Max Forward

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>

• In the phone web page, enter an appropriate value.

Allowed values: 1 to 255

Default: 70

Max Redirection

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

Max Auth

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 values: 0 to 255

Default: 2

SIP User Agent Name

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

SIP Server Name

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

SIP Reg User Agent Name

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

SIP Accept Language

Accept-Language the 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-US</SIP_Accept_Language>

• In the phone web page, enter an appropriate language.

There is no default. If empty, the header is not included.

DTMF Relay MIME Type

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

Hook Flash MIME Type

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: application/hook-flash

Remove Last Reg

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 | No

Default: No

Use Compact Header

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 | No

Default: No

Escape Display Name

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 | No

Default: No

Talk Package

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 | No

Default: No

Hold Package

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 | No

Default: No

Conference Package

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 | No

Default: No

RFC 2543 Call Hold

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 | No

Default: Yes

Random REG CID on Reboot

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.

Allowed values: Yes | No

Default: No

SIP TCP Port Min

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

SIP TCP Port Max

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

Caller ID Header

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 | FROM

Default: PAID-RPID-FROM

Hold Target Before Refer

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.

Allowed values: Yes | No

Default: No

Dialog SDP Enable

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 | No

Default: No

Keep Referee When Refer Failed

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 | No

Default: No

Display Diversion Info

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 | No

Default: No

Display Anonymous From Header

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 | No

Default: No

Sip Accept Encoding

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

SIP IP Preference

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

Disable Local Name To Header

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

User Preferred OffHook Timer

Time (in seconds) the phone waits after going off-hook before initiating dialing. When configured, this setting takes precedent over the P parameter in the dial plan.

Perform one of the following.

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <User_Preferred_OffHook_Timer ua="na">20</User_Preferred_OffHook_Timer>

• In the phone web page, enter the preferred value arranging from 0 to 30.

Allowed values: 0-30

Default: Empty

Enables the dialog-based shared line, so that the phones in the shared line can subscribe to the dialog event package.

Perform one of the following.

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Share_Line_Event_Package_Type ua="na">Dialog</Share_Line_Event_Package_Type>

• In the phone web page, select Dialog.

Allowed values: Call-Info | Dialog

Default: Call-Info

SIP T1

RFC 3261 T1 value (RTT estimate) that can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_T1 ua="na">.5</SIP_T1>

• In the phone web page, enter an appropriate value.

Default: 0.5 seconds

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses) that can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_T2 ua="na">4</SIP_T2>

• In the phone web page, enter an appropriate value.

Default: 4 seconds

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_T4 ua="na">5</SIP_T4>

• In the phone web page, enter an appropriate value.

Default: 5 seconds.

SIP Timer B

INVITE time-out value, which can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Timer_B ua="na">16</SIP_Timer_B>

• In the phone web page, enter an appropriate value.

Default: 16 seconds.

SIP Timer F

Non-INVITE time-out value, which can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Timer_F ua="na">16</SIP_Timer_F>

• In the phone web page, enter an appropriate value.

Default: 16 seconds.

SIP Timer H

INVITE final response, time-out value, which can from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Timer_H ua="na">16</SIP_Timer_H>

• In the phone web page, enter an appropriate value.

Default: 16 seconds.

SIP Timer D

ACK hang-around time, which can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Timer_D ua="na">16</SIP_Timer_D>

• In the phone web page, enter an appropriate value.

Default: 16 seconds.

SIP Timer J

Non-INVITE response hang-around time, which can range from 0 to 64 seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Timer_J ua="na">16</SIP_Timer_J>

• In the phone web page, enter an appropriate value.

Default: 16 seconds.

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <INVITE_Expires ua="na">240</INVITE_Expires>

• In the phone web page, enter an appropriate value.

Default: 240 seconds

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <ReINVITE_Expires ua="na">30</ReINVITE_Expires>

• In the phone web page, enter an appropriate value.

Default: 30

Reg Min Expires

Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Min_Expires ua="na">1</Reg_Min_Expires>

• In the phone web page, enter an appropriate value.

Default: 1

Reg Max Expires

Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Max_Expires ua="na">7200</Reg_Max_Expires>

• In the phone web page, enter an appropriate value.

Default: 7200

Reg Retry Intv

Interval to wait before the Cisco IP Phone retries registration after failing during the last registration. The range is from 1 to 2147483647

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Retry_Intvl ua="na">30</Reg_Retry_Intvl>

• In the phone web page, enter an appropriate value.

Default: 30

See the note below for additional details.

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match<Retry Reg RSC>, the Cisco IP Phone waits for the specified length of time before retrying. If this interval is 0, the phone stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Retry_Long_Intvl ua="na">1200</Reg_Retry_Long_Intvl>

• In the phone web page, enter an appropriate value.

Default: 1200

See the note below for additional details.

Reg Retry Random Delay

Random delay range (in seconds) to add to <Register Retry Intvl> when retrying REGISTER after a failure. Minimum and maximum random delay to be added to the short timer. The range is from 0 to 2147483647.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Retry_Random_Delay ua="na">3</Reg_Retry_Random_Delay>

• In the phone web page, enter an appropriate value.

Default: 0

Reg Retry Long Random Delay

Random delay range (in seconds) to add to the Register Retry Long Intvl parameter when retrying REGISTER after a failure.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Retry_Long_Random_Delay ua="na">5</Reg_Retry_Long_Random_Delay>

• In the phone web page, enter an appropriate value.

Default: 0

Reg Retry Intvl Cap

Maximum value of the exponential delay. The maximum value to cap the exponential backoff retry delay (which starts at the Register Retry Intvl and doubles every retry). Defaults to 0, which disables the exponential backoff (that is, the error retry interval is always at the Register Retry Intvl). When this feature is enabled, the Reg Retry Random Delay is added to the exponential backoff delay value. The range is from 0 to 2147483647.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Reg_Retry_Intvl_Cap ua="na">1000</Reg_Retry_Intvl_Cap>

• In the phone web page, enter an appropriate value.

Default: 0

Sub Min Expires

Sets the lower limit of the REGISTER expires value returned from the Proxy server.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Sub_Min_Expires ua="na">10</Sub_Min_Expires>

• In the phone web page, enter an appropriate value.

Default: 10

Sub Max Expires

Sets the upper limit of the REGISTER minexpires value returned from the Proxy server in the Min-Expires header.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Sub_Max_Expires ua="na">7200</Sub_Max_Expires>

• In the phone web page, enter an appropriate value.

Default: 7200

Sub Retry Intvl

This value (in seconds) determines the retry interval when the last Subscribe request fails.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Sub_Retry_Intvl ua="na">10</Sub_Retry_Intvl>

• In the phone web page, enter an appropriate value.

Default: 10

Try Backup RSC

Specifies the SIP response status codes that trigger the phone to fail over and attempt registration with a backup server.

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Try_Backup_RSC ua="na">503</Try_Backup_RSC>

• In the phone web page, enter an appropriate value.

Default: Empty

Retry Reg RSC

Specifies the SIP response status codes that trigger the phone to retry registration attempts.

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Retry_Reg_RSC ua="na">500</Retry_Reg_RSC>

• In the phone web page, enter an appropriate value.

Default: Empty

RTP Port Min

Minimum port number for RTP transmission and reception.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <RTP_Port_Min ua="na">16384</RTP_Port_Min>

• In the phone web page, enter an appropriate port number.

Allowed values: 2048 to 49151

If the value range (RTP Port Max - RTP Port Min) is less than 16 or you configure the parameter incorrectly, the RTP port range (16382 to 32766) is used instead.

Default: 16384

RTP Port Max

Maximum port number for RTP transmission and reception.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <RTP_Port_Max ua="na">16482</RTP_Port_Max>

• In the phone web page, enter an appropriate port number.

Allowed values: 2048 to 49151

If the value range (RTP Port Max - RTP Port Min) is less than 16 or you configure the parameter incorrectly, the RTP port range (16382 to 32766) is used instead.

Default: 16482

RTP Packet Size

Specifies the audio packetization interval (in seconds) for RTP voice streams

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <RTP_Packet_Size ua="na">0.02</RTP_Packet_Size>

• In the phone web page, enter an appropriate value to specify the packet size.

Allowed values: Ranges from 0.01 to 0.13. Valid values must be a multiple of 0.01 seconds.

Default: 0.02

Max RTP ICMP Err

Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the phone terminates the call. If value is set to 0, the phone ignores the limit on ICMP errors.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Max_RTP_ICMP_Err ua="na">0</Max_RTP_ICMP_Err>

• In the phone web page, enter an appropriate value.

Allowed values: 0-10

Default: 0

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <RTCP_Tx_Interval ua="na">5</RTCP_Tx_Interval>

• In the phone web page, enter an appropriate value.

Allowed values: 0 to 255 seconds

Default: 0

Call Statistics

Specifies whether the phone sends end-of-call statistics within SIP messages when a call terminates or is put on hold.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Call_Statistics ua="na">No</Call_Statistics>

• In the phone web page, select Yes to enable this feature.

Allowed values: Yes | No

Default: No

SDP IP Preferences

Select the preferred IP that the phone uses as RTP address.

If the phone is in dual-mode and has both ipv4 and ipv6 addresses, it will always include both addresses in SDP by attributes "a=altc …

If IPv4 address is selected, then ipv4 address has higher priority than ipv6 address in SDP and indicates that phone prefers using ipv4 RTP address.

If the phone has only ipv4 address or ipv6 address, SDP does not have ALTC attributes and RTP address is specified in “c=” line.

For information about the behavior in dual mode, see SIP and RTP behaviors in dual mode.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SDP_IP_Preference ua="na">IPv4</SDP_IP_Preference>

• In the phone web page, select the preferred IP .

Allowed values: IPv4 | IPv6

Default: IPv4

RTP Before ACK

Allows you to specify whether an RTP session starts before or after an ACK is received from the calling party.

• Yes: An RTP session doesn't await an ACK, but starts after a 200 OK message is sent.

• No: An RTP session doesn't start until an ACK is received from the calling party.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <RTP_Before_ACK ua="na">No</RTP_Before_ACK>

• In the phone web page, select Yes or No.

Allowed values: Yes | No

Default: No

SSRC Reset on Rx RE-INVITE

Controls whether to reset the Synchronization Source (SSRC) for the outgoing RTP and SRTP sessions on incoming RE-INVITE.

• Yes: the phone can avoid the call transfer error, where only one person on the call hears the audio. This occurs on calls of 30 minutes or longer, and often on three-way calls.

• No: the SSRC still remains during a long duration call. In this case, this error might occur.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SSRC_Reset_on_Rx_RE-INVITE ua="na">Yes</SSRC_Reset_on_Rx_RE-INVITE>

• In the phone web page select Yes or No.

Allowed values: Yes | No

Default: No

SSRC Reset on Tx RE-INVITE

Controls whether to reset the Synchronization Source (SSRC) for the outgoing RTP and SRTP sessions on outgoing RE-INVITE.

• Yes: the phone can avoid the one-way audio issue on a long duration call followed by a hold-resume action in certain Webex Calling environments where the SRTP is end-to-end encrypted.

• No: the SSRC still remains during a long duration call. In this case, this error might occur.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SSRC_Reset_on_Tx_RE-INVITE ua="na">Yes</SSRC_Reset_on_Tx_RE-INVITE>

• In the phone web page, select Yes or No.

Allowed values: Yes | No

Default: No

G711u Codec Name

G711u codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G711u_Codec_Name ua="na">PCMU</G711u_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: PCMU

G711a Codec Name

G711a codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G711a_Codec_Name ua="na">PCMU</G711a_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: PCMA

G729a Codec Name

G729a codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G729a_Codec_Name ua="na">PCMU</G729a_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: G729a

G722 Codec Name

G722 codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G722_Codec_Name ua="na">PCMU</G722_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: G722

G722.2 Codec Name

G722.2 codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G722.2_Codec_Name ua="na">AMR-WB</G722.2_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: AMR-WB

iLBC Codec Name

iLBC codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <iLBC_Codec_Name ua="na">iLBC</iLBC_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: iLBC

iSAC Codec Name

iSAC codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <iSAC_Codec_Name ua="na">iSAC</iSAC_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: iSAC

OPUS Codec Name

OPUS codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <OPUS_Codec_Name ua="na">OPUS</OPUS_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: OPUS

AVT Codec Name

AVT codec name used in SDP.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <AVT_Codec_Name ua="na">telephone-event</AVT_Codec_Name>

• In the phone web page, enter an appropriate codec name.

Default: telephone-event

G722.2 Dynamic Payload

G722 Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G722.2_Dynamic_Payload ua="na">96</G722.2_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 96

G722.2 OA Dynamic Payload

G722.2 OA Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <G722.2_OA_Dynamic_Payload ua="na">103</G722.2_OA_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 103

iLBC 20ms Dynamic Payload

iLBC 20ms Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <iLBC_Dynamic_Payload ua="na">97</iLBC_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 97

iLBC 30ms Dynamic Payload

iLBC 20ms Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <iLBC_30ms_Dynamic_Payload ua="na">105</iLBC_30ms_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 105

iSAC Dynamic Payload

iSAC Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <iSAC_Dynamic_Payload ua="na">98</iSAC_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 98

OPUS Dynamic Payload

OPUS Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <OPUS_Dynamic_Payload ua="na">99</OPUS_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 99

RSFEC Dynamic Payload

RSFEC Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <RSFEC_Dynamic_Payload ua="na">113</RSFEC_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 113

AVT Dynamic Payload

AVT dynamic payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <AVT_Dynamic_Payload ua="na">101</AVT_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 101

AVT 16kHz Dynamic Payload

AVT dynamic payload type for the 16 kHz clock rate.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <AVT_16kHz_Dynamic_Payload ua="na">101</AVT_16kHz_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 101

AVT 48kHz Dynamic Payload

AVT dynamic payload type for the 48 kHz clock rate.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <AVT_48kHz_Dynamic_Payload ua="na">101</AVT_48kHz_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 101

INFOREQ Dynamic Payload

INFOREQ Dynamic Payload type.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <INFOREQ_Dynamic_Payload ua="na">101</INFOREQ_Dynamic_Payload>

• In the phone web page, enter an appropriate value.

Allowed values: 96-127

Default: 101

SIP Transport

Specifies the transport protocol for SIP messages.

When set to AUTO, the phone selects the appropriate protocol automatically, based on the NAPTR records on the DNS server.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Transport_1_ ua="na">UDP</SIP_Transport_1_>

• In the phone web page, select the transport protocol type in the list.

Allowed values: UDP | TCP | TLS | AUTO

Default: UDP

SIP Port

The phone's port number for SIP message listening and transmission.

If you are using TCP, the system uses a random port within the range specified in SIP TCP Port Min and SIP TCP Port Max on the Voice > SIP tab.

If you need to specify a port of SIP proxy server, you can specify it using the Proxy field or the XSI Host Server field.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Port_1_ ua="na">5060</SIP_Port_1_>

• In the phone web page, enter an appropriate port number.

Default: 5060

SIP 100REL Enable

Individually enables the SIP 100REL feature.

When enabled, the phone supports the 100REL SIP extension for reliable transmission of provisional responses (18x) and uses the PRACK requests.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_100REL_Enable_1_ ua="na">Yes</SIP_100REL_Enable_1_>

• In the phone web page, select Yes to enable the feature.

Allowed values: Yes | No

Default: No

EXT SIP Port

The external SIP port number.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <EXT_SIP_Port_1_ ua="na">5060</EXT_SIP_Port_1_>

• In the phone web page, enter a port number.

Allowed values:

Default: 5060

Auth Resync-Reboot

The Cisco IP Phone authenticates the sender when it receives a NOTIFY message with the following requests:

• resync

• reboot

• report

• restart

• XML-service

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Auth_Resync-Reboot_1_ ua="na">No</Auth_Resync-Reboot_1_>

• In the phone web page, select Yes to enable the feature.

Allowed values: Yes | No

Default: Yes

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it receives the Proxy-Require header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_Proxy-Require_1_ ua="na">header<SIP_Proxy-Require_1_>

• In the phone web interface, enter the appropriate header in the field provided.

Default: Blank

SIP Remote-Party-ID

The Remote-Party-ID header to use instead of the From header. Select Yes to enable.

Default: Yes

Referor Bye Delay

Controls when the phone sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Referor_Bye_Delay_1_ ua="na">4</Referor_Bye_Delay_1_>

• In the phone web page, enter the appropriate period of time in seconds.

Allowed values: An integer from 0 through 65535

Default: 4

Refer-To Target Contact

Indicates the refer-to target.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Refer-To_Target_Contact_1_ ua="na">No</Refer-To_Target_Contact_1_>

• In the phone web page, select Yes to send the SIP Refer to the contact.

Allowed values: Yes | No

Default: No

Referee Bye Delay

Specifies the Referee Bye Delay time in seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Referee_Bye_Delay_1_ ua="na">0</Referee_Bye_Delay_1_>

• In the phone web page, enter the appropriate period of time in seconds.

Allowed values: An integer from 0 through 65535

Default: 0

Refer Target Bye Delay

Specifies the Refer Target Bye Delay time in seconds.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Refer_Target_Bye_Delay_1_ ua="na">0</Refer_Target_Bye_Delay_1_>

• In the phone web page, enter the appropriate period of time in seconds.

Allowed values: An integer from 0 through 65535

Default: 0

Sticky 183

Controls the first 183 SIP response for an outbound INVITE. To enable this feature,

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Sticky_183_1_ ua="na">No</Sticky_183_1_>

• In the phone web page, select Yes to enable this feature.

  When enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE.

Allowed values: Yes | No

Default: No

Auth INVITE

Controls if authorization is required for initial incoming INVITE requests from the SIP proxy. To enable this feature.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Auth_INVITE_1_ ua="na">No</Auth_INVITE_1_>

• In the phone web page, select Yes to enable this feature.

  When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.

Allowed values: Yes | No

Default: No

Ntfy Refer On 1xx-To-Inv

If set to Yes, as a transferee, the phone will send a NOTIFY with Event:Refer to the transferor for any 1xx response returned by the transfer target, on the transfer call leg.

If set to No, the phone will only send a NOTIFY for final responses (200 and higher).

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Ntfy_Refer_On_1xx-To-Inv_1_ ua="na">Yes</Ntfy_Refer_On_1xx-To-Inv_1_>

• In the phone web page, select Yes to enable this feature.

Allowed values: Yes | No

Default: Yes

Set G729 annexb

Configure G.729 Annex B settings.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Set_G729_annexb_1_ ua="na">Yes</Set_G729_annexb_1_>

• In the phone web page, select Yes to enable this feature.

Allowed values: None | No |Yes | Follow silence supp setting

Default: Yes

Use low-bandwidth OPUS

To improve bandwidth in your network, you can set up your phones to use the narrowband OPUS codec. The narrowband codec won't conflict with the wideband codec.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Use_low-bandwidth_OPUS_1_ ua="na">No</Use_low-bandwidth_OPUS_1_>

• In the phone web page, select Yes or No to enable or disable this feature.

Allowed values: No |Yes

Default: No

Voice Quality Report Address

Specifies the destination where the phone sends diagnostic reports about call quality. You can enter either a domain name or an IP address.

You can also add a port number along with the domain name or an IP address for this parameter. If you don't enter a port number, the value of the SIP UDP Port (5060) is used by default. If the collector server URL parameter is blank, a SIP PUBLISH message is not sent out.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Voice_Quality_Report_Address_1_ ua="na"/>100.100.1.1<Voice_Quality_Report_Address_1_>

• In the phone web page, enter the server address or domain name.

Default: Empty

Voice Quality Report Interval

Specifies how often the phone generates and sends voice quality reports to the configured report address. The interval is measured in minutes. Shorter intervals provide more frequent updates for monitoring but may increase reporting traffic, while longer intervals reduce overhead but provide less granularity.

The default interval is 0, meaning that no periodic reports will be send.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <VQ_Report_Interval_1_ ua="na">0</VQ_Report_Interval_1_>

• In the phone web page, specify the interval in .

Default: 0

Defines the group identifier used when sending voice quality reports from the phone. This parameter allows multiple phones to be organized under the same reporting group, so that administrators can easily sort, filter, and analyze call quality data.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Voice_Quality_Report_Group_1_ ua="na">vq_report_SupportTeam</Voice_Quality_Report_Group_1_>

• In the phone web page, enter the name of the group.

Default: Empty

User Equal Phone

When a tel URL is converted to a SIP URL and the phone number is represented by the user portion of the URL, the SIP URL includes the optional: user=phone parameter (RFC3261). For example:

To: sip:+12325551234@example.com; user=phone

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <User_Equal_Phone_1_ ua="na">Yes</User_Equal_Phone_1_>

• In the phone web page, select Yes to enable this feature.

Allowed values: Yes | No

Default: No

Privacy Header

Sets user privacy in the SIP message in the trusted network.

The privacy header options are:

• Disabled (default)

• none—The user requests that a privacy service applies no privacy functions to this SIP message.

• header—The user needs a privacy service to obscure headers which cannot be purged of identifying information.

• session—The user requests that a privacy service provide anonymity for the sessions.

• user—The user requests a privacy level only by intermediaries.

• id—The user requests that the system substitute an id that doesn't reveal the IP address or host name.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Privacy_Header_1_ ua="na">Disabled</Privacy_Header_1_>

• In the phone web page, select an option from the list.

Allowed values: Disabled | none | header | session | user | id

Default: Disabled

P-Early-Media Support

Controls whether the P-Early-Media header is included in the SIP message for an outgoing call.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <P-Early-Media_Support_1_ ua="na">No</P-Early-Media_Support_1_>

• In the phone web interface, to include the P-Early-Media header, select Yes.

Allowed values: Yes | No

Default: No

SIP SessionID Support

Enables the use of the SIP Session-ID header (defined in RFC 7989) to uniquely identify an end-to-end communication session. The Session-ID remains the same across call legs, even if the call is transferred, forwarded, or goes through multiple SIP intermediaries.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <SIP_SessionID_Support_1_ ua="na">Yes</SIP_SessionID_Support_1_>

• In the phone web interface, select Yes or No to enable or disable the feature.

Allowed values: Yes | No

Default: Yes

MediaSec Request

Specifies whether the phone requests the use of Media Security (MediaSec) during SIP call setup.

When enabled, the phone includes Security-Client headers in SIP messages to request secure media negotiation. When disabled, The phone does not request MediaSec; calls proceed without media security negotiation.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <MediaSec_Request_1_ ua="na">No</MediaSec_Request_1_>

• In the phone web interface, select Yes or No to enable or disable the feature.

Allowed values: Yes | No

Default: No

MediaSec Over TLS Only

Controls whether the phone uses Media Security (MediaSec, RFC 3329) only when the SIP signaling is protected with TLS.

When enabled, the phone includes MediaSec (Security-Client headers) only in TLS-based SIP sessions. If SIP is over UDP/TCP (unencrypted), MediaSec will not be requested. When disabled, The phone may request MediaSec regardless of whether SIP signaling is secured by TLS.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <MediaSec_Over_TLS_Only_1_ ua="na">No</MediaSec_Over_TLS_Only_1_>

• In the phone web interface, select Yes or No to enable or disable the feature.

Allowed values: Yes | No

Default: No

Precondition Support

Determines whether the phone includes the precondition tag (defined in RFC 3312) in the Supported header field.

• Disabled: The phone doesn't include the precondition tag in the Supported header filed. And the phone doesn't return the 183 response when it receives the INVITE request that contains the QoS precondition in the SDP description.

• Enabled: The phone includes the precondition tag in the Supported header field.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Precondition_Support_1_ ua="na">Enabled</Precondition_Support_1_>

• In the phone web page, select Enabled to enable the feature.

Allowed values: Disabled | Enabled

Default: Disabled

Auth Support RFC8760

Enables support for the authentication scheme defined in RFC 8760 – Digest Authentication for SIP (SHA-256/512). This standard extends SIP authentication to use stronger, more secure hash functions compared to the older MD5-based digest (RFC 2617).

When enabled, the phone supports SHA-256/SHA-512 digest authentication when challenged by the SIP server.

When disabled, the phone only supports legacy MD5 digest (RFC 2617) authentication.

Perform one of the following:

• In the phone configuration file with XML(cfg.xml), enter a string in this format:

  <Auth_Support_RFC8760_1_ ua="na">No</Auth_Support_RFC8760_1_>

• In the phone web interface, select Yes or No to enable or disable the feature.

Allowed values: Yes | No

Default: No