voipProt.SIP.anat.enabled

Enables or disables Alternative Network Address Types (ANAT).

0 (default) - ANAT is disabled.

1 - ANAT is enabled.

No

sip-interop.cfg

voIpProt.SIP.authOptimizedInFailover

0 (default) - The first new SIP request is sent to the server with the highest priority in the server list when failover occurs.

1 - The first new SIP request is sent to the server that sent the proxy authentication request when failover occurs.

No

features.cfg

voIpProt.SIP.callee.SourcePreference

Set priority order to display the callee's identity for outgoing calls.

Null (default)

Supported Headers Default Order: P-Asserted-Identity,Remote-Party-ID,From

String

features.cfg

features.cfg

voIpProt.SIP.Caller.SourcePreference

Set priority order to display the caller's identity for incoming calls.

Null (default)

Supported Headers Default Order: P-Asserted-Identity,Remote-Party-ID,From

String

features.cfg

sip-interop.cfg

voIpProt.SIP.callinfo.precedence.overAlertinfo

0 (default) - Give priority to call-info header with answer-after string over alert-info feature is disabled.

1 - Give priority to call-info header with answer-after string over alert-info feature is enabled.

No

sip-interop.cfg

voIpProt.SIP.callinfo.precedence.overAlertinfo

0 (default) - The alert-info is given priority over call-info header.

1 - The call-info header with answer-after string is given priority over alert-info header.

No

sip-interop.cfg

voIpProt.SIP.CID.request.sourceSipMessage

Specify which header in the SIP request to retrieve remote party caller ID from. You can use:
  • voIpProt.SIP.callee.sourcePreference
  • voIpProt.SIP.caller.sourcePreference
  • voIpProt.SIP.CID.sourcePreference

UPDATE takes precedence over the value of this parameter.

NULL (default) - Remote party caller ID information from INVITE is used.

INVITE

PRACK

ACK

This parameter does not apply to shared lines.

No

sip-interop.cfg

voIpProt.SIP.CID.response.sourceSipMessage

Specify which header in the SIP request to retrieve remote party caller ID from. You can use:
  • voIpProt.SIP.callee.sourcePreference
  • voIpProt.SIP.caller.sourcePreference
  • voIpProt.SIP.CID.sourcePreference

NULL (default) - The remote party caller ID information from the last SIP response is used.

100, 180, 183, 200

This parameter does not apply to shared lines.

No

sip-interop.cfg

voIpProt.SIP.CID.sourcePreference

Specify the priority order for the sources of caller ID information. The headers can be in any order.

Null (default) - Caller ID information comes from P-Asserted-Identity, Remote-Party-ID, and From in that order.

From,P-Asserted-Identity, Remote-Party-ID

P-Asserted-Identity,From,Remote-Party-ID

Supported Headers Default Order: P-Asserted-Identity,Remote-Party-ID,From

Note: By default callee and caller will take identity order from voIpProt.SIP.CID.sourcePreference.

If voIpProt.SIP.Caller.SourcePreference or voIpProt.SIP.Callee.SourcePreference are configured then the order set by voIpProt.SIP.CID.sourcePreference is ignored.

No

sip-interop.cfg

voIpProt.SIP.compliance.RFC3261.validate.contentLanguage

1 (default) - Validation of the SIP header content language is enabled.

0 - Validation of the SIP header content language is disabled

No

sip-interop.cfg

voIpProt.SIP.compliance.RFC3261.validate.contentLength

1 (default) - Validation of the SIP header content length is enabled.

0 - Validation of the SIP header content length is disabled

No

sip-interop.cfg

voIpProt.SIP.compliance.RFC3261.validate.uriScheme

1 (default) - Validation of the SIP header URI scheme is enabled.

0 - Validation of the SIP header URI scheme is disabled

No

sip-interop.cfg

voIpProt.SIP.conference.address

Null (default) - Conferences are set up on the phone locally.

String 128 max characters - Enter a conference address. Conferences are set up by the server using the conferencing agent specified by this address. Acceptable values depend on the conferencing server implementation policy.

No

sip-interop.cfg

voIpProt.SIP.conference.parallelRefer

0(deafult) - A parallel REFER is not sent to the call server.

1 - A parallel REFER is not sent to the call server.

Note: This parameter must be set for Siemens OpenScape Centralized Conferencing.

No

sip-interop.cfg

voIpProt.SIP.connectionReuse.useAlias

0 (default) - The alias parameter is not added to the via header

1 - The phone uses the connection reuse draft which introduces "alias".

No

sip-interop.cfg

voIpProt.SIP.dialog.strictXLineID

0 (default) - The phone will not look for x-line-id (call appearance index) in a SIP INVITE message.

1 - The phone will look for x-line-id (call appearance index) in a SIP INVITE message

No

sip-interop.cfg

voIpProt.SIP.dialog.usePvalue

0 (default) - Phone uses a pval field name in Dialog.

1 - Phone uses a pvalue field name in Dialog.

No

sip-interop.cfg

voIpProt.SIP.dialog.useSDP

0 (default) - A new dialog event package draft is used (no SDP in dialog body).

1 - Use this setting to send SDP in the dialog body for backwards compatibility

No

sip-interop.cfg

voIpProt.SIP.dtmfViaSignaling.rfc2976

Enable or disable DTMF relays for active SIP calls. Not supported for H.323 calls.

0 (default) - DTMF digit information is not sent

1 - DTMF digit information is sent in RFC2976 SIP INFO packets during a call.

Yes

sip-interop.cfg

voIpProt.SIP.dtmfViaSignaling.rfc2976.nonLegacyEncoding

Controls the behavior of the Star and Pound keys used for DTMF relays for active SIP calls. Not supported for H.323 calls.

0 (default) - The phone sends 10 when the Star key (*) is pressed and 11 when the Pound key (#) is pressed.

1 - The phone sends an asterisk (*) when the Star key is pressed and a hashtag (#) when the Pound key is pressed.

Yes

sip-basic.cfg

voIpProt.SIP.enable

A flag to determine if the SIP protocol is used for call routing, dial plan, DTMF, and URL dialing.

1 (default) - The SIP protocol is used.

0 - The SIP protocol is not used.

Yes

sip-interop.cfg

voIpProt.SIP.failoverOn503Response

A flag to determine whether or not to trigger a failover if the phone receives a 503 response. You must use a registration expiry of 66 seconds or greater for failover with a 503 response to work properly. This rule applies both to the phone configuration ( reg.x.server.y.expires and voIpProt.server.x.expires ) as well as the 200 OK register response from the server.

1 (default)

0

No

sip-interop.cfg

voIpProt.SIP.header.diversion.enable

0 (default) - If set to 0, the diversion header is not displayed.

1 - If set to 1, the diversion header is displayed if received.

Yes

sip-interop.cfg

voIpProt.SIP.header.diversion.list.useFirst

1 (default) - If set to 1, the first diversion header is displayed.

0 - If set to 0, the last diversion header is displayed.

Yes

sip-interop.cfg

voIpProt.SIP.header.pEarlyMedia.support

0 (Default) - The p-early-media header is not supported by the caller phone.

1 - The p-early-media header is supported by the caller phone.

sip-interop.cfg

voIpProt.SIP.header.warning.codes.accept

Specify a list of accepted warning codes.

Null (default) - All codes are accepted only codes between 300 and 399 are supported.

comma separated list

No

sip-interop.cfg

voIpProt.SIP.header.warning.codes.accept

Specify a list of accepted warning codes.

Null (default) - All codes are accepted. Only codes between 300 and 399 are supported.

For example, if you want to accept only codes 325 to 330: voIpProt.SIP.header.warning.codes.accept=325,326,327,328,329,330

No

sip-interop.cfg

voIpProt.SIP.header.warning.enable

0 (default) - The warning header is not displayed.

1 - The warning header is displayed if received.

No

sip-interop.cfg

voIpProt.SIP.IM.autoAnswerDelay

The time interval from receipt of the instant message invitation to automatically accepting the invitation.

10 (default)

0 to 40

No

sip-interop.cfg

voIpProt.SIP.IMS.enable

This parameter applies to all registered or unregistered SIP lines on the phone.

0 (Default) - The phone does not support IMS features introduced in UC Software 5.5.0.

1 - The phone supports IMS features introduced in UC Software 5.5.0.

sip-interop.cfg

voIpProt.SIP.intercom.alertInfo

The string you want to use in the Alert-Info header. You can use the following characters: '@', '-' ,'_' , '.' .

If you use any other characters, NULL, or empty spaces, the call is sent as normal without the Alert-Info header.

Intercom (default)

Alpha - Numeric string

No

sip-interop.cfg

voIpProt.SIP.keepalive.sessionTimers

0 (default) - The session timer is disabled.

1 - The session timer is enabled.

No

sip-interop.cfg

voIpProt.SIP.lineSeize.retries

Controls the number of times the phone will retry a notify when attempting to seize a line (BLA).

10 (default)

3 to 10

No

sip-interop.cfg

voIpProt.SIP.local.port

The local port for sending and receiving SIP signaling packets.

5060 - The value is used for the local port but is not advertised in the SIP signaling.

0 to 65535 - If set to 0,the 5060 value is used for the local port but is not advertised in the SIP signaling. For other values, that value is used for the local port and it is advertised in the SIP signaling

Yes

sip-interop.cfg

voIpProt.SIP.looseContact

0 (default) - The port parameter is added to the contact header in TLS case.

1 - The port parameter is not added to the contact header or SIP messages.

No

sip-interop.cfg

voIpProt.SIP.ms-forking

This parameter is applies when installing Microsoft Live Communications Server.

0 (default) - Support for MS-forking is disabled.

1 - Support for MS-forking is enabled.

Note: If any endpoint registered to the same account has MS-forking disabled, all other endpoints default back to non-forking mode. Windows Messenger does not use MS-forking so be aware of this behavior if one of the endpoints is using Windows Messenger.

No

sip-interop.cfg

voIpProt.SIP.musicOnHold.uri

A URI that provides the media stream to play for the remote party on hold. This parameter is used if reg.x.musicOnHold.uri is Null.

Null (default)

SIP URI

No

sip-interop.cfg

voIpProt.SIP.newCallOnUnRegister

1 (default) - The phone generate new Call-ID and From tag during re-registration.

0 - The phone does not generate new Call-ID and From tag during re-registration.

No

sip-basic.cfg

voIpProt.SIP.outboundProxy.address

The IP address or hostname of the SIP server to which the phone sends all requests.

Null (default)

IP address or hostname

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.failOver.failBack.mode

Duration (default) - The phone tries the primary server again after the time specified by reg.x.outboundProxy.failOver.failBack.timeout expires.

newRequests - All new requests are forwarded first to the primary server regardless of the last used server.

DNSTTL - The phone tries the primary server again after a timeout equal to the DNS TTL configured for the server that the phone is registered to.

registration - The phone tries the primary server again when the registration renewal signaling begins.

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.failOver.failBack.timeout

The time to wait (in seconds) before failback occurs (overrides voIpProt.server.x.failOver.failBack.timeout ).

3600 (default) -If the fail back mode is set to Duration, the phone waits this long after connecting to the current working server before selecting the primary server again.

0, 60 to 65535 -If set to 0, the phone will not fail-back until a fail-over event occurs with the current server.

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.failOver.failRegistrationOn

1 (default) - When set to 1, and the reRegisterOn parameter is enabled, the phone will silently invalidate an existing registration (if it exists), at the point of failing over.

0 - When set to 0, and the reRegisterOn parameter is enabled, existing registrations will remain active. This means that the phone will attempt failback without first attempting to register with the primary server to determine if it has recovered.

Note: voIpProt.SIP.outboundProxy.failOver.reRegisterOn must be enabled.

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.failOver.onlySignalWithRegistered

1 (default) - No signaling is accepted from or sent to a server that has failed until failback is attempted or failover occurs.

0 - signaling will be accepted from and sent to a server that has failed (even though failback hasn't been attempted or failover hasn't occurred). This parameter overrides voIpProt.server.x.failOver.onlySignalWithRegistered .

Note: reRegisterOn and failRegistrationOn parameters must be enabled

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.failOver.reRegisterOn

This parameter overrides the voIpProt.server.x.failOver.reRegisterOn .

0 (default) - The phone won't attempt to register with the secondary server, since the phone will assume that the primary and secondary servers share registration information.

1 - The phone will attempt to register with the secondary server. If the registration succeeds signaling will proceed with the secondary server.

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.port

The port of the SIP server to which the phone sends all requests.

0 (default)

0 to 65535

No

sip-interop.cfg

voIpProt.SIP.outboundProxy.transport

DNSnaptr (default) - If reg.x.outboundProxy.address is a hostname and reg.x.outboundProxy.port is 0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If reg.x.outboundProxy.address is an IP address, or a port is given, then UDP is used.

TCPpreferred - TCP is the preferred transport, UDP is used if TCP fails.

UDPOnly - Only UDP will be used.

TLS - If TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.

TCPOnly - Only TCP will be used.

No

sip-interop.cfg

voIpProt.SIP.pingInterval

The number in seconds to send PING message.

0 (default) - This feature is disabled.

0 to 3600 - This feature is enabled.

No

sip-interop.cfg

voIpProt.SIP.pingMethod

The ping method to be used.

PING (default)

OPTIONS

No

sip-interop.cfg

voIpProt.SIP.presence.nortelShortMode

This parameter is required when using the Presense feature with an Avaya or GENBAND server.

0 (default)

1 - Different headers are sent in SUBSCRIBE when used feature with an Avaya or GENBAND server. Support is indicated by adding a header Accept-Encoding: x-nortel-short. A PUBLISH is sent to indicate the status of the phone.

Yes

features.cfg

voIpProt.SIP.regevent

0 (default) - The phone is not subscribed to registration state change notifications for all phone lines.

1 - The phone is subscribed to registration state change notifications for all phone lines.

This parameter is overridden by the per-phone parameter reg.x.regevent.

sip-interop.cfg

voIpProt.SIP.rejectNDUBInvite

Specify whether or not the phone accepts a call for all registrations in case of a Network Determined User Busy (NDUB) event advertised by the SIP server.

0 (Default) - If an NDUB event occurs, the phone does not reject the call for all line registrations.

1 - If an NDUB event occurs, the phone rejects the call with a 603 Decline response code for all line registrations.

sip-interop.cfg

voIpProt.SIP.renewSubscribeOnTLSRefresh

1 (default) - For an as-feature-event, the SUBSCRIBE message is sent along with the RE-REGISTER when Transport Layer Security (TLS) breaks.

0 - The SUBSCRIBE and RE-REGISTER messages is sent at different times.

No

sip-interop.cfg

voIpProt.SIP.rport

0 (default) – The phone does not insert the rport parameter into the Via header of its requests.

1 – The phone inserts the rport parameter, as defined by RFC 3581, into the Via header of its requests.

No

sip-interop.cfg

voIpProt.SIP.requestURI.E164.addGlobalPrefix

0 (default) - ‘+' global prefix is not added to the E.164 user parts in sip: URIs.

1 - ‘+' global prefix is added to the E.164 user parts in sip: URIs.

No

sip-interop.cfg

voIpProt.SIP.requestValidation.digest.realm

Determines the string used for Realm.

PolycomSPIP (default)

string

Yes

sip-interop.cfg

voIpProt.SIP.requestValidation.x.method

Null (default) - no validation is made.

Source - ensure request is received from an IP address of a server belonging to the set of target registration servers.

digest: challenge requests with digest authentication using the local credentials for the associated registration (line).

both or all: apply both of the above methods.

Yes

sip-interop.cfg

voIpProt.SIP.requestValidation.x.method

Null (default) - no validation is made.

Source - ensure request is received from an IP address of a server belonging to the set of target registration servers.

digest: challenge requests with digest authentication using the local credentials for the associated registration (line).

both or all: apply both of the above methods.

Yes

sip-interop.cfg

voIpProt.SIP.requestValidation.x.request

Sets the name of the method for which validation will be applied.

Null (default)

INVITE, ACK, BYE, REGISTER, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, REFER, PRACK, UPDATE

Note: Intensive request validation may have a negative performance impact due to the additional signaling required in some cases.

Yes

sip-interop.cfg

voIpProt.SIP.requestValidation.x.request

Sets the name of the method for which validation will be applied.

Null (default)

INVITE, ACK , BYE, REGISTER, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, REFER, PRACK, UPDATE

Note: Intensive request validation may have a negative performance impact due to the additional signaling required in some cases.

Yes

sip-interop.cfg

voIpProt.SIP.requestValidation.x.request.y.event

Determines which events specified with the Event header should be validated; only applicable when voIpProt.SIP.requestValidation.x.request is set to SUBSCRIBE or NOTIFY .

Null (default) - all events will be validated.

A valid string - specified event will be validated.

Yes

sip-interop.cfg

voIpProt.SIP.requestValidation.x.request.y.event

Determines which events specified with the Event header should be validated; only applicable when voIpProt.SIP.requestValidation.x.request is set to SUBSCRIBE or NOTIFY .

Null (default) - all events will be validated.

A valid string - specified event will be validated.

Yes

sip-interop.cfg voIpProt.SIP.RFC3261TimerI

0 (default) - Timer I for reliable transport will be fired at five seconds. This parameter does not cause any change for unreliable transport.

1 - Timer I for reliable transport will be fired at zero seconds.

No

sip-interop.cfg

voIpProt.SIP.sendCompactHdrs

0 (default) - SIP header names generated by the phone use the long form, for example From .

1 - SIP header names generated by the phone use the short form, for example f .

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.callRecording

0 (default) - The BroadSoft BroadWorks v20 call recording feature for multiple phones is disabled.

1 - The BroadSoft BroadWorks v20 call recording feature for multiple phones is enabled.

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.cf

0 (default) - The server-based call forwarding is not enabled.

1 - The server-based call forwarding is enabled.

Yes

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.cf

0 (default) - Disable server-based call forwarding.

1 - Enable server-based call forwarding.

Yes

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.dnd

0 (default) - Disable server-based DND.

1 - Server-based DND is enabled. Server and local phone DND are synchronized.

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.localProcessing.cf

This parameter depends on the value of voIpProt.SIP.serverFeatureControl.cf .

1 (default) - If set to 1 and voIpProt.SIP.serverFeatureControl.cf is set to 1, the phone and the server perform call forwarding.

0 - If set to 0 and voIpProt.SIP.serverFeatureControl.cf is set to 1, call forwarding is performed on the server side only, and the phone does not perform local call forwarding.

If both voIpProt.SIP.serverFeatureControl.localProcessing.cf and voIpProt.SIP.serverFeatureControl.cf are set to 0, the phone performs local call forwarding and the localProcessing parameter is not used.

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.localProcessing.cf

1 (default) - Allows to use the value for voIpProt.SIP.serverFeatureControl.cf.

0 - Does not use the value for

This parameter depends on the value of voIpProt.SIP.serverFeatureControl.cf .

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.localProcessing.dnd

This parameter depends on the value of voIpProt.SIP.serverFeatureControl.dnd .

If set to 1 (default) and voIpProt.SIP.serverFeatureControl.dnd is set to 1, the phone and the server perform DND.

If set to 0 and voIpProt.SIP.serverFeatureControl.dnd is set to 1, DND is performed on the server-side only, and the phone does not perform local DND.

If both voIpProt.SIP.serverFeatureControl.localProcessing.dnd and voIpProt.SIP.serverFeatureControl.dnd are set to 0, the phone performs local DND and the localProcessing parameter is not used.

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.missedCalls

0 (default) - Server-based missed calls is not enabled.

1 - Server-based missed calls is enabled. The call server has control of missed calls.

Yes

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.securityClassification

0 (default) - The visual security classification feature for all lines on a phone is disabled.

1 - The visual security classification feature for all lines on a phone is enabled.

No

sip-interop.cfg

voIpProt.SIP.serverFeatureControl.securityClassification

0 (default) - The visual security classification feature for all lines on a phone is disabled.

1 - The visual security classification feature for all lines on a phone is enabled.

Yes

site.cfg

v oIpProt.SIP.specialEvent.checkSync.alwaysReboot

0 (default) - The phone will only reboot if necessary. Many configuration parameter changes can be applied dynamically without the need for a reboot.

1 - The phone always reboot when a NOTIFY message is received from the server with event equal to check-sync even if there has not been a change to software or configuration.

No

site.cfg

voIpProt.SIP.specialEvent.checkSync.downloadCallList

0 (default) - The phone does not download the call list for the logged-in user when a check sync event's NOTIFY message is received from the server.

1 - The phone downloads the call list for the logged-in user when a check sync event's NOTIFY message is received from the server.

No

site.cfg

voIpProt.SIP.specialEvent.checkSync.downloadCallList

0 (default) - The phone does not download the call list for the user after receiving a checksync event in the NOTIFY.

1 - The phone downloads the call list for the user after receiving a checksync event in the NOTIFY.

No

site.cfg

voIpProt.SIP.specialEvent.checkSync.downloadDirectory

0 (default) - The phone downloads updated directory files after receiving a checksync NOTIFY message.

1 - The phone downloads the updated directory files along with any software and configuration updates after receiving a checksync NOTIFY message. The files are downloaded when the phone restarts, reboots, or when the phone downloads any software or configuration updates.

Note: The parameter hotelingMode.type set to 2 or 3 overrides this parameter.

No

sip-interop.cfg

voIpProt.SIP.specialEvent.lineSeize.nonStandard

Controls the response for a line-seize event SUBSCRIBE.

1 (default) - This speeds up the processing of the response for line-seize event.

0 - This will process the response for the line seize event normally

Yes

sip-interop.cfg

voIpProt.SIP.strictLineSeize

0 (default) - Dial prompt is provided immediately when you attempt to seize a shared line without waiting for a successful OK from the call server.

1 - The phone is forced to wait for a 200 OK response when receiving a TRYING notify.

No

sip-interop.cfg

voIpProt.SIP.strictReplacesHeader

This parameter applies only to directed call pick-up attempts initiated against monitored BLF resources.

1 (default) - The phone requires call-id, to-tag, and from-tag to perform a directed call-pickup when call.directedCallPickupMethod is configured as native.

0 - Call pick-up requires a call id only.

No

sip-interop.cfg

voIpProt.SIP.strictReplacesHeader

This parameter applies only to directed call pick-up attempts initiated against monitored BLF resources.

1 (default) - The phone requires call-id, to-tag, and from-tag to perform a directed call-pickup when call.directedCallPickupMethod is configured as native.

0 - Call pick-up requires a call id only.

No

sip-interop.cfg

voIpProt.SIP.strictUserValidation

0 (default) - The phone is forced to match the user portion of signaling exactly.

1 - The phone will use the first registration if the user part does not match any registration.

No

sip-interop.cfg

voIpProt.SIP. supportFor100rel

1 (default) - The phone advertises support for reliable provisional responses in its offers and responses.

0 - The phone will not offer 100rel and will reject offers requiring 100rel.

No

reg-basic.cfg

voIpProt.SIP.supportFor199

Determine support for the 199 response code. For details on the 199 response code, see RFC 6228.

0 (Default) - The phone does not support the 199 response code.

1- The phone supports the 199 response code.

sip-interop.cfg

voIpProt.SIP.tcpFastFailover

0 (default) - A full 32 second RFC compliant timeout is used.

1 - A failover occurs based on the values of reg.x.server.y.retryMaxCount and voIpProt.server.x.retryTimeOut.

No

sip-interop.cfg

voIpProt.SIP.tlsDsk.enable

0 (default) - TLS DSK is disabled.

1 - TLS DSK is enabled.

No

sip-interop.cfg

voIpProt.SIP.turnOffNonSecureTransport

0 (default) - Stop listening to port 5060 when using AS-SIP feature is disabled.

1 - Stop listening to port 5060 when using AS-SIP feature is enabled.

Yes

sip-interop.cfg

voIpProt.SIP.use486forReject

0 (default) - The phone will not transmit 486 response.

1 - The phone will not transmit 486 response.

No

sip-interop.cfg

voIpProt.SIP.useContactInReferTo

0 (default) - The “To URI” is used in the REFER.

1 - The “Contact URI” is used in the REFER.

No

sip-interop.cfg

voIpProt.SIP.useLocalTargetUriforLegacyPickup

1 (default) - The target URI in BLF signaling uses the complete address as provided in the XML dialog document.

0 - Only the user portion of the target URI in the XML dialog document is used and the current registrar's domain is appended to create the address for pickup or retrieval.

No

sip-interop.cfg

voIpProt.SIP.useRFC2543hold

0 (default) - SDP media direction parameters (such as a=sendonly) per RFC 3264 when initiating a call.

1 - the obsolete c=0.0.0.0 RFC2543 technique is used when initiating a call.

No

sip-interop.cfg

voIpProt.SIP.useRFC2543hold

0 (default) - SDP media direction parameters (such as a=sendonly) per RFC 3264 when initiating a call.

1 - the obsolete c=0.0.0.0 RFC2543 technique is used when initiating a call.

No

sip-interop.cfg

voIpProt.SIP.useRFC3264HoldOnly

0 (default) - When set to 0, and no media direction is specified, the phone enters backward compatibility mode when negotiating SDP and responds using the c=0.0.0.0 RFC 2543 signaling method.

1 - When set to 1, and no media direction is specified, the phone uses sendrecv compliant with RFC 3264 when negotiating SDP and generates responses containing RFC 3264-compliant media attributes for calls placed on and off hold by either end.

Note: voIpProt.SIP.useSendonlyHold applies only to calls on phones that originate the hold.

No

sip-interop.cfg

voIpProt.SIP.useSendonlyHold

1 (default) - The phone will send a reinvite with a stream mode parameter of “sendonly” when a call is put on hold.

0 - The phone will send a reinvite with a stream mode parameter of “inactive” when a call is put on hold

Note: The phone will ignore the value of this parameter if set to 1 when the parameter voIpProt.SIP.useRFC2543hold is also set to 1 (default is 0).

No