The documentation set for this product strives to use bias-free language. For the purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on RFP documentation, or language that is used by a referenced third-party product. Learn more about how Cisco is using Inclusive Language.
You can view a variety of information about the phone using the phone status menu on the phone and the phone web pages. This information includes:
This chapter describes the information that you can obtain from the phone web page. You can use this information to remotely monitor the operation of a phone and to assist with troubleshooting.
The following sections describes how to view model information, status messages, and network statistics on the Cisco IP Phone.
Model Information: Displays hardware and software information about the phone.
Status menu: Provides access to screens that display the status messages, network statistics, and statistics for the current call.
You can use the information that displays on these screens to monitor the operation of a phone and to assist with troubleshooting.
You can also obtain much of this information, and obtain other related information, remotely through the phone web page.
You can access the Call Statistics screen on the phone to display counters, statistics, and voice-quality metrics of the most recent call.
Note | You can also remotely view the call statistics information by using a web browser to access the Streaming Statistics web page. This web page contains additional RTCP statistics that are not available on the phone. |
A single call can use multiple voice streams, but data is captured for only the last voice stream. A voice stream is a packet stream between two endpoints. If one endpoint is put on hold, the voice stream stops even though the call is still connected. When the call resumes, a new voice packet stream begins, and the new call data overwrites the former call data.
To display the Call Statistics screen for information about the latest voice stream, follow these steps:
The following table describes the items on the Call Statistics screen.
Item |
Description |
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---|---|---|---|
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. |
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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. |
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Receiver Size |
Size of voice packets, in milliseconds, in the receiving voice stream (RTP streaming audio). |
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Sender Size |
Size of voice packets, in milliseconds, in the transmitting voice stream. |
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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.
|
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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. |
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Max Jitter |
Maximum jitter, in milliseconds, that was observed since the receiving voice stream opened. |
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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). |
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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. |
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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. |
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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. |
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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. |
This section describes the information that you can obtain from the phone web page. You can use this information to remotely monitor the operation of a phone and to assist with troubleshooting.
Status
Secondary NTP Serve |
Indicates the type of internet connection for the phone: |
|
For information about Reboot History, see the Reboot Reasons.
Download Status |
Displays the downloaded locale package status. |
Download URL |
Displays the location from where the local package is downloaded. |
Parameter |
Description |
---|---|
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). |
Parameter |
Description |
---|---|
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. |
These fields display the status of provisioning using a custom Certificate Authority (CA).
Indicates whether provisioning using a custom CA succeeded or failed: |
|
Custom CA certificates are configured in the Provisioning tab. For more information about custom CA certificates, see the Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide.
Provisioning Profile |
Displays the profile file name of the phone. |
Provisioning Failure Reason |
Displays the reason for the failure of provisioning of the phone. |
Note | The Upgrade and Provisioning Status are displayed in reverse chronological order (like reboot history) displaying status with time and reason. |
Debug Info
Displays the syslog output of the phone in the reverse order, where messages is the latest one. The display includes hyperlinks to individual log files. The console log files include debug and error messages received on the phone and the time stamp reflects UTC time, regardless of time zone settings.
Attendant Console Status
Enter the programming information for each line key for the Attendant Console unit.
Unit Enable |
Indicates whether the key expansion module that is added to the phone is enabled. |
Unit Online |
Indicates whether the key expansion module that is added to the phone is active. |
HW Version |
Displays the hardware version of the key expansion module that is added to the phone.. |
SW Version |
Displays the software version of the key expansion module that is added to the phone. |
Network Statistics
Total number of broadcast packets that the phone transmitted. |
|
Total number of multicast packets that the phone transmitted. |
|
System
The network domain of the Cisco IP Phone. If you are using LDAP, see the LDAP Configuration. |
|
DNS Query Mode |
Specified mode of DNS query.
|
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:
|
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. |
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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. |
|
IP address or name of the primary NTP server used to synchronize its time. |
|
IP address or name of the secondary NTP server used to synchronize its time. |
|
Enable SSLv3 |
Choose Yes to enable SSLv3. Choose No to disable. Default: No |
SIP
RFC 3261 T1 value (RTT estimate) that can range from 0 to 64 seconds. |
|
RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses) that can range from 0 to 64 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 request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000. |
|
ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000. |
|
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. |
Interval to wait before the Cisco IP Phone retries registration after failing during the last registration.The range is from 1 to 2147483647 See the note below for additional details. |
|
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. See the note below for additional details. |
|
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. |
|
Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying REGISTER after a failure. |
|
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. |
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 |
INFOREQ Dynamic Payload |
INFOREQ Dynamic Payload type. |
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 |
Provisioning
For information about the Provisioning page, see the Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide.
The URL to download Custom CA. Default: Blank |
Allows you to enter a name for HTTP user. Default: Blank |
Method used to upload PRT logs to the remote server. Can be either HTTP POST or PUT. |
Regional
Parameter |
Description |
---|---|
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 |
Parameter |
Description |
---|---|
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. |
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. |
Makes this codec the preferred codec for the associated call. Defaults to *017110. |
|
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. |
Defines the location of the dictionary server, the languages available, and the associated dictionary. See the Dictionary Server Script. Default: Blank |
|
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> The maximum number of characters is 512. For example: <Language_Selection ua="na"> Spanish </Language_Selection> |
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Choose the locale that should be set in the HTTP Accept-Language header Default: en-US |
Phone
Station Name |
Name of the phone. |
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. |
|
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. |
Options to enable or to disable the extension mobility support for the phone. Default: No |
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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 is 10 |
Set to Yes to enable BroadSoft directory for the phone user. |
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Enter the name of the server; for example, xsi.iop1.broadworks.net. Default: Blank |
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Name of the directory. Displays on the phone as a directory choice. Default: Blank |
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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. |
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BroadSoft User ID of the phone user; for example, johndoe@xdp.broadsoft.com. Default: Blank |
|
Alphanumeric password associated with the User ID. Default: Blank |
Name of the XML Directory. Displays on the user’s phone as a directory choice Default: Blank |
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URL where the XML Directory is located. Default: Blank |
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XML service username for authentication purposes Default: Blank |
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XML service password for authentication purposes Default: Blank |
Default: No |
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Enter a free-form text name, such as “Corporate Directory.” Default: Blank |
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Enter a fully qualified domain name or IP address of an LDAP server in the following format: Enter the host name of the LDAP server if the MD5 authentication method is used. Default: Blank |
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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 |
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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 |
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User Name |
Enter the username for a credentialed user on the LDAP server. Default: Blank |
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Password |
Enter the password for the LDAP username. Default: Blank |
||
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 |
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This defines the search for surnames [sn], known as last name in some locations. For example, sn:(sn=*$VALUE*). This search allows the provided text to appear anywhere in a name: beginning, middle, or end. Default: Blank |
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This defines the search for the common name [cn]. For example, cn:(cn=*$VALUE*). This search allows the provided text to appear anywhere in a name: beginning, middle, or end. Default: Blank |
|||
Additional customized search item. Can be blank if not needed. Default: Blank |
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Customized filter for the searched item. Can be blank if not needed. Default: Blank |
|||
Additional customized search item. Can be blank if not needed. Default: Blank |
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Customized filter for the searched item. Can be blank if not needed. Default: Blank |
|||
Format of LDAP results displayed on phone, where: 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. 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. 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 |
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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. |
Softkeys that display when a call is active on a shared line. |
|
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. |
User
Specifies the time delay (in seconds), that a ring splash is heard on an active call when another call was placed on hold. Default: 0 |
|
Cfwd All Dest |
Enter the extensions to which the call is forwarded. |
Enter the extensions to forward calls to when the line is busy. |
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Enter the extension to forward calls to when the call is not answered. |
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Enter the delay in time (in seconds) to wait before forwarding a call that is unanswered. |
You can configure speed dials on the Cisco IP Phone from the LCD GUI or the web GUI.
Speed Dial 2 to 9: Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. Press the digit key (2-9) to dial out the assigned number.
Enables or disables the Call Waiting service. Default: Yes |
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Enables or disables the Block CID service. Default: No |
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Enables or disables the Block ANC service. Default: No |
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DND Setting |
Enables or disables the DND settings options for a user. |
Handset LED Alert |
Enables or disables LED alert on the handset. Options are: Voicemail and Voicemail, Missed Call. Default: Voicemail |
Secure Call Setting |
Enables or disables Secure Call. Default: No |
Auto Answer Page |
Enables or disables automatic answering of paged calls. Default: Yes |
Preferred Audio Device |
Choose the type of audio that the phone will use. Options are: Speaker and Headset. Default: None |
Choose the time format for the phone (12 or 24 hour). Default: 12hr |
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Choose the date format for the phone (month/day or day/month). Default: month/day |
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Miss Call Shortcut |
Enables or disables the option for creating a missed call shortcut. |
Alert Tone Off |
Enables or disables the alert tone. |
Log Missed Calls for EXT (n) |
Enables or disables the missed calls logs for a specific extension. |
Sets the default volume for the ringer. Default: 9 |
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Sets the default volume for the speakerphone. Default: 8 |
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Handset Volume |
Sets the default volume for the handset. Default: 10 |
Headset Volume |
Sets the default volume for the headset. Default: 10 |
Select the number of minutes before the back light should turn off (1m, 5m, or 30m) or Always On. Default: 5m |
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Brightness |
Enter a number value from 1 to 15. The higher the number, the greater the brightness on the IP phone screen. Default: 10 |
Extension
In a configuration profile, the Line parameters must be appended with the appropriate numeral to indicate the line to which the setting applies. For example:
[1] to specify line one [2] to specify line two
To enable this line for service, select yes. Otherwise, select No. |
Indicates whether this extension is to be shared with other Cisco IP phones or private. |
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The user identified assigned to the shared line appearance. Default: Blank |
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Subscription Expires |
Number of seconds before the SIP subscription expires. Before the subscription expiration, the phone gets NOTIFY messages from the SIP server on the status of the shared phone extension. Default: 3600 |
Restrict MWI |
When enabled, the message waiting indicator lights only for messages on private lines. Default: No |
To use externally mapped IP addresses and SIP/ RTP ports in SIP messages, select yes. Otherwise, select no. |
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To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. |
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Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. |
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NAT Keep Alive Dest |
Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. |
Time of service (ToS)/differentiated services (DiffServ) field value in UDP IP packets carrying a SIP message. Defaults to 0x68. |
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RTP ToS/DiffServ Value |
ToS/DiffServ field value in UDP IP packets carrying RTP data. Defaults to 0xb8. |
SIP Port |
Port number of the SIP message listening and transmission port. Default: 5060 |
Support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests. Select Yes to enable. |
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The Cisco IP Phone authenticates the sender when it receives a NOTIFY message with the following requests: |
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SIP Proxy-Require |
The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided. |
The Remote-Party-ID header to use instead of the From header. Select Yes to enable. |
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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. For the Referror Bye Delay, enter the appropriate period of time in seconds. Default: 4 |
Indicates the refer-to target. Select Yes to send the SIP Refer to the contact. |
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Referee Bye Delay |
For the Referee Bye Delay, enter the appropriate period of time in seconds. Default: 0 |
Refer Target Bye Delay |
For the Refer Target Bye Delay, enter the appropriate period of time in seconds. Default: 0 |
When enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select Yes. Otherwise, select No. |
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When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. To enable this feature, select Yes. |
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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). |
Set G729 annexb |
Configure G.729 Annex B settings. |
Set iLBC mode |
Select iLBC 20ms or 30ms frame size mode. Default: 20 |
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 |
Blind Attn-Xfer Enable |
Enables the phone to perform an attended transfer operation by ending the current call leg and performing a blind transfer of the other call leg. If this feature is disabled, the phone performs an attended transfer operation by referring the other call leg to the current call leg while maintaining both call legs. To use this feature, select Yes. Otherwise, select No. |
Indicates whether the Message Waiting Indicator on the phone is lit. This parameter toggles a message from the SIP proxy to indicate if a message is waiting. |
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Auth Page |
Specifies whether to authenticate the invite before auto answering a page. Default: No |
Type of ring heard. Choose from No Ring or 1 through 10. Ring options are Sunlight, Chirp 1, Chirp 2, Delight, Evolve, Mellow, Mischief, Reflections, Ringer, Ascent, Are you there, and Chime. |
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Auth Page Realm |
Identifies the Realm part of the Auth that is accepted when the Auth Page parameter is set to Yes. This parameter accepts alphanumeric characters. |
URL used to join into a conference call, generally in the form of the word conference or user@IPaddress:port. |
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Auth Page Password |
Identifies the password used when the Auth Page parameter is set to Yes. This parameter accepts alphanumeric characters. |
Mailbox ID |
Identifies the voice mailbox number/ID for the phone. |
Identifies the SpecVM server for the phone, generally the IP address, and port number of the VM server. |
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The expiration time, in seconds, of a subscription to a voice mail server. |
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Broadsoft ACD |
Enables support for basic BroadSoft Automatic CallDistribution (ACD). The supported values for this option are Yes and No. Default: No |
Auto Ans Page On Active Call |
Determines the behavior of the phone when a page call arrives. |
Enable/disable the Feature Key synchronization. Applies to DND and Call Forward All features. |
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Call Park Monitor Enable |
BroadSoft server only specific feature. If call park is enabled on the server or on any of the programmable line key, you need to enable this field for call park notification to work. Default: No |
Enable Broadsoft Hoteling |
When this parameter is set to yes, the phone sends out subscription message (without body) to the server. Default: No |
Hoteling Subscription Expires |
An expiration value that is added in the subscription message. Default value is 3600. |
SIP proxy server and port number set by the service provider for all outbound requests. For example: 192.168.2.100:6060. |
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All outbound requests are sent as the first hop. Enter an IP address or domain name. |
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This feature provides fast fall back when there is network partition at the Internet or when the primary proxy (or primary outbound proxy) is not responsive or available. The feature works well in a Verizon deployment environment as the alternate proxy is the Integrated Service Router (ISR) with analog outbound phone connection. Enter the proxy server addresses and port numbers in these fields. After the phone is registered to the primary proxy and the alternate proxy (or primary outbound proxy and alternate outbound proxy), the phone always sends out INVITE and Non-INVITE SIP messages (except registration) via the primary proxy. The phone always registers to both the primary and alternate proxies. If there is no response from the primary proxy after timeout (per the SIP RFC spec) for a new INVITE, the phone attempts to connect with the alternate proxy. The phone always tries the primary proxy first, and immediately tries the alternate proxy if the primary is unreachable. Active transactions (calls) never fall back between the primary and alternate proxies. If there is fall back for a new INVITE, the subscribe/notify transaction will fall back accordingly so that the phone's state can be maintained properly. You must also set Dual Registration in the Proxy and Registration section to Yes. |
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Use OB Proxy In Dialog |
Determines whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the Use Outbound Proxy field is set to No or if the Outbound Proxy field is empty. Default: Yes |
Enables periodic registration with the proxy. This parameter is ignored if a proxy is not specified. To enable this feature, select Yes. |
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Enables making outbound calls without successful (dynamic) registration by the phone. If set to No, the dial tone plays only when registration is successful. To enable this feature, select Yes. |
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Defines how often the phone renews registration with the proxy. If the proxy responds to a REGISTER with a lower expires value, the phone renews registration based on that lower value instead of the configured value. If registration fails with an “Expires too brief” error response, the phone retries with the value specified in the Min-Expires header of the error. |
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Ans Call Without Reg |
If enabled, the user does not have to be registered with the proxy to answer calls. Default: No |
Enables DNS SRV lookup for the proxy and outbound proxy. To enable this feature, select Yes. Otherwise, select No. |
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DNS SRV Auto Prefix |
Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Default: No |
Sets the delay after which the phone retries from the highest priority proxy (or outbound proxy) after it has failed over to a lower priority server. The phone should have the primary and backup proxy server list from a DNS SRV record lookup on the server name. It needs to know the proxy priority; otherwise, it does not retry. |
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Proxy Redundancy Method |
Select Normal or Based on SRV Port. The phone creates an internal list of proxies returned in the DNS SRV records. If you select Normal, the list contains proxies ranked by weight and priority. If you select Based on SRV Port, the phone uses normal, then inspects the port number based on the first-listed proxy port. Default: Normal |
Set to Yes to enable the Dual registration/Fast Fall back feature. To enable the feature you must also configure the alternate proxy/alternate outbound proxy fields in the Proxy and Registration section. |
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Auto Register When Failover |
If set to No, the fallback happens immediately and automatically. If the Proxy Fallback Intvl is exceeded, all the new SIP messages go to the primary proxy. If set to Yes, the fallback happens only when current registration expires, which means only a REGISTER message can trigger fallback. For example, when the value for Register Expires is 3600 seconds and Proxy Fallback Intvl is 600 seconds, the fallback is triggered 3600 seconds later and not 600 seconds later. When the value for Register Expires is 600 seconds and Proxy Fallback Intvl is 1000 seconds, the fallback is triggered at 1200 seconds. After successfully registering back to primary server, all the SIP messages go to primary server. |
SIP URI |
The parameter by which the user agent will identify itself for this line. If this field is blank, the actual URI used in the SIP signaling should be automatically formed as: sip:UserName@Domain where UserName is the username given for this line in the User ID, and Domain is the domain given for this profile in the User Agent Domain. If the User Agent Domain is an empty string, then the IP address of the phone should be used for the domain. If the URI field is not empty, but if a SIP or SIPS URI contains no @ character, the actual URI used in the SIP signaling should be automatically formed by appending this parameter with an @ character followed by the IP address of the device. |
Preferred codec for all calls. The actual codec used in a call still depends on the outcome of the codec negotiation protocol. Select one of the following: |
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Select No to use any code. Select Yes to use only the preferred codes. When you select Yes, calls fail if the far end does not support the preferred codecs. |
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Codec to use if the second codec fails. |
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To enable use of the G.729a codec at 8 kbps, select Yes. Otherwise, select No. |
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G722.2 Enable |
Enables use of the G.722.2 codec. Default: No |
OPUS Enable |
Enables the use of OPUS codec. Default: Yes |
To enable silence suppression so that silent audio frames are not transmitted, select Yes. Otherwise, select No. |
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The method for transmitting DTMF signals to the far end. The options are: |
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Codec Negotiation |
When set to Default, the Cisco IP phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the Cisco IP phone responds listing all the codecs that the phone supports. The default value is Default, or to respond with the preferred codec only. |
Encryption Method |
Encryption method to be used during secured call. Options are AES 128 and AES 256 GCM Default: 128. |
Dial plan script for the selected extension. Separate each parameter with a semi-colon (;). |
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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. |
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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). |
Att Console
Note | The attendant console tab, labeled Att Console, is only available in mode. |
Specifies how long the subscription remains valid. After the specified period of time elapses, the Cisco Attendant Console initiates a new subscription. Default: 1800 |
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Subscribe Retry Interval |
Specifies the length of time to wait to try again if the subscription fails. Default: 30 |
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 |
Options to enable or disable the line keys for BLF. Default: No |
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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 |
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 |
Options to select a mode which displays on the phone screen for BLF. Default: Blank |
Enter the programming information for each line key for the Attendant Console unit.
Unit Enable |
Indicates whether the key expansion module that is added to the phone is enabled. |
Unit Online |
Indicates whether the key expansion module that is added to the phone is active. |
HW Version |
Displays the hardware version of the key expansion module that is added to the phone.. |
SW Version |
Displays the software version of the key expansion module that is added to the phone. |
TR-069
Parameter |
Description |
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Enable TR-069 |
Settings that enables or disables the TR-069 function. |
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ACS URL |
URL of the ACS that uses the CPE WAN Management Protocol. This parameter must be in the form of a valid HTTP or HTTPS URL. The host portion of this URL is used by the CPE to validate the ACS certificate when it uses SSL or TLS. |
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ACS Username |
Username that authenticates the CPE to the ACS when ACS uses the CPE WAN Management Protocol. This username is used only for HTTP-based authentication of the CPE. If the user name is not configured, admin is used as default. |
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ACS Password |
Password to access to the ACS for a specific user. This password is used only for HTTP-based authentication of the CPE. If the password is not configured, admin is used as default. |
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ACS URL In Use |
URL of the ACS that is currently in use. This is a read-only field. |
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Connection Request URL |
URL of the ACS that makes the connection request to the CPE. |
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Connection Request Username |
Username that authenticates the ACS that makes the connection request to the CPE. |
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Connection Request Password |
Password used to authenticate the ACS that makes a connection request to the CPE. |
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Periodic Informal Interval |
Duration in seconds of the interval between CPE attempts to connect to the ACS when Periodic Inform Enable is set to yes. Default value is 20 seconds. |
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Periodic Inform Enable |
Settings that enables or disables the CPE connection requests. Default value is Yes. |
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TR-069 Traceability |
Settings that enables or disables TR-069 transaction logs. The default value is No. |
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CWMP V1.2 Support |
Settings that enables or disables CPE WAN Management Protocol (CWMP) support. If set to disable, the phone does not send any Inform messages to the ACS nor accept any connection requests from the ACS. Default value is Yes. |
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TR-069 VoiceObject Init |
Settings to modify voice objects. Select Yes to initialize all voice objects to factory default values or select No to retain the current values. |
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TR-069 DHCPOption Init |
Settings to modify DHCP settings. Select Yes to initialize the DHCP settings from the ACS or select No to retain the current DHCP settings. |
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TR-069 Fallback Support |
Settings that enables or disables the TR-069 fallback support. If the phone attempts to discover the ACS with DHCP and is unsuccessful, the phone next uses DNS to resolve the ACS IP address. |
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BACKUP ACS URL |
Backup URL of the ACS that uses the CPE WAN Management Protocol. This parameter must be in the form of a valid HTTP or HTTPS URL. The host portion of this URL is used by the CPE to validate the ACS certificate when it uses SSL or TLS. |
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BACKUP ACS User |
Backup username that authenticates the CPE to the ACS when ACS uses the CPE WAN Management Protocol. This username is used only for HTTP-based authentication of the CPE. |
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BACKUP ACS Password |
Backup password to access to the ACS for a specific user. This password is used only for HTTP-based authentication of the CPE. |
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Displays the call history for the phone. To change the information displayed, select the type of call history from the following tabs:
All Calls
Missed
Received
Placed
Select Add to Directory to add the call information to your Personal Directory.
The Personal Directory allows a user to store a set of personal numbers. Directory entries can include the following contact information:
No. (the directory number)
Name
Work
Mobile
Home
Speed Dials
To edit contact information, click Edit Contacts.