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Configure phone features
Feedback?You can use the ua attribute to control user access to menu items on the user interface. The ua attribute is attached to each parameters in the configuration file. See the following sections for the details.
The Cisco IP Phone firmware provides specific administrator and user accounts. These accounts provide specific login privileges. The administrator account name is admin; the user account name is user. These account names cannot be changed.
The admin account gives the service provider or Value-added Reseller (VAR) configuration access to the phone. The user account gives limited and configurable control to the device end user.
The user and admin accounts can be password protected independently. If the service provider sets an administrator account password, you are prompted for it when you click Admin Login. If the password does not yet exist, the screen refreshes and displays the administration parameters. No default passwords are assigned to either the administrator or the user account. Only the administrator account can assign or change passwords.
The administrator account can view and modify all web profile parameters, including web parameters, that are available to the user login. The phone system administrator can further restrict the parameters that a user account can view and modify through use of a provisioning profile.
Configuration parameters that are available to the user account are configurable on the phone. User access to the phone web user interface can be disabled.
The user access (ua) attribute controls may be used to change access by the User account. If the ua attribute is not specified, the existing user access setting is retained. This attribute does not affect access by the Admin account.
The ua attribute, if present, must have one of the following values:
-
na—No access
-
ro—Read-only
-
rw—Read and write
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y—Preserve value
The y value must be used together with na, ro, or rw.
The following example illustrates the ua attribute. Notice in the last line that the ua attribute is updated to rw, and the station name field (Travel Agent 1) is preserved. If y is not included, Travel Agent 1 is overwritten:
<flat-profile>
<SIP_TOS_DiffServ_Value_1_ ua=”na”/>
<Dial_Plan_1_ ua=”ro”/>
<Dial_Plan_2_ ua=”rw”/>
<Station_Name ua=“rw” preserve-value="y">Travel Agent 1</Station_Name></flat-profile>
Double quotes must enclose the value of the ua option.
The phone firmware provides mechanisms for restricting end-user access to some parameters. The firmware provides specific privileges for sign-in to an Admin account or a User account. Each can be independently password-protected.
-
Admin account–Allows the full access to all administration web server parameters
-
User account–Allows the access to a subset of the administration web server parameters
If your service provider has disabled access to the configuration utility, contact the service provider before proceeding.
| 1 |
Ensure that the computer can communicate with the phone. No VPN in use. |
| 2 |
Enter the IP address of the phone in your web browser address bar.
For example, |
| 3 |
Enter the password when prompted. |
You can bypass the phone Set password screen on the first boot or after a factory reset, based on these provisioning actions:
-
DHCP configuration
-
EDOS configuration
-
User password configuration using in the phone XML configuration file
After the User Password is configured, the set password screen doesn't appear.
| 1 |
Edit the phone |
| 2 |
Insert the
|
| 3 |
Save the changes to the |
You can configure the phone to allow or block access to the configuration parameters on the phone web page or the phone screen. The parameters for access control allow you to:
-
Indicate which configuration parameters are available to the user account when creating the configuration.
-
Enable or disable the access to the administration web server.
-
Enable or disable user access to the phone screen menus.
-
Bypass the Set password screen for the user.
-
Restrict the Internet domains that the phone accesses for resync, upgrades, or SIP registration for Line 1.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. To configure each parameter, see the syntax of the string in the following table of Access control parameters.
| 1 |
Access the phone administration web page. |
| 2 |
Click . |
| 3 |
In the System Configuration section, configure the parameters as defined in the following table of Access control parameters. |
| 4 |
Click Submit All Changes to apply the changes. |
The following table defines the function and usage of the access control parameters in the System Configuration section under the tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file (cfg.xml) with XML code to configure a parameter.
|
Parameter Name |
Description and Default Value |
|---|---|
|
Enable Web Server |
Enables or disables access to the phone web interface. Set this parameter to Yes to allow users or administrators to access the phone web interface. Otherwise, set it to No. When set to No, the phone web interface isn't accessible. Perform one of the following:
Allowed values: Yes|No Default: Yes. |
|
Enable Web Admin Access |
Allows or blocks the access to the phone administration pages:
When set to No, the web page for administrator is inaccessible. Only the web page for user is accessible. If you want to allow the access to the administration web page again after the access is blocked, you need to perform a factory reset from the phone. Perform one of the following:
Allowed values: Yes|No Default: Yes |
|
Admin Password |
Allows you to set or change the password for accessing the phone administration web pages. The Admin Password parameter is only available on the phone administration web page. A valid password must contain 4 to 127 characters from three out of the four types: capital letter, small letter, number, and special character. The password is set to empty after you perform a phone factory reset. Perform one of the following:
Default: Empty |
|
User Password |
Allows you or the phone user to set or change the password for accessing the phone web interfaces and the menus on the phone screen. A valid password must contain 4 to 127 characters from three out of the four types: capital letter, small letter, number, and special character. The password is set to empty after you perform a phone factory reset. In the configuration file (cfg.xml), you can use the Default: Empty |
|
Display Password Warnings |
Determines whether to display the The warning message disappears when both user password and admin password are set. If Enable Web Admin Access is set to No, the password warning doesn't display on the phone screen. Perform one of the following:
Allowed values: Yes|No Default: Yes |
As an administrator, you can configure the BLF shortcuts with speed dial and call pickup for your users. The BLF shortcuts allow users to monitor their coworkers lines. When BLF shortcuts are configured with call pickup and speed dial, users can use BLF shortcuts to answer calls for the monitored lines and call the monitored line with one tap.
The indicators on the monitoring lines vary with configurations. See the following tables for the status.
| Icon | Status |
|---|---|
|
|
The line is configured with call pickup and in idle mode. |
|
Alerting, or Alerting · Pickup enabled |
There's an incoming call waiting for pickup. If the Call Pickup feature is enabled for the BLF shortcut, the users are able to answer the call for their coworkers by tapping the BLF shortcut. If the Call Pickup feature isn't enabled, the users can't answer the call. Instead, tapping the shortcut initiates a call to the monitored line. You can't pick up the alerting call for the monitored line when there's an active call or outgoing call on your phone. |
|
In use |
The monitored line is in use. |
| Icon | Status | Description |
|---|---|---|
|
|
Idle |
The monitored line is in idle mode. You can tap the shortcut to call the speed dial number. |
|
|
Alerting Alerting · Pickup enabled |
A call is alerting on the monitored line. If the Call Pickup feature is enabled for the BLF shortcut, you’re able to answer the call for your coworker by tapping the BLF shortcut. If the Call Pickup feature isn't enabled, you can't answer the call. Instead, tapping the shortcut initiates a call to the monitored line. You can't pick up the alerting call for the monitored line when there's an active call or outgoing call on your phone. |
|
|
In use |
The monitored line is on a call. If you tap the shortcut, a call is placed to the speed dial number. You’ll hear a busy tone or be redirected to the voicemail depending on the configuration on your coworker's line. |
|
|
DND |
The monitored line is set to Do Not Disturb (DND). The calls to the line don't alert. |
|
|
Unregistered |
The monitored line isn’t registered. The calls to the line don't get connected. |
If the phone is registered to a BroadWorks server, you can configure the phone to monitor the entire BLF list. The phone assigns available line keys in sequence to monitor the BLF list entries, and starts showing the status of the monitored lines on the BLF keys.
You can also configure the parameters in the phone configuration file with XML (cfg.xml). To configure each parameter, see the syntax of the string in the following table of Parameters for monitoring multiple users' lines.
Before you begin
-
Ensure that the phone is registered to a BroadWorks server.
-
You set up a BLF list for a user of the phone on the BroadWorks server.
-
Ensure that the monitored lines on the BLF keys are not in the Inert mode.
| 1 |
Access the administration web interface. |
| 2 |
Select . |
| 3 |
Configure BLF List URI, Use Line Keys For BLF List, BLF List, and BLF Label Display Mode as described in the following table of Parameters for monitoring multiple users' lines. |
| 4 |
Click Submit All Changes. |
Parameters for monitoring multiple users' lines
Parameters for monitoring multiple users' lines
The following table defines the function and usage of the BLF parameters in the General section under the tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML (cfg.xml) to configure a parameter.
|
Parameter |
Description and default value |
|---|---|
| BLF List URI |
The Uniform Resource Identifier (URI) of the Busy Lamp Field (BLF) list that you have set up for a user of the phone, on the BroadWorks server. This field is only applicable if the phone is registered to a BroadWorks server. The BLF list is the list of users whose lines the phone is allowed to monitor. The BLF List URI must be specified in the format Perform one of the following:
Default: Blank |
| Use Line Keys For BLF List |
Controls whether the phone uses its line keys to monitor the BLF list, when monitoring of the BLF list is active. This setting only has significance when BLF List is set to Show. Perform one of the following:
Default: No |
| BLF List |
Determines whether to show or hide the BLF list on the line key. When set to Show, the phone assigns available line keys in sequence, to monitor the BLF list entries. The labels of the BLF list keys show the names of the monitored users and the status of the monitored lines. This setting only has significance when BLF List URI is configured. Perform one of the following:
Allowed values: Show|Hide Default: Show |
|
BLF Label Display Mode |
Specifies how the BLF entries are displayed on the line keys. The options are: Name, Ext (extension number), and Both. Perform one of the following:
Allowed values: Name|Ext|Both Default: Name |
You can configure busy lamp field on a phone line when a user needs to monitor a coworker's availability to handle calls.
You can configure the busy lamp field to work with any combination of speed dial or call pickup. For example, busy lamp field and speed dial, busy lamp field and call pickup, or busy lamp field with both speed dial and call pickup. But speed dial alone requires a different configuration.
You can also configure the parameters in the phone configuration file with XML (cfg.xml). To configure each parameter, see the syntax of the string in the table of Parameters for monitoring a specific line.
Before you begin
Ensure that the line key on which to configure a busy lamp field is not in the Inert mode.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
Select a line key on which to configure a busy lamp field. |
| 4 |
Configure the Extension, Extended Function, fields as defined in the following table of Parameters for monitoring a specific line. |
| 5 |
Click Submit All Changes. |
Parameters for monitoring a specific line
Parameters for monitoring a specific line
The following table defines the function and usage of the Busy Lamp Field (BLF) parameters in the Line Key (n) sections under the tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML (cfg.xml) to configure a parameter.
|
Parameter |
Description and default value |
|---|---|
|
Extension |
Assigns an extension number to a line key or disables the extension function on a line key. The number of line keys varies with phone models. When assigned with an extension number, you can configure the line key as a telephony extension. When you need to assign the line key with extended functions (for example, speed dial, BLF, call pickup) while the Direct PLK Configuration feature is disabled, you can either enable the feature or set the Extension parameter to Disabled. Perform one of the following:
Allowed values: Disabled|1|2|3|4, the allowed values vary with phones. Default: n, where n is the line key number. |
|
Extended Function |
This parameter functions only on the lines with the Extension parameter set to Disabled. Used to assign extended functions to a line on the phone. The supported functions are:
Where, Perform one of the following:
Default: Empty |
You can also configure the parameters in the phone configuration file with XML (cfg.xml) code. To configure each parameter, see the syntax of the string in the following table of Parameters for ringtones.
| 1 |
Access the phone administration web page. |
| 2 |
Select , where (n) is the number of a phone extension. |
| 3 |
In the Call Feature Settings section, select the Default Ring parameter from the list or select no ring. You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format: |
| 4 |
Select . |
| 5 |
In the Ringtone section, set the Ring(n) and Silent Ring Duration parameters as described in the following table of Parameters for ringtones. |
| 6 |
Click Submit All Changes. |
Parameters for ringtones
The following table describes the parameters in the Ringtone section in the tab of the phone web page.
|
Parameter |
Description |
|---|---|
|
Ring1 to Ring12 |
Ring tone scripts for various ringtones. If you want distinctive ringtong patterns, customize your ringtones with the ringtone scripts described in Customize ringtone patterns. In the phone configuration XML file (cfg.xml), enter a string in this format: |
|
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. In the phone configuration XML file (cfg.xml), enter a string in this format:
|
You can configure the characteristics of each ring tone using a ring tone script. When the 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.
|
In a ring tone script, assign a name for the ring tone and add the script to configure a distinctive ringtone in the format: n=ring-tone-name;h=hint;w=waveform-id-or-path;c=cadence-id;b=break-time;t=total-time
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:
c = 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. 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. In the phone configuration XML file (cfg.xml), enter a string in this format: |
When you enable the hoteling feature of BroadSoft on the phone, the user can sign in to the phone as a guest. After the guest sign out of the phone, the user will switch back to the host user.
You can also configure the parameters in the phone configuration file with XML (cfg.xml) code.
| 1 |
Access the phone administration web page. |
| 2 |
Select (where [n] is the extension number). |
| 3 |
In the Call Feature Settings section, set Enable Broadsoft Hoteling parameter to Yes. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: No |
| 4 |
Set the amount of time (in seconds) that the user can be signed in as a guest on the phone in Hoteling Subscription Expires. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Valid values: An integer from 10 through 86400 Default: 3600 |
| 5 |
Click Submit All Changes. |
You must use a server with an upload script to receive the problem reports that the user sends from the phone.
-
If the URL specified in the PRT Upload Rule field is valid, users get a notification alert on the phone UI saying that they have successfully submitted the problem report.
-
If the PRT Upload Rule field is empty or has an invalid URL, users get a notification alert on the phone UI saying that the data upload failed.
The phone uses an HTTP/HTTPS POST mechanism, with parameters similar to an HTTP form-based upload. The following parameters are included in the upload (utilizing multipart MIME encoding):
-
devicename (example: "SEP001122334455")
-
serialno (example: "FCH12345ABC")
-
username (The user name is either the Station Display Name or the User ID of the extension. The Station Display Name is first considered. If this field is empty, then the User ID is chosen.)
-
prt_file (example: "probrep-20141021-162840.tar.gz")
You can generate PRT automatically at specific intervals and can define the PRT file name.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. To configure each parameter, see the syntax of the string in the following table of Parameters for Problem Report Tool.
A sample script is shown below. This script is provided for reference only. Cisco does not provide support for the upload script installed on a customer's server.
<?php
// NOTE: you may need to edit your php.ini file to allow larger
// size file uploads to work.
// Modify the setting for upload_max_filesize
// I used: upload_max_filesize = 20M
// Retrieve the name of the uploaded file
$filename = basename($_FILES['prt_file']['name']);
// Get rid of quotes around the device name, serial number and username if they exist
$devicename = $_POST['devicename'];
$devicename = trim($devicename, "'\"");
$serialno = $_POST['serialno'];
$serialno = trim($serialno, "'\"");
$username = $_POST['username'];
$username = trim($username, "'\"");
// where to put the file
$fullfilename = "/var/prtuploads/".$filename;
// If the file upload is unsuccessful, return a 500 error and
// inform the user to try again
if(!move_uploaded_file($_FILES['prt_file']['tmp_name'], $fullfilename)) {
header("HTTP/1.0 500 Internal Server Error");
die("Error: You must select a file to upload.");
}
?>| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the Problem Report Tool section, set the fields as described in the following table of Parameters for Problem Report Tool. |
| 4 |
Click Submit All Changes. |
Parameters for problem report tool
|
Parameter |
Description |
|---|---|
|
PRT Upload Rule |
Specifies the path to the PRT upload script. If the PRT Max Timer and PRT Upload Rule fields are empty, the phone doesn't generate the problem reports automatically unless user manually performs the generation. Perform one of the following:
Default: Empty |
|
PRT Upload Method |
Determines the method used to upload PRT logs to the remote server.
Perform one of the following:
Valid values: POST and PUT Default: POST |
|
PRT Max Timer |
Determines at what interval (minutes) the phone starts generating problem report automatically. If the PRT Max Timer and PRT Upload Rule fields are empty, the phone doesn't generate the problem reports automatically unless user manually performs the generation.
Perform one of the following:
Valid value range: 15 minutes to 1440 minutes Default: Empty |
|
PRT Name |
Defines a name for the generated PRT file.
Perform one of the following:
Default: Empty |
|
PRT HTTP Header |
Specifies the HTTP header for the URL in PRT Upload Rule. The parameter value is associated with PRT HTTP Header Value. Only when both parameters are configured, the HTTP header is included in the HTTP request. Perform one of the following:
Valid value range: a-z, A-Z, 0-9, underscore (_), and hyphen (-) Default: Empty |
|
PRT HTTP Header Value |
Sets the value of the specified HTTP header. The parameter value is associated with PRT HTTP Header. Only when both parameters are configured, the HTTP header is included in the HTTP request. Perform one of the following:
Valid value range: a-z, A-Z, 0-9, underscore (_), comma (,), semicolon (;), equal (=), and hyphen (-) Except for the underscore (_), the first character must not be a special character. Default: Empty |
Users submit problem reports to you with the Problem Reporting Tool.
If you are working with Cisco TAC to troubleshoot a problem, they typically require the logs from the Problem Reporting Tool to help resolve the issue.
To issue a problem report, users access the Problem Reporting Tool and provide the date and time that the problem occurred, and a description of the problem. You need to download the problem report from the phone administration web page.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the Problem Reports area, click the problem report file to download. |
| 4 |
Save the file to your local system and open the file to access the problem reporting logs. |
You can use the protocols and standards defined in Technical Report 069 (TR-069) to manage phones. TR-069 explains the common platform for management of all phones and other customer-premises equipment (CPE) in large-scale deployments. The platform is independent of phone types and manufacturers.
As a bidirectional SOAP/HTTP-based protocol, TR-069 provides the communication between CPEs and Auto Configuration Servers (ACS).
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. To configure each parameter, see the syntax of the string in the following table of Parameters for TR-069 configuration.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
Set up the fields as described in the following table of Parameters for TR-069 configuration. |
| 4 |
Click Submit All Changes. |
Parameters for TR-069 configuration
Parameters for TR-069 configuration
The following table defines the function and usage of TR-069 parameters under the tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.
|
Parameter |
Description |
|---|---|
|
Enable TR-069 |
Settings that enables or disables the TR-069 function. Perform one of the following:
Valid values: Yes|No Default: No |
|
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. Perform one of the following:
Default: Blank |
|
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. Perform one of the following:
Default: Blank |
|
ACS Password |
Password to access to the ACS for a specific user. This password is used only for HTTP-based authentication of the CPE. Perform one of the following:
Default: Blank |
|
ACS URL In Use |
URL of the ACS that is currently in use. This is a read-only field. |
|
Connection Request URL |
This is read-only field showing the URL of the ACS that makes the connection request to the CPE. |
|
Connection Request Username |
Username that authenticates the ACS that makes the connection request to the CPE. Perform one of the following:
|
|
Connection Request Password |
Password used to authenticate the ACS that makes a connection request to the CPE. Perform one of the following:
Default: Blank |
|
Periodic Inform Interval |
Duration in seconds of the interval between CPE attempts to connect to the ACS when Periodic Inform Enable is set to yes. Perform one of the following:
Default: 20 |
|
Periodic Inform Enable |
Settings that enables or disables the CPE connection requests. Perform one of the following:
Valid values: Yes|No Default: Yes |
|
TR-069 Traceability |
Settings that enables or disables TR-069 transaction logs. Perform one of the following:
Valid values: Yes|No Default: No |
|
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.
Perform one of the following:
Valid values: Yes|No Default: Yes |
|
TR-069 VoiceObject Init |
Settings to modify voice objects. Perform one of the following:
Valid values: Yes|No Default: Yes |
|
TR-069 DHCPOption Init |
Settings to modify DHCP settings. Perform one of the following:
Valid values: Yes|No Default: Yes |
|
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 ACScertificate when it uses SSL or TLS. Perform one of the following:
Default: Blank |
|
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.
Perform one of the following:
Default: Blank |
|
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.
Perform one of the following:
Default: Blank |
|
If you do not configure the above parameters, you can also fetch them through DHCP options 60,43, and 125. | |
When you enable TR-069 on a user phone, you can view status of TR-069 parameters on the phone web interface.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
You can configure an extension to only accept secure calls. If the extension is configured to only accept secure calls then any calls the extension makes will be secure.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.
Before you begin
SIP transport with TLS can be set statically on the phone web page or automatically with information in the DNS NAPTR records. If the SIP transport parameter is set for the phone extension as TLS, the phone only allows SRTP. If the SIP transport parameter is set to AUTO, the phone performs a DNS query to get the transport method.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the Supplementary Services section, set Secure Call Serv to Yes. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: |
| 4 |
Select . |
| 5 |
In the Call Feature Settings section, set Secure Call Option to Optional, Required, or Strict. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: <Secure_Call_Option_1_ ua="na">Optional</Secure_Call_Option_1_>
Options: Optional, Required, and Strict
Default: Optional |
| 6 |
Click Submit All Changes. |
By default, the handset LED flashes when there's an unread voice message. You can configure the phone to flash the handset LED for missed calls.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the Supplementary Services section, for the Handset LED Alert parameter, select Voicemail, Missed Call. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Voicemail and Voicemail, Missed Call Default: Voicemail |
| 4 |
Click Submit All Changes. |
The phone supports multiple call appearances on a line at a time. You can configure the allowed number of calls on that a line can handle at a time. When a call is active, the other calls can be on hold or waiting for answer.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the Miscellaneous Line Key Settings section, for the Call Appearances Per Line parameter, specify the number of calls per line to allow. You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format: The allowed values range from 2 to 10. The default value is 2. |
| 4 |
Click Submit All Changes. |
For SIP messages, you can configure each extension to use:
-
a specific protocol
-
the protocol automatically selected by the phone
When you set up automatic selection, the phone determines the transport protocol based on the Name Authority Pointer (NAPTR) records on the DNS server. The phone uses the protocol with the highest priority in the records.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.
| 1 |
Access the phone administration web page. |
| 2 |
Select , where n is an extension number. |
| 3 |
In the SIP Settings section, set the SIP Transport parameter to select a transport protocol for SIP messages. You can configure this parameter in the phone configuration XML file (cfg.xml) with a string in this format: where n is the extension number. Options: UDP, TCP, TLS, and Auto AUTO allows the phone to select the appropriate protocol automatically, based on the NAPTR records on the DNS server. Default: UDP |
| 4 |
Click Submit All Changes. |
You can disable the ability of the phone to receive incoming SIP messages from a non-proxy server. When you enable this feature, the phone only accepts SIP messages from:
-
proxy server
-
outbound proxy server
-
alternative proxy server
-
alternative outbound proxy server
-
IN-Dialog message from proxy server and non-proxy server. For example: Call Session dialog and Subscribe dialog
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.
Before you begin
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the System Configuration section, set Block Nonproxy SIP to Yes to block any incoming non-proxy SIP messages except IN-dialog message. If you choose No, the phone does not block any incoming non-proxy SIP messages. Set Block Nonproxy SIP to No for phones that use TCP or TLS to transport SIP messages. Nonproxy SIP messages transported over TCP or TLS are blocked by default. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: No |
| 4 |
Click Submit All Changes. |
A user privacy header in the SIP message sets user privacy needs from the trusted network.
You can set the user privacy header value for each line extension.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the SIP Settings section, set the Privacy Header parameter to set user privacy in the SIP message in the trusted network. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options:
Default: Disabled |
| 4 |
Click Submit All Changes. |
You can determine whether to include the P-Early-Media header in the SIP message of outgoing calls. The P-Early-Media header contains the status of the early media stream. If the status indicates that the network is blocking the early media stream, the phone plays the local ringback tone. Otherwise, the phone plays the early media while waiting for the call to be connected.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.
| 1 |
Access the phone administration web page. |
| 2 |
Select Voice > Ext (n). |
| 3 |
In the SIP Settings section, set the P-Early-Media Support to Yes to control whether the P-Early-Media header is included in the SIP message for an outgoing call. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: No |
| 4 |
Click Submit All Changes. |
You can enable the phone to send end-of-call statistics in Session Initiation Protocol (SIP) messages (BYE and re‑INVITE messages). The phone sends call statistics to the other party of the call when the call terminates or when the call is on hold. The statistics include:
-
Real-time Transport Protocol (RTP) packets sent or received
-
Total bytes sent or received
-
Total number of lost packets
-
Delay jitter
-
Round-trip delay
-
Call duration
The call statistics are sent as headers in SIP BYE messages and SIP BYE response messages (200 OK and re-INVITE during hold). For audio sessions, the headers are RTP-RxStat and RTP-TxStat. For video sessions, the headers are RTP-VideoRxStat and RTP-VideoTxStat.
Example of call statistics in a SIP BYE message:
Rtp-Rxstat: Dur=13,Pkt=408,Oct=97680,LatePkt=8,LostPkt=0,AvgJit=0,VQMetrics="CCR=0.0017;
ICR=0.0000;ICRmx=0.0077;CS=2;SCS=0;VoRxCodec=PCMU;CID=4;VoPktSizeMs=30;VoPktLost=0;
VoPktDis=1;VoOneWayDelayMs=281;maxJitter=12;MOScq=4.21;MOSlq=3.52;network=ethernet;
hwType=CP-8865;rtpBitrate=60110;rtcpBitrate=0"
Rtp-Txstat: Dur=13,Pkt=417,Oct=100080,tvqMetrics="TxCodec=PCMU;rtpbitrate=61587;rtcpbitrate=0
Rtp-Videorxstat: Dur=12;pkt=5172;oct=3476480;lostpkt=5;avgjit=17;rtt=0;
ciscorxvm="RxCodec=H264 BP0;RxBw=2339;RxReso=1280x720;RxFrameRate=31;
RxFramesLost=5;rtpBitRate=2317653;rtcpBitrate=0"
Rtp-Videotxstat: Dur=12;pkt=5303;oct=3567031;ciscotxvm="TxCodec=H264 BP0;TxBw=2331;
TxReso=1280x720;TxFrameRate=31;rtpBitrate=2378020;rtcpBitrate=0"For description of the attributes in call statistics, see the following table of Attributes for call statistics in SIP messages.
You can also use the Call_Statistics parameter in the phone configuration file to enable this feature.
<Call_Statistics ua="na">Yes</Call_Statistics>| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the RTP Parameters section, set the Call Statistics field to Yes to enable the phone to send call statistics in SIP BYE and re‑INVITE messages. You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format: The allowed values are Yes|No. The default value is No. |
| 4 |
Click Submit All Changes. |
Attributes for call statistics in SIP messages
|
Attribute |
Description |
Mandatory |
|---|---|---|
|
Dur |
Duration of media session/call |
Yes |
|
Pkt |
Number of RTP packets received |
Yes |
|
Oct |
Number of RTP packets octets received |
No |
|
LatePkt |
Number of RTP packets received and discarded as late due to outside of buffer window |
Yes |
|
LostPkt |
Number of RTP packets lost |
Yes |
|
AvgJit |
Average Jitter over session duration |
Yes |
|
VoRxCodec |
Stream/session codec negotiated |
Yes |
|
VoPktSizeMs |
Packet size in milliseconds |
Yes |
|
maxJitter |
Max Jitter detected |
Yes |
|
VoOneWayDelayMs |
Latency/one way delay |
Yes |
|
MOScq |
Mean opinion score conversational quality for the session, per RFC https://tools.ietf.org/html/rfc3611 |
Yes |
|
maxBurstPktLost |
Maximum number of sequential packets lost |
No |
|
avgBurstPktLost |
Average number of sequential packets lost in a burst. The number can be used in conjunction with overall loss to compare the impact of loss on the call quality. |
No |
|
networkType |
Type of network the device is on (if possible). |
Yes |
|
hwType |
Hardware client that the session/media is running on. More relevant for soft clients but still useful for hard phones. For example, Model number CP-8865. |
Yes |
|
Attribute |
Description |
Mandatory |
|---|---|---|
|
Dur |
Duration of session |
Yes |
|
Pkt |
Number of RTP packets transmitted |
Yes |
|
Oct |
Number of RTP packets octets transmitted |
Yes |
|
TxCodec |
Transmit codec |
Yes |
|
rtpBitRate |
Total RTP transmit bit rate (bits/sec) |
Yes |
|
rctpBitRate |
Total RCTP transmit bit rate (bits/sec) |
Yes |
|
Attribute |
Description |
Mandatory |
|---|---|---|
|
Dur |
Duration of session in seconds |
Yes |
|
Pkt |
Number of packets received |
Yes |
|
Oct |
Number of octets received |
Yes |
|
LostPkt |
Number of packets lost |
Yes |
|
AvgJit |
Average Jitter over session duration |
Yes |
|
RTT |
End-to-end round trip time |
Yes |
|
CiscoRxVm.RxCodec |
Video codec used for received video stream |
Yes |
|
CiscoRxVm.RxBw |
Negotiated bandwidth for the received video stream |
No |
|
CiscoRxVm.RxReso |
Resolution of the received video stream |
Yes |
|
CiscoRxVm.RxFrameRate |
Frame rate for the received video stream |
Yes |
|
CiscoRxVm.RxFrameLost |
Frames lost for the received video stream |
Yes |
|
CiscoRxVm.rtpBitRate |
RTP bit rate in seconds (including any FEC, retransmits etc.). Used to estimate bandwidth usage (bits/sec). |
Yes |
|
CiscoRxVm.rtcpBitRate |
RTCP bit rate in seconds (including any FEC, retransmits etc.). Used to estimate bandwidth usage (bits/sec). |
Yes |
|
Attribute |
Description |
Mandatory |
|---|---|---|
|
Dur |
Duration of session in seconds |
Yes |
|
Pkt |
Number of packets transmitted |
Yes |
|
Oct |
Number of octets transmitted |
Yes |
|
CiscoTxVm.TxCodec |
Video codec used for the transmitted video stream |
Yes |
|
CiscoTxVm.TxBw |
Negotiated bandwidth for the transmitted video stream |
No |
|
CiscoTxVm.TxReso |
Resolution of the transmitted video stream |
Yes |
|
CiscoTxVm.TxFrameRate |
Frame rate for the transmitted video stream |
Yes |
|
CiscoRxVm.rtpBitRate |
RTP bit rate in seconds (including any FEC, retransmits etc.). Used to estimate bandwidth usage (bits/sec). |
Yes |
|
CiscoTxVm.rtcpBitRate |
RTCP bit rate in seconds (including any FEC, retransmits etc.). Used to estimate bandwidth usage (bits/sec). |
Yes |
The phone supports Session Identifier. This feature allows end-to-end tracking of a SIP session in IP-based multimedia communication systems in compliance with RFC 7989. To support session identifier, Session-ID header is added in the SIP request and response messages.
Session Identifier refers to the value of the identifier, whereas Session-ID refers to the header field used to convey the identifier.
-
When a user initiates the call, the phone while sending SIP INVITE message, generates the local-UUID.
-
When the UAS receives the SIP-INVITE, the phone picks up the local UUIDs with the incoming messages and appends it to the received Session-ID header and sends the header in reponses.
-
The same UUIDs are maintained in all the SIP messages of a particular session.
-
The phone maintains the same local-UUID during other features, such as conference or transfer.
-
This header is implemented in REGISTER method, the local-UUID remains same for all the REGISTER messages till the phone fails to REGISTER.
The Session-ID comprises of Universally Unique Identifier (UUID) for each user agent participating in a call. Each call consists of two UUID known as local UUID and remote UUID. Local UUID is the UUID generated from the originating user agent and remote UUID is generated from the terminating user agent. The UUID values are presented as strings of lower-case hexadecimal characters, with the most significant octet of the UUID appearing first. Session Identifier comprises of 32 characters and remains same for the entire session.
Session ID format
Components will implement Session-ID which is global session ID ready.
A sample current session ID passed in http header by phones (dashes are just included for clarity) is 00000000-0000-0000-0000-5ca48a65079a.
A session-ID format: UUUUUUUUSSSS5000y000DDDDDDDDDDDD where,
UUUUUUUU－A randomly generated unique ID[0-9a-f] for the session. Examples of new session IDs generated are:
-
Phone going off hook
-
Entry of the activation code through to first SIP first registration (the onboarding flow)
SSSS－The source that generates the session. For example, if the source type is "Cisco MPP" the source value (SSSS) can be "0100".
Y－Any of the values of 8, 9, A, or B and should be compliant with UUID v5 RFC.
DDDDDDDDDDDD－MAC address of the phone.
SessionID examples in SIP messages
This header is supported in the in-call dialog messages like INVITE/ACK/CANCEL/BYE/UPDATE/INFO/REFER and their responses as well as out-of-call messages essentially the REGISTER.
Request-Line: INVITE sip:901@10.89.107.37:5060 SIP/2.0
Session-ID: 298da61300105000a00000ebd5cbd5c1;remote=00000000000000000000000000000000
Status-Line: SIP/2.0 100 Trying
Session-ID: fbaa810a00105000a00000ebd5cc118b;remote=298da61300105000a00000ebd5cbd5c1
Status-Line: SIP/2.0 180 Ringing
Session-ID: fbaa810a00105000a00000ebd5cc118b;remote=298da61300105000a00000ebd5cbd5c1
Status-Line: SIP/2.0 200 OK
Session-ID: fbaa810a00105000a00000ebd5cc118b;remote=298da61300105000a00000ebd5cbd5c1
Request-Line: ACK sip:901@10.89.107.37:5060 SIP/2.0
Session-ID: 298da61300105000a00000ebd5cbd5c1;remote=fbaa810a00105000a00000ebd5cc118b
Request-Line: BYE sip:901@10.89.107.37:5060 SIP/2.0
Session-ID: 298da61300105000a00000ebd5cbd5c1;remote=fbaa810a00105000a00000ebd5cc118b
Status-Line: SIP/2.0 200 OK
Session-ID: fbaa810a00105000a00000ebd5cc118b;remote=298da61300105000a00000ebd5cbd5c1You can enable SIP session ID to overcome the limitations with the existing call-identifiers and to allow end-to-end tracking of a SIP session.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
Go to the SIP Settings section. |
| 4 |
Set SIP SessionID Support to Yes. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: <SIP_SessionID_Support_1_ ua="na">Yes</SIP_SessionID_Support_1_>
|
| 5 |
Click Submit All Changes. |
You can configure the settings on the phone administration web page to enable status synchronization of Do Not Disturb (DND) and call forward between the phone and the server.
There are two ways to synchronize the feature status:
-
Feature Key Synchronization (FKS)
-
XSI Synchronization
FKS uses SIP messages to communicate the feature status. XSI Synchronization uses HTTP messages. If both FKS and XSI synchronization are enabled, FKS takes precedent over XSI synchronization. See the table below for how FKS interacts with XSI synchronization.
|
Feature Key Sync |
XSI DND Enabled |
XSI CFWD Enabled |
DND Sync |
CFWD Sync |
|---|---|---|---|---|
|
Yes |
Yes |
Yes |
Yes (SIP) |
Yes (SIP) |
|
Yes |
No |
No |
Yes (SIP) |
Yes (SIP) |
|
Yes |
No |
Yes |
Yes (SIP) |
Yes (SIP) |
|
No |
Yes |
Yes |
Yes (HTTP) |
Yes (HTTP) |
|
No |
No |
Yes |
No |
Yes (HTTP) |
|
No |
Yes |
No |
Yes (HTTP) |
No |
|
No |
No |
No |
No |
No |
When you enable the Feature Key Synchronization (FKS), the settings of call forward and do not disturb (DND) on the server are synchronized to the phone. The changes in DND and call forward settings made on the phone will also be synchronized to the server.
When FKS is enabled on a line, the line get the DND and Call Forward settings from the server and doesn't sync with the settings on tab.
| 1 |
Access the phone administration web page. |
| 2 |
Select (where [n] is the extension number). |
| 3 |
In the Call Feature Settings section, set the Feature Key Sync field to Yes. |
| 4 |
Click Submit All Changes. |
When call forward sync is enabled, the settings related to call forward on the server are synchronized to the phone. The changes in call forward settings made on the phone will also be synchronized to the server.
Before you begin
-
Configure the XSI host server and the corresponding credentials on the tab.
-
When using Login Credentials for XSI server authentication, enter XSI Host Server, Login User ID, and Login Password in the XSI Line Service section.
-
When using SIP Credentials for XSI server authentication, enter XSI Host Server and Login User ID in the XSI Line Service section, and Auth ID and Password in the Subscriber Information section.
-
-
Disable Feature Key Sync (FKS) in Call Feature Settings section from .
| 1 |
Access the phone administration web page. |
| 2 |
Select (where [n] is the extension number). |
| 3 |
In the XSI Line Service section, set the CFWD Enable parameter to Yes. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: No |
| 4 |
Click Submit All Changes. |
When do not disturb (DND) sync is enabled, the DND setting on the server is synchronized to the phone. The changes in DND setting made on the phone will also be synchronized to the server.
Before you begin
-
Configure the XSI host server and the corresponding credentials on the tab.
-
When using Login Credentials for XSI server authentication, enter XSI Host Server, Login User ID, and Login Password in the XSI Line Service section.
-
When using SIP Credentials for XSI server authentication, enter XSI Host Server and Login User ID in the XSI Line Service section, and Auth ID and Password in the Subscriber Information section.
-
-
Disable Feature Key Synchronization (FKS) in Call Feature Settings section from .
| 1 |
Select (where [n] is the extension number). |
| 2 |
In the XSI Line Service section, set the DND Enable parameter to Yes. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: No |
| 3 |
Click Submit All Changes. |
You can sync the Block caller id status on the phone and the Line ID Blocking status on the BroadWorks XSI server. When you enable the synchronization, the changes that the user makes in the Block caller id settings also changes the BroadWorks server settings.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the XSI Line Service section, set the Block CID Enable parameter. Choose Yes to enable the synchronization of blocking caller id status with the server using XSI interface. Choose No to use the phone's local blocking caller id settings. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: No |
| 4 |
Click Submit All Changes. |
You can prioritize voice or video data in limited bandwidth conditions.
You need to configure the priorities individually on each line of a phone.
You can configure different priorities for different areas of traffic. For example, you can configure different priorities for internal and external traffic by setting up different configurations on internal and external lines. For effective traffic management, specify the same settings on all the phone lines in a group.
The Type of Service (ToS) field of a data packet determines the packet's priority in data traffic. You can configure the desired priorities by specifying appropriate values for the ToS fields of voice and video packets, for each phone line.
For voice data, the phone applies the ToS value that it receives by LLDP. When there is no ToS value available by LLDP, the phone applies the value that you specify for voice packets.
For video data, the phone always applies the ToS value that you specify for video packets.
The default values prioritize voice over video.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. To configure each parameter, see the syntax of the string in the following table of Parameters for prioritizing voice and video Data.
| 1 |
Access the phone administration web page. |
| 2 |
Select , where n is an extension number. |
| 3 |
In the Network Settings section, set the parameter values as described in the following table of Parameters for prioritizing voice and video Data. |
| 4 |
Click Submit All Changes. |
Parameters for prioritizing voice and video data
The following table defines the function and usage of Configure Priorities for Voice and Video Data parameters in the Network Settings section under the tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.
|
Parameter |
Description |
|---|---|
|
SIP TOS/DiffServ Value |
Time of service (ToS)/differentiated services (DiffServ) field value in UDP IP packets carrying a SIP message. Perform one of the following:
Default: 0x68 |
|
RTP ToS/DiffServ Value |
Value for the ToS field of voice data packets. Sets the priority for voice packets in data traffic. Perform one of the following:
Default: 0xb8 |
|
Video RTP ToS/DiffServ Value |
Value for the ToS field of video data packets. Sets the priority for video packets in data traffic. Perform one of the following:
Default: |
You can configure remote SDK for your phone. The remote SDK provides a WebSocket-based protocol through which the phone can be controlled.
Before you begin
A WebSocket server must be running with an address and port reachable from the phone.
| 1 |
Select . |
| 2 |
Go to the WebSocket API section. |
| 3 |
Set the Control Server URL and the Allowed APIs fields as described in the following table of WebSocket API parameters. |
| 4 |
Click Submit All Changes. |
WebSocket API parameters
The following table defines the function and usage of each parameter in the WebSocket API section in the tab of the phone web page. It also defines the syntax of the string that is added in the phone configuration file with XML (cfg.xml) code to configure a parameter.
|
Parameter Name |
Description and Default Value |
|---|---|
|
Control Server URL |
The URL of a WebSocket server to which the phone attempts to stay connected.
The URL should be in one of the following formats:
We recommend a secure connection. Default: Empty. |
By default, the phone's rollover counter (ROC) keeps its value after a re-keying occurs since SSRC, IP, or port is not changed. This is for compliance to the RFC 3711.
However, if the remote side is not fully compliance to the RFC 3711, this may result in certain interop issue (for example, one-way audio issue) when the IP phone and the remote side are on a security call.
To enhance the phone's compatibility for this situation, you can enable the ROC reset.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the RTP Parameters section, set the parameter RX ROC Reset on Re-Key to Yes. You can also configure this parameter in the configuration file:
Allowed values: Yes and No. Default: No If you set the parameter to Yes, the phone will reset Rx ROC value after re-keying without SSRC/IP/Port changes. If set to No, the phone will keep Rx ROC value after re-keying without SSRC/IP/Port changes. |
| 4 |
Click Submit All Changes. |
You can set up multicast paging to allow users to send paging messages to other phones. The page can go to all phones or a group of phones in the same network. Any phone in the group can initiate a multicast paging session. The page is received only by the phones that are set to listen for the paging group.
A phone can be added to up to10 paging groups. Each paging group has a unique multicast port and number. The phones within a paging group must subscribe to the same multicast IP address, port, and multicast number.
You configure the priority for the incoming page from a specific group. When a phone is active and an important page must be played, the user hears the page on the active audio path.
When multiple paging sessions occur, they are answered in chronological order. After the active page ends, the next page is automatically answered. When do not disturb (DND) applies to all lines on a phone instead of a specific line, the phone ignores any incoming paging.
You can specify a codec for the paging to use. The supported codecs are G711a, G711u, G722, and G729. If you don't specify the codec, paging uses G711u by default.
You can also enable phones to receive pages from a server to optionally display an image or other UI elements. With this feature, an XML service can be invoked during a multicast paging. To enable this feature, configure the parameter XML Application Service URL and add "xmlapp=yes" in the paging group scripts from . For more information about the parameters, see XML applications configuration for phones on BroadWorks and Parameters for multiple paging group.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. To configure each parameter, see the syntax of the string in Parameters for multiple paging group.
Before you begin
- Make sure that your network supports multicast so that all devices in the same paging group are able to receive paging.
- For Wi-Fi networks, enable and properly configure the access point for multicast.
- Make sure that all the phones in a paging group are in the same network.
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
Go to the Multiple Paging Group Parameters section. |
| 4 |
Enter multicast paging scripts as defined in the following table of Parameters for multiple paging group. |
| 5 |
Click Submit All Changes. |
Parameters for multiple paging group
The following table defines the function and usage of the multiple paging group parameters in the phone administration page.
It also defines the syntax of the string that is added in the phone configuration file (cfg.xml) to configure a parameter.
| Parameter | Description |
|---|---|
| Group 1 Paging Script – Group 10 Paging Script |
Enter a string to configure the phone to listen for and initiate multicast paging. You can add a phone to up to 10 paging groups. Enter the script in this format:
In the phone configuration file with XML(cfg.xml), enter a string in this format:
Default: Empty |
The paging feature enables a user to directly contact another user by phone. If the phone of the person being paged has been configured to answer pages automatically, the phone does not ring. Instead, a direct connection between the two phones is automatically established when paging is initiated.
You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.
Before you begin
| 1 |
Access the phone administration web page. |
| 2 |
Select . |
| 3 |
In the Supplementary Services section, choose Yes for the Auto Answer Page parameter. You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format: Options: Yes and No Default: Yes |
| 4 |
Click Submit All Changes. |
