Max Forward
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Specifies SIP Max Forward value.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Max_Forward ua="na">70</Max_Forward>
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In the phone web page, enter an appropriate value.
Value range: 1 to 255
Default: 70
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Max Redirection
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Specifies number of times an invite can be redirected to avoid an infinite loop.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Max_Redirection ua="na">5</Max_Redirection>
-
In the phone web page, enter an appropriate value.
Default: 5
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Max Auth
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Specifies the maximum number of times (from 0 to 255) a request can be challenged.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Max_Auth ua="na">2</Max_Auth>
-
In the phone web page, enter an appropriate value.
Allowed value: 0 to 255
Default: 2
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SIP User Agent Name
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Used in outbound requests.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <SIP_User_Agent_Name ua="na">$VERSION</SIP_User_Agent_Name>
-
In the phone web page, enter an appropriate name.
Default: $VERSION
If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed
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SIP Server Name
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Server header used in responses to inbound responses.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <SIP_Server_Name ua="na">$VERSION</SIP_Server_Name>
-
In the phone web page, enter an appropriate name.
Default: $VERSION
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SIP Reg User Agent Name
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User-Agent name to be used in a REGISTER request. If this is not specified, the SIP User Agent Name is also used for the REGISTER
request.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format:
<SIP_Reg_User_Agent_Name ua="na">agent name</SIP_Reg_User_Agent_Name>
-
In the phone web page, enter an appropriate name.
Default: Blank
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SIP Accept Language
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Accept-Language header used.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <SIP_Accept_Language ua="na">en</SIP_Accept_Language>
-
In the phone web page, enter an appropriate language.
There is no default. If empty, the header is not included.
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DTMF Relay MIME Type
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MIME Type used in a SIP INFO message to signal a DTMF event. This field must match that of the Service Provider.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <DTMF_Relay_MIME_Type ua="na">application/dtmf-relay</DTMF_Relay_MIME_Type>
-
In the phone web page, enter an appropriate MIME type.
Default: application/dtmf-relay
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Hook Flash MIME Type
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MIME Type used in a SIPINFO message to signal a hook flash event.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Hook_Flash_MIME_Type ua="na">application/hook-flash</Hook_Flash_MIME_Type>
-
In the phone web page, enter an appropriate MIME type for a SIPINFO message.
Default:
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Remove Last Reg
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Enables you to remove the last registration before registering a new one if the value is different.
Set to Yes to remove the last registration.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Remove_Last_Reg ua="na">No</Remove_Last_Reg>
-
In the phone web page, Select Yes or No.
Allowed values: Yes or No
Default: No
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Use Compact Header
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If set to yes, the phone uses compact SIP headers in outbound SIP messages. If inbound SIP requests contain normal headers,
the phone substitutes incoming headers with compact headers. If set to no, the phones use normal SIP headers. If inbound SIP
requests contain compact headers, the phones reuse the same compact headers when generating the response, regardless of this
setting.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Use_Compact_Header ua="na">No</Use_Compact_Header>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
Default: No
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Escape Display Name
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Enables you to keep the Display Name private.
Set to Yes if you want the IP phone to enclose the string (configured in the Display Name) in a pair of double quotes for
outbound SIP messages.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Escape_Display_Name ua="na">No</Escape_Display_Name>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
Default: Yes.
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Talk Package
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Enables support for the BroadSoft Talk Package that lets users answer or resume a call by clicking a button in an external
application.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Talk_Package ua="na">No</Talk_Package>
-
In the phone web page, select Yes to enable the Talk Package.
Allowed values: Yes or No
Default: No
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Hold Package
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Enables support for the BroadSoft Hold Package, which lets users place a call on hold by clicking a button in an external
application.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Hold_Package ua="na">No</Hold_Package>
-
In the phone web page, select Yes to enable support for the Hold Package.
Allowed values: Yes or No
Default: No
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Conference Package
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Enables support for the BroadSoft Conference Package that enables users to start a conference call by clicking a button in
an external application.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Conference_Package ua="na">No</Conference_Package>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
Default: No
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RFC 2543 Call Hold
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If set to yes, unit includes c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to
no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either
case.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <RFC_2543_Call_Hold ua="na">Yes</RFC_2543_Call_Hold>
-
In the phone web page, Yes or No.
Allowed values: Yes or No
Default: Yes
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Random REG CID on Reboot
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If set to yes, the phone uses a different random call-ID for registration after the next software reboot. If set to no, the
Cisco IP phone tries to use the same call-ID for registration after the next software reboot. The Cisco IP phone always uses
a new random Call-ID for registration after a power-cycle, regardless of this setting.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Random_REG_CID_on_Reboot ua="na">No</Random_REG_CID_on_Reboot>
-
In the phone web page, select Yes or No.
Default: No.
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SIP TCP Port Min
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Specifies the lowest TCP port number that can be used for SIP sessions.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <SIP_TCP_Port_Min ua="na">5060</SIP_TCP_Port_Min>
-
In the phone web page, enter an appropriate value.
Default: 5060
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SIP TCP Port Max
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Specifies the highest TCP port number that can be used for SIP sessions.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <SIP_TCP_Port_Max ua="na">5080</SIP_TCP_Port_Max>
-
In the phone web page, enter an appropriate value.
Default: 5080
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Caller ID Header
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Provides the option to take the caller ID from PAID-RPID-FROM, PAID-FROM, RPID-PAID-FROM, RPID-FROM, or FROM header.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Caller_ID_Header ua="na">PAID-RPID-FROM</Caller_ID_Header>
-
In the phone web page, select an option.
Allowed values: PAID-RPID-FROM, AID-FROM, RPID-PAID-FROM, RPID-FROM, and FROM
Default: PAID-RPID-FROM
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Hold Target Before Refer
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Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended
call transfer (where the transfer target has answered).
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Hold_Target_Before_Refer ua="na">No</Hold_Target_Before_Refer>
-
In the phone web page, select Yes or No.
Default: No
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Dialog SDP Enable
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When enabled and the Notify message body is too big causing fragmentation, the Notify message xml dialog is simplified; Session
Description Protocol (SDP) is not included in the dialog xml content.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Dialog_SDP_Enable ua="na">No</Dialog_SDP_Enable>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
Default: No
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Keep Referee When Refer Failed
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If set to yes, it configures the phone to immediately handle NOTIFY sipfrag messages.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Keep_Referee_When_Refer_Failed ua="na">No</Keep_Referee_When_Refer_Failed>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
Default: No
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Display Diversion Info
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Display the Diversion info included in SIP message on LCD or not.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Display_Diversion_Info ua="na">No</Display_Diversion_Info>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
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Display Anonymous From Header
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Show the caller ID from the SIP INVITE message “From” header when set to Yes, even if the call is an anonymous call. When
the parameter is set to no, the phone displays "Anonymous Caller" as the caller ID.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Display_Anonymous_From_Header ua="na">No</Display_Anonymous_From_Header>
-
In the phone web page, select Yes or No.
Allowed values: Yes or No
Default: No
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Sip Accept Encoding
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Supports the content-encoding gzip feature.
If gzip is selected, the SIP message header contains the string “Accept-Encoding: gzip”, and the phone is able to process
the SIP message body, which is encoded with the gzip format.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Sip_Accept_Encoding ua="na">none</Sip_Accept_Encoding>
-
In the phone web page, enter an appropriate MIME type for a SIPINFO message.
Allowed values: none and gzip
Default: none
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SIP IP Preference
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Sets if the phone uses IPv4 or IPv6.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <SIP_IP_Preference ua="na">IPv4</SIP_IP_Preference>
-
In the phone web page, select IPv4 or IPv6.
Allowed values: IPv4/IPv6
Default: IPv4.
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Disable Local Name To Header
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Controls the display name in “Directory”, “Call History”, and in the “To” header during an outgoing call.
Perform one of the following.
-
In the phone configuration file with XML(cfg.xml), enter a string in this format: <Disable_Local_Name_To_Header ua="na">No</Disable_Local_Name_To_Header>
-
In the phone web page, select Yes to disable the display name.
Allowed values: Yes/No
Default: No
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