Receiver Codec

Type of received voice stream (RTP streaming audio from codec): G.729, G.722, G.711 mu-law, G.711 A-law, OPUS, and iLBC.

Sender Codec

Type of transmitted voice stream (RTP streaming audio from codec): G.729, G.722, G.711 mu-law, G.711 A-law, OPUS, and iLBC.

Receiver Size

Size of voice packets, in milliseconds, in the receiving voice stream (RTP streaming audio).

Sender Size

Size of voice packets, in milliseconds, in the transmitting voice stream.

Rcvr Packets

Number of RTP voice packets that were received since voice stream opened.

Sender Packets

Number of RTP voice packets that were transmitted since voice stream opened.

Avg Jitter

Estimated average RTP packet jitter (dynamic delay that a packet encounters when going through the network), in milliseconds, that was observed since the receiving voice stream opened.

Max Jitter

Maximum jitter, in milliseconds, that was observed since the receiving voice stream opened.

Receiver Discarded

Number of RTP packets in the receiving voice stream that were discarded (bad packets, too late, and so on).

Rcvr Lost Packets

Missing RTP packets (lost in transit).

Voice-Quality Metrics

Cumulative Conceal Ratio

Total number of concealment frames divided by total number of speech frames that were received from start of the voice stream.

Interval Conceal Ratio

Ratio of concealment frames to speech frames in preceding 3-second interval of active speech. If using voice activity detection (VAD), a longer interval might be required to accumulate 3 seconds of active speech.

Max Conceal Ratio

Highest interval concealment ratio from start of the voice stream.

Conceal Seconds

Number of seconds that have concealment events (lost frames) from the start of the voice stream (includes severely concealed seconds).

Severely Conceal Seconds

Number of seconds that have more than 5 percent concealment events (lost frames) from the start of the voice stream.

Latency

Estimate of the network latency, expressed in milliseconds. Represents a running average of the round-trip delay, measured when RTCP receiver report blocks are received.

Host Name

Displays the current host name assigned to the phone.

Domain

Displays the network domain name of the phone.

Default: cisco.com

Primary NTP Server

Displays the primary NTP server assigned to the phone.

Secondary NTP Server

Displays the secondary NTP server assigned to the phone.

Bluetooth Enabled

Indicates if the phone has Bluetooth enabled to it.

Bluetooth Connected

Indicates if the phone has Bluetooth is connected to it.

Bluetooth MAC

Displays the MAC address of the Bluetooth device.

Connected Device ID

Displays the ID of the connected device.

Active Interface

Displays if the phone uses Ethernet cable as the deployment option.

Only for Cisco IP Phone 8861.

Wireless MAC

Displays MAC address of the phone.

Only for Cisco IP Phone 8861.

SSID

Displays the SSID of the phone.

Only for Cisco IP Phone 8861.

Mode 802.11

Displays if the phone uses 802.11 interface as the deployment option.

Only for Cisco IP Phone 8861.

Security Mode

Displays the type of authentication that the phone uses to access the WLAN.

Camera Shutter

Displays the state of the shutter.

Only for Cisco IP Phone 8845 and 8865.

IP Status

Indicates that the connection is established.

Connection Type

Indicates the type of internet connection for the phone:

• DHCP

• Static IP

Current IP

Displays the current IP address assigned to the IP phone.

Current Netmask

Displays the network mask assigned to the phone.

Current Gateway

Displays the default router assigned to the phone.

Primary DNS

Displays the primary DNS server assigned to the phone.

Secondary DNS

Displays the secondary DNS server assigned to the phone.

IP Status

Indicates that the connection is established.

Connection Type

Indicates the type of internet connection for the phone:

• Static IP

• DHCP

Current IP

Displays the current IPv6 address assigned to the IP phone.

Prefix Length

Identifies number of bits of a global unicast IPv6 address that are part of`the network. For example, if the IPv6 address is 2001:0DB8:0000:000b::/64, the number 64 identifies that the first 64 bits are part of the network.

Current Gateway

Displays the default router assigned to the phone.

Primary DNS

Displays the primary DNS server assigned to the phone.

Secondary DNS

Displays the secondary DNS server assigned to the phone.

Locale download status

Displays the downloaded locale package status.

Locale download URL

Displays the location from where the local package is downloaded.

Font download status

Displays the downloaded font file status.

Font download URL

Displays the location from where the font file is downloaded.

Current Time

Current date and time of the system; for example, 08/06/14 1:42:56 a.m.

Elapsed Time

Total time elapsed since the last reboot of the system; for example, 7 days, 02:13:02.

SIP Messages Sent

Total number of SIP messages sent (including retransmissions).

SIP Bytes Sent

Total number of SIP messages received (including retransmissions).

SIP Messages Recv

Total number of bytes of SIP messages sent which includes retransmissions.

SIP Bytes Recv

Total number of bytes of SIP messages received (including retransmissions).

Network Packets Sent

Total number of network packets sent.

Network Packets Recv

Total number of network packets received.

External IP

External IP of the phone.

Operational VLAN ID

ID of the VLAN currently in use if applicable.

SW Port

Displays the type of Ethernet connection from the IP phone to the switch.

PC Port

Displays the type of Ethernet connection from PC Port.

Upgrade Status

Displays status of the last phone upgrade.

SW Port Config

Displays the type of SW port configuration.

PC Port Config

Displays the type of PC port configuration.

Last Successful Login

Displays the time when the phone has last successful log in.

Last Failed Login

Displays the time when the phone has last failed log in.

Transaction status

Indicates if the phone is authenticated.

Protocol

Displays the protocol of the registered phone.

Registration State

Shows “Registered” if the phone is registered, or “Not Registered” if the phone is not registered to the ITSP.

Last Registration At

Last date and time the line was registered.

Next Registration In Seconds

Number of seconds before the next registration renewal.

Message Waiting

Indicates whether message waiting is enabled or disabled.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Hoteling State

Indicates whether Hoteling is enabled or disabled.

Extended Function Status

Indicates whether extended function is enabled.

Call State

Status of the call.

Tone

Type of tone that the call uses.

Encoder

Codec used for encoding.

Decoder

Codec used for decoding.

Type

Direction of the call.

Remote Hold

Indicates whether the far end placed the call on hold.

Callback

Indicates whether the call was triggered by a call back request.

Mapped RTP Port

The port mapped for Real Time Protocol traffic for the call.

Peer Name

Name of the internal phone.

Peer Phone

Phone number of the internal phone.

Duration

Duration of the call.

Packets Sent

Number of packets sent.

Packets Recv

Number of packets received.

Bytes Sent

Number of bytes sent.

Bytes Recv

Number of bytes received.

Decode Latency

Number of milliseconds for decoder latency.

Jitter

Number of milliseconds for receiver jitter.

Round Trip Delay

Number of milliseconds for delay in the RTP-to-RTP interface round trip.

Packets Lost

Number of packets lost.

Loss Rate

The fraction of RTP data packets from the source lost since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Packet Discarded

The fraction of RTP data packets from the source lost since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Discard Rate

The fraction of RTP data packets from the source that have been discarded since the beginning of reception, due to late or early arrival, under-run or overflow at the receiving jitter buffer. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Burst Duration

The mean duration, expressed in milliseconds, of the burst periods that have occurred since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Gap Duration

The mean duration, expressed in milliseconds, of the gap periods that have occurred since the beginning of reception. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

R-Factor

Voice quality metric that describes the segment of the call that is carried over this RTP session. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

MOS-LQ

The estimated mean opinion score for listening quality (MOS-LQ) is a voice quality metric on a scale from 1 to 5, in which 5 represents excellent and 1 represents unacceptable. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

MOS-CQ

The estimated mean opinion score for conversational quality (MOS-CQ) is defined as including the effects of delay and other effects that affect conversational quality. Defined in RFC-3611—RTP Control Protocol Extended Reports (RTCP XR).

Multicast Rx Pkts

Multicast Tx Pkts

TR-069 Feature

Indicates if TR-069 function is enabled or disabled.

Periodic Inform Time

Displays the inform time interval from CPE to ACS.

Last Inform Time

Indicates the last inform time.

Last Transaction Status

Displays the success or the failure status.

Last Session

Indicates the start and end time of the session.

ParameterKey

Displays the key for reference checkpoint for parameter set configured.

Custom CA Provisioning Status

Indicates whether provisioning using a custom CA succeeded or failed:

• Last provisioning succeeded on mm/dd/yyyy HH:MM:SS;

• Last provisioning failed on mm/dd/yyyy HH:MM:SS

Custom CA Info

Displays information about the custom CA:

• Installed—Displays the "CN Value", where "CN Value" is the value of the CN parameter for the Subject field in the first certificate.

• Not Installed—Displays if no custom CA certificate is installed.

Provisioning Profile

Displays the profile file name of the phone.

Provisioning Status 1

Displays the provisioning status (resync) of the phone.

Provisioning Status 2

Provisioning Status 3

Provisioning Failure Reason

Displays the reason for the failure of provisioning of the phone.

Debug Message 1

messages

Debug Message 2

messages.1

Debug Message 3

messages.2

Debug Message 4

messages.3

Debug Message 5

messages.4

Debug Message 6

messages.5

Debug Message 7

messages.6

Debug Message 8

messages.7

Firmware Upgrade Status 1

Displays the upgrade status (failed or succeeded) with reason for the same.

Firmware Upgrade Status 2

Firmware Upgrade Status 3

Provisioning Profile

Displays the profile file name of the phone.

Provisioning Status 1

Displays the provisioning status (resync) of the phone.

Provisioning Status 2

Provisioning Status 3

Provisioning Failure Reason

Displays the reason for the failure of provisioning of the phone.

Custom CA Provisioning Status

Indicates whether provisioning using a custom CA succeeded or failed:

• Last provisioning succeeded on mm/dd/yyyy HH:MM:SS;

• Last provisioning failed on mm/dd/yyyy HH:MM:SS

Custom CA Info

Displays information about the custom CA:

• Installed—Displays the "CN Value", where "CN Value" is the value of the CN parameter for the Subject field in the first certificate.

• Not Installed—Displays if no custom CA certificate is installed.

TxFrames

Total number of packets that the phone transmitted.

TxBroadcasts

Total number of broadcast packets that the phone transmitted.

TxMulticasts

Total number of multicast packets that the phone transmitted.

TxUnicasts

Total number of unicast packets that the phone transmitted.

RxFrames

Total number of packets received by the phone.

RxBroadcasts

Total number of broadcast packets that the phone received.

RxMulticasts

Total number of multicast packets that the phone received.

RxUnicasts

Total number of unicast packets that the phone received.

Restricted Access Domains

This feature is used when implementing software customization.

Enable Web Server

Enable/disable web server of the IP phone.

Default: Yes

Enable Protocol

Choose the type of protocol:

• Http

• Https

If you specify the HTTPS protocol, you must include https: in the URL.

Enable Direct Action Url

Enables the direct action of the URL.

Default: Yes

Session Max Timeout

Allows you to enter maximum timeout of the session.

Default: 3600

Session Idle Timeout

Allows you to enter idle timeout of the session.

Default: 3600

Web Server Port

Allows you to enter port number of the phone web user interface.

Default:

• 80 for protocol HTTP.

• 443 for protocol HTTPS.

If you specify a port number other than the default value for that protocol, you must include the nondefault port number in the server URL.

Example: https://192.0.2.1:999/admin/advanced

Enable Web Admin Access

Allows you to enable or disable local access to the phone web user interface. Select Yes or No from the drop-down menu.

Default: Yes

Admin Password

Allows you to enter password for the administrator.

Default: No password

User Password

Allows you to enter password for the user.

Default: Blank

Phone-UI-readonly

Allows you to make the phone menus and options that the phone users see as read-only fields.

Phone-UI-User-Mode

Allows you to restrict the menus and options that phone users see when they use the phone interface. Choose yes to enable this parameter and restrict access.

Default: No

Specific parameters are then designated as “na” or “ro” using provisioning files. Parameters designated as “na” will not appear on the phone interface. Parameters designated as “ro” will not be editable by the user.

IP Mode

Allows you to select the internet protocol mode in which the phone operates. Options are: IPv4 Only, IPv6 Only, and Dual Mode. In dual mode, the phone can have both IPv4 and IPv6 addresses.

Default: Dual Mode

Connection Type

Internet connection type that is configured for the phone. Options are DHCP and Static IP.

Default: DHCP

NetMask

Subnet mask of the phone.

Static IP

IP address of the phone.

Gateway

IP address of the gateway.

Primary DNS

Primary Domain Name Server (DNS) assigned to the phone.

Secondary DNS

Secondary Domain Name Server (DNS) if assigned to the phone.

Connection Type

Internet connection type that is configured for the phone. Options are DHCP and Static IP.

Default: DHCP

Static IP

IPv6 address of the phone.

Prefix Length

Identifies number of bits of a global unicast IPv6 address that are part of`the network. For example, if the IPv6 address is 2001:0DB8:0000:000b::/64, the number 64 identifies that the first 64 bits are part of the network.

Gateway

IP address of the gateway.

Primary DNS

Primary Domain Name Server (DNS) assigned to the phone.

Secondary DNS

Secondary Domain Name Server (DNS) if assigned to the phone.

Broadcast Echo

Options are Disabled and Enabled.

Default: Disabled

Auto Config

When enabled, phone generates an IPv6 address by default with the prefix length sent from the router. Options are Disabled and Enabled.

Default: Enabled

SIP IP Preference

SDP IP Preference

Host Name

The hostname of the Cisco IP Phone.

Domain

The network domain of the Cisco IP Phone.

If you are using LDAP, see LDAP Configuration.

DNS Server Order

Specifies the method for selecting the DNS server:

• Manual, DHCP

• Manual

• DHCP,Manual

DNS Query Mode

Specified mode of DNS query.

• Parallel

• Sequential

DNS Caching Enable

When set to Yes, the DNS query results are not cached.

Default: Yes

Switch Port Config

Allows you to select speed and duplex of the network port. Values are:

• Auto

• 10MB half

• 10MB full

• 100 MB half

• 100MB full

• 100 half

• 1000 full

PC Port Config

Allows you to select Speed and duplex of the Computer (access) port.

• Auto

• 10MB half

• 10MB full

• 100 MB half

• 100MB full

• 100 half

• 1000 full

PC PORT Enable

Specifies if PC port is enabled. Options are Yes or No.

Enable PC Port Mirror

Adds the ability to port mirror on the PC port. When enabled, you can see the packets on the phone. Select Yes to enable PC port mirroring and select No to disable it.

Syslog Server

Specify the syslog server name and port. This feature specifies the server for logging IP phone system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server.

Debug Level

The debug level from 0 to 2. The higher the level, the more debug information is generated. Zero (0) means that no debug information is generated. To log SIP messages, you must set the Debug Level to at least 2.

Default: 0

Primary NTP Server

IP address or name of the primary NTP server used to synchronize its time.

Default: Blank

Secondary NTP Server

IP address or name of the secondary NTP server used to synchronize its time.

Default: Blank

Enable SSLv3

Choose Yes to enable SSLv3. Choose No to disable.

Default: No

Enable VLAN

Choose Yes to enable VLAN. Choose No to disable.

Enable CDP

Enable CDP only if you are using a switch that has Cisco Discovery Protocol. CDP is negotiation based and determines which VLAN the IP phone resides in.

Enable LLDP-MED

Choose Yes to enable LLDP-MED for the phone to advertise itself to devices that use that discovery protocol.

When the LLDP-MED feature is enabled, after the phone has initialized and Layer 2 connectivity is established, the phone sends out LLDP-MED PDU frames. If the phone receives no acknowledgment, the manually configured VLAN or default VLAN will be used if applicable. If the CDP is used concurrently, the waiting period of 6 seconds is used. The waiting period will increase the overall startup time for the phone.

Network Startup Delay

Setting this value causes a delay for the switch to get to the forwarding state before the phone will send out the first LLDP-MED packet. The default delay is 3 seconds. For configuration of some switches, you might need to increase this value to a higher value for LLDP-MED to work. Configuring a delay can be important for networks that use Spanning Tree Protocol.

VLAN ID

If you use a VLAN without CDP (VLAN enabled and CDP disabled), enter a VLAN ID for the IP phone. Note that only voice packets are tagged with the VLAN ID. Do not use 1 for the VLAN ID.

PC Port VLAN ID

VLAN ID for the PC port.

Asset ID

Provides the ability to enter an asset ID for inventory management when using LLDP-MED. The default value for Asset ID is empty. Enter a string of less than 32 characters if you are using this field.

The Asset ID can be provisioned only by using the web management interface or remote provisioning. The Asset ID is not displayed on the phone screen.

Changing the Asset ID field causes the phone to reboot.

Max Forward

SIP Max Forward value, which can range from 1 to 255.

Default: 70

Max Redirection

Number of times an invite can be redirected to avoid an infinite loop.

Default: 5

Max Auth

Maximum number of times (from 0 to 255) a request can be challenged.

Default: 2

SIP User Agent Name

Used in outbound REGISTER requests.

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.

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.

Default: Blank

SIP Accept Language

Accept-Language header used. To access, click the SIP tab, and fill in the SIP Accept Language field.

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.

Default: application/dtmf-relay

Hook Flash MIME Type

MIME Type used in a SIPINFO message to signal a hook flash event.

Remove Last Reg

Enables you to remove the last registration before registering a new one if the value is different. Select yes or no from the drop-down menu.

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.

Default: No

Escape Display Name

Enables you to keep the Display Name private.

Select 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.

Default: Yes.

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.

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.

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.

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.

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.

Default: No.

SIP TCP Port Min

Specifies the lowest TCP port number that can be used for SIP sessions.

Default: 5060

SIP TCP Port Max

Specifies the highest TCP port number that can be used for SIP sessions.

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.

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).

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.

Keep Referee When Refer Failed

If set to yes, it configures the phone to immediately handle NOTIFY sipfrag messages.

Display Diversion Info

Display the Diversion info included in SIP message on LCD or not.

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.

Sip Accept Encoding

Supports the content-encoding gzip feature. The options are none and gzip.

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.

Disable Local Name To Header

The options are No and Yes. If No is selected, no changes are made. The default value is No.

If Yes is selected, it disables the display name in “Directory”, “Call History”, and in the “To” header during an outgoing call.

SIP IP Preference

Sets if the phone uses IPv4 or IPv6.

Default: IPv4.

Disable Local Name to Header

Select Yes to enable or No to disable.

Default: No

SIP T1

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

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.

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.

Default: 5 seconds.

SIP Timer B

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

Default: 16 seconds.

SIP Timer F

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

Default: 16 seconds.

SIP Timer H

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

Default: 16 seconds.

SIP Timer D

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

Default: 16 seconds.

SIP Timer J

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

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.

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.

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.

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.

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

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.

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.

Default: 0

Reg Retry Long Random Delay

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

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.

Default: 0

Sub Min Expires

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

Sub Max Expires

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

Default: 7200.

Sub Retry Intvl

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

Default: 10.

Try Backup RSC

This parameter may be set to invoke failover upon receiving specified response codes.

Default: Blank

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??

Retry Reg RSC

Interval to wait before the phone retries registration after failing during the last registration.

Default: Blank

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??

RTP Port Min

Minimum port number for RTP transmission and reception. Minimum port number for RTP transmission and reception. Should define a range that contains at least 10 even number ports (twice the number of lines); for example, configure RTP port min to 16384 and RTP port max to 16538.

Default: 16384

RTP Port Max

Maximum port number for RTP transmission and reception. Should define a range that contains at least 10 even number ports (twice the number of lines); for example, configure RTP port min to 16384 and RTP port max to 16538.

Default: 16538

RTP Packet Size

Packet size in seconds, which can range 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.

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds.

Default: 0

SDP IP Preferences

Select IPv4 or IPv6.

Default: IPv4

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.

G722.2 Dynamic Payload

G722 Dynamic Payload type.

Default: 96

iLBC Dynamic Payload

iLBC Dynamic Payload type.

Default: 97

iSAC Dynamic Payload

iSAC Dynamic Payload type.

Default: 98

OPUS Dynamic Payload

OPUS Dynamic Payload type.

Default: 99

AVT Dynamic Payload

AVT dynamic payload type. Ranges from 96-127.

Default: 101

INFOREQ Dynamic Payload

INFOREQ Dynamic Payload type.

H264 BP0 Dynamic Payload

H264 BPO Dynamic Payload type.

Default: 110

H264 HP Dynamic Payload

H264 HP Dynamic Payload type.

Default: 110

G711u Codec Name

G711u codec name used in SDP.

Default: PCMU

G711a Codec Name

G711a codec name used in SDP.

Default: PCMA

G729a Codec Name

G729a codec name used in SDP.

Default: G729a

G729b Codec Name

G729b codec name used in SDP.

Default: G729b

G722 Codec Name

G722 codec name used in SDP.

Default: G722

G722.2 Codec Name

G722.2 codec name used in SDP.

Default: G722.2

iLBC Codec Name

iLBC codec name used in SDP.

Default: iLBC

iSAC Codec Name

iSAC codec name used in SDP.

Default: iSAC

OPUS Codec Name

OPUS codec name used in SDP.

Default: OPUS

AVT Codec Name

AVT codec name used in SDP.

Default: telephone-event

Handle VIA received

Enables the phone to process the received parameter in the VIA header.

Default: No

Handle VIA rport

Enables the phone to process the rport parameter in the VIA header.

Default: No

Insert VIA received

Enables to insert the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Default: No

Insert VIA rport

Enables to insert the rport parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Default: No

Substitute VIA Addr

Enables the user to use NAT-mapped IP:port values in the VIA header.

Default: No

Send Resp To Src Port

Enables to send responses to the request source port instead of the VIA sent-by port.

Default: No

STUN Enable

Enables the use of STUN to discover NAT mapping.

Default: No

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the phone can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the phone detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

Default: No

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery. You can use a public STUN server or set up your own STUN server.

Default: Blank

EXT IP

External IP address to substitute for the actual IP address of phone in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, phone assumes this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line).

Default: Blank

EXT RTP Port Min

External port mapping number of the RTP Port Minimum number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range.

Default: 0

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Default: 15

Redirect Keep Alive

If enabled, the IP phone redirects the keepalive message when SIP_301_MOVED_PERMANENTLY is received as the registration response.

Provision Enable

Allows or denies resync actions.

Default: 160,159,66,150

Resync On Reset

The device performs a resync operation after power-up and after each upgrade attempt when set to Yes.

Default: Yes

Resync Random Delay

A random delay following the boot-up sequence before performing the reset, specified in seconds. In a pool of IP Telephony devices that are scheduled to simultaneously powered up, this introduces a spread in the times at which each unit sends a resync request to the provisioning server. This feature can be useful in a large residential deployment, in the case of a regional power failures.

Default: 2

Resync At (HHmm)

Time in 24-hour format (hhmm) to resync the device. When this parameter is provisioned, the Resync Periodic parameter is ignored.

Default: Blank

Resync At Random Delay

To avoid flooding the server with simultaneously resync requests from multiple phones set to resync at the same time, the phone triggers the resync up to ten minutes after the specified time.

The input value (in seconds) is converted to minutes.

The default value is 600 seconds (10 minutes). If the parameter value is set to less than 600, the default value is used.

Default: 600

Resync Periodic

Time in seconds between periodic resyncs. If this value is empty or zero, the device does not resync periodically.

Default: 3600

Resync Error Retry Delay

If a resync operation fails because the IP Telephony device was unable to retrieve a profile from the server, if the downloaded file is corrupt, or an internal error occurs, the device tries to resync again after a time specified in seconds.

If the delay is set to 0, the device does not try to resync again following a failed resync attempt.

Default: 3600

Forced Resync Delay

Forced resync delay typically takes place when it is time to a resync and you are in an active call. For example, if you set 5 minute for Periodic Resync and you place a call right after the resync, the resync happens while you are 6 minutes into the call (normal time of Resync + Forced Resync Delay).

Default: 14400

Resync From SIP

Controls requests for resync operations via a SIP NOTIFY event sent from the service provider proxy server to the IP Telephony device. If enabled, the proxy can request a resync by sending a SIP NOTIFY message containing the Event: resync header to the device.

Default: Yes

Resync After Upgrade Attempt

Enables or disables the resync operation after any upgrade occurs. If Yes is selected, sync is triggered.

Default: Yes

Resync Trigger 1

Resync Trigger 2

If the logical equation in these parameters evaluates to FALSE, Resync is not triggered even when Resync On Reset is set to TRUE. Only Resync via direct action URL and SIP notify ignores these Resync Trigger.

Default: Blank

Resync Fails On FNF

A resync is considered unsuccessful if a requested profile is not received from the server. This can be overridden by this parameter. When it is set to No, the device accepts a file-not-found response from the server as a successful resync.

Default: Yes

Profile Rule


Profile Rule B


Profile Rule C


Profile Rule D

Remote configuration profile rules evaluated in sequence. Each resync operation can retrieve multiple files, potentially managed by different servers.

Default: /$PSN.xml

DHCP Option To Use

DHCP options, delimited by commas, used to retrieve firmware and profiles.

Default: 66,160,159,150,60,43,125

DHCPv6 Option To Use

DHCP options, delimited by commas, used to retrieve firmware and profiles.

Default: 17,160,159

Log Request Msg

The message sent to the syslog server at the start of a resync attempt.

$PN $MAC –Requesting % $SCHEME://$SERVIP:$PORT$PATH

Log Success Msg

The syslog message issued upon successful completion of a resync attempt.

$PN $MAC -Successful Resync  % 

$SCHEME://$SERVIP:$PORT$PATH 

Log Failure Msg

The syslog message that is issued after a failed download attempt.

$PN $MAC -- Resync failed: $ERR

HTTP Report Method

Allows to select HTTP options. Options are POST and PUT.

Report Rule

Specifies the destination for the report of the current internal configuration that the phone sends to the provisioning server . The URL in this field can include an encryption key.

• If the report method is PUT, you can enter the URL for the report rule in this format:

  http://my_http_server/config-mpp.xml

• If the report method is POST, you can enter the URL for the report rule in this format:

  http://my_http_server/report_upload.php

User Configurable Resync

Allows a user to resync the phone from the phone screen.

Default: Yes

Custom CA Rule

The URL to download Custom CA.

Default: Blank

HTTP User Agent Name

Allows you to enter a name for HTTP user.

Default: Blank

PRT Upload Rule

Specifies the path to the PRT upload script. You can enter the path in the format:

or

If PRT Max Timer and PRT Upload Rule fields are empty, problem reports are not generated.

PRT Upload Method

Determines the method used to upload PRT logs to the remote server. Options are: HTTP POST and PUT.

Default: POST

PRT Max Timer

Determines at what interval (minutes) the phone starts generating problem report automatically. The interval range that you can set is 15 minutes to 1440 minutes.

Default: Empty

If PRT Max Timer and PRT Upload Rule fields are empty, problem reports are not generated.

PRT Name

Defines a name for the generated PRT file. Enter the name in the format:

GPP A - GPP P

The general purpose parameters GPP_* are used as free string, registers when configuring the Cisco IP phones to interact with a particular provisioning server solution. They can be configured to contain diverse values, including the following:

• Encryption keys

• URLs

• Multistage provisioning status information

• Post request templates

• Parameter name alias maps

• Partial string values, eventually combined into complete parameter values

Default: Blank

Cadence 1

Cadence script for distinctive ring 1.

Defaults to 60(2/4).

Cadence 2

Cadence script for distinctive ring 2.

Defaults to 60(.3/.2, 1/.2,.3/4).

Cadence 3

Cadence script for distinctive ring 3.

Defaults to 60(.8/.4,.8/4).

Cadence 4

Cadence script for distinctive ring 4.

Defaults to 60(.4/.2,.3/.2,.8/4).

Cadence 5

Cadence script for distinctive ring 5.

Defaults to 60(.2/.2,.2/.2,.2/.2,1/4).

Cadence 6

Cadence script for distinctive ring 6.

Defaults to 60(.2/.4,.2/.4,.2/4).

Cadence 7

Cadence script for distinctive ring 7.

Defaults to 60(4.5/4).

Cadence 8

Cadence script for distinctive ring 8.

Defaults to 60(0.25/9.75)

Cadence 9

Cadence script for distinctive ring 9.

Defaults to 60(.4/.2,.4/2).

Reorder Delay

Delay after far end hangs up before reorder (busy) tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds. Set to 255 to return the phone immediately to on-hook status and to not play the tone.

Interdigit Long Timer

Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds.

Default: 10

Interdigit Short Timer

Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.

Default: 3

Call Return Code

This code calls the last caller.

Defaults to *69.

Blind Transfer Code

Begins a blind transfer of the current call to the extension specified after the activation code.

Defaults to *88.

Cfwd All Act Code

Forwards all calls to the extension specified after the activation code.

Defaults to *72.

Cfwd All Deact Code

Cancels call forwarding of all calls.

Defaults to *73.

Cfwd Busy Act Code

Forwards busy calls to the extension specified after the activation code.

Defaults to *90.

Cfwd Busy Deact Code

Cancels call forwarding of busy calls.

Defaults to *91.

Cfwd No Ans Act Code

Forwards no-answer calls to the extension specified after the activation code.

Defaults to *92.

Cfwd No Ans Deact Code

Cancels call forwarding of no-answer calls.

Defaults to *93.

CW Act Code

Enables call waiting on all calls.

Defaults to *56.

CW Deact Code

Disables call waiting on all calls.

Defaults to *57.

CW Per Call Act Code

Enables call waiting for the next call.

Defaults to *71.

CW Per Call Deact Code

Disables call waiting for the next call.

Defaults to *70.

Block CID Act Code

Blocks caller ID on all outbound calls.

Defaults to *67.

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

Defaults to *68.

Block CID Per Call Act Code

Removes caller ID blocking on the next inbound call.

Defaults to *81.

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

Defaults to *82.

Block ANC Act Code

Blocks all anonymous calls.

Defaults to *77.

Block ANC Deact Code

Removes blocking of all anonymous calls.

Defaults to *87.

DND Act Code

Enables the do not disturb feature.

Defaults to *78.

DND Deact Code

Disables the do not disturb feature.

Defaults to *79.

Secure All Call Act Code

Makes all outbound calls secure.

Defaults to *16.

Secure No Call Act Code

Makes all outbound calls not secure.

Defaults to *17.

Secure One Call Act Code

Secure One Call Deact Code

Paging Code

The star code used for paging the other clients in the group.

Defaults to *96.

Call Park Code

The star code used for parking the current call.

Defaults to *38.

Call Pickup Code

The star code used for picking up a ringing call.

Defaults to *36.

Call Unpark Code

The star code used for picking up a call from the call park.

Defaults to *39.

Group Call Pickup Code

The star code used for picking up a group call.

Defaults to *37.

Referral Services Codes

These codes tell the IP phone what to do when the user places the current call on hold and is listening to the second dial tone.

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, and so on. Max total length is 79 chars. This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the phone to perform a blind transfer to a target number that is prepended by the service *code.

For example, after the user dials *98, the IP phone plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the phone sends a blind REFER to the holding party with the Refer-To target equals to *98<target_number>. This feature allows the phone to hand off a call to an application server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally processed by the IP phone. You can empty the corresponding *code that you do not want to the phone to process.

Feature Dial Services Codes

These codes tell the phone what to do when the user is listening to the first or second dial tone.

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, and so forth. The maximum total length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the phone to call the target number prepended by the *code. For example, after user dials *72, the phone plays a prompt tone awaiting the user to enter a valid target number. When a complete number is entered, the phone sends a INVITE to *72<target_number> as in a normal call. This feature allows the proxy to process features like call forward (*72) or BLock Caller ID (*67).

The *codes should not conflict with any of the other vertical service codes internally processed by the phone. You can empty the corresponding *code that you do not want to the phone to process.

You can add a parameter to each *code in Features Dial Services Codes to indicate what tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed tone parameters (note the use of back quotes surrounding the parameter without spaces)

• c = Cfwd Dial Tone

• d = Dial Tone

• m = MWI Dial Tone

• o = Outside Dial Tone

• p = Prompt Dial Tone

• s = Second Dial Tone

• x = No tones are place, x is any digit not used above

If no tone parameter is specified, the phone plays Prompt tone by default.

If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include it in this parameter. In that case, simple add that *code in the dial plan and the phone sends INVITE *73@..... as usual when user dials *73.

Service Annc Base Number

Defaults to blank.

Service Annc Extension Codes

Defaults to blank.

Prefer G711u Code

Makes this codec the preferred codec for the associated call.

Defaults to *017110.

Force G711u Code

Makes this codec the only codec that can be used for the associated call.

Defaults to *027110.

Prefer G711a Code

Makes this codec the preferred codec for the associated call.

Defaults to *017111

Force G711a Code

Makes this codec the only codec that can be used for the associated call.

Defaults to *027111.

Prefer G722 Code

Makes this codec the preferred codec for the associated call.

Defaults to *01722.

Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio.

Force G722 Code

Makes this codec the only codec that can be used for the associated call.

Defaults to *02722.

Only one G.722 call at a time is allowed. If a conference call is placed, a SIP re-invite message is sent to switch the calls to narrowband audio.

Prefer G722.2 Code

Makes this codec the preferred codec for the associated call.

Force G722.2 Code

Makes this codec the only codec that can be used for the associated call.

Prefer G729a Code

Makes this codec the preferred codec for the associated call.

Defaults to *01729.

Force G729a Code

Makes this codec the only codec that can be used for the associated call.

Defaults to *02729.

Prefer iLBC Code

Makes this codec the preferred codec for the associated call.

Force iLBC Code

Makes this codec the only codec that can be used for the associated call.

Prefer ISAC Code

Makes this codec the preferred codec for the associated call.

Force ISAC Code

Makes this codec the only codec that can be used for the associated call.

Prefer OPUS Code

Makes this codec the preferred codec for the associated call.

Force OPUS Code

Makes this codec the only codec that can be used for the associated call.

Set Local Date (mm/dd/yyyy)

Sets the local date (mm represents the month and dd represents the day). The year is optional and uses two or four digits.

Default: Blank

Set Local Time (HH/mm)

Sets the local time (hh represents hours and mm represents minutes). Seconds are optional.

Default: Blank

Time Zone

Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00, …, GMT+13:00.

Default: GMT-08:00

Time Offset (HH/mm)

This specifies the offset from GMT to use for the local system time.

Default: 00/00

Ignore DHCP Time Offset

When used with some routers that have DHCP with time offset values configured, the IP phone uses the router settings and ignores the IP phone time zone and offset settings. To ignore the router DHCP time offset value, and use the local time zone and offset settings, choose yes for this option. Choosing no causes the IP phone to use the router's DHCP time offset value.

Default: Yes.

Daylight Saving Time Rule

Enter the rule for calculating daylight saving time; it should include the start, end, and save values. This rule is comprised of three fields. Each field is separated by ; (a semicolon) as shown below. Optional values inside [ ] (the brackets) are assumed to be 0 if they are not specified. Midnight is represented by 0:0:0 of the given date.

This is the format of the rule: Start = <start-time>; end=<end-time>; save = <save-time>.

The <start-time> and <end-time> values specify the start and end dates and times of daylight saving time. Each value is in this format: <month> /<day> / <weekday>[/HH:[mm[:ss]]]

The <save-time> value is the number of hours, minutes, and/or seconds to add to the current time during daylight saving time. The <save-time> value can be preceded by a negative (-) sign if subtraction is desired instead of addition. The <save-time> value is in this format: [/[+|-]HH:[mm[:ss]]]

The <month> value equals any value in the range 1-12 (January-December).

The <day> value equals [+|-] any value in the range 1-31.

If <day> is 1, it means the <weekday> on or before the end of the month (in other words the last occurrence of < weekday> in that month).

Daylight Saving Time Rule (continued)

The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can also equal 0. If the <weekday> value is 0, this means that the date to start or end daylight saving is exactly the date given. In that case, the <day> value must not be negative. If the <weekday> value is not 0 and the <day> value is positive, then daylight saving starts or ends on the <weekday> value on or after the date given. If the <weekday> value is not 0 and the <day> value is negative, then daylight saving starts or ends on the <weekday> value on or before the date given. Where:

• HH stands for hours (0-23).

• mm stands for minutes (0-59).

• ss stands for seconds (0-59).

Default: 3/-1/7/2;end=10/-1/7/2;save=1.

Daylight Saving Time Enable

Enables Daylight Saving Time.

Default: Yes

Dictionary Server Script

Defines the location of the dictionary server, the languages available, and the associated dictionary. See Dictionary Server Script.

Default: Blank

Language Selection

Specifies the default language. The value must match one of the languages supported by the dictionary server. The script (dx value) is:

<Language_Selection ua="na">
</Language_Selection>

Default: Blank

The maximum number of characters is 512. For example:

<Language_Selection ua="na"> Spanish
</Language_Selection>

Locale

Choose the locale that should be set in the HTTP Accept-Language header

Default: en-US

Station Name

Name of the phone.

Station Display Name

Name to identify the phone; appears on the phone screen. You can use spaces in this field and the name does not have to be unique.

Voice Mail Number

A phone number or URL to check voice mail.

Default: None

Select Logo

Select from None, PNG Picture, or Text Logo.

Default: None

Bandwidth Allowance

Enables you to restrict the maximum amount of information that the phone can transmit or receive. Options are:

• Auto

• 2 Mbps

• 1 Mbps

• 750 Kbps

• 500 Kbps

• 250 Kbps

Default: Auto

Extension

Specifies the n extension to be assigned to Line Key n.

Default: n

Short Name

Specifies the user name for Line Key.

Default: $USER

Share Call Appearance

Specifies whether the incoming call appearance is shared with other phones or it is private.

Extended Function

Use to assign Busy Lamp Field, Call Pickup, and Speed Dial Functions to Idle Lines on the IP phone.

Line ID Mapping

Specifies the shared call appearance line ID mapping. If Vertical First is set, the second call makes the next available line ID LED flash. If Horizontal first is set, the second call will make the same LED flash on which the first call is received. Also, the behavior is same for both outgoing and incoming calls.

Default: Horizontal First

SCA Barge-In Enable

Enables the SCA Barge-In.

Default: No

SCA Sticky Auto Line Seize

If enabled, restricts to automatically pick up an incoming call on a shared line when you take the phone off-hook.

Call Appearances Per Line

This parameter allows you to choose the number of calls per line button. You can choose a value from 2 to 10.

Default: 2

Conference Serv

Enable or disable three-way conference service.

Default: Yes

Attn Transfer Serv

Enable or disable attended-call-transfer service.

Default: Yes

Blind Transfer Serv

Enable or disable blind-call-transfer service.

Default: Yes

DND Serv

Enable or disable do not disturb service.

Default: Yes

Block ANC Serv

Enable or disable block-anonymous-call service.

Default: Yes

Block CID Serv

Enable or disable blocking outbound Caller-ID service.

Default: Yes

Secure Call Serv

Enable or disable secured call services.

Default: Yes

Cfwd All Serv

Enable or disable call-forward-all service.

Default: Yes

Cfwd Busy Serv

Enable or disable call-forward-on-busy service.

Default: Yes

Cfwd No Ans Serv

Enable or disable call-forward-no-answer service.

Default: Yes

Paging Serv

Enable or disable paging service on the phone.

Default: Yes

Call Park Serv

Enable or disable call park services on the phone.

Default: Yes

Call Pick Up Serv

Enable or disable call pick up services on the phone.

Default: Yes

ACD Login Serv

Enable or disable ACD login services on the phone.

Default: Yes

Group Call Pick Up Serv

Enable or disable group call pick up services on the phone.

Default: Yes

Service Annc Serv

Enable or disable the vertical service announcement services on the phone.

Default: No

Call Recording Serv

Enable or disable the call recording services on the phone.

Default: No

Video Serv

Enable or disable video services on the phone.

When enabled, the Video Enable field is displayed in the User tab. When disabled, the Video Enable field is not displayed.

Default: No

Ring1 to Ring12

Ring tone scripts for different rings.

Silent Ring Duration

Controls the duration of the silent ring.

For example, if the parameter is set to 20 seconds, the phone plays the silent ring for 20 seconds then sends 480 response to INVITE message.

EM Enable

Options to enable or to disable the extension mobility support for the phone.

Default: No

EM User Domain

Name of the domain for the phone or the authentication server.

Default: Blank

Inactivity Timer(m)

Specifies the duration for which the extension mobility remains inactive.

Countdown Timer(s)

Specifies the duration for which it waits before it logs out.

Default: 10

Preferred Password Input Mode

Options to specify the password input method of extension mobility PIN. Options are: Alpha-numeric and Numeric.

Default: Alpha-numeric

XSI Host Server

Enter the name of the server; for example, xsi.iop1.broadworks.net.

Default: Blank

XSI Authentication Type

Determines the XSI authentication type. Select Login Credentials to authenticate access with XSI id and password. Select SIP Credentials to authenticate access with the register user ID and password of the SIP account registered on the phone.

Default: Login Credentials

Login User ID

BroadSoft User ID of the phone user; for example, johndoe@xdp.broadsoft.com.

Enter SIP Auth ID when you select Login Credentials or SIP Credentials for XSI authentication type.

When you choose SIP Auth ID as SIP Credentials, you must enter Login User ID. Without Login User ID, the BroadSoft directory will not appear under the phone Directory list.

Default: Blank

Login Password

Alphanumeric password associated with the User ID.

Enter login password, when you select Login Credentials for XSI authentication type.

Default: Blank

SIP Auth ID

The registered user ID of the SIP account registered on the phone.

Enter SIP Auth ID when you select SIP Credentials for XSI authentication type.

SIP Password

The password of the SIP account registered on the phone.

Enter SIP password when you select SIP Credentials for XSI authentication type.

Directory Enable

Enables BroadSoft directory for the phone user. Select Yes to enable the directory and select No to disable it.

Default: No

Directory Name

Name of the directory. Displays on the phone as a directory choice.

Default: Blank

Directory Type

Select the type of BroadSoft directory:

Enterprise: Allows users to search on last name, first name, user or group ID, phone number, extension, department, or email address.

Group: Allows users to search on last name, first name, user ID, phone number, extension, department, or email address.

Personal: Allows users to search on last name, first name, or telephone number.

Default: Enterprise

CallLog Enable

Enables to log XSI calls. Select Yes to log XSI calls and select No to disable it.

Default: No

XMPP Enable

Set to Yes to enable the BroadSoft XMPP directory for the phone user.

Default: No

Server

Enter the name of the XMPP server; for example, xsi.iop1.broadworks.net.

Default: Blank

Port

Server port for the directory.

Default: Blank

User ID

BroadSoft User ID of the phone user; for example, johndoe@xdp.broadsoft.com.

Default: Blank

Password

Alphanumeric password associated with the User ID.

Default: Blank

Login Invisible

When enabled, the user's presence information is not published when the user signs in.

Default: No

Retry Intvl

Interval, in seconds, to allow a reconnect without a log in after the client disconnects from the server. After this interval, the client needs to reauthenticate.

Default: 30

XML Directory Service Name

Name of the XML Directory. Displays on the user’s phone as a directory choice

Default: Blank

XML Directory Service URL

URL where the XML Directory is located.

Default: Blank

XML User Name

XML service username for authentication purposes

Default: Blank

XML Password

XML service password for authentication purposes

Default: Blank

Group Paging Script

Enter a string to configure group paging and priority paging (out of band paging) that does not required the phone registration.

LDAP Dir Enable

Choose Yes to enable LDAP.

Default: No

Corp Dir Name

Enter a free-form text name, such as “Corporate Directory.”

Default: Blank

Server

Enter a fully qualified domain name or IP address of an LDAP server in the following format:

nnn.nnn.nnn.nnn

Enter the host name of the LDAP server if the MD5 authentication method is used.

Default: Blank

Search Base

Specify a starting point in the directory tree from which to search. Separate domain components [dc] with a comma. For example:

dc=cv2bu,dc=com

Default: Blank

Client DN

Enter the distinguished name domain components [dc]; for example:

dc=cv2bu,dc=com

If you are using the default Active Directory schema (Name(cn)->Users->Domain), an example of the client DN follows:

cn=”David Lee”,dc=users,dc=cv2bu,dc=com

Default: Blank

User Name

Enter the username for a credentialed user on the LDAP server.

Default: Blank

Password

Enter the password for the LDAP username.

Default: Blank

Auth Method

Select the authentication method that the LDAP server requires. Choices are:

None—No authentication is used between the client and the server.

Simple—The client sends its fully-qualified domain name and password to the LDAP server. Might present security issues.

Digest-MD5—The LDAP server sends authentication options and a token to the client. The client returns an encrypted response that is decrypted and verified by the server.

Default: None

Last Name Filter

This defines the search for surnames [sn], known as last name in some locations. For example, sn:(sn=*$VALUE*). This search allows the provided text to appear anywhere in a name: beginning, middle, or end.

Default: Blank

First Name Filter

This defines the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This search allows the provided text to appear anywhere in a name: beginning, middle, or end.

Default: Blank

Search Item 3

Additional customized search item. Can be blank if not needed.

Default: Blank

Search Item 3 Filter

Customized filter for the searched item. Can be blank if not needed.

Default: Blank

Search Item 4

Additional customized search item. Can be blank if not needed.

Default: Blank

Search Item 4 Filter

Customized filter for the searched item. Can be blank if not needed.

Default: Blank

Display Attrs

Format of LDAP results displayed on phone, where:

• a—Attribute name

• cn—Common name

• sn—Surname (last name)

• telephoneNumber—Phone number

• n—Display name

For example, n=Phone causes “Phone:” to be displayed in front of the phone number of an LDAP query result when the detail soft button is pressed.

• t—type

When t=p, that is, t is of type phone number, the retrieved number can be dialed. Only one number can be made dialable. If two numbers are defined as dialable, only the first number is used. For example, a=ipPhone, t=p; a=mobile, t=p;

This example results in only the IP Phone number being dialable and the mobile number is ignored.

• p—phone number

When p is assigned to a type attribute, example t=p, the retrieved number is dialable by the phone.

For example, a=givenName,n=firstname;a=sn,n=lastname;a=cn,n=cn;a=telephoneNumber,n=tele,t=p

Default: Blank

Number Mapping

Can be blank if not needed.

If you do not manipulate the number in this fashion, a user can use the Edit Dial feature to edit the number before dialing out.

Default: Blank

Programmable Softkey Enable

Enables programmable softkeys.

Idle Key List

Softkeys that display when the phone is idle.

Off Hook Key List

Softkeys that display when the phone is off-hook.

Dialing Input Key List

Softkeys that display when the user must enter dialing data.

Progressing Key List

Softkeys that display when a call is attempting to connect.

Connected Key List

Softkeys that display when a call is connected.

Start-Xfer Key List

Softkeys that display when a call transfer has been initiated.

Start-Conf Key List

Softkeys that display when a conference call has been initiated.

Conferencing Key List

Softkeys that display when a conference call is in progress.

Releasing Key List

Softkeys that display when a call is released.

Hold Key List

Softkeys that display when one or more calls are on hold.

Ringing Key List

Softkeys that display when a call is incoming.

Shared Active Key List

Softkeys that display when a call is active on a shared line.

Shared Held Key List

Softkeys that display when a call is on hold on a shared line.

PSK 1 through PSK 16

Programmable softkey fields. Enter a string in these fields to configure softkeys that display on the phone screen. You can create softkeys for speed dials to numbers or extensions, vertical service activation codes (* codes), or XML scripts.

Dial Plan

Dial plan script for the selected extension.

Separate each parameter with a semi-colon (;).

Caller ID Map

Inbound caller ID numbers can be mapped to a different string. For example, a number that begins with +44xxxxxx can be mapped to 0xxxxxx. This feature has the same syntax as the Dial Plan parameter. With this parameter, you can specify how to map a caller ID number for display on screen and recorded into call logs.

Enable URI Dialing

Enables or disables URI dialing.

Emergency Number

Enter a comma-separated list of emergency numbers. When one of these numbers is dialed, the unit disables processing of CONF, HOLD, and other similar softkeys or buttons to avoid accidentally putting the current call on hold. The phone also disables hook flash event handling.

Only the far end can terminate an emergency call. The phone is restored to normalcy after the call is terminated and the receiver is back on-hook.

Maximum number length is 63 characters. Defaults to blank (no emergency number).

Subscribe Expires

Specifies how long the subscription remains valid. After the specified period of time elapses, the Cisco Attendant Console initiates a new subscription.

Default: 1800

Subscribe Retry Interval

Specifies the length of time to wait to try again if the subscription fails.

Default: 30

Number of Units

Specifies the number of Cisco Attendant Console units.

Default: 0

Subscribe Delay

Length of delay before attempting to subscribe.

Default: 1

BLF List URL

Domain name or user name that is defined in the Broadsoft server for the phone.

Default: Blank

Use Line Keys For BLF List

Options to enable or disable the line keys for BLF.

Default: No

Call Pickup Audio Notification

By default, this parameter is set to No. If you set it to Yes, the phone plays the Call Pickup tone when there are incoming calls to any of the lines that the user is monitoring with the Call Pickup function.

Default: No

Attendant Console LCD Brightness

The contrast between the text, lines, and background on the attendant console display. Enter a number value from 1 to 30. The higher the number, the greater the contrast on the display.

Default: 12

BXfer to Starcode Enable

When set to Yes, the phone performs a blind transfer when the *code is defined in a speed dial extended function,. If set to No, the current call is held and a new call is started to the speed dial destination.

Default: No

BXfer On Speed Dial Enable

When set to Yes, the phone performs a blind transfer when the speed dial function key is selected. When set to no, the current connected call is held and a new call to the speed dial destination is started.

For example, when a user parks a call using the speed dial function, if the parameter is enabled, a blind transfer is performed to the parking lot. If the parameter is not enabled, an attended transfer is performed to the parking lot.

Default: No

BLF Label Display Mode

Options to select a mode which displays on the phone screen for BLF.

Default: Blank