- Phone Information and Display Settings
- Call Features Configuration
- Enable Call Transfer
- Call Forward
- Enable Conferencing
- Enable Remote Call Recording with SIP REC
- Enable Remote Call Recording with SIP INFO
- Configure Missed Call Indication with the Configuration Utility
- Enable Do Not Disturb
- Configure Synchronization of DND and Call Forward
- Configure Star Codes for DND
- Set Up a Call Center Agent Phone
- Set Up a Phone for Presence
- Bluetooth Handsfree Profile Audio Gateway
- Shared Lines
Cisco IP Phone Customization
Phone Information and Display Settings
The phone web user interface allows you to customize settings such as the phone name, background picture, logo, and screen saver.
- Configure the Phone Name
- Customize the Startup Screen with Text and Picture
- Download Wallpaper
- Configure the Screen Saver with the Phone Web Page
- Add Logo as Boot Display
- Adjust Backlight Timer from Configuration Utility
- Configure the Number of Call Appearances Per Line
Configure the Phone Name
Customize the Startup Screen with Text and Picture
You can create a text or 128-by-48 pixel by 1-bit deep image logo to display when the Cisco IP Phone boots up. A logo displays during the boot sequence for a short period after the Cisco logo displays.
Step 1 | Click . |
Step 2 | In the Screen section, select any option from the Boot Display field.
|
Step 3 | To display a text logo, enter text in the Text Logo field with following requirements:
|
Step 4 | In the Screen section, enter text in the Text Logo field with following requirements: |
Step 5 | To display a picture logo:
|
Step 6 | Click Submit All Changes. The phone reboots, retrieves the .png file, and displays the picture when it next boots. |
Download Wallpaper
You can download a picture to customize the background on the phone screen.
Configure the Screen Saver with the Phone Web Page
You can configure a screen saver for the phone. When the phone is idle for a specified time, it enters screen saver mode.
Any button press returns the phone to normal mode. If a user password is set, the user must enter it to exit screen saver mode.
Add Logo as Boot Display
Step 1 | On the phone web page, select . |
Step 2 | In the Screen section, select Logo from the Boot Display field. In the Logo URL field, enter a URL or path for the location where the logo image is saved.
You can also download a picture and add it as a boot display: select Download Picture from the Boot Display field. In the Picture Download URL field, enter a URL or path for the location where the picture is saved. The logo must be a .jpg or a .png file. The phone has a fixed display area. So, if the original logo size doesn't fit into the display area, you need to scale it to fit the screen. For the Cisco IP Phone 8800 series, the logo display area is at the mid-center of the phone screen. The display area size of the Cisco IP Phone 8800 series is 128x128. |
Step 3 | Click Submit All Changes. |
Adjust Backlight Timer from Configuration Utility
You can save energy by disabling the backlight on each phone at a preset time.
Step 1 | On the Configuration Utility page, select . |
Step 2 | Under Screen, select a duration for the Back Light Timer paramter. |
Step 3 | In the Display Brightness field, enter a number for the desired brightness. |
Configure the Number of Call Appearances Per Line
Phones that support multiple call appearances on a line can be configured to specify the number of calls to allow on the line.
Call Features Configuration
Enable Call Transfer
Call Forward
To enable call forwarding, you can enable the feature in two places: on the Voice tab and the User tab of the phone web page.
Enable Call Forwarding on Voice Tab
Enable Call Forwarding on User Tab
Enable Conferencing
Enable Remote Call Recording with SIP REC
You can enable call recording on a phone so that your user can record an active call. The recording mode configured on the server controls the display of the recording softkeys for each phone.
Recording Mode in Server |
Recording Softkeys Available on the Phone |
---|---|
Always |
No softkeys available. Your user can't control recording from the phone. Recording starts automatically when a call is connected. |
Always with Pause/Resume |
PauseRec ResumeRec When a call is connected, recording starts automatically and your user can control the recording. |
Never |
PauseRec ResumeRec When a call is connected, recording starts automatically and your user can control the recording. |
On Demand |
Record PauseRec ResumeRec When a call is connected, recording starts automatically but the recording is not saved until the user presses the Record softkey. Your user sees a message when recording state changes. |
On Demand with User Initiated Start |
Record PauseRec StopRec ResumeRec The recording only starts when your user presses the Record softkey. Your user sees a message when recording state changes. |
During a recording, your user sees different icons which depend on the recording state. The icons are displayed on the Calls screen and also on the line key on which the user is recording a call.
Icon |
Meaning |
---|---|
![]() |
Recording in progress |
![]() |
Recording in progress (8811) |
![]() |
Recording paused |
![]() |
Recording paused (8811) |
Step 1 | On the phone web page, select . |
Step 2 | In the Supplementary Services section, click Yes or click No to enable or to disable call recording in the Call Recording Serv field. |
Step 3 | (Optional)
In the Programmable Softkeys section, to enable softkeys, add a string in this format in the Connected Key List and Conferencing Key List fields.
crdstart;crdstop;crdpause;crdresume |
Step 4 | In the phone web page, click the Ext(n) tab that requires call recording. |
Step 5 | In the SIP Settings section, in the Call Recording Protocol, select SIPREC as the call recording protocol.
For details on the SIP Settings fields, see SIP Settings. |
Step 6 | Click Submit All Changes. |
Enable Remote Call Recording with SIP INFO
You can enable call recording on a phone so that your user can record an active call. The recording mode configured on the server controls the display of the recording softkeys for each phone.
Recording Mode in Server |
Recording Softkeys Available on the Phone |
---|---|
Always |
No softkeys available. Your user can't control recording from the phone. Recording starts automatically when a call is connected. |
On Demand |
Record When a call is connected, recording starts automatically but the recording is not saved until the user presses the Record softkey. Your user sees a message when recording state changes. |
On Demand with User Initiated Start |
Record StopRec The recording only starts when your user presses the Record softkey. Your user sees a message when recording state changes. |
During a recording, your user sees different icons which depend on the recording state. The icons are displayed on the Calls screen and also on the line key on which the user is recording a call.
Icon |
Meaning |
---|---|
![]() |
Recording in progress |
![]() |
Recording in progress (8811) |
You need to set up call recording on the call control system.
Step 1 | On the phone web page, select . |
Step 2 | In the Supplementary Services section, click Yes or click No to enable or to disable call recording in the Call Recording Serv field. |
Step 3 | (Optional)
In the Programmable Softkeys section, to enable softkeys, add a string in this format in the Connected Key List and Conferencing Key List fields.
crdstart;crdstop;crdpause;crdresume |
Step 4 | In the phone web page, click the Ext(n) tab that requires call recording. |
Step 5 | In the SIP Settings section, in the Call Recording Protocol, select SIPINFO as the call recording protocol.
For details on SIP Settings fields, see SIP Settings. |
Step 6 | Click Submit All Changes. |
Configure Missed Call Indication with the Configuration Utility
If a user is not on an active or held call and misses a call, the user needs to know about the missed call. To alert the user, configure the Handset LED Alert field on the Configuration Utility page. If you set this field to Voicemail, Missed Call, the LED on the Handset will turn on when the user has recently missed a call.
Enable Do Not Disturb
You can allow users to turn the do not disturb feature on or off. The caller receives a message that the user is unavailable. Users can press the Ignore softkey on their phones to divert a ringing call to another destination.
If the feature is enabled for the phone, users turn the feature on or off with the DND softkey.
Configure Synchronization of DND and Call Forward
Step 1 | On the Configuration Utility page, select (where [n] is the extension number). |
Step 2 | In the Call Feature Settings section, set the Feature Key Sync field to Yes. |
Step 3 | Click Submit All Changes. |
Configure Star Codes for DND
You can configure star codes that a user dials to turn on or off the do not disturb (DND) feature on a phone.
Set Up a Call Center Agent Phone
You can enable a phone with Automatic Call Distribution (ACD) features. This phone acts as a call center agent's phone and can be used to trace a customer call, to escalate any customer call to a supervisor in emergency, to categorize contact numbers using disposition codes, and to view customer call details.
Set up the phone as a call center phone on the BroadSoft server.
Step 1 | On the phone web page, select . |
Step 2 | In the ACD Settings section, set up the fields as described in ACD Settings. |
Step 3 | Click Submit All Changes. |
Set Up a Phone for Presence
Set up the Broadsoft server for XMPP.
Step 1 | In the phone web page, click . |
Step 2 | In the Broadsoft XMPP section, set the fields as described in Broadsoft XMPP. |
Step 3 | Click Submit All Changes. |
Bluetooth Handsfree Profile Audio Gateway
Cisco IP Phones 8851 and 8861 support Hands-free Audio Gateway mode to work with your Bluetooth headset.
Configure Bluetooth Handsfree from Configuration Utility
Step 1 | On the Configuration Utility page, click . |
Step 2 | Under Handsfree, select a Bluetooth Mode. |
Step 3 | Select a line.
You can select a line from 1 to 10 for Handsfree. When a line is configured as Handsfree line, it displays mobile phone number and you can only use it for mobile phone. You cannot use it for shared line or speed dial. |
Step 4 | Click Submit All Changes. |
Shared Lines
A shared line is a directory number that appears on more than one phone. You can create a shared line by assigning the same directory number to different phones.
Incoming calls display on all phones that share a line, and anyone can answer the call. Only one call remains active at a time on a phone.
Call information displays on all phones that are sharing a line. If somebody turns on the privacy feature, you do not see the outbound calls made from the phone. However, you see inbound calls to the shared line.
All phones with a shared line ring when a call is made to the line. If you place the shared call on hold, anyone can resume the call by pressing the corresponding line key from a phone that shares the line. You can also press the Select button if the Resume icon is displayed.
The following shared line features are supported:
-
Line Seizure
-
Public Hold
-
Private Hold
-
Silent Barge (only through enabled programmable softkey)
The following features are supported as for a private line
-
Transfer
-
Conference
-
Call Park / Call Retrieve
-
Call Pickup
-
Do Not Disturb
-
Call Forward
You can configure each phone independently. Account information is usually the same for all IP phones, but settings such as the dial plan or preferred codec information can vary.
Configure a Shared Line
You can create a shared line by assigning the same directory number to different phones on the phone web page.
Step 1 | On the Configuration Utility page, click . |
Step 2 | Click the Ext_n tab of the extension that is shared. |
Step 3 | Under General in the Line Enable list, choose Yes. |
Step 4 | Under Share Line Appearance in the Share Ext list, select Shared.
If you set this extension to Private, the extension does not share calls, regardless of the Share Call Appearance setting on the Phone tab. If you set this extension to Shared, calls follow the Share Call Appearance setting on the Phone tab. |
Step 5 | In the Shared User ID field, enter the user ID of the phone with the extension that is being shared. |
Step 6 | In the Subscription Expires field, enter the number of seconds before the SIP subscription expires. The default is 60 seconds.
Until the subscription expires, the phone gets NOTIFY messages from the SIP server on the status of the shared phone extension. |
Step 7 | In the Restrict MWI field, set the message waiting indicator:
|
Step 8 | Under Proxy and Registration, enter the IP address of the proxy server in the Proxy field. |
Step 9 | Under Subscriber Information, enter a Display Name and User ID (extension number) for the shared extension. |
Step 10 | In the Phone tab, under Miscellaneous Line Key Settings, configure SCA Barge-In Enable:
|
Step 11 | Click Submit All Changes. |
Configure Voice Mail
Configure Voice Mail for each Extension
Step 1 | Click Admin Login > advanced > Voice > Extn. |
Step 2 | Under Call Feature Settings, enter the Voice Mail Server. |
Step 3 | (Optional) Enter the Voice Mail Subscribe Interval; the expiration time in seconds, of a subscription to a voice mail server. |
Step 4 | Click Submit All Changes. The phone reboots. |
Configure the Message Waiting Indicator
You can configure the Message Waiting Indicator for separate extensions on the phone. The Message Waiting Indicator lights based on the presence of new voicemail messages in the mailbox.
You can enable the indicator at the top of your IP phone to light when voice mail is left, or display a seeing message waiting notification.
Assign a Ring Tone to an Extension
Add Distinctive Ringtone
You can configure the characteristics of each ring tone using a ring tone script. When phone receives SIP Alert-INFO message and the message format is correct, then the phone plays the specified ringtone. Otherwise, the phone plays the default ringtone.
where: n = ring-tone-name that identifies this ring tone. This name appears on the Ring Tone menu of the phone. The same name can be used in a SIP Alert-Info header in an inbound INVITE request to tell the phone to play the corresponding ring tone. The name should contain the same characters allowed in a URL only. h = hint used to SIP Alert-INFO rule. w = waveform-id-or-path which is the index of the desired waveform to use in this ring tone. The built-in waveforms are:
You can also enter a network path (url) to download a ring tone data file from a server. Add the path in this format: w=[tftp://]hostname[:port]/pathc = is the index of the desired cadence to play the given waveform. 8 cadences (1–8) as defined in <Cadence 1> through <Cadence 8>. Cadence-id can be 0 If w=3,4, or an url. Setting c=0 implies the on-time is the natural length of the ring tone file. b = break-time that specifies the number of seconds to break between two bursts of ring tone, such as b=2.5. t = total-time that specifies the total number of seconds to play the ring tone before it times out. |
Configure the Audio Settings
The user can modify volume settings by pressing the volume control button on the phone, then pressing the Save softkey.
Step 1 | Click . |
Step 2 | In the Audio Volume section, configure a volume level between 1 and 10, with 1 being the lowest level:
|
Step 3 | Click Submit All Changes. |
User Access Control
The Cisco IP Phone respects only the “ua” user access attribute. For a specific parameter, the “ua” attribute defines access by the user account to the administration web server. If the “ua” attribute is not specified, the phone applies the factory default user access for the corresponding parameter. This attribute does not affect access by the admin account.
![]() Note | The value of the element attribute encloses within double quotes. |
Disable Video Services
You can disable or hide all video settings on the phone to disable the video capability of the phone. When you disable video services, your user can't see any video settings menu on their phone and the Video and Camera Exposure parameters don't appear on the phone web page. For information on camera exposure, see Adjust the Camera Exposure.
Step 1 | On the phone web page, select . |
Step 2 | Under Supplementary Services section, from the Video Serv list, select Yes to enable video services or No to disable the service. |
Step 3 | Click Submit All Changes to save your settings. |
Control the Video Bandwidth
If you have a busy network or have limited network resources, users may complain about video issues; for example, the video may lag or suddenly stop.
By default, the phone automatically selects a bandwidth setting that balances the audio and video network requirements.
You can configure a fixed bandwidth setting to override the automatic selection, if required for your network conditions. If you configure a fixed bandwidth, select a setting and adjust downwards until there is no video lag.
Step 1 | On the phone web page, select . |
Step 2 | In the Video Configuration section, choose a bandwidth from the Bandwidth Allowance list to restrict the maximum amount of information that the phone can transmit or receive. For more information see Video Configuration and Video Transmit Resolution Setup. |
Step 3 | Click Submit All Changes. |
Adjust the Camera Exposure
You can adjust the camera exposure for the ambient light in your office. Adjust the exposure to change the brightness of the transmitted video.
Your users can also adjust the exposure on the phone from
menu.
The camera shutter must be open.
Step 1 | On the phone web page, select . |
Step 2 | In the Video Configuration section, enter a value in the Camera Exposure field.
The exposure range is 0 to 15, and the default value is 8. |
Step 3 | Click Submit All Changes. |
Phone Web Server
The web server allows administrators and users to log in to the phone by using a phone web user interface. Administrators and users have different privileges and see different options for the phone based on their role.
- Configure the Web Server from the Phone Screen Interface
- Direct Action URL
- Enable Access to Phone Web Interface
Configure the Web Server from the Phone Screen Interface
Use this procedure to enable the phone web user interface from the phone screen.
Direct Action URL
If the Enable Direct Action URL setting is set to "Yes ", these Direct action URLs are accessible only for the admin. If Admin user is password protected, the client provides a login prompt before these are accessed. The Direct Action URLs are accessible via the phone web page via the path /admin/<direct_action>. The syntax is:
http[s]://<ip_or_hostname>/admin/<direct_action>[?<url>]
For example, http://10.1.1.1/admin/resync?http://server_path/config.xml
The following table provides a list of the different direct avtion URLs that are supported.
direct_action |
Description |
---|---|
resync |
Initiates a one-time resync of the config file specified by URL. The URL to resync is provided by appending ? followed by the URL. The URL specified here will not be saved anywhere in the phone settings. Example http://10.1.1.1/admin/resync?http://my_provision_server.com/cfg/device.cfg |
upgrade |
Initiates an upgrade of a phone to the specified load. The load is specified via the upgrade rule. the rule is specified by appending ? followed by URL path to the load. The upgrade rule specified is one time only and will not be saved in any property setting. Example http://10.1.1.1/admin/upgrade?http://my_upgrade_server.com/loads/sip88xx.11.0.0MP2.123.loads |
updateca |
Initiates a one-time install of the Custom Certificate Authority (Custom CA) specified by the URL. The URL to download is provided by appending ? followed by the URL. The URL specified here will not be saved anywhere in the phone settings. Example http://10.1.1.1/admin/updateca?http://my_cert_server.com/certs/myCompanyCA.pem |
reboot |
Initiates a reboot of the phone. Does not take any parameter with ? Example http://10.1.1.1/admin/reboot |
cfg.xml |
Downloads a snapshot of the phone configuration in XML format. The passwords are hidden for security. Most of the information here corresponds to the properties on the phone web page under Voice tab. Example http://10.1.1.1/admin/cfg.xml |
status.xml |
Downloads a snapshot of the phone status in XML format. Most of the information here corresponds to the Status tab in the phone web page. Example http://10.1.1.1/admin/status.xml |
screendump.bmp |
Downloads a screenshot of the phone LCD UI at the time when this action is initiated. Example http://10.1.1.1/admin/screendump.bmp |
log.tar |
Downloads a set of archived logs stored on the phone. Example http://10.1.1.1/admin/log.tar |
Enable Access to Phone Web Interface
Step 1 | Click . | ||
Step 2 | Under the System Configuration section, choose Yes from the Enable Web Server drop-down list box. | ||
Step 3 | In the Enable Protocol drop-down list box, choose Http or Https. | ||
Step 4 | In the Web Server Port field, enter the port to access the web server. The default is port 80 for HTTP or port 443 for HTTPS. | ||
Step 5 | In the Enable Web Admin Access drop-down list box, you can enable or disable local access to the Admin Login of the phone web user interface. Defaults to Yes (enabled). | ||
Step 6 | In the Admin Password field, enter a password if you want the system administrator to log in to the phone web user interface with a password. The password prompt appears when an administrator clicks Admin Login. The minimum password length can be 4 characters or the maximum password length is 127 characters.
| ||
Step 7 | In the User Password field, enter a password if you want users to log in to the phone web user interface with a password. The password prompt appears when users click User Login. The minimum password length can be 4 characters or the maximum password length is 127 characters.
| ||
Step 8 | Click Submit All Changes. |
XML Services
The phones provide support for XML services, such as an XML Directory Service or other XML applications. For XML services, only HTTP and HTTPS support are available.
The following Cisco XML objects are supported:
CiscoIPPhoneMenu
CiscoIPPhoneText
CiscoIPPhoneInput
CiscoIPPhoneDirectory
CiscoIPPhoneIconMenu
CiscoIPPhoneStatus
CiscoIPPhoneExecute
CiscoIPPhoneImage
CiscoIPPhoneImageFile
CiscoIPPhoneGraphicMenu
CiscoIPPhoneFileMenu
CiscoIPPhoneStatusFile
CiscoIPPhoneResponse
CiscoIPPhoneError
CiscoIPPhoneGraphicFileMenu
Init:CallHistory
Key:Headset
EditDial:n
The full list of supported URIs is contained in Cisco Unified IP Phone Services Application Development Notes for Cisco Unified Communications Manager and Mutiplatform Phones, located here:
- XML Directory Service
- XML Applications
- Macro Variables
- Configure a Phone to Connect to an XML Application
- Configure a Phone to Connect to an XML Directory Service
XML Directory Service
When an XML URL requires authentication, use the parameters XML UserName and XML Password.
The parameter XML UserName in XML URL is replaced by $XML UserName.
For example:
The parameter XML UserName is cisco. The XML Directory Service URL is http://www.sipurash.compath?username=$XML_User_Name.
This results in the request URL: http://www.sipurash.com/path?username=cisco.
XML Applications
When authentication is required for CGI/Execute URL via Post from an external application (for example, a web application) to the phones, the parameter CISCO XML EXE Auth Mode is used in 3 different scenarios:
Trusted—No authentication is performed (local user password is set or not). This is the default.
Local Credential—Authentication is based on digest authentication using the local user password, if the local user password is set. If not set, then no authentication is performed.
Remote Credential—Authentication is based on digest authentication using the remote username/password as set in the XML application on the web page (to access an XML application server).
Macro Variables
You can use macro variables in XML URLs. The following macro variables are supported:
User ID—UID1, UID2 to UIDn
Display name—DISPLAYNAME1, DISPLAYNAME2 to DISPLAYNAMEn
Auth ID—AUTHID1, AUTHID2 to AUTHIDn
Proxy—PROXY1, PROXY2 to PROXYn
MAC Address using lower case hex digits—MA
Product Name—PN
Product Series Numbe—PSN
Serial Number—SERIAL_NUMBER
The following table shows the list of macros supported on the phones:
Macro Name |
Macro Expansion |
||
---|---|---|---|
$ |
The form $$ expands to a single $ character. |
||
A through P |
Replaced by general purpose parameters GPP_A through GPP_P. |
||
SA through SD |
Replaced by special purpose parameters GPP_SA through GPP_SD. These parameters hold keys or passwords used in provisioning.
|
||
MA |
MAC address using lower case hex digits (000e08aabbcc). |
||
MAU |
MAC address using upper case hex digits (000E08AABBCC). |
||
MAC |
MAC address using lower case hex digits with colon to separate hex digit pairs (00:0e:08:aa:bb:cc). |
||
PN |
Product Name; for example, IP Phone 8861. |
||
PSN |
Product Series Number; for example, 8861. |
||
SN |
Serial Number string; for example 88012BA01234. |
||
CCERT |
SSL Client Certificate status, installed or not installed. |
||
IP |
IP address of the phone within its local subnet; for example 192.168.1.100. |
||
EXTIP |
External IP of the phone, as seen on the internet; for example 66.43.16.52. |
||
SWVER |
Software version string; for example 2.0.6(b). |
||
HWVER |
Hardware version string; for example 1.88.1. |
||
PRVST |
Provisioning State (a numeric string): |
||
UPGST |
Upgrade State (a numeric string): |
||
UPGERR |
Result message (ERR) of previous upgrade attempt; for example http_get failed. |
||
PRVTMR |
Seconds since last resync attempt. |
||
UPGTMR |
Seconds since last upgrade attempt. |
||
REGTMR1 |
Seconds since Line 1 lost registration with SIP server. |
||
REGTMR2 |
Seconds since Line 2 lost registration with SIP server. |
||
UPGCOND |
Legacy macro name. |
||
SCHEME |
File access scheme (TFTP, HTTP, or HTTPS, obtained after parsing resync or upgrade URL). |
||
METH |
Deprecated alias for SCHEME, do not use. |
||
SERV |
Request target server host name. |
||
SERVIP |
Request target server IP address (following DNS lookup). |
||
PORT |
Request target UDP/TCP port. |
||
PATH |
Request target file path. |
||
ERR |
Result message of resync or upgrade attempt. |
||
UIDn |
The contents of the Line n UserID configuration parameter. |
||
ISCUST |
If unit is customized, value=1, otherwise 0.
|
||
INCOMINGNAME |
Name associated with first connected, ringing, or inbound call. |
||
REMOTENUMBER |
Phone number of first connected, ringing, or inbound call. If there are multiple calls, the data associated with the first call found will be provided. |
||
DISPLAYNAMEn |
The contents of the Line N Display Name configuration parameter. |
||
AUTHIDn |
The contents of the Line N auth ID configuration parameter. |
Configure a Phone to Connect to an XML Application
Step 1 | In the Configuration Utility, select . |
Step 2 | Enter this information:
If you configure an unused line button to connect to an XML application, the button connects to the URL configured above. If this is not what you want, you need to enter a different URL when you configure the line button. |
Step 3 | Click Submit All Changes. |