Chapter 1. Cisco IP Phone Hardware

- Overview of the Cisco IP Phone 6800 Series Multiplatform Phones

- Cisco IP Phone 6821 Multiplatform Phones Connections

- Cisco IP Phone 6841 Multiplatform Phones Connections

- Cisco IP Phone 6851 Multiplatform Phones Connections

- Cisco IP Phone 6861 Multiplatform Phones Connections

- Cisco IP Phone 6871 Multiplatform Phones Connections

- Cisco IP Phone 6821 Multiplatform Phones Buttons and Hardware

- Cisco IP Phone 6841, 6851, and 6861 Multiplatform Phones Buttons and Hardware

- Cisco IP Phone 6871 Multiplatform Phones Buttons and Hardware

- Softkey, Line, and Feature Buttons

- Terminology Differences

Overview of the Cisco IP Phone 6800 Series Multiplatform Phones

The Cisco IP Phone 6800 Series Multiplatform Phones comprises a set of full-featured VoIP (Voice-over-Internet Protocol) phones that provide voice communication over an IP network. The phones provide all the features of traditional business phones, such as call forwarding, redialing, speed dialing, transferring calls, and conference calling. The Cisco IP Phone 6800 Series Multiplatform Phones is targeted for solutions that are centered on third-party SIP-based IP PBX.

Note

This document doesn't include the DECT phones.

The Cisco IP Phone 6800 Series Multiplatform Phones deliver easy-to-use, highly-secure voice communications. In the following figure, you see (Clockwise from the top left):

- Cisco IP Phone 6841 Multiplatform Phone

- Cisco IP Phone 6851 Multiplatform Phone with Key Expansion Module

- Cisco IP Phone 6851 Multiplatform Phone

- Cisco IP Phone 6861 Multiplatform Phone

- Cisco IP Phone 6871 Multiplatform Phone

- Cisco IP Phone 6821 Multiplatform Phone

Cisco IP Phone 6800 Series Multiplatform Phones

Note

In this document, the terms Cisco IP Phone, phone, or device mean Cisco IP Phone 6800 Series Multiplatform Phones.

Cisco IP Phone 6821 Multiplatform Phones Connections

Connect your phone to your LAN with an Ethernet cable to enable full functionality of your phone. If your Ethernet port is equipped with Power over Ethernet (PoE), you can power the phone through the LAN port. If you don't have PoE available, then you need to use a power adapter to power the phone. Do not extend the Ethernet cable outside the building. For your phone to work, it must be connected to the IP telephony network.

Cisco IP Phone 6821 Multiplatform Phones Connections

1

DC adapter port (optional)

4

Access port (10/100 PC) connection (optional)

2

Power adapter (optional)

5

Handset connection

3

Network port (10/100 SW) connection; IEEE 802.3af power enabled

6

Analog headset connection (optional)

Cisco IP Phone 6841 Multiplatform Phones Connections

Connect your phone to your LAN with an Ethernet cable to enable full functionality of your phone. You need to use a power adapter to power the phone. Do not extend the LAN Ethernet cable outside the building. For your phone to work, it must be connected to the IP telephony network.

1

DC adapter port

4

Access port (10/100/1000 PC) connection (optional)

2

Power adapter

5

Handset connection

3

Network port (10/100/1000 SW) connection

6

Analog headset connection (optional)

Cisco IP Phone 6851 Multiplatform Phones Connections

Connect your phone to your LAN with an Ethernet cable to enable full functionality of your phone. If your Ethernet port is equipped with Power over Ethernet (PoE), you can power the phone through the LAN port. If you don't have PoE available, then you need to use a power adapter to power the phone. Do not extend the LAN Ethernet cable outside the building. For your phone to work, it must be connected to the IP telephony network.

1

Key Expansion Module port

5

Access port (10/100/1000 PC) connection (optional)

2

DC adapter port (optional)

6

Auxiliary port (optional)

3

Power adapter (optional)

7

Handset connection

4

Network port (10/100/1000 SW) connection; IEEE 802.3af power enabled

8

Analog headset connection (optional)

Cisco IP Phone 6861 Multiplatform Phones Connections

Connect your phone to your LAN with an Ethernet cable or with Wi-Fi to enable full functionality of your phone. You need to use a power adapter to power the phone. If you connect your phone to a wired network, do not extend the LAN Ethernet cable outside the building. For your phone to work, it must be connected to the IP telephony network.

1

DC adapter port

4

Auxiliary port (optional)

2

Power adapter

5

Handset connection

3

Network port (10/100 SW) connection

6

Analog headset connection (optional)

Cisco IP Phone 6871 Multiplatform Phones Connections

Connect your phone to your LAN with an Ethernet cable to enable full functionality of your phone. If your Ethernet port is equipped with Power over Ethernet (PoE), you can power the phone through the LAN port. If you don't have PoE available, then you need to use a power adapter to power the phone. Do not extend the LAN Ethernet cable outside the building. For your phone to work, it must be connected to the IP telephony network.

1

USB Headsets

5

Access port (10/100/1000 PC) connection (optional)

2

DC adapter port (optional)

6

Auxiliary port (optional)

3

Power adapter (optional)

7

Handset connection

4

Network port (10/100/1000 SW) connection; IEEE 802.3af power enabled

8

Analog headset connection (optional)

Cisco IP Phone 6821 Multiplatform Phones Buttons and Hardware

The following figure shows the Cisco IP Phone 6821 Multiplatform Phones.

Cisco IP Phone 6821 Multiplatform Phones

1

Light strip

Indicates whether you have an incoming call (flashing red) or a new voice message (steady red).

2

Programmable feature buttons and line buttons

Access your phone lines, features, and call sessions.

For more information, see Softkey, Line, and Feature Buttons.

3

Softkey buttons

Access functions and services such as Conference and Transfer.

For more information, see Softkey, Line, and Feature Buttons.

4

Navigation cluster

Navigation ring and Select button. Scroll through menus, highlight items, and select the highlighted item.

5

Applications and Headset

Applications Access call history, user preferences, phone settings, and phone model information.

Headset Toggle the headset on or off. When a headset is active, a headset icon is displayed in the header.

6

Mute and Speakerphone

Mute Toggle the microphone on or off. When the microphone is muted, a mute icon flashes on the screen.

Speakerphone Toggle the speakerphone on or off.

7

Volume button

Adjust the handset, headset, and speakerphone volume (off hook) and the ringer volume (on hook).

Cisco IP Phone 6841, 6851, and 6861 Multiplatform Phones Buttons and Hardware

The following figure shows the Cisco IP Phone 6841.

Cisco IP Phone 6841, 6851, and 6861 Multiplatform Phones Buttons and Features

1

Handset and Handset light strip

Indicates whether you have an incoming call (flashing red) or a new voice message (steady red).

2

Programmable feature buttons and line buttons

Access your phone lines, features, and call sessions.

For more information, see Softkey, Line, and Feature Buttons.

3

Softkey buttons

Access functions and services.

For more information, see Softkey, Line, and Feature Buttons.

4

Navigation cluster

Navigation ring and Select button. Scroll through menus, highlight items, and select the highlighted item.

5

Hold/Resume, Conference, and Transfer

Hold/Resume Place an active call on hold and resume the held call.

Conference Create a conference call.

Transfer Transfer a call.

6

Speakerphone, Mute, and Headset

Speakerphone Toggle the speakerphone on or off. When the speakerphone is on, the button is lit.

Mute Toggle the microphone on or off. When the microphone is muted, the button is lit.

Headset Toggle the headset on or off. When the headset is on, the button islit.

7

Contacts, Applications, and Messages

Contacts Access personal and corporate directories.

Applications Access call history, user preferences, phone settings, and phone model information.

Messages Autodial your voice messaging system.

8

Volume button

Adjust the handset, headset, and speakerphone volume (off hook) and the ringer volume (on hook).

Cisco IP Phone 6871 Multiplatform Phones Buttons and Hardware

The following figure shows the Cisco IP Phone 6871.

Cisco IP Phone 6871 Multiplatform Phones Buttons and Features

1

Handset and Handset light strip

Indicates whether you have an incoming call (flashing red) or a new voice message (steady red).

2

Programmable feature buttons and line buttons

Access your phone lines, features, and call sessions.

For more information, see Softkey, Line, and Feature Buttons.

3

Softkey buttons

Access functions and services.

For more information, see Softkey, Line, and Feature Buttons.

4

Navigation cluster

Navigation ring and Select button. Scroll through menus, highlight items, and select the highlighted item.

5

Hold/Resume, Conference, and Transfer

Hold/Resume Place an active call on hold and resume the held call.

Conference Create a conference call.

Transfer Transfer a call.

6

Speakerphone, Mute, and Headset

Speakerphone Toggle the speakerphone on or off. When the speakerphone is on, the button is lit.

Mute Toggle the microphone on or off. When the microphone is muted, the button is lit.

Headset Toggle the headset on or off. When the headset is on, the button islit.

7

Contacts, Applications, and Messages

Contacts Access personal and corporate directories.

Applications Access call history, user preferences, phone settings, and phone model information.

Messages Autodial your voice messaging system.

8

Volume button

Adjust the handset, headset, and speakerphone volume (off hook) and the ringer volume (on hook).

Softkey, Line, and Feature Buttons

You can interact with the features on your phone in several ways:

- Softkeys, located below the screen, give you access to the function displayed on the screen above the softkey. The softkeys change depending on what you are doing at the time. The More ... softkey shows you that more functions are available.

- Feature and line buttons give you access to phone features and phone lines. On the Cisco IP Phone 6821, these are the buttons on the left side of the screen. On the Cisco IP Phone 6841, 6851, 6861, and 6871, these are the buttons on either side of the screen.

- Feature buttons—Used for features such as Speed dial or Call pickup, and to view your status on another line.

- Line buttons—Used to initiate or answer a call. You can also use a line key to open and close the call session window, and to navigate through the call session window. Open the call session window to see the calls on the line.

Feature and line buttons illuminate to indicate status:

- or Green—Line is idle.

- or Red, steady—Line is active or in use.

- or Red, flashing—Line is on hold or there is an inbound call.

- or Amber, steady—Line is unregistered (cannot be used).

Some functions can be set up as softkeys or as feature buttons. You can also access some functions with softkeys or the associated hard button.

The Cisco IP Phone 6821 has a limited number of hard buttons. You use the softkeys to access most call features.

Terminology Differences

The following table highlights some of the terminology differences in the Cisco IP Phone 6800 Series Multiplatform Phones User Guide and the Cisco IP Phone 6800 Series Multiplatform Phones Administration Guide

Terminology Differences

User Guide

Administration Guide

Message Indicators

Message Waiting Indicator (MWI) or Message Waiting Lamp

Voicemail System

Voice Messaging System

Phone Web Page

Configuration Utility

Chapter 2. New and Changed Information

- New and Changed for Firmware Release 11.3(7)

- New and Changed for Firmware Release 11.3(6)

- New and Changed for Firmware Release 11.3(5)

- New and Changed for Firmware Release 11.3(4)

- New and Changed for Firmware Release 11.3(3)

- New and Changed for Firmware Release 11.3(2)

- New and Changed for Cisco IP Phone 6871 Multiplatform Phones with Firmware Release 11.3(1)

- New and Changed for Firmware Release 11.3(1)

- New and Changed for Firmware Release 11.2(4)

- New and Changed for Firmware Release 11.2(3)SR1

- New and Changed for Firmware Release 11.2(3)

- New and Changed for Firmware Release 11.2(2)

- New and Changed for Firmware Release 11.2(1)

- New and Changed for Firmware Release 11.1(2)

- New and Changed for Firmware Release 11.1(1)

New and Changed for Firmware Release 11.3(7)

Revision

New and Changed

Updated the topic to add the new feature Inert

Configurable Features on Line Keys

Added the task for the new feature Inert Mode for PLK Configuration

Shut Down a Line Key

Shut Down a Line Key on a Key Expansion Module

Updated the topics to add a prerequisite for the new feature Inert Mode for PLK Configuration

Enable the User to Configure Features on Line Keys

Enable the User to Configure Features on Key Expansion Module Line Keys

Select the Line Key LED Behavior

Configure the Phone to Monitor Multiple Users' Lines

Configure a Line Key on the Phone to Monitor a Single User's Line

Added the task about how to support Spam indication for incoming calls

Spam Indication for Incoming Webex Calls

Updated the topic to add a reference to the topic to "Spam Indication for Incoming Webex Calls"

Spam Indication for Incoming Calls

Added the topic for the feature VPN Connection Support

Set Up a VPN Connection from the Phone

View the VPN Status

Set Up a VPN Connection from the Phone Web Page

Parameters for VPN Settings

Reboot the Phone from the Phone Web Page

Updated the topic to add the feature VPN Connection Support

Telephony Features

Updated the topic to add the shortcut strings of the phone menus VPN settings and VPN status

Menu Shortcuts Mapping on PLK and PSK

Updated the topic to add new fields of the feature VPN Connection Support

Network Configuration Fields

Added the task about how to enable support for LDAP unified search

Enable LDAP Unified Search

Added the task about how to enable the feature Direct PLK Configuration

Enable Direct PLK Configuration

Updated the topics for the feature Direct PLK Configuration

Enable the User to Configure Features on Line Keys

Parameters for Features on Line Keys

Configure Speed Dial on a Line Key

Parameters for Monitoring a Single Line

Add Call Park to a Line Key

Configure Executive and Assistant Menu Access on a Line Key

Add a Menu Shortcut to a Line Key

Add an Extended Feature to a Line Key

Enable the Extension of a Line Key

Add an XML Service to a Line Key

Configure Webex Contacts on a Line Key

Configure the Voicemail PLK on a Line Key

Configure the Voicemail PLK on the Phone

Enable the User to Configure Features on Key Expansion Module Line Keys

Added the task about how to disable a USB port

Disable the USB Port

Updated the topic to mention the general call records that the users can view

Display Call Statistics Screen

Updated the topic to update the table

Call Statistics Fields

Updated the topic to add more steps

Set the User Password

Added the task about how to set up a proxy server in the phone web page

Set Up a Proxy Server

Added the topic for the feature HTTP Proxy Support

Parameters for HTTP Proxy Settings

Updated the topic to add the feature HTTP Proxy

Telephony Features

Updated the topic to add the shortcut string of the phone menu HTTP proxy settings

Menu Shortcuts Mapping on PLK and PSK

Updated the topic to add new fields of the feature HTTP Proxy

Network Configuration Fields

Updated the topic to mention the HTTP proxy settings

Onboard Your Phone with the Activation Code

Use DNS SRV for HTTP Provisioning

Enable Auto Provisioning with Short Activation Code

Set Up Your Phones for Activation Code Onboarding

Added the task for Webex contact support

Enable Webex Contacts on the Phone

Added the task for Webex contact support on a line key

Configure Webex Contacts on a Line Key

Added the taskfor Webex contact support on a softkey

Add a Softkey for Webex Contacts

Updated the topic for Webex contact on PSK and PLK

Menu Shortcuts Mapping on PLK and PSK

Added the topic for Webex call log support

Enable Webex Call Logs on the Phone

Added the topic to troubleshoot an issue about the phone onboarding to Webex

Failed to Onboard the Phone to Webex

New and Changed for Firmware Release 11.3(6)

Revision

New and Changed

Updated the task to add the situations in which the focus moves to the incoming call

Keep Focus on the Active Call

Updated the description of Call Forward

Telephony Features

Updated the topic to add the new parameter Forward Softkey

Parameters for Enable Call Forward on User Tab

Updated the softkey in the topic for the new feature

Enable Feature Activation Code Synchronization for Forward All Calls

Updated the topic to add the support for French (Canada) language.

Setup for Latin and Cyrillic Languages

Supported Languages for the Phone Display

Set Up Dictionaries and Fonts

Updated the description of the parameter Display Attrs

Parameters for LDAP Directory

Added the new task topic on how to restrict users from controlling the ringer volume

Restrict User from Controlling the Ringer Volume

Updated the task topic on non-availability of DND softkey

Enable DND Status Sync via XSI Service

Updated the task topic on non-availability of Forward softkey

Enable Call Forward Status Sync via XSI Service

Updated the task topic on non-availability of Save softkey on the Add BroadSoft personal contact screen

Parameters for XSI Phone Service

Updated the task topic on Select softkey on the Select feature screen

Enable the User to Configure Features on Line Keys

Added the new topic for Webex cloud onboarding

Phone Onboarding to Webex Cloud

Enable a Phone to Onboarding to Webex Cloud

Added the new topic for PRT generation from Cisco Webex Control Hub

Report Phone Issues from Webex Control Hub

Added the new topic to reboot from Cisco Webex Control Hub

Reboot the Phone from the Webex Control Hub

Added the new topic about how to resolve a downgrade issue

Saved Passwords Become Invalid after Downgrade

New and Changed for Firmware Release 11.3(5)

Revision

New and Changed

Updated the topic to remove a duplicated sentence

Configure Voicemail

Rewrote the topic.

Configure Voicemail for An Extension

Updated the topic to add a reference link.

Configure the Message Waiting Indicator

Updated the topic to add new parameters

Parameters for Voicemail Server and Message Waiting

Added the task about how to enable the feature

Keep Focus on the Active Call

Updated the topic to add MIC Cert Refresh Status

View Download Status

Added the task about how to renew the MIC certificate

Renew the MIC Certificate

Added the topic for the feature MIC Certificate Renewal by SUDI Service

Parameters for MIC Certificate Renewal by SUDI Service

Added the topic to support STIR/SHAKEN

Spam Indication for Incoming Calls

Added the task for dialog-based shared line

Add Dialog-Based Shared Line Appearance

Added the topic about how to set the upgrade rule for the Cisco Headsets

Set the Upgrade Rule for the Cisco Headset

Updated the topic for the Cisco Headset 700 Series

Report Headset Inventory

New and Changed for Firmware Release 11.3(4)

Revision

New and Changed

Added the topic about how to configure a phone to report the connected or disconnected peripheral information to the server.

Report Headset Inventory

Added a new topic for RTL language support

Setup for RTL Languages

Updated the existing topic with RTL language entries

Supported Languages for the Phone Display

Updated the existing topic with RTL language entries

Set Up Dictionaries and Fonts

Added the task about how to enable SSRC reset to avoid a call transfer error

Enable SSRC Reset for the New RTP and SRTP Sessions

Updated the topic to add the new parameter SSRC Reset on RE-INVITE

RTP Parameters

Updated the number of the DNS SRV records

SIP Proxy Redundancy

Added the task about how to disable or enable the Don't Fragment Bit feature

Disable or Enable DF Bit

New and Changed for Firmware Release 11.3(3)

Revision

New and Changed

Updated the topic to add the new parameter Add Contacts to Directory Personal

Parameters for XSI Phone Service

Added Executive and Assistant relevant topics for Cisco IP Phone 6871 Multiplatform Phones only

Enable Alphanumeric Dialing for Executives

Set Up Preference for the Executive-Assistant Role

Selection Logic of the Executive-Assistant Role

Executives and Assistants

Executive-Assistant Setting Synchronization

Dial Plan for Executives and Assistants

Configure Executive and Assistant Menu Access on a Line Key

Service Activation Codes for Executives and Assistants

Programmable Softkeys for Executives and Assistants

Hide a Menu Item from Being Displayed on the Phone Screen

Parameters for Menu Visibility

Executive or Assistant Menu Does Not Appear

Updated the topics regarding the Executive and Assistant for Cisco IP Phone 6871 Multiplatform Phones only

Telephony Features

Feature Buttons and Softkeys

Enable Feature Key Sync

DND and Call Forward Status Sync

Line Key LED Behaviour Customization

Hide a Menu Item from Being Displayed on the Phone Screen

Parameters for Menu Visibility

Menu Shortcuts Mapping on PLK and PSK

Configurable Features on Line Keys

Enable Synchronization of Settings Between the Phone and the Server

Programmable Line Keys

Programmable Softkeys

Digit Sequence Examples

Vertical Service Activation Codes

Added the topics for the Synchronization of Call Waiting and Anonymous Call Rejection feature

Enable Synchronization of Anonymous Call Rejection via XSI Service

Set Feature Activation Code for Anonymous Call Rejection

Enable Synchronization of Call Waiting via XSI Service

Set Feature Activation Code for Call Waiting

Added the task topic on how to display or hide the Unavailability menu text box of agent status on the phone

Display or Hide Unavailable Menu Text Box of Agent Status on the Phone

Added the task topic on how to configure softkeys for different types of calls history list

Enable Softkeys to Calls History List Menu

Updated the topic to add new parameters PRT HTTP Header and PRT HTTP Header Value

Parameters for Configure Problem Report Tool

Updated the topic to add the parameter Precondition Support and update the parameter SIP 100REL Enable

Parameters for SIP Settings on Extensions

Updated the topic for the Product information screen on the phone

Display the Phone Information Window

Added the topic on how to customize the product configuration version

Customize the Product Configuration Version

Updated the parameters Port and User ID in the topic

Parameters for Set Up Presence

Added the topic on how to use the DNS SRV records for the BroadSoft XMPP

Use DNS SRV for XMPP

New and Changed for Firmware Release 11.3(2)

Revision

New and Changed

Added the tasks about the menu shortcuts of features on PLK and PSK

Add a Menu Shortcut to a Line Key

Menu Shortcuts Mapping on PLK and PSK

Add an Extended Feature to a Line Key
Configurable Features on Line Keys

Add a Menu Shortcut to a Programmable Softkey

Added the topics for the user authentication control feature

Control the Authentication Requirement to Access the Phone Menus

Parameters for User Authentication Control

Added the tasks about adding menu shortcuts or features to the keys of an key expansion module

Enable the User to Configure Features on Key Expansion Module Line Keys

Add a Menu Shortcut to a Key Expansion Module Line Key

Add an Extended Feature to a Key Expansion Module Line Key

Updated the topics about features on PLK and PSK with menu shortcuts

Enable the User to Configure Features on Line Keys

Parameters for Programmable Softkeys

Added the topics for the Feature Activation Code Synchronization feature

Enable Feature Activation Code Synchronization for Forward All Calls

Set Feature Activation Code for the Call Forward All Service

Added the topics introducing the SIP Proxy Redundancy enhancements

SIP Proxy Redundancy

SIP Proxy Failover

SIP Proxy Fallback

Updated the task context to support the SIP proxy redundancy enhancement

Configure the SIP Transport

Updated the description for the feature of Show Caller Name and Caller Number

Telephony Features

Added the task about configuring the caller name and number display in incoming call alerts

Display Caller Number Instead of Unresolved Caller Name

Added the task about how to disable contact search in all directories

Disable Contact Search in All Directories

Added the task about how to disable the personal directory

Disable Personal Directory

Added the task about how to hide menu items on the phone screen

Hide a Menu Item from Being Displayed on the Phone Screen

Add the reference topic about the menu visibility function

Parameters for Menu Visibility

Added the task about how to configure the directory services on the phone web page

Configure Directory Services

Added the reference topic about the directory services

Parameters for Directory Services

Updated the reference topic to add the new parameters for the directory enhancement feature

Parameters for XSI Phone Service

Updated the reference topic for the directory enhancement feature

Feature Buttons and Softkeys

Added the reference topic about a troubleshooting case for the directory enhancement feature

Phone Doesn't Show Contacts

Updated the task to add more information about how to enable call forward setting on user tab

Enable Call Forward on User Tab

Added the reference topic about the parameters for call forward settings on user tab

Parameters for Enable Call Forward on User Tab

Updated the reference topic to add new features

Feature Buttons and Softkeys

Update the description of the Message Waiting Indicator

Telephony Features

Updated the topic to add Voicemail

Programmable Line Keys

Added the topic on how to configure the voicemail PLK on a line key

Configure the Voicemail PLK on a Line Key

Added the topic on how to the voicemail PLK on a KEM button

Configure the Voicemail PLK on a Key Expansion Module Button

Added the topic on how to the voicemail PLK on the phone

Configure the Voicemail PLK on the Phone

Added the reference topic for the string syntax used in the voicemail PLK feature

String Syntax for Voicemail PLK

Updated the topic to add the voicemail PLK feature

Enable the User to Configure Features on Line Keys

Updated the topic to add new features

Configurable Features on Line Keys

Updated the topic to add the new step for MWI

Select the Line Key LED Behavior

Updated the topic to add a table

Line Key LED Behaviour Customization

Added the topics for troubleshooting voicemail PLK relevant issues

SIP Subscription Failure Message

Number of Voicemail Messages Doesn't Display

Unable to Place a Call with Speed Dial for Voicemail Messages

Failed to Sign Into a Voicemail Account

Voicemail PLK Options Don't Display on the Phone

Updated the topic to support the automatic RTP (Real-time Transport Protocol) transport selection

Set up a Secure Extension

Updated the Client DN, User Name, Password, and Auth Method parameters for LDAP directory

Parameters for LDAP Directory

Added the topic to show the logic of the LDAP directory access

Overview of LDAP Directory Access

Updated the firmware version (SWVER) format

Macro Expansion Variables

Macro Variables

Conditional Expressions

Updated the topic for the enhancement of the call park feature

Configure One-Button Call Park

Updated the topic to add prerequisites and updated the short description

Enable Hoteling on a Phone

Added the topic to describe the Flexible Seating feature of BroadWorks and how to enable it

Enable Flexible Seating on a Phone

Added the topic on how to enable EM for the user

Enable Extension Mobility on a Phone

Updated the topic to support the reverse name lookup against local contacts for BroadWorks server call logs

Enable Viewing BroadWorks XSI Call Logs on a Line

Added the task on how to configure StartTLS

Configure StartTLS

Updated the topic to add "StartTLS"

Enable Hostname Verification for SIP over TLS

Updated the topic to add the new parameter of the feature

Configure the Cipher List

Updated the topic for "StartTLS"

Parameters for LDAP Directory

Updated the topic for "StartTLS"

XML and TR-069 Parameter Comparison

Updated the topic to support the reverse name lookup

Enable Reverse Name Lookup

Updated the topic to add the new parameter of the feature

Parameters for Call Center Agent Setup

Updated the description of Agent Sign-in and Sign-out for the feature

Configurable Features on Line Keys

Added the task for the ACD feature sync

Restore the ACD Status

New and Changed for Cisco IP Phone 6871 Multiplatform Phones with Firmware Release 11.3(1)

Revision

New and Changed Sections

Updated the hardware-related topics with Cisco IP Phone 6871 Multiplatform Phones

Overview of the Cisco IP Phone 6800 Series Multiplatform Phones

Cisco IP Phone 6871 Multiplatform Phones Connections

Cisco IP Phone 6871 Multiplatform Phones Buttons and Hardware

Install Your New Cisco IP Phone 6800 Series Multiplatform Phones

Physical and Operating Environment Specifications

Phone Power Requirements

Added supported Cisco headsets

Headsets

New and Changed for Firmware Release 11.3(1)

Revision

New and Changed

Added a new task to support the feature Auto Provisioning with Short Activation Code.

Enable Auto Provisioning with Short Activation Code

Added topics to support HTTP provisioning with DNS servers

DNS SRV for HTTP Provisioning

Add tasks to support MPP OS Hardening

Enable the Firewall

Configure Your Firewall with Additional Options

Added a new task about how to configure a cipher list

Configure the Cipher List

Added a task and the relevant parameters to support client-initiated mode for media plane security negotiations

Enable Client-Initiated Mode for Media Plane Security Negotiations

Added a task about how to enable hostname verification for a line that uses SIP over TLS

Enable Hostname Verification for SIP over TLS

Added a task to support One-Button Call Park.

Configure One-Button Call Park

Added a task and the parameter about multicast paging

Configure Multicast Paging

Parameters for Multiple Paging Group

Added a task and the relevant parameters to support the remote SDK feature

Set Up a Phone for Remote SDK

WebSocket API Parameters

Add Task for configuring a programmable softkey (PSK) with DTMF support.

Configure a PSK with DTMF Support

Added a task about how to enable call statistics report in SIP BYE messages

Enable End-of-Call Statistics Reports in SIP Messages

Added a task to support the new feature VQM SIP Publish Message New Fields

Configure Voice Quality Reporting

Added new topics to support the SIP Session ID feature

SIP Session ID

Enable SIP Session ID

Session ID Parameters

Added a topics to support Line Key LED Behavior Customization.

Line Key LED Behaviour Customization

Select the Line Key LED Behavior

Added description for the new field RTP Before ACK

RTP Parameters

Updated the task on how to configure the SDP Payload Types

Configure the SDP Payload Types

Added a task to support OPUS Codec Narrowband.

Set Up Your Phone to Use OPUS Codec Narrowband

New and Changed for Firmware Release 11.2(4)

Revision

New and Changed Sections

Updated the hardware-related topics with Cisco IP Phone 6861 Multiplatform Phones

Physical and Operating Environment Specifications

Network and Computer Port Pinouts

Phone Power Requirements

Power Negotiation Over LLDP

Overview of the Cisco IP Phone 6800 Series Multiplatform Phones

Cisco IP Phone 6841, 6851, and 6861 Multiplatform Phones Buttons and Hardware

Install the Cisco IP Phone

Uppdated the supported accessories for Cisco IP Phone 6861 Multiplatform Phones

Cisco IP Phone 6800 Series Multiplatform Phones Supported Accessories

Headsets

Install Cisco IP Phone 6841, 6851, and 6861 with the Wall Mount Kit

Enable Electronic Hookswitch

Added a topic about how to connect Cisco IP Phone 6861 Multiplatform Phones

Cisco IP Phone 6800 Series Multiplatform Phones Connections

Added topics about Wi-Fi settings on Cisco IP Phone 6861 Multiplatform Phones

Wi-Fi Settings

Turn the Wi-Fi On or Off from Your Phone

Turn the Wi-Fi On or Off from the Phone Web Page

Connect the Phone to a Wi-Fi Manually

Connect Your Phone to a Wireless Network with WPS

Set Up a Wi-Fi Profile from the phone

Set Up a Wi-Fi Profile

Delete a Wi-Fi Profile

Change the Order of a Wi-Fi Profile

Scan and Save a Wi-Fi Network

View the Wi-Fi Status

View Wi-Fi Status Messages on the Phone

New and Changed for Firmware Release 11.2(3)SR1

Revision

New and Changed Sections

Added a new task to support Activation Code Onboarding

Onboard Your Phone with the Activation Code

New and Changed for Firmware Release 11.2(3)

Revisions

New and Changed Sections

Added Catalan to the supported language list

Supported Languages for the Phone Display

Updated the --key keyword details and added a note about RFC 8188-based encryption.

Report Current Phone Configuration to the Provisioning Server

Added a new task and a new field to support audio compliance standards: ETSI and TIA

Specify Audio Compliance Standard

Audio Compliance

Added a new task on how to phone users to configure speed dial and monitor a coworker's line.

Enable Users to Configure Features on Line Keys

Added a new topic replacing the existing Busy Lamp Field Configuration on a Monitoring Phone topic.

Phone Configuration for Monitoring Other Phones

Added a new topic replacing the existing Configure the Phone to Monitor Multiple Users' Lines topic.

Configure the Phone to Monitor Multiple Users' Lines

Added a new task on how to enable early media support

Enable P-Early-Media Support

Added topics replacing the "Set up Profile Account" topic to support the enhancements in profile authentication

Profile Authentication

Specify the Profile Authentication

Added new fields and topics to support the feature of DND and call forward status sync

DND and Call Forward Status Sync

Enable Feature Key Sync

Enable Call Forward Status Sync via XSI Service

Enable DND Status Sync via XSI Service

XSI Line Service

Added a new topic about including a device identifier in uploaded syslog messages.

Include a Device Identifier in Uploaded Syslog Messages

Added new fields and a new task on how to remotely report phone problems.

Report a Phone Problem Remotely

PRT Status

Added the Syslog Identifier field.

Optional Network Configuration

Replaced the Profile Account Enable parameter with the Profile Authentication Type field

Updated the description of the Profile Rule parameter.

Configuration Profile

Updated Report Rule and added new parameters: Report to Server, periodic Upload to Server, and Upload Delay on Local Change.

Upload Configuration Options

Updated the Wideband Handset Support parameter to support the wideband handset enhancement.

Voice > Phone > General

Updated the description for the Extension field.

Line Key

Updated the description of the First Name Filter and Last Name Filter fields.

LDAP Directory Parameters

Example XML parameter configuration is added for the Line Enable parameter.

Voice > Ext (n) > General

The new fields Customizable PLK Options and BLF List, and their descriptions are added to the table.

The description for the BLF List URI and Use Line Keys For BLF List fields are updated.

Voice > Att Console > General

New and Changed for Firmware Release 11.2(2)

Revisions

New and Changed Sections

Added a new topic to introduce the phone buttons and hardware of Cisco IP Phone 6821 Multiplatform Phones

Cisco IP Phone 6821 Multiplatform Phones Buttons and Hardware

Added a new topic to explain how to connect your of Cisco IP Phone 6821 Multiplatform Phones

Cisco IP Phone 6821 Multiplatform Phones Connections

Added a new topic to show how to install your Cisco IP Phone 6821 Multiplatform Phones on a wall

Install the Cisco IP Phone 6821 Multiplatform Phones on a Wall

Updated the topic to include the information about the new phone model of 6821

Overview of the Cisco IP Phone 6800 Series Multiplatform Phones

Updated the topic with the information of 6821 softkeys, line, and feature buttons

Softkey, Line, and Feature Buttons

Updated the topic with 6821 specifications

Physical and Operating Environment Specifications

Updated the topic with power requirements for Cisico IP Phone 6821 Multiplatform Phones

Phone Power Requirements

Updated the supported accessories for Cisico IP Phone 6821 Multiplatform Phones

Cisco IP Phone 6800 Series Multiplatform Phones Supported Accessories

Updated the topic with 6821 wall mount kit information

Wall Mount Kit Components

Updated the topic with the information of 6821 display screen size

Add a Logo as the Boot Display

Updated the topic to indicate that only 6851 support the EHS feature

Enable Electronic Hookswitch

Along with the above changes, some existing sections have been retitled to include Cisco IP Phone 6841 and 6851 Multiplatform Phones.

New and Changed for Firmware Release 11.2(1)

Revisions

New or Changed Sections

Updated the topics to support LCD screen does not honor "ro" and "na" attribute

Enable User Access to the Phone Interface Menus

System Configuration

Added a new topic to support NAPTR

Configure the SIP Transport

Updated the topics to support NAPTR

Telephony Features for Cisco IP Phone

SIP Settings

Added a new topic to support SIP privacy header

Configure a Privacy Header.

Updated the topic to support SIP privacy header

SIP Settings

Added a new topic to support blocking of SIP message from a non-proxy device

Block Non-Proxy SIP Messages to a Phone

Updated the topic to support blocking of SIP message from a non-proxy device

System Configuration

Added a new topic to support Peer Firmware Sharing

Peer Firmware Sharing

Enable Peer Firmware Sharing

Updated the topic to support Peer Firmware Sharing

Telephony Features for Cisco IP Phone

Firmware Upgrade

Updated the topic to support profile account

Configuration Profile

Added new topic to support call silence

Silence an Incoming Call with Ignore Soft key

Updated the topics to support call silence

Programmable Softkeys

Telephony Features for Cisco IP Phone

Added new topics to support XSI BroadWorks Anywhere

Move an Active Call from a Phone to Other Phones (Locations)

XSI Line Services

Added new topics to support XSI caller ID blocking

Sync the Block Caller ID Feature with the Phone and the BroadWords XSI Server

XSI Line Service

Added new topics to support XSI call logs

Enable Viewing BroadWorks XSI Call Logs on a Line

Updated the topics to support XSI call logs

Parameters for XSI Phone Service

Telephony Features for Cisco IP Phone

Updated the topic to support screen saver type "lock" removal

Configure the Screen Save with the Phone Web Interface

New and Changed for Firmware Release 11.1(2)

Feature

New or Changed Sections

Add support for LDAP over TLS(LDAPS). Updated administration information with a new task to configure LDAP over TLS

Configure LDAP over TLS

Add support for DHCP VLAN options. Updated administration information with new tasks to set up DHCP VLAN options

Set Up DHCP VLAN Option from Phone Web Page

Update field description with a new field DHCP VLAN Option.

VLAN Settings

Update Ethernet configuration menu on the phone with a new field DHCP VLAN Option.

Network Configuration Fields

Add XSI over HTTPS support. Updated XSI Host Server parameter in the XSI Service section.

Parameters for XSI Phone Service

Add a troubleshooting scenario when phone cannot access BroadSoft directory for XSI

Phone Cannot Access BroadSoft Directory for XSI

Add Cisco IP Phone 6800 Key Expansion Module support

Cisco 6800 Series Key Expansion Module Setup Overview

Add controlling reverse name lookup for incoming and outgoing calls

Reverse name Lookup for Incoming and Outgoing Calls

Emergency Call Support

Emergency Call Support Background

Emergency Call Support Terminology

Configure a Phone to make Emergency Calls

E911 Geolocation Configuration

An Emergency Call Doesn't Connect to Emergency Services

New and Changed for Firmware Release 11.1(1)

Feature

New or Changed Sections

Asian Language Support

Phone Display Language

Phone Display Problems

Phone Displays Irregular Fonts

Phone Screen Displays Boxes instead of Asian Characters

Phone Locale is Not Displayed

Softkey Labels are Truncated

Call Center Support

Set Up a Call Center Agent Phone

ACD Call Information Missing

ACD Settings

Phone Doesn't Show ACD Softkeys

Call Recording

Enable Remote Call Recording with SIP REC

Enable Remote Call Recording with SIP INFO

Cal Doesn't Record

Factory Reset Button in the Phone Web Page

Factory Reset the Phone from the Phone Web Interface

Factory Reset

Gigabit Support

Network Configuration Fields

IPv6 Support

Network Configuration Fields

IPv6 Information

Network Settings

IPv6 Settings

Presence

Set Up a Phone for Presence

Presence Status Doesn't Work

Phone Presence Message: Disconnected from Server

BroadSoft XMPP

Wideband Handset Support

General

Part I. Cisco IP Phone Provisioning

- Provisioning

- Provisioning Methods

- Provisioning Parameters

- Provisioning Formats

Chapter 3. Provisioning

- Provisioning Overview

- Provisioning

- TR69 Provisioning

- Communication Encryption

- Phone Behavior During Times of Network Congestion

- In-House Preprovisioning and Provisioning Servers

- Server Preparation and Software Tools

- In-House Device Preprovisioning

- Provisioning Server Setup

Provisioning Overview

Cisco IP Phones are intended for high-volume deployments by Voice-over-IP (VoIP) service providers to customers in home, business, or enterprise environments. Hence, provisioning the phone using remote management and configuration ensures the proper operation of the phone at the customer site.

Cisco supports the customized, ongoing feature configuration of the phone by using:

- Reliable remote control of the phone.

- Encryption of the communication that controls the phone.

- Streamlined phone account binding.

Phones can be provisioned to download configuration profiles or updated firmware from a remote server. Downloads can happen when the phones are connected to a network, when they are powered up, and at set intervals. Provisioning is typically part of the high-volume, VoIP deployments common by service providers. Configuration profiles or updated firmware is transferred to the device using TFTP, HTTP, or HTTPS.

At a high level, the phone provisioning process is as follows:

1. If the phone is not configured, the provisioning server information is applied to the phone using one of the following options:

- A–Downloaded from the Cisco Enablement Data Orchestration System (EDOS) Remote Customization (RC) server using HTTPS, DNS SRV, GDS (Activation code onboarding), EDOS device activation.

- B–Queried from a local DHCP server.

- C–Entered manually using the Cisco phone web-based configuration utility or Phone UI.

2. The phone downloads the provisioning server information and applies the configuration XML using the HTTPS, HTTP, or TFTP protocol.

3. The phone downloads and applies the updated firmware, if needed, using HTTPS, HTTP, or TFTP.

4. The VoIP service is established using the specified configuration and firmware.

VoIP service providers intend to deploy many phones to residential and small business customers. In business or enterprise environments, phones can serve as terminal nodes. Providers widely distribute these devices across the Internet, which are connected through routers and firewalls at the customer premises.

The phone can be used as a remote extension of the service provider back-end equipment. Remote management and configuration ensure the proper operation of the phone at the customer premises.

Provisioning

A phone can be configured to resynchronize its internal configuration state to match a remote profile periodically and on power-up. The phone contacts a normal provisioning server (NPS) or an access control server (ACS).

By default, a profile resync is only attempted when the phone is idle. This practice prevents an upgrade that would trigger a software reboot and interrupt a call. If intermediate upgrades are required to reach a current upgrade state from an older release, the upgrade logic can automate multistage upgrades.

- Normal Provisioning Server

- Phone Provisioning Practices

- Onboard Your Phone with the Activation Code

- Phone Onboarding to Webex Cloud

- Enable Auto Provisioning with Short Activation Code

- Manually Provision a Phone from the Keypad

- DNS SRV for HTTP Provisioning

Normal Provisioning Server

The Normal Provisioning Server (NPS) can be a TFTP, HTTP, or HTTPS server. A remote firmware upgrade is achieved by using TFTP or HTTP, or HTTPS, because the firmware does not contain sensitive information.

Although HTTPS is recommended, communication with the NPS does not require the use of a secure protocol because the updated profile can be encrypted by a shared secret key. For more information about utilizing HTTPS, see Communication Encryption. Secure first-time provisioning is provided through a mechanism that uses SSL functionality. An unprovisioned phone can receive a 256-bit symmetric key encrypted profile that is targeted for that device.

Phone Provisioning Practices

Typically, the Cisco IP Phone is configured for provisioning when it first connects to the network. The phone is also provisioned at the scheduled intervals that are set when the service provider or the VAR preprovisions (configures) the phone. Service providers can authorize VARs or advanced users to manually provision the phone by using the phone keypad. You can also configure provisioning using the Phone Web UI.

Check the Status > Phone Status > Provisioning from the Phone LCD UI, or Provisioning Status in the Status tab of the web-based Configuration Utility.

Onboard Your Phone with the Activation Code

This feature is available in firmware release 11-2-3MSR1, BroadWorks Application Server Release 22.0 (patch AP.as.22.0.1123.ap368163 and its dependencies). However, you can change phones with older firmware to use this feature. You instruct the phone to upgrade to the new firmware and to use the gds:// profile rule to trigger the activation code screen. A user enters a 16-digit code in the provided field to onboard the phone automatically.

Before you begin

Ensure that you allow the activation.webex.com service through your firewall to support onboarding via activation code.

If you want to set up a proxy server for the onboarding, ensure that the proxy server is configured correctly. See Set Up a Proxy Server .

Procedure

Step 1

Edit the phone config.xml file in a text or XML editor.

Step 2

Follow the example below in your config.xml file to set the profile rule for Activation Code Onboarding.

<?xml version="1.0" encoding="UTF-8"?>
<device>
<flat-profile>
<!-- System Configuration -->
<Profile_Rule ua="na">gds://</Profile_Rule>
<!--  Firmware Upgrade  -->
<Upgrade_Enable ua="na">Yes</Upgrade_Enable>
<Upgrade_Error_Retry_Delay ua="na">3600</Upgrade_Error_Retry_Delay>
<Upgrade_Rule ua="na">http://<server ip address>/sip88xx.11-2-3MSR1-1.loads</Upgrade_Rule>
<!-- <BACKUP_ACS_Password ua="na"/> -->
</flat-profile>
</device>

Note

For the firmware release after the 11.2(3) SR1, the setting of Firmware Upgrade is optional.

Step 3

Save the changes to the config.xml file.

Phone Onboarding to Webex Cloud

Phone onboarding provides a simple and secure way to onboard Webex-aware phones to Webex cloud. You can achieve the onboarding process either with activation code onboarding (GDS) or with phone MAC address (EDOS device activation).

For more information on how to generate the activation code, see Cisco BroadWorks Partner Configuration Guide, Cisco Multi-Platform Phones.

For more information on Webex-aware phone onboarding, see Webex for Cisco BroadWorks Solution Guide.

- Enable a Phone to Onboarding to Webex Cloud

Enable a Phone to Onboarding to Webex Cloud

After the successful registration of the phone to the Webex cloud, a cloud symbol appears on the phone screen.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Webex section, set the Onboard Enable parameter to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Default value: Yes

Step 3

Click Submit All Changes.

Enable Auto Provisioning with Short Activation Code

Use the steps below to enable auto provisioning with a short activation code.

Before you begin

Ensure that your phones are updated with Firmware Release 11.3(1) or later.

If you want to set up a proxy server for the phone, ensure that the proxy server is configured correctly. See Set Up a Proxy Server.

Review how to set up the CDA server for redirection profile:

https://community.cisco.com/t5/collaboration-voice-and-video/cisco-multi-platform-phones-cloud-provisioning-process/ta-p/3910244

Procedure

Step 1

Create a redirection profile name that contains a any number of digits between three and 16, inclusive. This becomes the activation code, later. Use one of these formats:

- nnn.

- nnnnnnnnnnnnnnnn

- Any number of digits between three and sixteen, inclusive. Example, 123456

Step 2

Provide the profile name that you created in step 1 to the Customer Device Activation (CDA) support team at cdap-support@cisco.com.

Step 3

Ask the CDA support team to enable your profile for discovery.

Step 4

When you get confirmation from the CDA support team, distribute the activation code to the users.

Step 5

Instruct users to press pound (#) before entering the digits at the activation screen.

Manually Provision a Phone from the Keypad

Procedure

Step 1

Press Applications .

Step 2

Select Device administration > Profile Rule.

Step 3

Enter the profile rule using the following format:

protocol://server[:port]/profile_pathname

For example:

tftp://192.168.1.5/CP_x8xx_MPP.cfg

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the URL is used as the server name. If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, or 443 for HTTPS).

Step 4

Press Resync.

DNS SRV for HTTP Provisioning

The DNS SRV for HTTP Provisioning feature enables auto provisioning of your multiplatform phone. Domain Name System Service (DNS SRV) records establish connections between a service and a hostname. When the phone looks for the location of the provisioning service, it first queries on the given DNS SRV domain name, then it queries for SRV records. The phone validates the records to confirm that the server is accessible. Then, it continues to the actual provisioning flow. Service providers can utilize this DNS SRV provisioning flow to provide auto provisioning.

DNS SRV bases the hostname validation on the certificate of the DHCP provided domain name. It is important that all SRV records use a valid certificate containing the DHCP provided domain name.

The DNS SRV query includes the DHCP domain name in its construction as follows: _<servicename>._<transport>.<domainName>.

For example, _ciscoprov-https._tls.example.com, instructs the phone to do a lookup for example.com. The phone uses the hostname and port number that's retrieved by the DNS SRV query to build the URL that it uses to download the initial configuration.

DNS SRV is one of many auto provisioning mechanisms that the phone uses. The phone tries the mechanisms in the following order:

1. DHCP

2. DNS SRV

3. EDOS

4. GDS (Activation Code Onboarding), or EDOS Device Activation

The following table describes the SRV record fields.

SRV Record Fields

Field

Description

Example

<_servicename.>

The service name begins with an underscore. Server services use symbolic names in SRV records.

After the service, a period (.) signifies that the service is established and the next section is beginning.

_ciscoprov-https. Or _ciscoprov-http.

DNS SRV doesn't support the TFTP protocol. If you use TFTP, you receive the following error message: Error - TFTP Scheme not supported in SRV lookups.

<_proto.>

The transport protocol begins with an underscore.

The period that follows the protocol signals that the protocol section has ended.

_tls.You must use HTTPS with TLS.

Or
_tcp.You must use HTTP with TCP.

<domainName>

The service domain name follows the protocol.

Hostname validation: All SRV records are validated based on the original DHCP-provided domain name. It is important that all records use a valid certificate containing the original domain name.

example.com

TTL (Time to Live)

Expiration value of the record, in seconds.

86400

Class

Internet-type—Standard BIND notation indicating that it’s an SRV record.

IN

<priority>

Each line contains a priority number. The lower the number, the earlier the phone will attempt the target hostname and port included in this DNS SRV record.

10

<weight>

If two or more services have the same priority, the weight number determines which line comes first. The lower the number, the earlier the phone will attempt the target hostname and port included in this DNS SRV record.

20

<port>

optional port number

5060

<target>

The A record of the machine providing the service.

A Records are the most basic type of DNS record and are used to point a domain or subdomain to an IP address.

pr1.example.com

Example SRV Configurations

_service._proto.name. TTL class SRV priority weight port target.

_ciscoprov-https._tls.example.com. 86400 IN SRV 10 60 5060 pr1.example.com.

_ciscoprov-https._tls.example.com. 86400 IN SRV 10 20 5060 pr2.example.com.

_ciscoprov-http._tcp.example.com. 86400 IN SRV 10 50 5060 px1.example.com.

_ciscoprov-http._tcp.example.com. 86400 IN SRV 10 30 5060 px2.example.com.

- Use DNS SRV for HTTP Provisioning

- Set the Profile Rule with the SRV Option on the Web Page

- Set the Profile Rule with the SRV Option on the Phone

Use DNS SRV for HTTP Provisioning

New phones use DNS SRV as one method of auto provisioning. For existing phones, if your network is set up for provisioning with DNS SRV for HTTP, you can use this feature to resync your phone. Sample configuration file:

<flat-profile>
<!-- System Configuration -->
<Primary_DNS ua="rw">10.89.68.150</Primary_DNS>
<Back_Light_Timer ua="rw">Always On</Back_Light_Timer>
<Peer_Firmware_Sharing ua="na">Yes</Peer_Firmware_Sharing>
<Profile_Authentication_Type ua="na">Basic Http Authentication </Profile_Authentication_Type>
<Proxy_1_ ua="na">example.com</Proxy_1_>
<Display_Name_1_ ua="na">4081001141</Display_Name_1_>
<User_ID_1_ ua="na">4081001141</User_ID_1_>
</flat-profile>

Before you begin

If you want to set up a proxy server for the HTTP provisioning, ensure that the proxy server is configured correctly. See Set Up a Proxy Server.

Procedure

Perform one of the following actions. Then, Set the Profile Rule with the SRV Option on the Web Page or Set the Profile Rule with the SRV Option on the Phone

- Place the XML configuration file, $PSN.xml, in the web server root directory.

- Place the XML configuration file, $MA.cfg, in the web server root directory/Cisco/.

Set the Profile Rule with the SRV Option on the Web Page

You can use the SRV option to download a configuration file to your phone.

Before you begin

Access the Phone Web Interface

Procedure

Step 1

Select Voice > Provisioning

Step 2

In the Profile Rule field, enter the profile rule with the SRV option. Only HTTP and HTTPS are supported.

Example:

[--srv] https://example.com/$PSN.xml

Set the Profile Rule with the SRV Option on the Phone

You can use the SRV option on your phone to download a configuration file.

Procedure

Step 1

Press Applications .

Step 2

Select Device administration > Profile rule.

Step 3

Enter the profile rule with the [--srv] parameter. Only HTTP and HTTPS are supported.

Example:

[--srv] https://example.com/$PSN.xml

Step 4

Press Resync.

TR69 Provisioning

The Cisco IP Phone helps the administrator to configure the TR69 parameters using the Web UI. For information related to the parameters, including a comparison of the XML and TR69 parameters, see the Administration Guide for the corresponding phone series.

The phones support Auto Configuration Server (ACS) discovery from DHCP Option 43, 60, and 125.

- Option 43–Vendor-specific information for the ACS URL.

- Option 60–Vendor class identifier, for the phone to identify itself with dslforum.org to the ACS.

- Option 125–Vendor-specific information for the gateway association.

RPC Methods Supported

The phones support only a limited set of Remote Procedure Call (RPC) methods as follows:

- GetRPCMethods

- SetParameterValues

- GetParameterValues

- SetParameterAttributes

- GetParameterAttributes

- GetParameterNames

- AddObject

- DeleteObject

- Reboot

- FactoryReset

- Inform

- Download: Download RPC method, the file types supported are:

- Firmware upgrade image

- Vendor configuration file

- Custom Certificate Authority (CA) file

- Transfer Complete

Event Types Supported

The phones support event types based on features and methods supported. Only the following event types are supported:

- Bootstrap

- Boot

- value change

- connection request

- Periodic

- Transfer Complete

- M Download

- M Reboot

Communication Encryption

The configuration parameters that are communicated to the device can contain authorization codes or other information that protect the system from unauthorized access. It is in the service provider’s interest to prevent unauthorized customer activity. It is in the customer’s interest to prevent the unauthorized use of the account. The service provider can encrypt the configuration profile communication between the provisioning server and the device, in addition to restricting access to the administration web server.

Phone Behavior During Times of Network Congestion

Anything that degrades network performance can affect phone audio and, in some cases, can cause a call to drop. Sources of network degradation can include, but are not limited to, the following activities:

- Administrative tasks, such as an internal port scan or security scan.

- Attacks that occur on your network, such as a Denial of Service attack.

In-House Preprovisioning and Provisioning Servers

The service provider preprovisions phones, other than RC units, with a profile. The preprovision profile can comprise a limited set of parameters that resynchronizes the phone. The profile can also comprise a complete set of parameters that the remote server delivers. By default, the phone resynchronizes on power-up and at intervals that are configured in the profile. When the user connects the phone at the customer premises, the device downloads the updated profile and any firmware updates.

This process of preprovisioning, deployment, and remote provisioning can be accomplished in many ways.

Server Preparation and Software Tools

The examples in this chapter require the availability of one or more servers. These servers can be installed and run on a local PC:

- TFTP (UDP port 69)

- syslog (UDP port 514)

- HTTP (TCP port 80)

- HTTPS (TCP port 443).

To troubleshoot server configuration, it is helpful to install clients for each type of server on a separate server machine. This practice establishes proper server operation, independent of the interaction with the phones.

We also recommend that you install these software tools:

- To generate configuration profiles, install the open source gzip compression utility.

- For profile encryption and HTTPS operations, install the open source OpenSSL software package.

- To test the dynamic profile generation and one-step remote provisioning using HTTPS, we recommend a scripting language with CGI scripting support. Open source Perl language tools is an example of such a scripting language.

- To verify secure exchanges between provisioning servers and the phones, install an Ethernet packet sniffer (such as the freely downloadable Ethereal/Wireshark). Capture an Ethernet packet trace of the interaction between the phone and the provisioning server. To do so, run the packet sniffer on a PC that is connected to a switch with port mirroring enabled. For HTTPS transactions, you can use the ssldump utility.

- Remote Customization (RC) Distribution

Remote Customization (RC) Distribution

All phones contact the Cisco EDOS RC server until they are provisioned initially.

In an RC distribution model, a customer purchases a phone that has already been associated with a specific Service Provider in the Cisco EDOS RC Server. The Internet Telephony Service Provider (ITSP) sets up and maintains a provisioning server, and registers their provisioning server information with the Cisco EDOS RC Server.

When the phone is powered on with an internet connection, the customization state for the unprovisioned phone is Open. The phone first queries the local DHCP server for provisioning server information and sets the customization state of the phone. If DHCP query is successful, Customization State is set to Aborted and RC is not attempted due to DHCP providing the needed provisioning server information.

When a phone connects to a network for the first time or after a factory reset, if there are no DHCP options setup, it contacts a device activation server for zero touch provisioning. New phones will use "activate.cisco.com" instead of "webapps.cisco.com" for provisioning. Phones with firmware release prior to 11.2(1), will continue to use webapps.cisco.com. Cisco recommends that you allow both the domain names through your firewall.

If DHCP server does not provide provisioning server information, the phone queries the Cisco EDOS RC Server and provides its MAC address and model and the Customization State is set to Pending. The Cisco EDOS server responds with the associated service provider's provisioning server information including provisioning server URL and the phone's Customization State is set to Custom Pending. The phone then performs a resync URL command to retrieve the Service Provider's configuration and, if successful, the Customization State is set to Acquired.

If the Cisco EDOS RC Server does not have a service provider associated with the phone, the customization state of the phone is set to Unavailable. The phone can be manually configured or an association added for the service provider of the phone to the Cisco EDOS Server.

If a phone is provisioned via either the LCD or Web Configuration Utility, prior to the Customization State becoming Acquired, the Customization State is set to Aborted and the Cisco EDOS Server will not be queried unless the phone is factory reset.

Once the phone has been provisioned, the Cisco EDOS RC Server is not utilized unless the phone is factory reset.

In-House Device Preprovisioning

With the Cisco factory default configuration, the phone automatically tries to resync to a profile on a TFTP server. A managed DHCP server on a LAN delivers the information about the profile and TFTP server that is configured for preprovisioning to the device. The service provider connects each new phone to the LAN. The phone automatically resyncs to the local TFTP server and initializes its internal state in preparation for deployment. This preprovisioning profile typically includes the URL of a remote provisioning server. The provisioning server keeps the device updated after the device is deployed and connected to the customer network.

The preprovisioned device bar code can be scanned to record its MAC address or serial number before the phone is shipped to the customer. This information can be used to create the profile to which the phone resynchronizes.

Upon receiving the phone, the customer connects it to the broadband link. On power-up, the phone contacts the provisioning server through the URL that is configured through preprovisioning. The phone can thus resync and update the profile and firmware, as necessary.

Provisioning Server Setup

This section describes setup requirements for provisioning a phone by using various servers and different scenarios. For the purposes of this document and for testing, provisioning servers are installed and run on a local PC. Also, generally available software tools are useful for provisioning the phones.

- TFTP Provisioning

- HTTP Provisioning

TFTP Provisioning

The phones support TFTP for both provisioning resync and firmware upgrade operations. When devices are deployed remotely, HTTPS is recommended, but HTTP and TFTP can also be used. This then requires provisioning file encryption to add security, as it offers greater reliability, given NAT and router protection mechanisms. TFTP is useful for the in-house preprovisioning of a large number of unprovisioned devices.

The phone is able to obtain a TFTP server IP address directly from the DHCP server through DHCP option 66. If a Profile_Rule is configured with the filepath of that TFTP server, the device downloads its profile from the TFTP server. The download occurs when the device is connected to a LAN and powered up.

The Profile_Rule provided with the factory default configuration is &PN.cfg, where &PN represents the phone model name.

For example, for a CP-6841-3PCC, the filename is CP-6841-3PCC.cfg.

For a device with the factory default profile, upon powering up, the device resyncs to this file on the local TFTP server that DHCP option 66 specifies. The filepath is relative to the TFTP server virtual root directory.

- Remote Endpoint Control and NAT

Remote Endpoint Control and NAT

The phone is compatible with network address translation (NAT) to access the Internet through a router. For enhanced security, the router might attempt to block unauthorized incoming packets by implementing symmetric NAT, a packet-filtering strategy that severely restricts the packets that are allowed to enter the protected network from the Internet. For this reason, remote provisioning by using TFTP is not recommended.

VoIP can coexist with NAT only when some form of NAT traversal is provided. Configure Simple Traversal of UDP through NAT (STUN). This option requires that the user have:

- A dynamic external (public) IP address from your service

- A computer that is running STUN server software

- An edge device with an asymmetric NAT mechanism

HTTP Provisioning

The phone behaves like a browser that requests web pages from a remote Internet site. This provides a reliable means of reaching the provisioning server, even when a customer router implements symmetric NAT or other protection mechanisms. HTTP and HTTPS work more reliably than TFTP in remote deployments, especially when the deployed units are connected behind residential firewalls or NAT-enabled routers. HTTP and HTTPs are used interchangeably in the following request type descriptions.

Basic HTTP-based provisioning relies on the HTTP GET method to retrieve configuration profiles. Typically, a configuration file is created for each deployed phone, and these files are stored within an HTTP server directory. When the server receives the GET request, it simply returns the file that is specified in the GET request header.

Rather than a static profile, the configuration profile can be generated dynamically by querying a customer database and producing the profile on-the-fly.

When the phone requests a resynch, it can use the HTTP POST method to request the resync configuration data. The device can be configured to convey certain status and identification information to the server within the body of the HTTP POST request. The server uses this information to generate a desired response configuration profile, or to store the status information for later analysis and tracking.

As part of both GET and POST requests, the phone automatically includes basic identifying information in the User-Agent field of the request header. This information conveys the manufacturer, product name, current firmware version, and product serial number of the device.

The following example is the User-Agent request field from a CP-6841-3PCC:

User-Agent: Cisco-CP-6841-3PCC/11.0 (00562b043615)

User Agent is configurable, and the phone uses this the value if it has not be configured (still at default).

When the phone is configured to resync to a configuration profile by using HTTP, it is recommended that HTTPS be used or the profile be encrypted to protect confidential information. Encrypted profiles that the phone downloads by using HTTP avoid the danger of exposing confidential information that is contained in the configuration profile. This resync mode produces a lower computational load on the provisioning server when compared to using HTTPS.

The phone can decrypt profiles encrypted with one of these encryption methods:

- AES-256-CBC encryption

- RFC-8188 based encryption with AES-128-GCM ciphering

Note

The phones support HTTP Version 1.0, HTTP Version 1.1, and Chunk Encoding when HTTP Version 1.1 is the negotiated transport protocol.

- HTTP Status Code Handling on Resync and Upgrade

HTTP Status Code Handling on Resync and Upgrade

The phone supports HTTP response for remote provisioning (Resync). Current phone behavior is categorized in three ways:

- A—Success, where the “Resync Periodic” and “Resync Random Delay” values determine subsequent requests.

- B—Failure when File Not Found or corrupt profile. The “Resync Error Retry Delay” value determines subsequent requests.

- C—Other failure when a bad URL or IP address causes a connection error. The “Resync Error Retry Delay” value determines subsequent requests.

Phone Behavior for HTTP Responses

HTTP Status Code

Description

Phone Behavior

301 Moved Permanently

This and future requests should be directed to a new location.

Retry request immediately with new location.

302 Found

Known as Temporarily Moved.

Retry request immediately with new location.

3xx

Other 3xx responses not processed.

C

400 Bad Request

The request cannot be fulfilled due to bad syntax.

C

401 Unauthorized

Basic or digest access authentication challenge.

Immediately retry request with authentication credentials. Maximum 2 retries. Upon failure, the phone behavior is C.

403 Forbidden

Server refuses to respond.

C

404 Not Found

Requested resource not found. Subsequent requests by client are permissible.

B

407 Proxy Authentication Required

Basic or digest access authentication challenge.

Immediately retry request with authentication credentials. Maximum two retries. Upon failure, the phone behavior is C.

4xx

Other client error status codes are not processed.

C

500 Internal Server Error

Generic error message.

Phone behavior is C.

501 Not Implemented

The server does not recognize the request method, or it lacks the ability to fulfill the request.

Phone behavior is C.

502 Bad Gateway

The server is acting as a gateway or proxy and receives an invalid response from the upstream server.

Phone behavior is C.

503 Service Unavailable

The server is currently unavailable (overloaded or down for maintenance). This is a temporary state.

Phone behavior is C.

504 Gateway Timeout

The server behaves as a gateway or proxy and does not receive timely response from the upstream server.

C

5xx

Other server error

C

Chapter 4. Provisioning Methods

- Provision a Phone with BroadSoft Server

- Provisioning Examples Overview

- Basic Resync

- TFTP Resync

- Unique Profiles, Macro Expansion, and HTTP

- Resync a Device Automatically

- Set Up Your Phones for Activation Code Onboarding

- Secure HTTPS Resync

- Profile Management

- Set the Phone Privacy Header

- Renew the MIC Certificate

- Set the Upgrade Rule for the Cisco Headset

Provision a Phone with BroadSoft Server

BroadSoft Server user only.

You can register your Cisco IP multiplatform phones to a BroadWorks platform.

Procedure

Step 1

Download the CPE Kit from BroadSoft Xchange. To get the latest CPE kits, go to this URL: https://xchange.broadsoft.com.

Step 2

Upload the most recent DTAF file to the BroadWorks (system level) server.

For more information, go to this URL: (https://xchange.broadsoft.com/node/1031047). Access the BroadSoft Partner Configuration Guide and see the section “Configure BroadWorks Device Profile Type”.

Step 3

Configure Broadworks Device Profile Type.

For more information on how to configure the device profile type, go to this URL:

https://xchange.broadsoft.com/node/1031047. Access the BroadSoft Partner Configuration Guide and see the section “Broadworks Device Profile Type Configuration”.

Provisioning Examples Overview

This chapter provides example procedures for transferring configuration profiles between the phone and the provisioning server.

For information about creating configuration profiles, refer to Provisioning Formats.

Basic Resync

This section demonstrates the basic resync functionality of the phones.

- Use Syslog to Log Messages

Use Syslog to Log Messages

A phone can be configured to send logging messages to a syslog server over UDP, including messages related to provisioning. To identify this server, you can access the phone Web interface (see Access the Phone Web Interface), select Voice > System and identify the server in the Syslog Server parameter of the Optional Network Configuration section. Configure the syslog server IP address into the device and observe the messages that are generated during the remaining procedures.

To get the information, you can access the phone Web interface, select Info > Debug Info > Control Logs and click messages.

Before you begin

Procedure

Step 1

Install and activate a syslog server on the local PC.

Step 2

Program the PC IP address into the Syslog_Server parameter of the profile and submit the change:

<Syslog_Server>192.168.1.210</Syslog_Server>

Step 3

Click the System tab and enter the value of your local syslog server into the Syslog_Server parameter.

Step 4

Repeat the resync operation as described in TFTP Resync.

The device generates two syslog messages during the resync. The first message indicates that a request is in progress. The second message marks success or failure of the resync.

Step 5

Verify that your syslog server received messages similar to the following:

CP-68xx-3PCC 00:0e:08:ab:cd:ef –- Requesting resync tftp://192.168.1.200/basic.txtc.txt

Detailed messages are available by programming a Debug_Server parameter (instead of the Syslog_Server parameter) with the IP address of the syslog server, and by setting the Debug_Level to a value between 0 and 3 (3 being the most verbose):

<Debug_Server>192.168.1.210</Debug_Server>
<Debug_Level>3</Debug_Level>

The contents of these messages can be configured by using the following parameters:

- Log_Request_Msg

- Log_Success_Msg

- Log_Failure_Msg

If any of these parameters are cleared, the corresponding syslog message is not generated.

TFTP Resync

The phone supports multiple network protocols for retrieving configuration profiles. The most basic profile transfer protocol is TFTP (RFC1350). TFTP is widely used for the provisioning of network devices within private LAN networks. Although not recommended for the deployment of remote endpoints across the Internet, TFTP can be convenient for deployment within small organizations, for in-house preprovisioning, and for development and testing. See In-House Device Preprovisioning for more information on in-house preprovisioning. In the following procedure, a profile is modified after downloading a file from a TFTP server.

Procedure

Step 1

Within a LAN environment, connect a PC and a phone to a hub, switch, or small router.

Step 2

On the PC, install and activate a TFTP server.

Step 3

Use a text editor to create a configuration profile that sets the value for GPP_A to 12345678 as shown in the example.

<flat-profile>
<GPP_A> 12345678
</GPP_A>
</flat-profile>

Step 4

Save the profile with the name basic.txt in the root directory of the TFTP server.

You can verify that the TFTP server is properly configured: request the basic.txt file by using a TFTP client other than the phone. Preferably, use a TFTP client that is running on a separate host from the provisioning server.

Step 5

Open the PC web browser to the admin/advanced configuration page. For example, if the IP address of the phone is 192.168.1.100:

http://192.168.1.100/admin/advanced

Step 6

Select the Voice > Provisioning tab, and inspect the values of the general purpose parameters GPP_A through GPP_P. These should be empty.

Step 7

Resync the test phone to the basic.txt configuration profile by opening the resync URL in a web browser window.

If the IP address of the TFTP server is 192.168.1.200, the command should be similar to the following example:

http://192.168.1.100/admin/resync?tftp://192.168.1.200/basic.txt

When the phone receives this command, the device at address 192.168.1.100 requests the file basic.txt from the TFTP server at IP address 192.168.1.200. The phone then parses the downloaded file and updates the GPP_A parameter with the value 12345678.

Step 8

Verify that the parameter was correctly updated: Refresh the configuration page on the PC web browser and select the Voice > Provisioning tab.

The GPP_A parameter should now contain the value 12345678.

- Log Messages to the Syslog Server

Log Messages to the Syslog Server

If a syslog server is configured on the phone through the use of the parameters, the resync and upgrade operations send messages to the syslog server. A message can be generated at the start of a remote file request (configuration profile or firmware load), and at the conclusion of the operation (indicating either success or failure).

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 System Log Parameters.

Before you begin

- A syslog server is installed and configured.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Click Voice > System.

Step 2

In the Optional Network Configuration section, enter the server IP in Syslog Server and optionally specify a Syslog Identifier as defined in System Log Parameters.

Step 3

Optionally define the content of the syslog messages using Log Request Msg, Log Success Msg, and Log Failure Msg as defined in System Log Parameters.

The fields defining syslog message content are located in the Configuration Profile section on the Voice > Provisioning tab. If you don't specify the message content, the default settings in the fields are used. If any of the fields are cleared, the corresponding message is not generated.

Step 4

Click Submit All Changes to apply the configuration.

Step 5

Verify the validity of the configuration.

1. Perform a TFTP resync. See TFTP Resync.

The device generates two syslog messages during the resync. The first message indicates that a request is in progress. The second message marks the success or failure of the resync.

2. Verify that your syslog server received messages similar to the following:

CP-78xx-3PCC 00:0e:08:ab:cd:ef –- Requesting resync tftp://192.168.1.200/basic.txt

CP-88xx-3PCC 00:0e:08:ab:cd:ef –- Successful resync tftp://192.168.1.200/basic.txt

- System Log Parameters

System Log Parameters

The following table defines the function and usage of the syslog parameters in the Optional Network Configuration section under the Voice > System tab in the phone web page. 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.

Syslog Parameters

Parameter Name

Description and Default Value

Syslog Server

Specify the server for logging the phone system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server.

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

<Syslog_Server ua="na">10.74.30.84</Syslog_Server>

- On the phone web page, specify the Syslog server.

Syslog Identifier

Select the device identifier to include in syslog messages that are uploaded to the syslog server. The device identifier appears after the timestamp in each message. The options for the identifiers are:

- None: No device identifier.

- $MA: The MAC address of the phone, expressed as continuous lower case letters and digits. Example: c4b9cd811e29

- $MAU: The MAC address of the phone, expressed as continuous upper case letters and digits. Example: C4B9CD811E29

- $MAC: The MAC address of the phone in the standard colon-separated format. Example: c4:b9:cd:81:1e:29

- $SN: The product serial number of the phone.

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

<Syslog_Identifier ua="na">$MAC</Syslog_Identifier>

- On the phone web page, select an identifier from the list.

Default: None

Log Request Msg

The message that is sent to the syslog server at the start of a resync attempt. If no value is specified, the syslog message is not generated.

The default value is $PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH

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

<Log_Request_Msg ua="na">$PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH</Log_Request_Msg>

Log Success Msg

The syslog message that is issued upon successful completion of a resync attempt. If no value is specified, the syslog message is not generated.

In the phone configuration file with XML(cfg.xml), enter a string in this format: <Log_Success_Msg ua="na">$PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH</Log_Success_Msg>

Log Failure Msg

The syslog message that is issued after a failed resync attempt. If no value is specified, the syslog message is not generated.

The default value is $PN $MAC -- Resync failed: $ERR

In the phone configuration file with XML(cfg.xml), enter a string in this format: <Log_Failure_Msg ua="na">$PN $MAC -- Resync failed: $ERR</Log_Failure_Msg>

Unique Profiles, Macro Expansion, and HTTP

In a deployment where each phone must be configured with distinct values for some parameters, such as User_ID or Display_Name, the service provider can create a unique profile for each deployed device and host those profiles on a provisioning server. Each phone, in turn, must be configured to resync to its own profile according to a predetermined profile naming convention.

The profile URL syntax can include identifying information that is specific to each phone, such as MAC address or serial number, by using the macro expansion of built-in variables. Macro expansion eliminates the need to specify these values in multiple locations within each profile.

A profile rule undergoes macro expansion before the rule is applied to the phone. The macro expansion controls a number of values, for example:

- $MA expands to the unit 12-digit MAC address (using lower case hex digits). For example, 000e08abcdef.

- $SN expands to the unit serial number. For example, 88012BA01234.

Other values can be macro expanded in this way, including all the general purpose parameters, GPP_A through GPP_P. An example of this process can be seen in TFTP Resync. Macro expansion is not limited to the URL file name, but can also be applied to any portion of the profile rule parameter. These parameters are referenced as $A through $P. For a complete list of variables that are available for macro expansion, see Macro Expansion Variables.

In this exercise, a profile specific to a phone is provisioned on a TFTP server.

- Provision a Specific IP Phone Profile on a TFTP Server

- HTTP GET Resync

- Provisioning Through Cisco XML

- URL Resolution with Macro Expansion

Provision a Specific IP Phone Profile on a TFTP Server

Procedure

Step 1

Obtain the MAC address of the phone from its product label. (The MAC address is the number, using numbers and lower–case hex digits, such as 000e08aabbcc.

Step 2

Copy the basic.txt configuration file (described in TFTP Resync) to a new file named CP-xxxx-3PCC macaddress.cfg (replacing xxxx with the model number and macaddress with the MAC address of the phone).

Step 3

Move the new file in the virtual root directory of the TFTP server.

Step 4

Access the phone administration web page. See Access the Phone Web Interface.

Step 5

Select Voice  > Provisioning.

Step 6

Enter tftp://192.168.1.200/CP-6841-3PCC$MA.cfg in the Profile Rule field.

<Profile_Rule>
tftp://192.168.1.200/CP-6841-3PCC$MA.cfg
</Profile_Rule>

Step 7

Click Submit All Changes. This causes an immediate reboot and resync.

When the next resync occurs, the phone retrieves the new file by expanding the $MA macro expression into its MAC address.

HTTP GET Resync

HTTP provides a more reliable resync mechanism than TFTP because HTTP establishes a TCP connection and TFTP uses the less reliable UDP. In addition, HTTP servers offer improved filtering and logging features compared to TFTP servers.

On the client side, the phone does not require any special configuration setting on the server to be able to resync by using HTTP. The Profile_Rule parameter syntax for using HTTP with the GET method is similar to the syntax that is used for TFTP. If a standard web browser can retrieve a profile from your HTTP server, the phone should be able to do so as well.

- Resync with HTTP GET

Resync with HTTP GET

Procedure

Step 1

Install an HTTP server on the local PC or other accessible host.

The open source Apache server can be downloaded from the internet.

Step 2

Copy the basic.txt configuration profile (described in TFTP Resync) onto the virtual root directory of the installed server.

Step 3

To verify proper server installation and file access to basic.txt, access the profile with a web browser.

Step 4

Modify the Profile_Rule of the test phone to point to the HTTP server in place of the TFTP server, so as to download its profile periodically.

For example, assuming the HTTP server is at 192.168.1.300, enter the following value:

<Profile_Rule>
http://192.168.1.200/basic.txt
</Profile_Rule>

Step 5

Click Submit All Changes. This causes an immediate reboot and resync.

Step 6

Observe the syslog messages that the phone sends. The periodic resyncs should now be obtaining the profile from the HTTP server.

Step 7

In the HTTP server logs, observe how information that identifies the test phone appears in the log of user agents.

This information should include the manufacturer, product name, current firmware version, and serial number.

Provisioning Through Cisco XML

For each of the phones, designated as xxxx here, you can provision through Cisco XML functions.

You can send an XML object to the phone by a SIP Notify packet or an HTTP Post to the CGI interface of the phone: http://IPAddressPhone/CGI/Execute.

The CP-xxxx-3PCC extends the Cisco XML feature to support provisioning via an XML object:

<CP-xxxx-3PCCExecute>
<ExecuteItem URL=Resync:[profile-rule]/>
</CP-xxxx-3PCCExecute>

After the phone receives the XML object, it downloads the provisioning file from [profile-rule]. This rule uses macros to simplify the development of the XML services application.

URL Resolution with Macro Expansion

Subdirectories with multiple profiles on the server provide a convenient method for managing a large number of deployed devices. The profile URL can contain:

- A provisioning server name or an explicit IP address. If the profile identifies the provisioning server by name, the phone performs a DNS lookup to resolve the name.

- A nonstandard server port that is specified in the URL by using the standard syntax :port following the server name.

- The subdirectory of the server virtual root directory where the profile is stored, specified by using standard URL notation and managed by macro expansion.

For example, the following Profile_Rule requests the profile file ($PN.cfg), in the server subdirectory /cisco/config, from the TFTP server that is running on host prov.telco.com listening for a connection on port 6900:

<Profile_Rule>
tftp://prov.telco.com:6900/cisco/config/$PN.cfg
</Profile_Rule>

A profile for each phone can be identified in a general purpose parameter, with its value referred within a common profile rule by using macro expansion.

For example, assume GPP_B is defined as Dj6Lmp23Q.

The Profile_Rule has the value:

tftp://prov.telco.com/cisco/$B/$MA.cfg

When the device resyncs and the macros are expanded, the phone with a MAC address of 000e08012345 requests the profile with the name that contains the device MAC address at the following URL:

tftp://prov.telco.com/cisco/Dj6Lmp23Q/000e08012345.cfg

Resync a Device Automatically

A device can resync periodically to the provisioning server to ensure that any profile changes made on the server are propagated to the endpoint device (as opposed to sending an explicit resync request to the endpoint).

To cause the phone to periodically resync to a server, a configuration profile URL is defined by using the Profile_Rule parameter, and a resync period is defined by using the Resync_Periodic parameter.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice  > Provisioning.

Step 2

Define the Profile_Rule parameter. This example assumes a TFTP server IP address of 192.168.1.200.

Step 3

In the Resync Periodic field, enter a small value for testing, such as 30 seconds.

Step 4

Click Submit all Changes.

With the new parameter settings, the phone resyncs twice a minute to the configuration file that the URL specifies.

Step 5

Observe the resulting messages in the syslog trace (as described in the Use Syslog to Log Messages section).

Step 6

Ensure that the Resync On Reset field is set to Yes.

<Resync_On_Reset>Yes</Resync_On_Reset>

Step 7

Power cycle the phone to force it to resync to the provisioning server.

If the resync operation fails for any reason, such as if the server is not responding, the unit waits (for the number of seconds configured in Resync Error Retry Delay) before it attempts to resync again. If Resync Error Retry Delay is zero, the phone does not try to resync after a failed resync attempt.

Step 8

(Optional) Set the value of Resync Error Retry Delay field to a small number, such as 30.

<Resync_Error_Retry_Delay>30</Resync_Error_Retry_Delay>

Step 9

Disable the TFTP server, and observe the results in the syslog output.

- Profile Resync Parameters

Profile Resync Parameters

The following table defines the function and usage of the profile resync parameters in the Configuration Profile section under the Voice > Provisioning tab in the phone web page. 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

Description

Provision Enable

Allows or denies configuration profile resync actions.

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

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

- On the phone web page, set this field to Yes to allow resync actions, or No to block resync actions.

Default: Yes

Resync On Reset

Specifies whether the phone resynchronizes configurations with the provisioning server after power-up and after each upgrade attempt.

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

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

- On the phone web page, set this field to Yes to allow resync on power-up or reset, or No to block resync on power-up or reset.

Default: Yes

Resync Random Delay

Prevents an overload of the provisioning server when a large number of devices power-on simultaneously and attempt initial configuration. This delay is effective only on the initial configuration attempt, following a device power-on or reset.

The parameter is the maximum time interval that the device waits before making contact with the provisioning server. The actual delay is a pseudo-random number between 0 and this value.

This parameter is in units of 20 seconds.

The valid value ranges between 0 and 65535.

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

<Resync_Random_Delay ua="na">2</Resync_Random_Delay>

- On the phone web page, specify the number of the units (20 seconds) for the phone to delay resync after power-up or reset.

The default value is 2 (40 seconds).

Resync At (HHmm)

The time (HHmm) that the phone resynchronizes with the provisioning server.

The value for this field must be a four-digit number ranging from 0000 to 2400 to indicate the time in HHmm format. For example, 0959 indicates 09:59.

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

<Resync_At__HHmm_ ua="na">0959</Resync_At__HHmm_>

- On the phone web page, specify the time in HHMM format for the phone to start resync.

The default value is empty. If the value is invalid, the parameter is ignored. If this parameter is set with a valid value, the Resync Periodic parameter is ignored.

Resync At Random Delay

Prevents an overload of the provisioning server when a large number of devices power on simultaneously.

To avoid flooding resync requests to the server from multiple phones, the phone resynchronizes in the range between the hours and minutes, and the hours and minutes plus the random delay (hhmm, hhmm+random_delay). For example, if the random delay = (Resync At Random Delay + 30)/60minutes, the input value in seconds is converted to minutes, rounding up to the next minute to calculate the final random_delay interval.

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

<Resync_At_Random_Delay ua="na">600</Resync_At_Random_Delay>

- On the phone web page, specify the time period in seconds.

The valid value ranges between 600 and 65535.

If the value is less than 600, the random delay internal is between 0 and 600.

The default value is 600 seconds (10minutes).

Resync Periodic

The time interval between periodic resynchronization with the provisioning server. The associated resync timer is active only after the first successful sync with the server.

The valid formats are as follows:

- An integer

Example: An input of 3000 indicates that the next resync occurs in 3000 seconds.

- Multiple integers

Example: An input of 600,1200,300 indicates that the first resync occurs in 600 seconds, the second resync occurs in 1200 seconds after the first one, and the third resync occurs in 300 seconds after the second one.

- A time range

Example, an input of 2400+30 indicates that the next resync occurs in between 2400 and 2430 seconds after a successful resync.

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

<Resync_Periodic ua="na">3600</Resync_Periodic>

- On the phone web page, specify the time period in seconds.

Set this parameter to zero to disable periodic resynchronization.

The default value is 3600 seconds.

Resync Error Retry Delay

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

The valid formats are as follows:

- An integer

Example: An input of 300 indicates that the next retry for resync occurs in 300 seconds.

- Multiple integers

Example: An input of 600,1200,300 indicates that the first retry occurs in 600 seconds after the failure, the second retry occurs in 1200 seconds after the failure of the first retry, and the third retry occurs in 300 seconds after the failure of the second retry.

- A time range

Example, an input of 2400+30 indicates that the next retry occurs in between 2400 and 2430 seconds after a resync failure.

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

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

<Resync_Error_Retry_Delay ua="na">60,120,240,480,960,1920,3840,7680,15360,30720,61440,86400</Resync_Error_Retry_Delay>

- On the phone web page, specify the time period in seconds.

Default: 60,120,240,480,960,1920,3840,7680,15360,30720,61440,86400

Forced Resync Delay

Maximum delay (in seconds) the phone waits before performing a resynchronization.

The device does not resync while one of its phone lines is active. Because a resync can take several seconds, it is desirable to wait until the device has been idle for an extended period before resynchronizing. This allows a user to make calls in succession without interruption.

The device has a timer that begins counting down when all of its lines become idle. This parameter is the initial value of the counter. Resync events are delayed until this counter decrements to zero.

The valid value ranges between 0 and 65535.

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

<Forced_Resync_Delay ua="na">14400</Forced_Resync_Delay>

- On the phone web page, specify the time period in seconds.

The default value is 14,400seconds.

Resync From SIP

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

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

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

- On the phone web page, select Yes to enable this feature, or No to disalbe it.

Default: Yes

Resync After Upgrade Attempt

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

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

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

- On the phone web page, select Yes to trigger resync after a firmware upgrade, or No to not resync.

Default: Yes

Resync Trigger 1

Resync Trigger 2

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

The parameters can be programmed with a conditional expression that undergoes macro expansion. For the valid macro expansions, see Macro Expansion Variables.

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

<Resync_Trigger_1 ua="na">$UPGTMR gt 300 and $PRVTMR ge 600</Resync_Trigger_1>

<Resync_Trigger_2 ua="na"/>

- On the phone web page, specify the triggers.

Default: Blank

User Configurable Resync

Allows a user to resync the phone from the phone screen menu. When set to Yes, a user can resync the phone configuration by entering the profile rule from the phone. When set to No, the Profile rule parameter isn't displayed on the phone screen menu. The Profile rule parameter is loacated under Applications > Device administration.

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

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

- On the phone web page, select Yes to show the Profile rule parameter on the phone menu, or select No to hide this parameter.

Default: Yes

Resync Fails On FNF

A resync is typically considered unsuccessful if a requested profile is not received from the server. This parameter override this behavior. When set to No, the device accepts a file-not-found response from the server as a successful resync.

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

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

- On the phone web page, select Yes to take a file-not-found response as an unsuccessful resync, or select No to take a file-not-found response as a successful resync.

Default: Yes

Profile Authentication Type

Specifies the credentials to use for profile account authentication. The available options are:

- Disabled: Disables the profile account feature. When this feature is disabled, the Profile account setup menu doesn't display on the phone screen.

- Basic HTTP Authentication: The HTTP login credentials are used to authenticate the profile account.

- XSI Authentication: XSI login credentials or XSI SIP credentials are used to authenticate the profile account. The authentication credentials depend on the XSI Authentication Type for the phone:

- When the XSI Authentication Type for the phone is set to Login Credentials, the XSI login credentials are used.

- When the XSI Authentication Type for the phone is set to SIP Credentials, the XSI SIP credentials are used.

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

<Profile_Authentication_Type ua="na">Basic Http Authentication</Profile_Authentication_Type>

- On the phone web page, select an option from the list for the phone to authenticate profile resync.

Default: Basic HTTP Authentication

Profile Rule

Profile Rule B

Profile Rule C

Profile Rule D

Each profile rule informs the phone of a source from which to obtain a profile (configuration file). During every resync operation, the phone applies all the profiles in sequence.

If you are applying AES-256-CBC encryption to the configuration files, specify the encryption key with the --key keyword as follows:

[--key <encryption key>]

You can enclose the encryption key in double-quotes (") optionally.

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

<Profile_Rule ua="na">/$PSN.xml</Profile_Rule>

<Profile_Rule_B ua="na"/>

<Profile_Rule_C ua="na"/>

<Profile_Rule_D ua="na"/>

- On the phone web page, specify the profile rule.

Default: /$PSN.xml

DHCP Option To Use

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

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

DHCPv6 Option To Use

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

Default: 17,160,159

Set Up Your Phones for Activation Code Onboarding

If your network is configured for Activation Code Onboarding, you can set up new phones to register automatically in a secure way. You generate and provide each user with a unique 16-digit activation code. The user enters the activation code, and the phone automatically registers. This feature keeps your network secure because the phone can't register until the user enters a valid activation code.

Activation codes can be used only once, and have an expiry date. If a user enters an expired code, the phone displays Invalid activation code on the screen. If this happens, provide the user with a new code.

This feature is available in firmware release 11-2-3MSR1, BroadWorks Application Server Release 22.0 (patch AP.as.22.0.1123.ap368163 and its dependencies). However, you can change phones with older firmware to use this feature. To do this, use the following procedure.

Before you begin

Ensure that you allow the activation.webex.com service through your firewall to support onboarding via activation code.

If you want to set up a proxy server for the onboarding, ensure that the proxy server is configured correctly. See Set Up a Proxy Server.

Access the phone web page. Access the Phone Web Interface

Procedure

Step 1

Reset the phone to the factory settings.

Step 2

Select Voice > Provisioning > Configuration Profile.

Step 3

Enter the profile rule in the Profile Rule field as described in the Activation Code Provisioning Parameters table.

Step 4

(Optional) In the Firmware Upgrade section, enter the upgrade rule in the Upgrade Rule field as described in the Activation Code Provisioning Parameters table.

Step 5

Submit All Changes.

- Activation Code Provisioning Parameters

Activation Code Provisioning Parameters

The following table defines the function and usage of the activation code parameters in the Configuration Profile section under the Voice > Provisioning tab in the phone web page. 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

Description

Profile Rule

Profile Rule B

Profile Rule C

Profile Rule D

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

Perform one of the following:

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

<Profile_Rule ua="na">gds://</Profile_Rule>

- In the phone web interface, enter a string in this format:

gds://

Default: /$PSN.xml

Upgrade Rule

Specifies the firmware upgrade script that defines upgrade conditions and associated firmware URLs. It uses the same syntax as Profile Rule.

Perform one of the following:

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

<Upgrade_Rule ua="na">http://<server ip address>/
sip88xx.11-2-3MSR1-1.loads</Upgrade_Rule>

- In the phone web interface, enter the upgrade rule:

protocol://server[:port]/profile_pathname

For example:

tftp://192.168.1.5/image/sip88xx.11-2-3MSR1-1.loads

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the URL is used as the server name. If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, or 443 for HTTPS).

Default: Blank

Secure HTTPS Resync

These mechanisms are available on the phone for resyncing by using a secure communication process:

- Basic HTTPS Resync

- HTTPS with Client Certificate Authentication

- HTTPS Client Filtering and Dynamic Content

- Basic HTTPS Resync

- HTTPS with Client Certificate Authentication

- Configure a HTTPS Server for Client Filtering and Dynamic Content

- HTTPS Certificates

Basic HTTPS Resync

HTTPS adds SSL to HTTP for remote provisioning so that the:

- The phone can authenticate the provisioning server.

- Provisioning server can authenticate the phone.

- Confidentiality of information exchanged between the phone and the provisioning server is ensured.

SSL generates and exchanges secret (symmetric) keys for each connection between the phone and the server, using public/private key pairs that are pre-installed in the phone and the provisioning server.

On the client side, the phone does not require any special configuration setting on the server to be able to resync using HTTPS. The Profile_Rule parameter syntax for using HTTPS with the GET method is similar to the syntax that is used for HTTP or TFTP. If a standard web browser can retrieve a profile from a your HTTPS server, the phone should be able to do so as well.

In addition to installing a HTTPS server, a SSL server certificate that Cisco signs must be installed on the provisioning server. The devices cannot resync to a server that is using HTTPS unless the server supplies a Cisco-signed server certificate. Instructions for creating signed SSL Certificates for Voice products can be found at https://supportforums.cisco.com/docs/DOC-9852.

- Authenticate with Basic HTTPS Resync

Authenticate with Basic HTTPS Resync

Procedure

Step 1

Install an HTTPS server on a host whose IP address is known to the network DNS server through normal hostname translation.

The open source Apache server can be configured to operate as an HTTPS server when installed with the open source mod_ssl package.

Step 2

Generate a server Certificate Signing Request for the server. For this step, you might need to install the open source OpenSSL package or equivalent software. If using OpenSSL, the command to generate the basic CSR file is as follows:

openssl req –new –out provserver.csr

This command generates a public/private key pair, which is saved in the privkey.pem file.

Step 3

Submit the CSR file (provserver.csr) to Cisco for signing.

A signed server certificate is returned (provserver.cert) along with a Sipura CA Client Root Certificate, spacroot.cert.

See https://supportforums.cisco.com/docs/DOC-9852 for more information

Step 4

Store the signed server certificate, the private key pair file, and the client root certificate in the appropriate locations on the server.

In the case of an Apache installation on Linux, these locations are typically as follows:

# Server Certificate:
SSLCertificateFile /etc/httpd/conf/provserver.cert
# Server Private Key:
SSLCertificateKeyFile /etc/httpd/conf/pivkey.pem
# Certificate Authority:
SSLCACertificateFile /etc/httpd/conf/spacroot.cert

Step 5

Restart the server.

Step 6

Copy the basic.txt configuration file (described in TFTP Resync) onto the virtual root directory of the HTTPS server.

Step 7

Verify proper server operation by downloading basic.txt from the HTTPS server by using a standard browser from the local PC.

Step 8

Inspect the server certificate that the server supplies.

The browser probably does not recognize the certificate as valid unless the browser has been pre-configured to accept Cisco as a root CA. However, the phones expect the certificate to be signed this way.

Modify the Profile_Rule of the test device to contain a reference to the HTTPS server, for example:

<Profile_Rule>
https://my.server.com/basic.txt
</Profile_Rule>

This example assumes the name of the HTTPS server is my.server.com.

Step 9

Click Submit All Changes.

Step 10

Observe the syslog trace that the phone sends.

The syslog message should indicate that the resync obtained the profile from the HTTPS server.

Step 11

(Optional) Use an Ethernet protocol analyzer on the phone subnet to verify that the packets are encrypted.

In this exercise, client certificate verification was not enabled. The connection between the phone and server is encrypted. However, the transfer is not secure because any client can connect to the server and request the file, given knowledge of the file name and directory location. For secure resync, the server must also authenticate the client, as demonstrated in the exercise described in HTTPS with Client Certificate Authentication.

HTTPS with Client Certificate Authentication

In the factory default configuration, the server does not request an SSL client certificate from a client. Transfer of the profile is not secure because any client can connect to the server and request the profile. You can edit the configuration to enable client authentication; the server requires a client certificate to authenticate the phone before it accepts a connection request.

Because of this requirement, the resync operation cannot be independently tested by using a browser that lacks the proper credentials. The SSL key exchange within the HTTPS connection between the test phone and the server can be observed with the ssldump utility. The utility trace shows the interaction between client and server.

- Authenticate HTTPS with Client Certificate

Authenticate HTTPS with Client Certificate

Procedure

Step 1

Enable client certificate authentication on the HTTPS server.

Step 2

In Apache (v.2), set the following in the server configuration file:

SSLVerifyClient  require

Also, ensure that the spacroot.cert has been stored as shown in the Basic HTTPS Resync exercise.

Step 3

Restart the HTTPS server and observe the syslog trace from the phone.

Each resync to the server now performs symmetric authentication, so that both the server certificate and the client certificate are verified before the profile is transferred.

Step 4

Use ssldump to capture a resync connection between the phone and the HTTPS server.

If client certificate verification is properly enabled on the server, the ssldump trace shows the symmetric exchange of certificates (first server-to-client, then client-to-server) before the encrypted packets that contain the profile.

With client authentication enabled, only a phone with a MAC address that matches a valid client certificate can request the profile from the provisioning server. The server rejects a request from an ordinary browser or other unauthorized device.

Configure a HTTPS Server for Client Filtering and Dynamic Content

If the HTTPS server is configured to require a client certificate, the information in the certificate identifies the resyncing phone and supplies it with the correct configuration information.

The HTTPS server makes the certificate information available to CGI scripts (or compiled CGI programs) that are invoked as part of the resync request. For the purpose of illustration, this exercise uses the open source Perl scripting language, and assumes that Apache (v.2) is used as the HTTPS server.

Procedure

Step 1

Install Perl on the host that is running the HTTPS server.

Step 2

Generate the following Perl reflector script:

#!/usr/bin/perl -wT
use strict;
print “Content-Type: text/plain\n\n”;
print “<flat-profile><GPP_D>”;

print “OU=$ENV{‘SSL_CLIENT_I_DN_OU’},\n”;
print “L=$ENV{‘SSL_CLIENT_I_DN_L’},\n”;
print “S=$ENV{‘SSL_CLIENT_I_DN_S’}\n”;
print “</GPP_D></flat-profile>”;

Step 3

Save this file with the file name reflect.pl, with executable permission (chmod 755 on Linux), in the CGI scripts directory of the HTTPS server.

Step 4

Verify accessibility of CGI scripts on the server (that is, /cgi-bin/…).

Step 5

Modify the Profile_Rule on the test device to resync to the reflector script, as in the following example:

https://prov.server.com/cgi-bin/reflect.pl?

Step 6

Click Submit All Changes.

Step 7

Observe the syslog trace to ensure a successful resync.

Step 8

Access the phone administration web page. See Access the Phone Web Interface.

Step 9

Select Voice  > Provisioning.

Step 10

Verify that the GPP_D parameter contains the information that the script captured.

This information contains the product name, MAC address, and serial number if the test device carries a unique certificate from the manufacturer. The information contains generic strings if the unit was manufactured before firmware release 2.0.

A similar script can determine information about the resyncing device and then provide the device with appropriate configuration parameter values.

HTTPS Certificates

The phone provides a reliable and secure provisioning strategy that is based on HTTPS requests from the device to the provisioning server. Both a server certificate and a client certificate are used to authenticate the phone to the server and the server to the phone.

In addition to Cisco issued certifications, the phone also accepts server certificates from a set of commonly used SSL certificate providers.

To use HTTPS with the phone, you must generate a Certificate Signing Request (CSR) and submit it to Cisco. The phone generates a certificate for installation on the provisioning server. The phone accepts the certificate when it seeks to establish an HTTPS connection with the provisioning server.

- HTTPS Methodology

- SSL Server Certificate

- Obtain a Server Certificate

- Client Certificate

- Certificate Structure

- Configure a Custom Certificate Authority

HTTPS Methodology

HTTPS encrypts the communication between a client and a server, thus protecting the message contents from other network devices. The encryption method for the body of the communication between a client and a server is based on symmetric key cryptography. With symmetric key cryptography, a client and a server share a single secret key over a secure channel that is protected by Public/Private key encryption.

Messages encrypted by the secret key can only be decrypted by using the same key. HTTPS supports a wide range of symmetric encryption algorithms. The phone implements up to 256-bit symmetric encryption, using the American Encryption Standard (AES), in addition to 128-bit RC4.

HTTPS also provides for the authentication of a server and a client engaged in a secure transaction. This feature ensures that a provisioning server and an individual client cannot be spoofed by other devices on the network. This capability is essential in the context of remote endpoint provisioning.

Server and client authentication is performed by using public/private key encryption with a certificate that contains the public key. Text that is encrypted with a public key can be decrypted only by its corresponding private key (and vice versa). The phone supports the Rivest-Shamir-Adleman (RSA) algorithm for public/private key cryptography.

SSL Server Certificate

Each secure provisioning server is issued a secure sockets layer (SSL) server certificate that Cisco signs directly. The firmware that runs on the phone recognizes only a Cisco certificate as valid. When a client connects to a server by using HTTPS, it rejects any server certificate that is not signed by Cisco.

This mechanism protects the service provider from unauthorized access to the phone, or any attempt to spoof the provisioning server. Without such protection, an attacker might be able to reprovision the phone, to gain configuration information, or to use a different VoIP service. Without the private key that corresponds to a valid server certificate, the attacker is unable to establish communication with a phone.

Obtain a Server Certificate

Procedure

Step 1

Contact a Cisco support person who will work with you on the certificate process. If you are not working with a specific support person, email your request to ciscosb-certadmin@cisco.com.

Step 2

Generate a private key that will be used in a CSR (Certificate Signing Request). This key is private and you do not need to provide this key to Cisco support. Use open source “openssl” to generate the key. For example:

openssl genrsa -out <file.key> 1024

Step 3

Generate a CSR that contains fields that identify your organization and location. For example:

openssl req -new -key <file.key> -out <file.csr>

You must have the following information:

- Subject field—Enter the Common Name (CN) that must be an FQDN (Fully Qualified Domain Name) syntax. During SSL authentication handshake, the phone verifies that the certificate it receives is from the machine that presented it.

- Server hostname—For example, provserv.domain.com.

- Email address—Enter an email address so that customer support can contact you if needed. This email address is visible in the CSR.

Step 4

Email the CSR (in zip file format) to the Cisco support person or to
ciscosb-certadmin@cisco.com. The certificate is signed by Cisco. Cisco sends the certificate to you to install on your system.

Client Certificate

In addition to a direct attack on a phone, an attacker might attempt to contact a provisioning server through a standard web browser or another HTTPS client to obtain the configuration profile from the provisioning server. To prevent this kind of attack, each phone also carries a unique client certificate, signed by Cisco, that includes identifying information about each individual endpoint. A certificate authority root certificate that is capable of authenticating the device client certificate is given to each service provider. This authentication path allows the provisioning server to reject unauthorized requests for configuration profiles.

Certificate Structure

The combination of a server certificate and a client certificate ensures secure communication between a remote phone and its provisioning server. The figure below illustrates the relationship and placement of certificates, public/private key pairs, and signing root authorities, among the Cisco client, the provisioning server, and the certification authority.

The upper half of the diagram shows the Provisioning Server Root Authority that is used to sign the individual provisioning server certificate. The corresponding root certificate is compiled into the firmware, which allows the phone to authenticate authorized provisioning servers.

Certificate Authority Flow

Configure a Custom Certificate Authority

Digital certificates can be used to authenticate network devices and users on the network. They can be used to negotiate IPSec sessions between network nodes.

A third party uses a Certificate Authority certificate to validate and authenticate two or more nodes that are attempting to communicate. Each node has a public and private key. The public key encrypts data. The private key decrypts data. Because the nodes have obtained their certificates from the same source, they are assured of their respective identities.

The device can use digital certificates provided by a third-party Certificate Authority (CA) to authenticate IPSec connections.

The phones support a set of preloaded Root Certificate Authority embedded in the firmware:

- Cisco Small Business CA Certificate

- CyberTrust CA Certificate

- Verisign CA certificate

- Sipura Root CA Certificate

- Linksys Root CA Certificate

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info  > Status.

Step 2

Scroll to Custom CA Status and see the following fields:

- Custom CA Provisioning Status—Indicates the provisioning status.

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

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

- Custom CA Info—Displays information about the custom CA.

- Installed—Displays the “CN Value,” where “CN Value” is the value of the CN parameter for the Subject field in the first certificate.

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

Profile Management

This section demonstrates the formation of configuration profiles in preparation for downloading. To explain the functionality, TFTP from a local PC is used as the resync method, although HTTP or HTTPS can be used as well.

- Compress an Open Profile with Gzip

- Encrypt a Profile with OpenSSL

- Create Partitioned Profiles

Compress an Open Profile with Gzip

A configuration profile in XML format can become quite large if the profile specifies all parameters individually. To reduce the load on the provisioning server, the phone supports compression of the XML file, by using the deflate compression format that the gzip utility (RFC 1951) supports.

Note

Compression must precede encryption for the phone to recognize a compressed and encrypted XML profile.

For integration into customized back-end provisioning server solutions, the open source zlib compression library can be used in place of the standalone gzip utility to perform the profile compression. However, the phone expects the file to contain a valid gzip header.

Procedure

Step 1

Install gzip on the local PC.

Step 2

Compress the basic.txt configuration profile (described in TFTP Resync) by invoking gzip from the command line:

gzip basic.txt

This generates the deflated file basic.txt.gz.

Step 3

Save the basic.txt.gz file in the TFTP server virtual root directory.

Step 4

Modify the Profile_Rule on the test device to resync to the deflated file in place of the original XML file, as shown in the following example:

tftp://192.168.1.200/basic.txt.gz

Step 5

Click Submit All Changes.

Step 6

Observe the syslog trace from the phone.

Upon resync, the phone downloads the new file and uses it to update its parameters.

Encrypt a Profile with OpenSSL

A compressed or uncompressed profile can be encrypted (however, a file must be compressed before it is encrypted). Encryption is useful when the confidentiality of the profile information is of particular concern, such as when TFTP or HTTP is used for communication between the phone and the provisioning server.

The phone supports symmetric key encryption by using the 256-bit AES algorithm. This encryption can be performed by using the open source OpenSSL package.

Procedure

Step 1

Install OpenSSL on a local PC. This might require that the OpenSSL application be recompiled to enable AES.

Step 2

Using the basic.txt configuration file (described in TFTP Resync), generate an encrypted file with the following command:

>openssl enc –aes-256-cbc –k MyOwnSecret –in basic.txt –out basic.cfg

The compressed basic.txt.gz file that was created in Compress an Open Profile with Gzip also can be used, because the XML profile can be both compressed and encrypted.

Step 3

Store the encrypted basic.cfg file in the TFTP server virtual root directory.

Step 4

Modify the Profile_Rule on the test device to resync to the encrypted file in place of the original XML file. The encryption key is made known to the phone with the following URL option:

[--key MyOwnSecret ] tftp://192.168.1.200/basic.cfg

Step 5

Click Submit All Changes.

Step 6

Observe the syslog trace from the phone.

Upon resync, the phone downloads the new file and uses it to update its parameters.

Create Partitioned Profiles

A phone downloads multiple separate profiles during each resync. This practice allows management of different kinds of profile information on separate servers and maintenance of common configuration parameter values that are separate from account specific values.

Procedure

Step 1

Create a new XML profile, basic2.txt, that specifies a value for a parameter that makes it distinct from the earlier exercises. For instance, to the basic.txt profile, add the following:

<GPP_B>ABCD</GPP_B>

Step 2

Store the basic2.txt profile in the virtual root directory of the TFTP server.

Step 3

Leave the first profile rule from the earlier exercises in the folder, but configure the second profile rule (Profile_Rule_B) to point to the new file:

<Profile_Rule_B>tftp://192.168.1.200/basic2.txt
</Profile_Rule_B>

Step 4

Click Submit All Changes.

The phone now resyncs to both the first and second profiles, in that order, whenever a resync operation is due.

Step 5

Observe the syslog trace to confirm the expected behavior.

Set the Phone Privacy Header

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 using an XML tag in the config.xml file.

The privacy header options are:

- Disabled (default)

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

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

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

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

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

Procedure

Step 1

Edit the phone config.xml file in a text or XML editor.

Step 2

Insert the <Privacy_Header_N_ ua="na">Value</Privacy_Header_N_> tag, where N is the line extension number (1–10), and use one of the following values.

- Default value: Disabled

- none

- header

- session

- user

- id

Step 3

(Optional) Provision any addition line extensions using the same tag with the required line extension number.

Step 4

Save the changes to the config.xml file.

Renew the MIC Certificate

You can renew the Manufacture Installed Certificate (MIC) by a specified or default Secure Unique Device Identifier (SUDI) service. If the MIC certificate expires, the features that use SSL/TLS don't work.

Before you begin

- Ensure that you allow the sudirenewal.cisco.com service (port 80) through your firewall to support the MIC certificate renewal.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

Under the MIC Cert Settings section, set the parameters as defined in Parameters for MIC Certificate Renewal by SUDI Service.

Step 3

Click Submit All Changes.

After the certificate renewal is completed successfully, the phone reboots.

Step 4

(Optional) Check the latest status of the MIC certificate renewal under the MIC Cert Refresh Status section from Info > Download Status.

Note

If you restore the phone to factory settings, the phone still uses the renewed certificate.

- Parameters for MIC Certificate Renewal by SUDI Service

Parameters for MIC Certificate Renewal by SUDI Service

The following table defines the function and usage of each parameter in the MIC Cert Settings section of the Voice > Provisioning tab.

Parameters for MIC Certificate Renewal by SUDI Service

Parameter Name

Description and Default Value

MIC Cert Refresh Enable

Controls whether to enable the Manufacture Installed Certificate (MIC) renewal by the default or specified Secure Unique Device Identifier (SUDI) service.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to enable or disable the MIC certificate renewal.

Valid values: Yes and No

Default: No

MIC Cert Refresh Rule

Enter the HTTP URL of the SUDI service that provides the renewed MIC certificate, for example,

http://sudirenewal.cisco.com/

Note

Don't change the URL. Only the default URL is supported for the MIC certificate renewal.

Perform one of the following:

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

<MIC_Cert_Refresh_Rule ua="na">http://sudirenewal.cisco.com/</MIC_Cert_Refresh_Rule>

- In the phone web interface, enter the HTTP URL to use.

Allowed values: A valid URL not exceeding 1024 characters

Default: http://sudirenewal.cisco.com/

Set the Upgrade Rule for the Cisco Headset

You can upgrade the firmware of a Cisco headset by connecting it to a Cisco IP Phone Multiplatform Phone. Before the user performs the upgrade, you must set the upgrade rule on the phone administration web page. When the headset is connected to the phone, the phone automatically detects the new version of the headset firmware, then prompts the user to perform the upgrade.

The supported connections for the upgrade are:

- Cisco Headset 520 Series: USB cable

- Cisco Headset 560 Series: USB cable and Y-Cable (RJ-9 and AUX connector)

- Cisco Headset 700 Series: USB cable

The headset settings are not erased by a phone reset. The upgrade rule supports HTTP, HTTPS, and TFTP protocols.

The release of the Cisco Headset provides the headset XML file that can be used for the firmware upgrade. If the software version in the file is later than the firmware on the headset, the user is prompted to upgrade the headset on the phone screen. The user can choose to upgrade the headset immediately or postpone it to a later time.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

Under the Cisco Headset Firmware Upgrade section, find the parameter Cisco Headset Upgrade Rule.

Step 3

Specify the TFTP, HTTP, or HTTPS protocol, an IP address of the headset upgrade device, and the name of the headset XML file. Enter the values as a single string in the parameter.

Caution

Do not change the headset XML file contents.

For example, tftp://10.74.51.81/prov/headset/1-6-0-162/ciscoheadsetfirmware.xml

You can also configure this parameter in the configuration file (cfg.xml):

<Cisco_Headset_Upgrade_Rule ua="na">tftp://10.74.51.81/prov/headset/1-6-0-162/ciscoheadsetfirmware.xml</Cisco_Headset_Upgrade_Rule>

Step 4

Click Submit All Changes.

If a new version of the headset firmware is detected, the phone displays an upgrade prompt.

Chapter 5. Provisioning Parameters

- Provisioning Parameters Overview

- Configuration Profile Parameters

- Firmware Upgrade Parameters

- General Purpose Parameters

- Macro Expansion Variables

- Internal Error Codes

Provisioning Parameters Overview

This chapter describes the provisioning parameters that can be used in configuration profile scripts

Configuration Profile Parameters

The following table defines the function and usage of each parameter in the Configuration Profile Parameters section under the Provisioning tab.

Parameter Name

Description and Default Value

Provision Enable

Controls all resync actions independently of firmware upgrade actions. Set to Yes to enable remote provisioning.

The default value is Yes.

Resync On Reset

Triggers a resync after every reboot except for reboots caused by parameter updates and firmware upgrades.

The default value is Yes.

Resync Random Delay

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

The value for this field must be an integer ranging between 0 and 65535.

The default value is 2.

Resync At (HHmm)

The time (HHmm) that the device resynchronizes with the provisioning server.

The value for this field must be a four-digit number ranging from 0000 to 2400 to indicate the time in HHmm format. For example, 0959 indicates 09:59.

The default value is empty. If the value is invalid, the parameter is ignored. If this parameter is set with a valid value, the Resync Periodic parameter is ignored.

Resync At Random Delay

Prevents an overload of the provisioning server when a large number of devices power-on simultaneously.

To avoid flooding resync requests to the server from multiple phones, the phone resynchronizes in the range between the hours and minutes, and the hours and minutes plus the random delay (hhmm, hhmm+random_delay). For example, if the random delay = (Resync At Random Delay + 30)/60 minutes, the input value in seconds is converted to minutes, rounding up to the next minute to calculate the final random_delay interval.

The valid value ranges between 600 and 65535.

If the value is less than 600, the random delay internal is between 0 and 600.

The default value is 600 seconds (10 minutes).

Resync Periodic

The time interval between periodic resynchronizes with the provisioning server. The associated resync timer is active only after the first successful sync with the server.

The valid formats are as follows:

- An integer

Example: An input of 3000 indicates that the next resync occurs in 3000 seconds.

- Multiple integers

Example: An input of 600,1200,300 indicates that the first resync occurs in 600 seconds, the second resync occurs in 1200 seconds after the first one, and the third resync occurs in 300 seconds after the second one.

- A time range

Example, an input of 2400+30 indicates that the next resync occurs in between 2400 and 2430 seconds after a successful resync.

Set this parameter to zero to disable periodic resynchronization.

The default value is 3600 seconds.

Resync Error Retry Delay

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

The valid formats are as follows:

- An integer

Example: An input of 300 indicates that the next retry for resync occurs in 300 seconds.

- Multiple integers

Example: An input of 600,1200,300 indicates that the first retry occurs in 600 seconds after the failure, the second retry occurs in 1200 seconds after the failure of the first retry, and the third retry occurs in 300 seconds after the failure of the second retry.

- A time range

Example, an input of 2400+30 indicates that the next retry occurs in between 2400 and 2430 seconds after a resync failure.

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

Forced Resync Delay

Maximum delay (in seconds) the phone waits before performing a resynchronization.

The device does not resync while one of its phone lines is active. Because a resync can take several seconds, it is desirable to wait until the device has been idle for an extended period before resynchronizing. This allows a user to make calls in succession without interruption.

The device has a timer that begins counting down when all of its lines become idle. This parameter is the initial value of the counter. Resync events are delayed until this counter decrements to zero.

The valid value ranges between 0 and 65535.

The default value is 14,400 seconds.

Resync From SIP

Enables a resync to be triggered via a SIP NOTIFY message.

The default value is Yes.

Resync After Upgrade Attempt

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

The default value is Yes.

Resync Trigger 1, Resync Trigger 2

Configurable resync trigger conditions. A resync is triggered when the logic equation in these parameters evaluates to TRUE.

The default value is (empty).

Resync Fails On FNF

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

The default value is Yes.

Profile Rule

Profile Rule B

Profile Rule C

Profile Rule D

Each profile rule informs the phone of a source from which to obtain a profile (configuration file). During every resync operation, the phone applies all the profiles in sequence.

Default: /$PSN.xml

If you are applying AES-256-CBC encryption to the configuration files, specify the encryption key with the --key keyword as follows:

[--key <encryption key>]

You can enclose the encryption key in double-quotes (") optionally.

DHCP Option To Use

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

The default value is 66,160,159,150,60,43,125.

Log Request Msg

This parameter contains the message that is sent to the syslog server at the start of a resync attempt.

The default value is $PN $MAC –Requesting % $SCHEME://$SERVIP:$PORT$PATH.

Log Success Msg

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

The default value is $PN $MAC –Successful Resync % $SCHEME://$SERVIP:$PORT$PATH -- $ERR.

Log Failure Msg

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

The default value is $PN $MAC -- Resync failed: $ERR.

User Configurable Resync

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

The default value is Yes.

Firmware Upgrade Parameters

The following table defines the function and usage of each parameter in the Firmware Upgrade section of the Provisioning tab.

Parameter Name

Description and Default Value

Upgrade Enable

Enables firmware upgrade operations independently of resync actions.

The default value is Yes.

Upgrade Error Retry Delay

The upgrade retry interval (in seconds) applied in case of upgrade failure. The device has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero.

The default value is 3600 seconds.

Upgrade Rule

A firmware upgrade script that defines upgrade conditions and associated firmware URLs. It uses the same syntax as Profile Rule.

Use the following format to enter the upgrade rule:

<tftp|http|https>://<ip address><:port>/<path>/<load name>

For example:

tftp://192.168.1.5/firmware/sip68xx.11-0-1MPP-321.loads

If no protocol is specified, TFTP is assumed. If no server-name is specified, the host that requests the URL is used as the server name. If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, or 443 for HTTPS).

The default value is blank.

Log Upgrade Request Msg

Syslog message issued at the start of a firmware upgrade attempt.

Default: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH

Log Upgrade Success Msg

Syslog message issued after a firmware upgrade attempt completes successfully.

The default value is $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR

Log Upgrade Failure Msg

Syslog message issued after a failed firmware upgrade attempt.

The default value is $PN $MAC -- Upgrade failed: $ERR

Peer Firmware Sharing

Enables or disables the Peer Firmware Sharing feature. Select Yes or No to enable or to disable the feature.

Default: Yes

Peer Firmware Sharing Log Server

Indicates the IP address and the port to which the UDP message is sent.

For example: 10.98.76.123:514 where, 10.98.76.123 is the IP address and 514 is the port number.

General Purpose Parameters

The following table defines the function and usage of each parameter in the General Purpose Parameters section of the Provisioning tab.

Parameter Name

Description and Default Value

GPP A - GPP P

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

- Encryption keys.

- URLs.

- Multistage provisioning status information.

- Post request templates.

- Parameter name alias maps.

- Partial string values, eventually combined into complete parameter values.

The default value is blank.

Macro Expansion Variables

Certain macro variables are recognized within the following provisioning parameters:

- Profile_Rule

- Profile_Rule_*

- Resync_Trigger_*

- Upgrade_Rule

- Log_*

- GPP_* (under specific conditions)

Within these parameters, syntax types, such as $NAME or $(NAME), are recognized and expanded.

Macro variable substrings can be specified with the notation $(NAME:p) and $(NAME:p:q), where p and q are non-negative integers (available in revision 2.0.11 and above). The resulting macro expansion is the substring starting at character offset p, with length q (or else till end-of-string if q is not specified). For example, if GPP_A contains ABCDEF, then $(A:2) expands to CDEF, and $(A:2:3) expands to CDE.

An unrecognized name is not translated, and the $NAME or $(NAME) form remains unchanged in the parameter value after expansion.

Parameter Name

Description and Default Value

$

The form $$ expands to a single $ character.

A through P

Replaced by the contents of the 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.

Note

$SA through $SD are recognized as arguments to the optional resync URL qualifier, --key.

MA

MAC address using lower case hex digits, for example, 000e08aabbcc.

MAU

MAC address using upper case hex digits, for example 000E08AABBCC.

MAC

MAC address using lower case hex digits, and colons to separate hex digit pairs. For example 00:0e:08:aa:bb:cc.

PN

Product Name. For example, CP-6841-3PCC.

PSN

Product Series Number. For example, 6841-3PCC.

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,

- For Firmware Release 11.3(1)SR1 and previous: sip68xx.11-0-1MPP-312

- For Firmware Release 11.3(2) and later: sip68xx.11-3-2MPP0001-609

HWVER

Hardware version string. For example, 2.0.1

PRVST

Provisioning State (a numeric string):

-1 = explicit resync request

0 = power-up resync

1 = periodic resync

2 = resync failed, retry attempt

UPGST

Upgrade State (a numeric string):

1 = first upgrade attempt

2 = upgrade failed, retry attempt

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, one of TFTP, HTTP, or HTTPS, as obtained after parsing resync or upgrade URL.

SERV

Request target server host name, as obtained after parsing resync or upgrade URL.

SERVIP

Request target server IP address, as obtained after parsing resync or upgrade URL, possibly following DNS lookup.

PORT

Request target UDP/TCP port, as obtained after parsing resync or upgrade URL.

PATH

Request target file path, as obtained after parsing resync or upgrade URL.

ERR

Result message of resync or upgrade attempt. Only useful in generating result syslog messages. The value is preserved in the UPGERR variable in the case of upgrade attempts.

UIDn

The contents of the Line n UserID configuration parameter.

EMS

Extension Mobility Status

MUID

Extension Mobility User ID

MPWD

Extension Mobility Password

Internal Error Codes

The phone defines a number of internal error codes (X00–X99) to facilitate configuration in providing finer control over the behavior of the unit under certain error conditions.

Parameter Name

Description and Default Value

X00

Transport layer (or ICMP) error when sending a SIP request.

X20

SIP request times out while waiting for a response.

X40

General SIP protocol error (for example, unacceptable codec in SDP in 200 and ACK messages, or times out while waiting for ACK).

X60

Dialed number invalid according to given dial plan.

Chapter 6. Provisioning Formats

- Configuration Profiles

- Configuration Profile Formats

- Open Profile (XML) Compression and Encryption

- Application of a Profile to the Phone

- Provisioning Parameter Types

- Data Types

- Profile Updates and Firmware Upgrades

Configuration Profiles

The phone accepts configuration in an XML format.

The examples in this document use configuration profiles with an XML format (XML) syntax.

For detailed information about your phone, refer to the administration guide for your particular device. Each guide describes the parameters that can be configured through the administration web server.

Configuration Profile Formats

The configuration profile defines the parameter values for the phone.

The configuration profile XML format uses standard XML authoring tools to compile the parameters and values.

Note

Only the UTF-8 charset is supported. If you modify the profile in an editor, do not change the encoding format; otherwise, the phone cannot recognize the file.

Each phone has a different feature set and therefore, a different set of parameters.

XML Format (XML) Profile

The open format profile is a text file with XML-like syntax in a hierarchy of elements, with element attributes and values. This format lets you use standard tools to create the configuration file. A configuration file in this format can be sent from the provisioning server to the phone during a resync operation. The file can be sent without compilation as a binary object.

The phone can accept configuration formats that standard tools generate. This feature eases the development of back-end provisioning server software that generates configuration profiles from existing databases.

To protect confidential information in the configuration profile, the provisioning server delivers this type of file to the phone over a channel secured by TLS. Optionally, the file can be compressed by using the gzip deflate algorithm (RFC1951).

The file can be encrypted with one of these encryption methods:

- AES-256-CBC encryption

- RFC-8188 based HTTP content encryption with AES-128-GCM ciphering

Example: Open Profile Format

<flat-profile>
<Resync_On_Reset> Yes </Resync_On_Reset>
<Resync_Periodic> 7200 </Resync_Periodic>
<Profile_Rule> tftp://prov.telco.com:6900/cisco/config/CP_xxxx_MPP.cfg</Profile_Rule>
</flat-profile>

The <flat-profile> element tag encloses all parameter elements that the phone recognizes.

- Configuration File Components

Configuration File Components

A configuration file can include these components:

- Element tags

- Attributes

- Parameters

- Formatting features

- XML comments

- Element Tag Properties

- Parameter Properties

- String Formats

Element Tag Properties

- The XML provisioning format and the Web UI allow the configuration of the same settings. The XML tag name and the field names in the Web UI are similar but vary due to XML element name restrictions. For example, underscores (_) instead of " ".

- The phone recognizes elements with proper parameter names that are encapsulated in the special <flat-profile> element.

- Element names are enclosed in angle brackets.

- Most element names are similar to the field names in the administration web pages for the device, with the following modifications:

- Element names may not include spaces or special characters. To derive the element name from the administration web field name, substitute an underscore for every space or the special characters [, ], (, ), or /.

Example: The <Resync_On_Reset> element represents the Resync On Reset field.

- Each element name must be unique. In the administration web pages, the same fields can appear on multiple web pages, such as the Line, User, and Extension pages. Append [n] to the element name to indicate the number that is shown in the page tab.

Example: The <Dial_Plan_1_> element represents the Dial Plan for Line 1.

- Each opening element tag must have a matching closing element tag. For example:

<flat-profile>
<Resync_On_Reset> Yes
</Resync_On_Reset>
<Resync_Periodic> 7200
</Resync_Periodic>
<Profile_Rule>tftp://prov.telco.com: 6900/cisco/config/CP_xxxx_MPP.cfg
</Profile_Rule>
</flat-profile>

- Element tags are case-sensitive.

- Empty element tags are allowed and will be interpreted as configuring the value to be empty. Enter the opening element tag without a corresponding element tag, and insert a space and a forward slash before the closing angle bracket (>). In this example, Profile Rule B is empty:

<Profile_Rule_B />

- An empty element tag can be used to prevent the overwriting of any user-supplied values during a resync operation. In the following example, the user speed dial settings are unchanged:

<flat-profile>
<Speed_Dial_2_Name ua="rw"/>
<Speed_Dial_2_Number ua="rw"/>
<Speed_Dial_3_Name ua="rw"/>
<Speed_Dial_3_Number ua="rw"/>
<Speed_Dial_4_Name ua="rw"/>
<Speed_Dial_4_Number ua="rw"/>
<Speed_Dial_5_Name ua="rw"/>
<Speed_Dial_5_Number ua="rw"/>
<Speed_Dial_6_Name ua="rw"/>
<Speed_Dial_6_Number ua="rw"/>
<Speed_Dial_7_Name ua="rw"/>
<Speed_Dial_7_Number ua="rw"/>
<Speed_Dial_8_Name ua="rw"/>
<Speed_Dial_8_Number ua="rw"/>
<Speed_Dial_9_Name ua="rw"/>
<Speed_Dial_9_Number ua="rw"/>
</flat-profile>

- Use an empty value to set the corresponding parameter to an empty string. Enter an opening and closing element without any value between them. In the following example, the GPP_A parameter is set to an empty string.

<flat-profile>
<GPP_A>
</GPP_A>
</flat-profile>

- Unrecognized element names are ignored.

Parameter Properties

These properties apply to the parameters:

- Any parameters that are not specified by a profile are left unchanged in the phone.

- Unrecognized parameters are ignored.

- If the Open format profile contains multiple occurrences of the same parameter tag, the last such occurrence overrides any earlier ones. To avoid inadvertent override of configuration values for a parameter, we recommend that each profile specify at most one instance of a parameter.

- The last profile processed takes precedence. If multiple profiles specify the same configuration parameter, the value of the latter profile takes precedence.

String Formats

These properties apply to the formatting of the strings:

- Comments are allowed through standard XML syntax.

<!-- My comment is typed here -->

- Leading and trailing white space is allowed for readability but is removed from the parameter value.

- New lines within a value are converted to spaces.

- An XML header of the form <? ?> is allowed, but the phone ignores it.

- To enter special characters, use basic XML character escapes, as shown in the following table.

Special Character

XML Escape Sequence

& (ampersand)

&amp;

< (less than)

&lt;

> (greater than)

&gt;

’ (apostrophe)

&apos;

” (double quote)

&quot;

In the following example, character escapes are entered to represent the greater than and less than symbols that are required in a dial plan rule. This example defines an information hotline dial plan that sets the <Dial_Plan_1_> parameter (Admin Login > advanced > Voice > Ext (n)) equal to (S0 <:18005551212>).

<flat-profile>
<Dial_Plan_1_>
(S0 &lt;:18005551212&gt;)
</Dial_Plan_1_>
</flat-profile>

- Numeric character escapes, using decimal and hexadecimal values (s.a. &#40; and &#x2e;), are translated.

- The phone firmware only supports ASCII characters.

Open Profile (XML) Compression and Encryption

The Open configuration profile can be compressed to reduce the network load on the provisioning server. The profile can also be encrypted to protect confidential information. Compression is not required, but it must precede encryption.

- Open Profile Compression

- Open Profile Encryption

- Optional Resync Arguments

Open Profile Compression

The supported compression method is the gzip deflate algorithm (RFC1951). The gzip utility and the compression library that implements the same algorithm (zlib) are available from Internet sites.

To identify compression, the phone expects the compressed file to contain a gzip compatible header. Invocation of the gzip utility on the original Open profile generates the header. The phone inspects the downloaded file header to determine the file format.

For example, if profile.xml is a valid profile, the file profile.xml.gz is also accepted. Either of the following commands can generate this profile type:

- >gzip profile.xml

Replaces original file with compressed file.

- >cat profile.xml | gzip > profile.xml.gz

Leaves original file in place, produces new compressed file.

A tutorial on compression is provided in the Compress an Open Profile with Gzip section.

Open Profile Encryption

Symmetric key encryption can be used to encrypt an open configuration profile, whether the file is compressed or not. Compression, if applied, must be applied before encryption.

The provisioning server uses HTTPS to handle initial provisioning of the phone after deployment. Pre-encrypting configuration profiles offline allows the use of HTTP for resyncing profiles subsequently. This reduces the load on the HTTPS server in large-scale deployments.

The phone supports two methods of encryption for configuration files:

- AES-256-CBC encryption

- RFC 8188-based HTTP content encryption with AES-128-GCM ciphering

The key or Input Keying Material (IKM) must be preprovisioned into the unit at an earlier time. Bootstrap of the secret key can be accomplished securely by using HTTPS.

The configuration file name does not require a specific format, but a file name that ends with the .cfg extension normally indicates a configuration profile.

- AES-256-CBC Encryption

- RFC 8188-Based HTTP Content Encryption

AES-256-CBC Encryption

The phone supports AES-256-CBC encryption for configuration files.

The OpenSSL encryption tool, available for download from various Internet sites, can perform the encryption. Support for 256-bit AES encryption may require recompilation of the tool to enable the AES code. The firmware has been tested against version openssl-1.1.1d.

Encrypt a Profile with OpenSSL provides a tutorial on encryption.

For an encrypted file, the profile expects the file to have the same format as generated by the following command:

# example encryption key = SecretPhrase1234

openssl enc –e –aes-256-cbc –k SecretPhrase1234 –in profile.xml –out profile.cfg

# analogous invocation for a compressed xml file

openssl enc –e –aes-256-cbc –k SecretPhrase1234 –in profile.xml.gz –out profile.cfg

A lowercase -k precedes the secret key, which can be any plain text phrase, and which is used to generate a random 64-bit salt. With the secret specified by the -k argument, the encryption tool derives a random 128-bit initial vector and the actual 256-bit encryption key.

When this form of encryption is used on a configuration profile, the phone must be informed of the secret key value to decrypt the file. This value is specified as a qualifier in the profile URL. The syntax is as follows, using an explicit URL:

[--key “SecretPhrase1234”] http://prov.telco.com/path/profile.cfg

This value is programmed by using one of the Profile_Rule parameters.

- Macro Expansion

- Conditional Expressions

- URL Syntax

Macro Expansion

Several provisioning parameters undergo macro expansion internally prior to being evaluated. This preevaluation step provides greater flexibility in controlling the phone resync and upgrade activities.

These parameter groups undergo macro expansion before evaluation:

- Resync_Trigger_*

- Profile_Rule*

- Log_xxx_Msg

- Upgrade_Rule

Under certain conditions, some general-purpose parameters (GPP_*) also undergo macro expansion, as explicitly indicated in Optional Resync Arguments.

During macro expansion, the contents of the named variables replace expressions of the form $NAME and $(NAME). These variables include general-purpose parameters, several product identifiers, certain event timers, and provisioning state values. For a complete list, see Macro Expansion Variables.

In the following example, the expression $(MAU) is used to insert the MAC address 000E08012345.

The administrator enters: $(MAU)config.cfg

The resulting macro expansion for a device with MAC address 000E08012345 is: 000E08012345config.cfg

If a macro name is not recognized, it remains unexpanded. For example, the name STRANGE is not recognized as a valid macro name, while MAU is recognized as a valid macro name.

The administrator enters: $STRANGE$MAU.cfg

The resulting macro expansion for a device with MAC address 000E08012345 is: $STRANGE000E08012345.cfg

Macro expansion is not applied recursively. For example, $$MAU” expands into $MAU” (the $$ is expanded), and does not result in the MAC address.

The contents of the special purpose parameters, GPP_SA through GPP_SD, are mapped to the macro expressions $SA through $SD. These parameters are only macro expanded as the argument of the --key , --uid, and --pwd options in a resync URL.

Conditional Expressions

Conditional expressions can trigger resync events and select from alternate URLs for resync and upgrade operations.

Conditional expressions consist of a list of comparisons, separated by the and operator. All comparisons must be satisfied for the condition to be true.

Each comparison can relate to one of the following three types of literals:

- Integer values

- Software or hardware version numbers

- Doubled-quoted strings

Version Numbers

The software version for Cisco IP Phones with Multiplatform Firmware uses this format (where BN is the Build Number):

- For Firmware Release 11.3(1)SR1 and previous: sipyyyy.11-0-1MPP-376

where yyyy indicates the phone model or phone series; 11 is the major version; 0 is the minor version; 1MPP is the micro version; and 376 is the build number.

- For Firmware Release 11.3(2) and later: sipyyyy.11-3-2MPP0001-609

where yyyy indicates the phone model or phone series; 11 is the major version; 3 is the minor version; 2MPP0001 is the micro version; and 609 is the build number.

The comparing string must use the same format. Otherwise, a format parsing error results.

When comparing the software version, the major version, minor version, and micro version are compared in sequence, and the leftmost digits take precedence over the latter ones. When version numbers are identical, the build number is compared.

Examples of Valid Version Number

- For Firmware Release 11.3(1)SR1 and previous:

sip68xx.11-0-1MPP-312

- For Firmware Release 11.3(2) and later:

sip68xx.11-3-2MPP0001-609

Comparison

- For Firmware Release 11.3(1)SR1 and previous:

sipyyyy.11-3-1MPP-110 ˃ sipyyyy.11-2-3MPP-256

- For Firmware Release 11.3(2) and later:

sipyyyy.11-3-2MPP0002-256 ˃ sipyyyy.11-3-2MPP0001-609

Quoted strings can be compared for equality or inequality. Integers and version numbers can also be compared arithmetically. The comparison operators can be expressed as symbols or as acronyms. Acronyms are convenient for expressing the condition in an Open format profile.

Operator

Alternate Syntax

Description

Applicable to Integer and Version Operands

Applicable to Quoted String Operands

=

eq

equal to

Yes

Yes

!=

ne

not equal to

Yes

Yes

<

lt

less than

Yes

No

<=

le

less than or equal to

Yes

No

>

gt

greater than

Yes

No

>=

ge

greater than or equal to

Yes

No

AND

and

Yes

Yes

It is important to enclose macro variables in double quotes where a string literal is expected. Don't do so where a number or version number is expected.

When used in the context of the Profile_Rule* and Upgrade_Rule parameters, conditional expressions must be enclosed within the syntax “(expr)?” as in this upgrade rule example. Remember to replace BN with the build number of your firmware load to upgrade to.

- For Firmware Release 11.3(1)SR1 and previous

($SWVER ne sip68xx.11-0-0MPP-256)? http://ps.tell.com/sw/sip68xx.11-0-0MPP-BN.loads

- For Firmware Release 11.3(2) and later

($SWVER ne sip68xx.11-3-2MPP0001-609)? http://ps.tell.com/sw/sip68xx.11-3-2MPP0001-BN.loads

Do not use the preceding syntax with parentheses to configure the Resync_Trigger_* parameters.

URL Syntax

Use the Standard URL syntax to specify how to retrieve configuration files and firmware loads in Profile_Rule* and Upgrade_Rule parameters, respectively. The syntax is as follows:

[ scheme:// ] [ server [:port]] filepath

Where scheme is one of these values:

- tftp

- http

- https

If scheme is omitted, tftp is assumed. The server can be a DNS-recognized hostname or a numeric IP address. The port is the destination UDP or TCP port number. The filepath must begin with the root directory (/); it must be an absolute path.

If server is missing, the tftp server specified through DHCP (option 66) is used.

Note

For upgrade rules, the server must be specified.

If port is missing, the standard port for the specified scheme is used. Tftp uses UDP port 69, http uses TCP port 80, https uses TCP port 443.

A filepath must be present. It need not necessarily refer to a static file, but can indicate dynamic content obtained through CGI.

Macro expansion applies within URLs. The following are examples of valid URLs:

/$MA.cfg
/cisco/cfg.xml
192.168.1.130/profiles/init.cfg
tftp://prov.call.com/cpe/cisco$MA.cfg
http://neptune.speak.net:8080/prov/$D/$E.cfg
https://secure.me.com/profile?Linksys

/$MA.cfg
/cisco/cfg.xml
192.168.1.130/profiles/init.cfg
tftp://prov.call.com/cpe/cisco$MA.cfg

When using DHCP option 66, the empty syntax is not supported by upgrade rules. It is only applicable for Profile Rule*.

RFC 8188-Based HTTP Content Encryption

The phone supports RFC 8188-based HTTP content encryption with AES-128-GCM ciphering for configuration files. With this encryption method, any entity can read the HTTP message headers. However, only the entities that know the Input Keying Material (IKM) can read the payload. When the phone is provisioned with the IKM, the phone and the provisioning server can exchange configuration files securely, while allowing third-party network elements to use the message headers for analytic and monitoring purposes.

The XML configuration parameter IKM_HTTP_Encrypt_Content holds the IKM on the phone. For security reasons, this parameter is not accessible on the phone administration web page. It is also not visible in the phone's configuration file, which you can access from the phone's IP address or from the phone's configuration reports sent to the provisioning server.

If you want to use the RFC 8188-based encryption, ensure the following:

- Provision the phone with the IKM by specifying the IKM with the XML parameter IKM_HTTP_Encrypt_Content in the configuration file that is sent from the provisioning server to the phone.

- If this encryption is applied to the configuration files sent from the provisioning server to the phone, ensure that the Content-Encoding HTTP header in the configuration file has "aes128gcm".

In the absence of this header, the AES-256-CBC method is given precedence. The phone applies AES-256-CBC decryption if a AES-256-CBC key is present in a profile rule, regardless of IKM.

- If you want the phone to apply this encryption to the configuration reports that it sends to the provisioning server, ensure that there is no AES-256-CBC key specified in the report rule.

Optional Resync Arguments

Optional arguments, key, uid, and pwd, can precede the URLs entered in Profile_Rule* parameters, collectively enclosed by square brackets.

- key

- uid and pwd

key

The --key option tells the phone that the configuration file that it receives from the provisioning server is encrypted with AES-256-CBC encryption, unless the Content-Encoding header in the file indicates "aes128gcm" encryption. The key itself is specified as a string following the term --key. The key can be enclosed in double-quotes (") optionally. The phone uses the key to decrypt the configuration file.

Usage Examples

[--key VerySecretValue]
[--key “my secret phrase”]
[--key a37d2fb9055c1d04883a0745eb0917a4]

The bracketed optional arguments are macro expanded. Special purpose parameters, GPP_SA through GPP_SD, are macro expanded into macro variables, $SA through $SD, only when they are used as key option arguments. See these examples:

[--key $SC]
[--key “$SD”]

In Open format profiles, the argument to --key must be the same as the argument to the -k option that is given to openssl.

uid and pwd

The uid and pwd options can be used to specify the userID and password that will be sent in response to HTTP Basic and Digest authentication challenges when the specified URL is requested. The bracketed optional arguments are macro expanded. Special purpose parameters, GPP_SA through GPP_SD, are macro expanded into macro variables, $SA through $SD, only when they are used as key option arguments. See these examples:

GPP_SA = MyUserID
GPP_SB = MySecretPassword

[--uid $SA --pwd $SB] https://provisioning_server_url/path_to_your_config/your_config.xml

would then expand to:

[--uid MyUserID --pwdMySecretPassword] https://provisioning_server_url/path_to_your_config/your_config.xml

Application of a Profile to the Phone

After you create an XML configuration script, it must be passed to the phone for application. To apply the configuration, you can either download the configuration file to the phone from a TFTP, HTTP, or HTTPS server using a web browser or by using cURL command line utility.

- Download the Configuration File to the Phone from a TFTP Server

- Download the Configuration File to the Phone with cURL

Download the Configuration File to the Phone from a TFTP Server

Complete these steps to download the configuration file to a TFTP server application on your PC.

Procedure

Step 1

Connect your PC to the phone LAN.

Step 2

Run a TFTP server application on the PC and ensure that the configuration file is available in the TFTP root directory.

Step 3

In a web browser, enter the phone LAN IP address, the IP address of the computer, the filename, and the login credentials. Use this format:

http://<WAN_IP_Address>/admin/resync?tftp://<PC_IP_Address>/<file_name>&xuser=admin&xpassword=<password>

Example:

http://192.168.15.1/admin/resync?tftp://192.168.15.100/my_config.xml&xuser=admin&xpassword=admin

Download the Configuration File to the Phone with cURL

Complete these steps to download the configuration to the phone by using cURL. This command-line tool is used to transfer data with a URL syntax. To download cURL, visit:

https://curl.haxx.se/download.html

Note

We recommend that you do not use cURL to post the configuration to the phone because the username and password might get captured while using cURL.

Procedure

Step 1

Connect your PC to the LAN port of the phone.

Step 2

Download the configuration file to the phone by entering the following cURL command:

curl –d @my_config.xml
“http://192.168.15.1/admin/config.xml&xuser=admin&xpassword=admin”

Provisioning Parameter Types

This section describes the provisioning parameters broadly organized according to function:

These provisioning parameter types exist:

- General Purpose

- Enables

- Triggers

- Configurable Schedules

- Profile Rules

- Upgrade Rule

- General Purpose Parameters

- Enable Parameters

- Triggers

- Configurable Schedules

- Profile Rules

- Upgrade Rule

General Purpose Parameters

The general-purpose parameters GPP_* (Admin Login > advanced > Voice > Provisioning) are used as free string registers when configuring the phone to interact with a particular provisioning server solution. The GPP_* parameters are empty by default. They can be configured to contain diverse values, including the following:

- Encryption keys

- URLs

- Multistage provisioning status information.

- Post request templates

- Parameter name alias maps

- Partial string values, eventually combined into complete parameter values.

The GPP_* parameters are available for macro expansion within other provisioning parameters. For this purpose, single-letter uppercase macro names (A through P) suffice to identify the contents of GPP_A through GPP_P. Also, the two-letter uppercase macro names SA through SD identify GPP_SA through GPP_SD as a special case when used as arguments of the following URL options:

key, uid, and pwd

These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prefixing the variable name with a ‘$’ character, such as $GPP_A.

- Use General Purpose Parameters

Use General Purpose Parameters

For example, if GPP_A contains the string ABC, and GPP_B contains 123, the expression $A$B macro expands into ABC123.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

Scroll to the General Purpose Parameters section.

Step 3

Enter valid values in the fields, GPP A through GPP P.

Step 4

Click Submit All Changes.

Enable Parameters

The Provision_Enable and Upgrade_Enable parameters control all profile resync and firmware upgrade operations. These parameters control resyncs and upgrades independently of each other. These parameters also control resync and upgrade URL commands that are issued through the administration web server. Both of these parameters are set to Yes by default.

The Resync_From_SIP parameter controls requests for resync operations. A SIP NOTIFY event is sent from the service provider proxy server to the phone. If enabled, the proxy can request a resync. To do so, the proxy sends a SIP NOTIFY message that contains the Event: resync header to the device.

The device challenges the request with a 401 response (authorization refused for used credentials). The device expects an authenticated subsequent request before it honors the resync request from the proxy. The Event: reboot_now and Event: restart_now headers perform cold and warm restarts, respectively, which are also challenged.

The two remaining enables are Resync_On_Reset and Resync_After_Upgrade_Attempt. These parameters determine whether the device performs a resync operation after power-up software reboots and after each upgrade attempt.

When Resync_On_Reset is enabled, the device introduces a random delay that follows the boot-up sequence before the reset is performed. The delay is a random time up to the value that the Resync_Random_Delay (in seconds) specifies. In a pool of phones that power up simultaneously, this delay spreads out the start times of the resync requests from each unit. This feature can be useful in a large residential deployment, in the case of a regional power failure.

Triggers

The phone allows you to resync at specific intervals or at a specific time.

- Resync at Specific Intervals

- Resync at a Specific Time

Resync at Specific Intervals

The phone is designed to resync with the provisioning server periodically. The resync interval is configured in Resync_Periodic (seconds). If this value is left empty, the device does not resync periodically.

The resync typically takes place when the voice lines are idle. If a voice line is active when a resync is due, the phone delays the resync procedure until the line becomes idle again. A resync can cause configuration parameter values to change.

A resync operation can fail because the phone is unable to retrieve a profile from the server, the downloaded file is corrupt, or an internal error occurred. The device tries to resync again after a time that is specified in Resync_Error_Retry_Delay (seconds). If Resync_Error_Retry_Delay is set to 0, the device does not try to resync again after a failed resync attempt.

If an upgrade fails, a retry is performed after Upgrade_Error_Retry_Delay seconds.

Two configurable parameters are available to conditionally trigger a resync: Resync_Trigger_1 and Resync_Trigger_2. Each parameter can be programmed with a conditional expression that undergoes macro expansion. When the resync interval expires (time for the next resync) the triggers, if set, will prevent resync unless one or more triggers evaluates to true.

The following example condition triggers a resync. In the example, the last phone upgrade attempt has elapsed more than 5 minutes (300 seconds), and at least 10 minutes (600 seconds) have elapsed since the last resync attempt.

$UPGTMR gt 300 and $PRVTMR ge 600

Resync at a Specific Time

The Resync_At parameter allows the phone to resync at a specific time. This parameter uses the 24-hour format (hhmm) to specify the time.

The Resync_At_Random_Delay parameter allows the phone to resync at an unspecified delay in time. This parameter uses a positive integer format to specify the time.

Flooding the server with resync requests from multiple phones that are set to resync at the same time should be avoided. To do so, the phone triggers the resync up to 10 minutes after the specified time.

For example, if you set the resync time to 1000 (10 a.m.), the phone triggers the resync anytime between 10:00 a.m. and 10:10 a.m.

By default, this feature is disabled. If the Resync_At parameter is provisioned, the Resync_Periodic parameter is ignored.

Configurable Schedules

You can configure schedules for periodic resyncs, and you can specify the retry intervals for resync and upgrade failures by using these provisioning parameters:

- Resync_Periodic

- Resync_Error_Retry_Delay

- Upgrade_Error_Retry_Delay

Each parameter accepts a single delay value (seconds). The new extended syntax allows for a comma-separated list of consecutive delay elements. The last element in the sequence is implicitly repeated forever.

Optionally, you can use a plus sign to specify another numeric value that appends a random extra delay.

Example 1

In this example, the phone periodically resyncs every 2 hours. If a resync failure occurs, the device retries at these intervals: 30 minutes, 1 hour, 2 hours, 4 hours. The device continues to try at 4-hour intervals until it resyncs successfully.

Resync_Periodic=7200
Resync_Error_Retry_Delay=1800,3600,7200,14400

Example 2

In this example, the device periodically resyncs every hour (plus an extra random delay of up to 10 minutes). In the case of a resync failure, the device retries at these intervals: 30 minutes (plus up to 5 minutes). 1 hour (plus up to 10 minutes), 2 hours (plus up to 15 minutes). The device continues to try at 2-hour intervals (plus up to 15 minutes) until it successfully resyncs.

Resync_Periodic=3600+600
Resync_Error_Retry_Delay=1800+300,3600+600,7200+900

Example 3

In this example, if a remote upgrade attempt fails, the device retries the upgrade in 30 minutes, then again after one more hour, then in two hours. If the upgrade still fails, the device retries every four to five hours until the upgrade succeeds.

Upgrade_Error_Retry_Delay  =  1800,3600,7200,14400+3600

Profile Rules

The phone provides multiple remote configuration profile parameters (Profile_Rule*). Thus, each resync operation can retrieve multiple files that different servers manage.

In the simplest scenario, the device resyncs periodically to a single profile on a central server, which updates all pertinent internal parameters. Alternatively, the profile can be split between different files. One file is common for all the phones in a deployment. A separate, unique file is provided for each account. Encryption keys and certificate information can be supplied by still another profile, stored on a separate server.

Whenever a resync operation is due, the phone evaluates the four Profile_Rule* parameters in sequence:

1. Profile_Rule

2. Profile_Rule_B

3. Profile_Rule_C

4. Profile_Rule_D

Each evaluation can result in a profile retrieval from a remote provisioning server, with a possible update of some number of internal parameters. If an evaluation fails, the resync sequence is interrupted, and is retried again from the beginning specified by the Resync_Error_Retry_Delay parameter (seconds). If all evaluations succeed, the device waits for the second specified by the Resync_Periodic parameter and then performs another resync.

The contents of each Profile_Rule* parameter consist of a set of alternatives. The alternatives are separated by the | (pipe) character. Each alternative consists of a conditional expression, an assignment expression, a profile URL, and any associated URL options. All these components are optional within each alternative. The following are the valid combinations, and the order in which they must appear, if present:

[ conditional-expr ] [ assignment-expr ] [[ options ] URL ]

Within each Profile_Rule* parameter, all alternatives except the last one must provide a conditional expression. This expression is evaluated and is processed as follows:

1. Conditions are evaluated from left to right, until one is found that evaluates as true (or until one alternative is found with no conditional expression).

2. Any accompanying assignment expression is evaluated, if present.

3. If a URL is specified as part of that alternative, an attempt is made to download the profile that is located at the specified URL. The system attempts to update the internal parameters accordingly.

If all alternatives have conditional expressions and none evaluates to true (or if the whole profile rule is empty), the entire Profile_Rule* parameter is skipped. The next profile rule parameter in the sequence is evaluated.

Example 1

This example resyncs unconditionally to the profile at the specified URL, and performs an HTTP GET request to the remote provisioning server:

http://remote.server.com/cisco/$MA.cfg

Example 2

In this example, the device resyncs to two different URLs, depending on the registration state of Line 1. In case of lost registration, the device performs an HTTP POST to a CGI script. The device sends the contents of the macro expanded GPP_A, which may provide additional information on the device state:

($PRVTMR ge 600)? http://p.tel.com/has-reg.cfg
| [--post a] http://p.tel.com/lost-reg?

Example 3

In this example, the device resyncs to the same server. The device provides additional information if a certificate is not installed in the unit (for legacy pre-2.0 units):

(“$CCERT” eq “Installed”)? https://p.tel.com/config?
| https://p.tel.com/config?cisco$MAU

Example 4

In this example, Line 1 is disabled until GPP_A is set equal to Provisioned through the first URL. Afterwards, it resyncs to the second URL:

(“$A” ne “Provisioned”)? (Line_Enable_1_ = “No”;)! https://p.tel.com/init-prov
| https://p.tel.com/configs

Example 5

In this example, the profile that the server returns is assumed to contain XML element tags. These tags must be remapped to proper parameter names by the aliases map stored in GPP_B:

[--alias b] https://p.tel.com/account/$PN$MA.xml

A resync is typically considered unsuccessful if a requested profile is not received from the server. The Resync_Fails_On_FNF parameter can override this default behavior. If Resync_Fails_On_FNF is set to No, the device accepts a file-not-found response from the server as a successful resync. The default value for Resync_Fails_On_FNF is Yes.

Upgrade Rule

Upgrade rule is to tell the device to activate to a new load and from where to get the load, if necessary. If the load is already on the device, it will not try to get the load. So, validity of the load location does not matter when the desired load is in the inactive partition.

The Upgrade_Rule specifies a firmware load which, if different from the current load, will be downloaded and applied unless limited by a conditional expression or Upgrade_Enable is set to No.

The phone provides one configurable remote upgrade parameter, Upgrade_Rule. This parameter accepts syntax similar to the profile rule parameters. URL options are not supported for upgrades, but conditional expressions and assignment expressions can be used. If conditional expressions are used, the parameter can be populated with multiple alternatives, separated by the | character. The syntax for each alternative is as follows:

[ conditional-expr ] [ assignment-expr ] URL

As in the case of Profile_Rule* parameters, the Upgrade_Rule parameter evaluates each alternative until a conditional expression is satisfied or an alternative has no conditional expression. The accompanying assignment expression is evaluated, if specified. Then, an upgrade to the specified URL is attempted.

If the Upgrade_Rule contains a URL without a conditional expression, the device upgrades to the firmware image that the URL specifies. After macro expansion and evaluation of the rule, the device does not reattempt to upgrade until the rule is modified or the effective combination of scheme + server + port + filepath is changed.

To attempt a firmware upgrade, the device disables audio at the start of the procedure and reboots at the end of the procedure. The device automatically begins an upgrade that is driven by the contents of Upgrade_Rule only if all voice lines are currently inactive.

For example,

- For Cisco IP Phone 6821:

http://p.tel.com/firmware/sip6821.11-3-1MPP-678.loads

- For the other phones in Cisco IP 6800 Series:

http://p.tel.com/firmware/sip68xx.11-3-1MPP-678.loads

In this example, the Upgrade_Rule upgrades the firmware to the image that is stored at the indicated URL.

Here is another example:

- For Cisco IP Phone 6821:

(“$F” ne “beta-customer”)? http://p.tel.com/firmware/sip6821.11-3-1MPP-678.loads
| http://p.tel.com/firmware/sip6821.11-3-1MPP-678.loads

- For the other models in Cisco IP Phone 6800 Series:

(“$F” ne “beta-customer”)? http://p.tel.com/firmware/sip68xx.11-3-1MPP-678.loads
| http://p.tel.com/firmware/sip68xx.11-3-1MPP-678.loads

This example directs the unit to load one of two images, based on the contents of a general-purpose parameter, GPP_F.

The device can enforce a downgrade limit regarding firmware revision number, which can be a useful customization option. If a valid firmware revision number is configured in the Downgrade_Rev_Limit parameter, the device rejects upgrade attempts for firmware versions earlier than the specified limit.

Data Types

These data types are used with configuration profile parameters:

- {a,b,c,…}—A choice among a, b, c, …

- Bool—Boolean value of either “yes” or “no.”

- CadScript—A miniscript that specifies the cadence parameters of a signal. Up to 127 characters.

Syntax: S_1[;S_2], where:

- Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/offi,5[,oni,6/offi,6]]]]]) and is known as a section.

- oni,j and offi,j are the on/off duration in seconds of a segment. i = 1 or 2, and j = 1 to 6.

- Di is the total duration of the section in seconds.

All durations can have up to three decimal places to provide 1 ms resolution. The wildcard character “*” stands for infinite duration. The segments within a section are played in order and repeated until the total duration is played.

Example 1:

60(2/4)

Number of Cadence Sections = 1
Cadence Section 1: Section Length = 60 s
Number of Segments = 1
Segment 1: On=2s, Off=4s

Total Ring Length = 60s

Example 2—Distinctive ring (short,short,short,long):

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

Number of Cadence Sections = 1
Cadence Section 1: Section Length = 60s
Number of Segments = 4
Segment 1: On=0.2s, Off=0.2s
Segment 2: On=0.2s, Off=0.2s
Segment 3: On=0.2s, Off=0.2s
Segment 4: On=1.0s, Off=4.0s

Total Ring Length = 60s

- DialPlanScript—Scripting syntax that is used to specify Line 1 and Line 2 dial plans.

- Float<n>—A floating point value with up to n decimal places.

- FQDN—Fully Qualified Domain Name. It can contain up to 63 characters. Examples are as follows:

- sip.Cisco.com:5060 or 109.12.14.12:12345

- sip.Cisco.com or 109.12.14.12

- FreqScript—A miniscript that specifics the frequency and level parameters of a tone. Contains up to 127 characters.

Syntax: F_1@L_1[,F_2@L_2[,F_3@L_3[,F_4@L_4[,F_5@L_5[,F_6@L_6]]]]], where:

- F_1–F_6 are frequency in Hz (unsigned integers only).

- L_1–L_6 are corresponding levels in dBm (with up to one decimal place).

White spaces before and after the comma are allowed but not recommended.

Example 1—Call Waiting Tone:

440@-10

Number of Frequencies = 1
Frequency 1 = 440 Hz at –10 dBm

Example 2—Dial Tone:

350@-19,440@-19

Number of Frequencies = 2
Frequency 1 = 350 Hz at –19 dBm
Frequency 2 = 440 Hz at –19 dBm

- IP— Valid IPv4 Address in the form of x.x.x.x, where x is between 0 and 255. Example: 10.1.2.100.

- UserID—User ID as it appears in a URL; up to 63 characters.

- Phone—A phone number string, such as 14081234567, *69, *72, 345678; or a generic URL, such as, 1234@10.10.10.100:5068 or jsmith@Cisco.com. The string can contain up to 39 characters.

- PhTmplt—A phone number template. Each template may contain one or more patterns that are separated by a comma (,). White space at the beginning of each pattern is ignored. “?” and “*” represent wildcard characters. To represent literally, use %xx. For example, %2a represents *. The template can contain up to 39 characters. Examples: “1408*, 1510*”, “1408123????, 555?1.”.

- Port—TCP/UDP Port number (0-65535). It can be specified in decimal or hex format.

- ProvisioningRuleSyntax—Scripting syntax that is used to define configuration resync and firmware upgrade rules.

- PwrLevel—Power level expressed in dBm with one decimal place, such as –13.5 or 1.5 (dBm).

- RscTmplt—A template of SIP Response Status Code, such as “404, 5*”, “61?”, “407, 408, 487, 481”. It can contain up to 39 characters.

- Sig<n>—Signed n-bit value. It can be specified in decimal or hex format. A “-” sign must precede negative values. A + sign before positive values is optional.

- Star Codes—Activation code for a supplementary service, such as *69. The code can contain up to 7 characters.

- Str<n>—A generic string with up to n nonreserved characters.

- Time<n>—Time duration in seconds, with up to n decimal places. Extra specified decimal places are ignored.

- ToneScript—A miniscript that specifies the frequency, level, and cadence parameters of a call progress tone. Script may contain up to 127 characters.

Syntax: FreqScript;Z_1[;Z_2].

The section Z_1 is similar to the S_1 section in a CadScript, except that each on/off segment is followed by a frequency components parameter: Z_1 = D_1(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2 [,oni,3/offi,3/fi,3 [,oni,4/offi,4/fi,4 [,oni,5/offi,5/fi,5 [,oni,6/offi,6/fi,6]]]]]) where:

- fi,j = n_1[+n_2]+n_3[+n_4[+n_5[+n_6]]]]].

- 1 < nk < 6 specifies the frequency components in the FreqScript that are used in that segment.

If more than one frequency component is used in a segment, the components are summed together.

Example 1—Dial tone:

350@-19,440@-19;10(*/0/1+2)

Number of Frequencies = 2
Frequency 1 = 350 Hz at –19 dBm
Frequency 2 = 440 Hz at –19 dBm
Number of Cadence Sections = 1
Cadence Section 1: Section Length = 10 s
Number of Segments = 1
Segment 1: On=forever, with Frequencies 1 and 2

Total Tone Length = 10s

Example 2—Stutter tone:

350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Number of Frequencies = 2
Frequency 1 = 350 Hz at –19 dBm
Frequency 2 = 440 Hz at –19 dBm
Number of Cadence Sections = 2
Cadence Section 1: Section Length = 2s
Number of Segments = 1
Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2
Cadence Section 2: Section Length = 10s
Number of Segments = 1
Segment 1: On=forever, with Frequencies 1 and 2

Total Tone Length = 12s

- Uns<n>—Unsigned n-bit value, where n = 8, 16, or 32. It can be specified in decimal or hex format, such as 12 or 0x18, as long as the value can fit into n bits.

Note

Keep these under consideration:

- <Par Name> represents a configuration parameter name. In a profile, the corresponding tag is formed by replacing the space with an underscore “_”, such as Par_Name.

- An empty default value field implies an empty string < “” >.

- The phone continues to use the last configured values for tags that are not present in a given profile.

- Templates are compared in the order given. The first, not the closest, match is selected. The parameter name must match exactly.

- If more than one definition for a parameter is given in a profile, the last such definition in the file is the one that takes effect in the phone.

- A parameter specification with an empty parameter value forces the parameter back to its default value. To specify an empty string instead, use the empty string "" as the parameter value.

Profile Updates and Firmware Upgrades

The phone supports secure remote provisioning (configuration) and firmware upgrades. An unprovisioned phone can receive an encrypted profile targeted for that device. The phone does not require an explicit key due to a secure first-time provisioning mechanism that uses SSL functionality.

User intervention is not required to either start or complete a profile update, or firmware upgrade, or if intermediate upgrades are required to reach a future upgrade state from an older release. A profile resync is only attempted when the phone is idle, because a resync can trigger a software reboot and disconnect a call.

General-purpose parameters manage the provisioning process. Each phone can be configured to periodically contact a normal provisioning server (NPS). Communication with the NPS does not require the use of a secure protocol because the updated profile is encrypted by a shared secret key. The NPS can be a standard TFTP, HTTP, or HTTPS server with client certificates.

The administrator can upgrade, reboot, restart, or resync phones by using the phone web user interface. The administrator can also perform these tasks by using a SIP notify message.

Configuration profiles are generated by using common, open-source tools that integrate with service provider provisioning systems.

- Allow Profile Updates

- Allow and Configure Firmware Upgrades

- Upgrade Firmware by TFTP, HTTP, or HTTPS

- Upgrade Firmware With a Browser Command

Allow Profile Updates

Profile updates can be allowed at specified intervals. Updated profiles are sent from a server to the phone by using TFTP, HTTP, or HTTPS.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice  > Provisioning.

Step 2

In the Configuration Profile section, choose Yes from the Provision Enable parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Default: Yes

Step 3

Set the parameters as described in the Profile Resync Parameters table.

Step 4

Click Submit All Changes.

Allow and Configure Firmware Upgrades

Firmware updates can be allowed at specified intervals. Updated firmware is sent from a server to the phone by using TFTP or HTTP. Security is less of an issue with a firmware upgrade, because firmware does not contain personal information.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

In the Firmware Upgrade section, choose Yes from the Upgrade Enable parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: Yes

Step 3

Set the Upgrade Error Retry Delay parameter in seconds.

The upgrade retry interval (in seconds) applied in case of upgrade failure. The device has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Upgrade_Error_Retry_Delay ua="na">3600</Upgrade_Error_Retry_Delay>

Default: 3600

:

<tftp|http|https>://<ip address>/image/<load name>

Step 4

Set the Upgrade Rule parameter by entering a firmware upgrade script that defines upgrade conditions and associated firmware URLs. It uses the same syntax as Profile Rule. Enter a script and use the following format to enter the upgrade rule:

<tftp|http|https>://<ipaddress>/image/<load name>

For example:

tftp://192.168.1.5/image/sip88xx.11-0-0MPP-BN.loads

tftp://192.168.1.5/image/sip78xx.11-0-1MPP-BN.loads

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Upgrade_Rule ua="na">http://10.74.10.205:6970/sip8845_65.0104-MPP-9875dev.loads
</Upgrade_Rule>

Step 5

Click Submit All Changes.

Upgrade Firmware by TFTP, HTTP, or HTTPS

The phone supports firmware upgrade by TFTP, HTTP, or HTTPS.

Note

Downgrades to earlier releases may not be available for all devices. For more information, see the release notes for your phone and firmware version.

Before you begin

The firmware load file must be downloaded to an accessible server.

Procedure

Step 1

Rename the image as follows:

cp-x8xx-sip.aa-b-cMPP.cop to cp-x8xx-sip.aa-b-cMPP.tar.gz

where:

x8xx is the phone series, such as 6841.

aa-b-c is the release number, such as 10-4-1

Step 2

Use the tar –xzvf command to untar the tar ball.

Step 3

Copy the folder to a TFTP, HTTP, or HTTPS download directory.

Step 4

Access the phone administration web page. See Access the Phone Web Interface.

Step 5

Select Voice  > Provisioning.

Step 6

Find the load filename which ends in .loads and append it to the valid URL.

Step 7

Click Submit All Changes.

Upgrade Firmware With a Browser Command

An upgrade command entered into the browser address bar can be used to upgrade firmware on a phone. The phone updates only when it is idle. The update is attempted automatically after the call is complete.

Procedure

To upgrade the phone with a URL in a web browser, enter this command:

http://<phone_ip>/admin/upgrade?<schema>://<serv_ip[:port]>/filepath

Part II. Cisco IP Phone Configuration

- Access Control Configuration

- Third-Party Call Control Setup

- Cisco IP Phone Security

- Phone Features and Setup

- Phone Information and Display Configuration

- Call Features Configuration

- Audio Configuration

- Voicemail Configuration

- Corporate and Personal Directory Setup

- Cisco IP Phone 6800 Key Expansion Module

Chapter 7. Access Control Configuration

- Access Control

- Administrator and User Accounts

- User Access Attribute

- Access the Phone Web Interface

- Control Access to the Phone Settings

- Bypass the Set Password Screen

Access Control

If the <Phone-UI-User-Mode> parameter is enabled, the phone GUI honors the user access attribute of the relevant parameters when the GUI presents a menu item.

For menu entries that are associated with a single configuration parameter:

- Provisioning the parameter with “ua=na” (“ua” stands for “user access”) attribute makes the entry disappear.

- Provisioning the parameter with “ua=ro” attribute makes the entry read-only and non-editable.

For menu entries that are associated with multiple configuration parameters:

- Provisioning all concerned parameters with “ua=na” attribute makes the entries disappear.

Administrator and User Accounts

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 Cisco IP 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 Cisco IP 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 Cisco IP Phone. User access to the phone web user interface can be disabled.

User Access Attribute

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

- 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.

Access the Phone Web Interface

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.

Procedure

Step 1

Ensure that the computer can communicate with the phone. No VPN in use.

Step 2

Start a web browser.

Step 3

Enter the IP address of the phone in your web browser address bar.

- User Access: http://<ip address>

- Admin Access: http://<ip address>/admin/advanced

- Admin Access: http://<ip address>, click Admin Login and click advanced

For example, http://10.64.84.147/admin

Step 4

Enter the password when prompted.

Related Tasks

Assign a Speed Dial Number

Control Access to the Phone Settings

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 Access Control Parameters.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Click Voice > System.

Step 2

In the System Configuration section, configure the parameters as defined in the Access Control Parameters table.

Step 3

Click Submit All Changes to apply the changes.

- Access Control Parameters

Access Control Parameters

The following table defines the function and usage of the access control parameters in the System Configuration section under the Voice > System 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.

Access Control Parameters

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:

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

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

- In the phone web interface, set to Yes to allow the access.

Allowed values: Yes|No

Default: Yes.

Enable Web Admin Access

Allows or blocks the access to the phone administration pages:

http://<phone_IP>/admin

When set to No, the web page for administrator is inaccessible. Only the web page for user is accessible.

Note

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:

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

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

- In the phone web interface, set this parameter to Yes to allow the access. Otherwise, set it to No.

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.

Perform one of the following:

- In the phone configuration file with XML(cfg.xml), enter a string in this format: <Admin_Password ua="na">P0ssw0rd_tes89</Admin_Password>

- In the phone web interface, enter the password for administrator access.

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.

You can also set or change the user password from the phone screen menu Applications > Device administration > Set password.

A valid password must contain 4 to 127 characters from three out of the four types: capital letter, small letter, number, and special character.

In the configuration file (cfg.xml), you can use the User_Password parameter to bypass the Set password screen that prompts on the first boot or after a factory reset. For more information, see Bypass the Set Password Screen.

Default: Empty

Phone-UI-User-Mode

This parameter works only with the user access the (ua) attribute attached to an element tag in the configuration file (cfg.xml). You can restrict the parameters that the phone users see on the phone screen.

When set to Yes, you can use the ua attribute to control user access to specific parameters on the phone screen menu. When set to No, the ua attribute isn’t working.

The options for the ua attribute are “na”, “ro”, and “rw”. Parameters designated as “na” don't appear on the phone screen. Parameters designated as “ro” aren't editable by the user. Parameters designated as “rw” are editable by the user.

Perform one of the following:

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

<Phone-UI-User-Mode ua="na">No</Phone-UI-User-Mode>

- In the phone web interface, set to Yes and then set the ua attribute of the desired parameter in the phone configuration file.

Example:

<Phone-UI-User-Mode ua="na">Yes</Phone-UI-User-Mode>
<Enable_VLAN ua="ro">Yes</Enable_VLAN>
<Preferred_Audio_Device ua="rw">Headset</Preferred_Audio_Device>
<Block_ANC_Setting ua="na">Yes</Block_ANC_Setting>

With the settings in the example, the user:

- Can see but can't change the setting of VLAN (Enable_VLAN) on the phone screen menu

- Can change the setting of Preferred audio device (Preferred_Audio_Device)

- Can't see the menu item Block anonymous call (Block_ANC_Setting) on the phone screen.

Allowed values: Yes|No

Default: No

User Password Prompt

Controls whether the user password setup screen prompts.

Perform one of the following:

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

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

- In the phone web interface, set to Yes to make the prompt available to the user.

Allowed values: Yes|No

Default: Yes

Bypass the Set Password Screen

Note

This feature isn’t available from firmware release 11.2.3 and later.

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.

Procedure

Step 1

Edit the phone cfg.xml file in a text or XML editor.

Step 2

Insert the <User_Password> tag using one of these options.

- No password (start and end tag)–<User_Password></User_Password>

- Password value (4-127 characters)–<User_Password >Abc123</User_Password>

- No password (start tag only)–<User_Password />

Step 3

Save the changes to the cfg.xml file.

The Set password screen doesn't appear on the first boot or after a factory reset. If a password is specified, the user is prompted to enter the password when accessing the phone web interface or the phone screen menus.

Chapter 8. Third-Party Call Control Setup

- Determine the Phone MAC Address

- Network Configuration

- Provisioning

- Report Current Phone Configuration to the Provisioning Server

Determine the Phone MAC Address

To add phones to the Third-Party Call Control system, determine the MAC address of a Cisco IP Phone.

Procedure

Perform one of the following actions:

- On the phone, press Applications > Status > Product Information, and look at the MAC address field.

- Look at the MAC label on the back of the phone.

- Display the web page for the phone and select Info > Status > Product Information.

Network Configuration

The Cisco IP Phone is used as a part of a SIP network, because the phone supports Session Initiation Protocol (SIP). The Cisco IP Phone is compatible with other SIP IP PBX call control systems, such as BroadSoft, MetaSwitch, and Asterisk.

Configuration of these systems is not described in this document. For more information, see the documentation for the SIP PBX system to which you are connecting the Cisco IP Phone.

This document describes some common network configurations; however, your configuration can vary, depending on the type of equipment that your service provider uses.

Provisioning

Phones can be provisioned to download configuration profiles or updated firmware from a remote server when they are connected to a network, when they are powered up, and at set intervals. Provisioning is typically part of high-volume, Voice-over-IP (VoIP) deployments and is limited to service providers. Configuration profiles or updated firmware are transferred to the device through use of TFTP, HTTP, or HTTPS.

Report Current Phone Configuration to the Provisioning Server

You can configure the phone to report its full configuration, delta changes in the configuration, or the status data to the server. You can add up to two URLs in the Report Rule field to specify the destination for the report, and include an optional encryption key.

When requesting delta configuration and status reports at once, separate report rules with a space. Include a destination upload-URL in each of the report rules. You can optionally precede the report rule by one or more content arguments that are enclosed in square brackets [ ].

When a report upload is attempted, the HTTP Report Method field specifies whether the HTTP Request that the phone sends should be an HTTP PUT or an HTTP POST. Choose:

- PUT Method–To create a new report or overwrite an existing report at a known location on the server. For example, you may want to keep overwriting each report that you send and only store the most current configuration on the server.

- POST Method–To send the report data to the server for processing, such as, by a PHP script. This approach provides more flexibility for storing the configuration information. For example, you may want to send a series of phone status reports and store all the reports on the server.

Use the following content arguments in the Report Rule field to send specific configuration reports:

Content Argument

Report Content

Default: Blank

Full Configuration report

[--delta]

Configuration report contains only the latest changed fields

For example,

- Report 1 contains ABC changes.

- Report 2 contains XYZ changes (not ABC and XYZ).

[--status]

Full Phone Status report

Note

The preceding arguments can be combined with other arguments, such as, --key,--uid, and --pwd. These arguments control upload authentication and encryption, and are documented in the Profile Rule field.

- When you specify the [--key <encryption key>] argument in the Report Rule, the phone applies AES-256-CBC encryption to the file (configuration, status, or delta), with the specified encryption key.

Note

If you have provisioned the phone with Input Keying Material (IKM) and want the phone to apply RFC 8188-based encryption to the file, do not specify the --key argument.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning > Upload Configuration Options.

Step 2

Set the parameter for each of the five fields as described in Parameters for Reporting the Phone Configuration to the Server.

Step 3

Click Submit All Changes.

Example of user inputs and the resulting actions of the phone and provisioning server for the Report Rule:

- HTTP PUT ALL configuration:

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

http://my_http_server/config-mpp.xml
Then the phone will report the configuration data to http://my_http_server/config-mpp.xml.

- HTTP PUT Changed Configuration

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

[--delta]http://my_http_server/config-mpp-delta.xml;
Then the phone will report changed configuration to http://my_http_server/config-mpp-delta.xml.

- HTTP PUT Encrypted Delta Configuration

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

[--delta --key test123]http://my_http_server/config-mpp-delta.enc.xml;
The phone will report status data to http://my_http_server/config-mpp-delta.enc.xml

On the report server side, the file can be decrypted like this: # openssl enc -d -aes-256-cbc -k test123 -in config-mpp-delta.enc-delta.enc -out cfg.xml

- HTTP PUT Status Data

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

[--status]http://my_http_server/config-mpp-status.xml;
The phone will report status data to http://my_http_server/config-mpp-status.xml

- HTTP PUT Changed Configuration and Status

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

[--status]http://my_http_server/config-mpp-status.xml [--delta]http://my_http_server/config-mpp-delta.xml
The phone will report status data to http://my_http_server/config-mpp-status.xml and http://my_http_server/config-mpp-delta.xml

- HTTP POST Changed Configuration

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

[--delta]http://my_http_server/report_upload.php
The report upload file format"

// report_upload.php content
<?php
$filename = “report_cfg.xml”;  // report file name
// where to put the file
$file = “/path/to/file”.$filename;
// get data from http post
$report_data = file_get_contents(‘php://input');
// save the post data to file
$file_put_contents($file, $report_data);
?>

The phone will upload changed data to http://my_http_server/report_cfg.xml

- Parameters for Reporting the Phone Configuration to the Server

Parameters for Reporting the Phone Configuration to the Server

Parameters for Reporting the Phone Configuration to the Server

Field

Description

Report Rule

Specifies how the phone reports its current internal configuration to the provisioning server. The URLs in this field specify the destination for a report and can include an encryption key.

You can use the following keywords, encryption key, and file locations and names to control how you store the phone configuration information:

- No keywords and only an XML file reports the entire configuration data to server.

- [--status] keyword reports the status data to server.

- [--delta] keyword reports the changed configuration to server.

- [--key <encryption key>] keyword tells the phone to apply AES-256-CBC encryption with the specified encryption key to the configuration report, before sending it to the server.

You can enclose the encryption key in double-quotes (") optionally.

Note

If you have provisioned the phone with Input Keying Material (IKM) and want the phone to apply RFC 8188-based encryption to the file, do not specify a AES-256-CBC encryption key.

- Two rules used together as:

[--delta]http://my_http_server/config-mpp-delta.xml
[--status]http://my_http_server/config-mpp-status.xml

Caution

If you need to use the [--delta]xml-delta file rule and the [--status]xml-status file rule together, you must separate the two rules with a space.

Perform one of the following:

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

<Profile_Rule ua="na">
[--delta]http://my_http_server/config-mpp-delta.xml
[--status]http://my_http_server/config-mpp-status.xml
</Profile_Rule>

- In the phone web interface, enter the profile rule in this field.

HTTP Report method:

Specifies whether the HTTP Request that the phone sends should be an PUT or an POST.

- PUT–To create a new report or overwrite an existing report at a known location on the server. For example, you may want to keep overwriting each report that you send and only store the most current configuration on the server.

- POST–To send the report data to the server for processing, such as, by a PHP script. This approach provides more flexibility for storing the configuration information. For example, you may want to send a series of phone status reports and store all the reports on the server.

Perform one of the following:

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

<HTTP_Report_Method ua="na">PUT</HTTP_Report_Method>

- In the phone web interface, select an HTTP report method.

Allowed values: PUT|POST

Default: POST

Report to Server:

Defines when the phone reports its configuration to the provisioning servers.

- On Request: The phone reports its configuration only when an administrator sends a sip notify event, or the phone restarts.

- On Local Change: The phone reports its configuration when any configuration parameter changes by an action on the phone or on the phone administration web page. The phone waits for a few seconds after a change is made, and then reports the configuration. This delay ensures that changes are reported to the web server in batches, rather than reporting a single change at a time.

- Periodically: The phone reports its configuration at regular intervals. The interval is expressed in seconds.

Perform one of the following:

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

<Report_to_Server ua="na">Periodically</Report_to_Server>

- In the phone web interface, select an option from the list.

Allowed values: On Request|On Local Change|Periodically

Default: On Request

Periodic Upload to Server:

Defines the interval (in seconds) that the phone reports its configuration to the provisioning servers.

This field is used only when Report to Server is set to Periodically.

Perform one of the following:

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

<periodic_upload_to_server ua="na">3600</periodic_upload_to_server>

- In the phone web interface, specify the interval in seconds.

Allowed values: An integer ranging between 600 and 259200

Default: 3600

Upload Delay On Local Change:

Defines the delay (in seconds) that the phone waits after a change is made, and then reports the configuration.

This field is used only when Report to Server is set to On Local Change.

Perform one of the following:

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

<Upload_Delay_On_Local_Change ua="na">60</Upload_Delay_On_Local_Change>

- In the phone web interface, specify the delay in seconds.

Allowed values: An integer ranging between 10 and 900

Default: 60

Chapter 9. Cisco IP Phone Security

- Domain and Internet Setting

- Configure the Challenge for SIP INVITE Messages

- Transport Layer Security

- HTTPS Provisioning

- Enable the Firewall

- Configure Your Firewall with Additional Options

- Configure the Cipher List

- Enable Hostname Verification for SIP over TLS

- Enable Client-Initiated Mode for Media Plane Security Negotiations

- 802.1X Authentication

- Set Up a Proxy Server

- Set Up a VPN Connection from the Phone

- Set Up a VPN Connection from the Phone Web Page

- Cisco Product Security Overview

Domain and Internet Setting

- Configure Restricted Access Domains

- Configure the DHCP Options

Configure Restricted Access Domains

You can configure the phone to register, provision, firmware upgrade, and send reports using only the specified servers. Any registration, provisioning, upgrade, and report that don't use the specified servers can't be performed on the phone. If you specify the servers to use, ensure that the servers you enter in the following fields are included in the list:

- Profile Rule, Profile Rule B, Profile Rule C, and Profile Rule D on the Provisioning tab

- Upgrade Rule and Cisco Headset Upgrade Rule on the Provisioning tab

- Report Rule on the Provisioning tab

- Custom CA Rule on the Provisioning tab

- Proxy and Outbound Proxy on the Ext (n) tab

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the System Configuration section, locate the Restricted Access Domains field and enter fully qualified domain names (FQDNs) for each server. Separate FQDNs with commas.

Example:

voiceip.com, voiceip1.com

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Restricted_Access_Domains ua="na">voiceip.com, voiceip1.com</Restricted_Access_Domains>

Step 3

Click Submit All Changes.

Configure the DHCP Options

You can set the order in which your phone uses the DHCP options. For help with DHCP Options, see DHCP Option Support.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

In the Configuration Profile section, set the DHCP Option To Use and DHCPv6 Option To Use parameters as described in the Parameters for DHCP Options Configuration table.

Step 3

Click Submit All Changes.

- Parameters for DHCP Options Configuration

- DHCP Option Support

Parameters for DHCP Options Configuration

The following table defines the function and usage of parameters for DHCP Options Configuration in the Configuration Profile section under the Voice>Provisioning 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.

Parameters for DHCP Options Configuration

Parameter

Description

DHCP Option To Use

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

Perform one of the following:

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

<DHCP_Option_To_Use ua="na">66,160,159,150,60,43,125</DHCP_Option_To_Use>

- In the phone web page, enter the DHCP options separated by commas.

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

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

DHCPv6 Option To Use

DHCPv6 options, delimited by commas, used to retrieve firmware and profiles.

Perform one of the following:

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

<DHCPv6_Option_To_Use ua="na">17,160,159</DHCPv6_Option_To_Use>

- In the phone web page, enter the DHCP options separated by commas.

Example: 17,160,159

Default: 17,160,159

DHCP Option Support

The following table lists the DHCP options that are supported on the multiplatform phones.

Network Standard

Description

DHCP option 1

Subnet mask

DHCP option 2

Time offset

DHCP option 3

Router

DHCP option 6

Domain name server

DHCP option 15

Domain name

DHCP option 41

IP address lease time

DHCP option 42

NTP server

DHCP option 43

Vendor-specific information

Can be used for TR.69 Auto Configurations Server (ACS) discovery.

DHCP option 56

NTP server

NTP server configuration with IPv6

DHCP option 60

Vendor class identifier

DHCP option 66

TFTP server name

DHCP option 125

Vendor-identifying vendor-specific information

Can be used for TR.69 Auto Configurations Server (ACS) discovery.

DHCP option 150

TFTP server

DHCP option 159

Provisioning server IP

DHCP option 160

Provisioning URL

Configure the Challenge for SIP INVITE Messages

You can set up the phone to challenge the SIP INVITE (initial) message in a session. The challenge restricts the SIP servers that are permitted to interact with devices on a service provider network. This practice prevents malicious attacks against the phone. When you enable this feature, authorization is required for initial incoming INVITE requests from the SIP proxy.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, select Yes from the Auth INVITE list to enable this feature or select No to disable it.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Auth_INVITE_1>Yes</Auth_INVITE_1_>

Default: No.

Step 3

Click Submit All Changes.

Transport Layer Security

Transport Layer Security (TLS) is a standard protocol for securing and authenticating communications over the Internet. SIP over TLS encrypts the SIP signaling messages between the service provider SIP proxy and the end user.

The Cisco IP Phone uses UDP as the standard for SIP transport, but the phone also supports SIP over TLS for added security.

The following table describes the two TLS layers.

TLS Layers

Protocol Name

Description

TLS Record Protocol

Layered on a reliable transport protocol, such as SIP or TCH, this layer ensures that the connection is private through use of symmetric data encryption and it ensures that the connection is reliable.

TLS Handshake Protocol

Authenticates the server and client, and negotiates the encryption algorithm and cryptographic keys before the application protocol transmits or receives data.

- Encrypt Signaling with SIP Over TLS

- Configure LDAP over TLS

- Configure StartTLS

Encrypt Signaling with SIP Over TLS

You can configure added security when you encrypt signaling messages with SIP over TLS.

Before you begin

Access the Phone Web Interface. See Transport Layer Security

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, select TLS from the SIP Transport list.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<SIP_Transport_1_ ua="na">TLS</SIP_Transport_1_>
.

Options available:

- UDP

- TCP

- TLS

- Auto

Default: UDP.

Step 3

Click Submit All Changes.

Configure LDAP over TLS

You can configure LDAP over TLS (LDAPS) to enable secure data transmission between the server and a specific phone.

Attention

Cisco recommends leaving the authentication method to the default value of None. Next to the server field is an authentication field that uses the values None, Simple, or DIGEST-MD5. There is no TLS value for authentication. The software determines the authentication method from the LDAPS protocol in the server string.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the LDAP section, enter a server address in the Server field.

You can also configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<LDAP_Server ua="na">ldaps://10.45.76.79</LDAP_Server>

For example, enter ldaps://<ldaps_server>[:port].

where:

- ldaps://= The start of the server address string.

- ldaps_server = IP address or domain name

- port = Port number. Default: 636

Step 3

Click Submit All Changes.

Configure StartTLS

You can enable Start Transport Layer Security (StartTLS) for the communications between the phone and the LDAP server. It uses the same network port (default 389) for both secure and insecure communications. If the LDAP server supports StartTLS, TLS encrypts the communications. Otherwise, the communications are in plaintext.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the LDAP section, enter a server address in the Server field.

For example, enter ldap://<ldap_server>[:port].

Where:

- ldap:// = The start of the server address string, scheme of the URL

- ldap_server = IP address or domain name

- port = Port number

You can also configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<LDAP_Server ua="na">ldap://<ldap_server>[:port]</LDAP_Server>

Step 3

Set the StartTLS Enable field to Yes.

You can also configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Step 4

Click Submit All Changes.

Related References

Parameters for LDAP Directory

HTTPS Provisioning

The phone supports HTTPS for provisioning for increased security in managing remotely deployed units. Each phone carries a unique SLL Client Certificate (and associated private key), in addition to a Sipura CA server root certificate. The latter allows the phone to recognize authorized provisioning servers, and reject non-authorized servers. On the other hand, the client certificate allows the provisioning server to identify the individual device that issues the request.

For a service provider to manage deployment by using HTTPS, a server certificate must be generated for each provisioning server to which a phone resyncs by using HTTPS. The server certificate must be signed by the Cisco Server CA Root Key, whose certificate is carried by all deployed units. To obtain a signed server certificate, the service provider must forward a certificate signing request to Cisco, which signs and returns the server certificate for installation on the provisioning server.

The provisioning server certificate must contain the Common Name (CN) field, and the FQDN of the host running the server in the subject. It might optionally contain information following the host FQDN, separated by a slash (/) character. The following examples are of CN entries that are accepted as valid by the phone:

CN=sprov.callme.com
CN=pv.telco.net/mailto:admin@telco.net
CN=prof.voice.com/info@voice.com

In addition to verifying the server certificate, the phone tests the server IP address against a DNS lookup of the server name that is specified in the server certificate.

- Get a Signed Server Certificate

- Multiplatform Phone CA Client Root Certificate

- Redundant Provisioning Servers

- Syslog Server

Get a Signed Server Certificate

The OpenSSL utility can generate a certificate signing request. The following example shows the openssl command that produces a 1024-bit RSA public/private key pair and a certificate signing request:

openssl req –new –out provserver.csr

This command generates the server private key in privkey.pem and a corresponding certificate signing request in provserver.csr. The service provider keeps the privkey.pem secret and submits provserver.csr to Cisco for signing. Upon receiving the provserver.csr file, Cisco generates provserver.crt, the signed server certificate.

Procedure

Step 1

Navigate to https://software.cisco.com/software/cda/home and log in with your CCO credentials.

Note

When a phone connects to a network for the first time or after a factory reset, and there are no DHCP options set up, it contacts a device activation server for zero touch provisioning. New phones use "activate.cisco.com" instead of "webapps.cisco.com" for provisioning. Phones with firmware release earlier than 11.2(1) continues to use "webapps.cisco.com". We recommend that you allow both the domain names through your firewall.

Step 2

Select Certificate Management.

On the Sign CSR tab, the CSR of the previous step is uploaded for signing.

Step 3

From the Select Product drop-down list box, select SPA1xx firmware 1.3.3 and newer/SPA232D firmware 1.3.3 and newer/SPA5xx firmware 7.5.6 and newer/CP-78xx-3PCC/CP-88xx-3PCC.

Note

This product includes the Cisco IP Phone 6800 Series Multiplatform Phones.

Step 4

In the CSR File field, click Browse and select the CSR for signing.

Step 5

Select the encryption method:

- MD5

- SHA1

- SHA256

Cisco recommends that you select SHA256 encryption.

Step 6

From the Sign in Duration drop-down list box, select the applicable duration (for example, 1 year).

Step 7

Click Sign Certificate Request.

Step 8

Select one of the following options to receive the signed certificate:

- Enter Recipient’s Email Address—If you wish to receive the certificate via email, enter your email address in this field.

- Download—If you wish to download the signed certificate, select this option.

Step 9

Click Submit.

The signed server certificate is either emailed to the email address previously provided or downloaded.

Multiplatform Phone CA Client Root Certificate

Cisco also provides a Multiplatform Phone Client Root Certificate to the service provider. This root certificate certifies the authenticity of the client certificate that each phone carries. The Multiplatform Phones also support third-party signed certificates such as those provided by Verisign, Cybertrust, and so on.

The unique client certificate that each device offers during an HTTPS session carries identifying information that is embedded in its subject field. This information can be made available by the HTTPS server to a CGI script invoked to handle secure requests. In particular, the certificate subject indicates the unit product name (OU element), MAC address (S element), and serial number (L element).

The following example from the Cisco IP Phone 6841 Multiplatform Phones client certificate subject field shows these elements:

OU=CP-6841-3PCC, L=88012BA01234, S=000e08abcdef

To determine if a phone carries an individualized certificate, use the $CCERT provisioning macro variable. The variable value expands to either Installed or Not Installed, according to the presence or absence of a unique client certificate. In the case of a generic certificate, it is possible to obtain the serial number of the unit from the HTTP request header in the User-Agent field.

HTTPS servers can be configured to request SSL certificates from connecting clients. If enabled, the server can use the Multiplatform Phone Client Root Certificate that Cisco supplies to verify the client certificate. The server can then provide the certificate information to a CGI for further processing.

The location for certificate storage may vary. For example, in an Apache installation, the file paths for storage of the provisioning server-signed certificate, its associated private key, and the Multiplatform Phone CA client root certificate are as follows:

# Server Certificate:
SSLCertificateFile /etc/httpd/conf/provserver.crt

# Server Private Key:
SSLCertificateKeyFile /etc/httpd/conf/provserver.key

# Certificate Authority (CA):
SSLCACertificateFile /etc/httpd/conf/spacroot.crt

For specific information, refer to the documentation for an HTTPS server.

The Cisco Client Certificate Root Authority signs each unique certificate. The corresponding root certificate is made available to service providers for client authentication purposes.

Redundant Provisioning Servers

The provisioning server can be specified as an IP address or as a Fully Qualified Domain Name (FQDN). The use of an FQDN facilitates the deployment of redundant provisioning servers. When the provisioning server is identified through an FQDN, the phone attempts to resolve the FQDN to an IP address through DNS. Only DNS A-records are supported for provisioning; DNS SRV address resolution is not available for provisioning. The phone continues to process A-records until a server responds. If no server that is associated with the A-records responds, the phone logs an error to the syslog server.

Syslog Server

If a syslog server is configured on the phone through use of the <Syslog Server> parameters, the resync and upgrade operations send messages to the syslog server. A message can be generated at the start of a remote file request (configuration profile or firmware load), and at the conclusion of the operation (indicating either success or failure).

The logged messages are configured in the following parameters and macro expanded into the actual syslog messages:

- Log_Request_Msg

- Log_Success_Msg

- Log_Failure_Msg

Enable the Firewall

We have improved phone security by hardening the operating system. Hardening ensures that the phone has a firewall to protect it from malicious incoming traffic. The firewall tracks the ports for incoming and outgoing data. It detects incoming traffic from unexpected sources and blocks the access. Your firewall allows all outgoing traffic.

The firewall may dynamically unblock normally blocked ports. The outgoing TCP connection or UDP flow unblocks the port for return and continued traffic. The port is kept unblocked while flow is alive. The port reverts to blocked state when flow terminates or ages out.

The legacy setting, IPv6 Multicast Ping Voice > System > IPv6 Settings > Broadcast Echo continues to work independently of the new firewall settings.

Firewall configuration changes generally don't result in a phone restart. Phone soft restarts generally don't affect firewall operation.

The firewall is enabled by default. If it is disabled, you can enable it from the phone web page.

Before you begin

Access the Phone Web Interface

Procedure

Step 1

Select Voice > System > Security Settings.

Step 2

In the Firewall drop down list, select Enabled.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

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

The allowed values are Disabled|Enabled. The default value is Enabled.

Step 3

Click Submit All Changes.

This enables the firewall with its default open UDP and TCP ports.

Step 4

Select Disabled to disable the firewall if you wish your network to return to its prior behavior.

The following table describes the default open UDP ports.

Firewall Default Open UDP Ports

Default Open UDP Port

Description

DHCP/DHCPv6

DHCP client Port 68

DHCPv6 client Port 546

SIP/UDP

Configure the Port in Voice > Ext<n> > SIP Settings > SIP Port (example: 5060), when Line Enable is set to Yes, and SIP Transport is set to UDP or Auto.

RTP/RTCP

UDP port range from RTP Port Min to RTP Port Max+1

PFS (Peer Firmware Sharing)

Port 4051, when Upgrade Enable and Peer Firmware Sharing are set to Yes.

TFTP clients

Ports 53240-53245. You need this port range if the remote server uses a port other than the standard TFTP port 69. You may turn it off if the server uses standard port 69. See Configure Your Firewall with Additional Options.

TR-069

UDP/STUN port 7999, when Enable TR-069 is set to Yes.

The following table describes the default open TCP ports.

Firewall Default Open TCP Ports

Default Open TCP Port

Description

Web server

Port configured via Web Server Port (default 80), when Enable Web Server is set to Yes.

PFS (Peer Firmware Sharing)

Ports 4051 and 6970, when both Upgrade Enable and Peer Firmware Sharing are set to Yes.

TR-069

HTTP/SOAP port in TR-069 Connection Request URL, when Enable TR-069 is set to Yes.

The port is chosen randomly from the range 8000-9999.

Configure Your Firewall with Additional Options

You can configure additional options in the Firewall Options field. Type the keyword for each option in the field, and separate the keywords by commas (,). Some keywords have values. Separate the values by colons (:).

Before you begin

Access the Phone Web Interface

Procedure

Step 1

Go to Voice > System > Security Settings.

Step 2

Select Enabled for the Firewall field.

Step 3

In the Firewall Options field, enter the keywords. The list of ports applies to both IPv4 and IPv6 protocols.

When you enter the keywords,

- separate the keywords with commas (,).

- separate keywords values with colons (:).

Firewall Optional Settings

Firewall Options Keywords

Description

Field is empty.

The firewall runs with default open ports.

NO_ICMP_PING

The firewall blocks incoming ICMP/ICMPv6 Echo requests (Ping).

This option may break some types of traceroute requests to the phone. Windows tracert is one example.

Example Firewall Options entry with a combination of options:

NO_ICMP_PING,TCP:12000,UDP:8000:8010

The firewall runs with default settings and the following additional options:

- Drops incoming ICMP/ICMPv6 Echo (Ping) requests.

- Opens TCP port 12000 (IPv4 and IPv6) for incoming connections.

- Opens UDP port range 8000-8010 (IPv4 and IPv6) for incoming requests.

NO_ICMP_UNREACHABLE

The phone doesn't send ICMP/ICMPv6 Destination Unreachable for UDP ports.

Note

The exception is to always send Destination Unreachable for ports in the RTP port range.

This option may break some types of traceroute requests to the device. For example, Linux traceroute may break.

NO_CISCO_TFTP

- The phone doesn't open TFTP-client port-range (UDP 53240:53245).

- Requests to non-standard (non 69) TFTP server ports fail.

- Requests to standard TFTP server port 69 work.

The following keywords and options apply when the phone runs custom apps that handle incoming requests.

UDP:<xxx>

Opens UDP port <xxx>.

UDP:<xxx:yyy>

Opens UDP port-range, <xxx to yyy>, inclusive.

You can have up to 5 UDP port options (single ports and port ranges). For example, you can have 3 UDP:<xxx> and 2 UDP:<xxx:yyy>.

TCP:<xxx>

Opens TCP port <xxx>.

TCP:<xxx:yyy>

Opens TCP port-range <xxx to yyy>, inclusive.

You can have up to 5 TCP port options (single ports and port ranges). For example, you can have 4 TCP:<xxx> and one TCP:<xxx:yyy>.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Firewall_Config ua="na">NO_ICMP_PING</Firewall_Config>

Step 4

Click Submit All Changes.

Configure the Cipher List

You can specify the cipher suites that the phone TLS applications use. The specified cipher list applies to all the applications that use the TLS protocol. The TLS applications on your phone include:

- Customer CA Provisioning

- E911 Geolocation

- Firmware/Cisco Headset Upgrade

- LDAPS

- LDAP (StartTLS)

- Picture Download

- Logo Download

- Dictionary Download

- Provisioning

- Report Upload

- PRT Upload

- SIP over TLS

- TR-069

- WebSocket API

- XML Services

- XSI Services

You can also specify the cipher suites with the TR-069 parameter (Device.X_CISCO_SecuritySettings.TLSCipherList) or with the the configuration file (cfg.xml). Enter a string in the configuration file in this format:

<TLS_Cipher_List ua="na">RSA:!aNULL:!eNULL</TLS_Cipher_List>

Before you begin

Access the phone administration web page, see Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the Security Settings section, enter the cipher suite or the combination of cipher suites in the TLS Cipher List field.

Example:

RSA:!aNULL:!eNULL
supports those cipher suites using RSA authentication, but excludes those cipher suites offering no encryption and authentication.

Note

A valid cipher list must follow the format defined at https://www.openssl.org/docs/man1.1.1/man1/ciphers.html. Your phone doesn't support all the cipher strings listed on the OpenSSL web page. For the supported strings, see Supported Cipher Strings.

If the value in the TLS Cipher List field is blank or invalid, the cipher suites used vary with applications. See the following list for the suites that the applications use when this field is with a blank or an invalid value.

- Web Server (HTTPS) applications use the following cipher suites:

- ECDHE-RSA-AES256-GCM-SHA384

- ECDHE-RSA-AES128-GCM-SHA256

- AES256-SHA

- AES128-SHA

- DES-CBC3-SHA

- XMPP uses the cipher list HIGH:MEDIUM:AES:@STRENGTH.

- SIP, TR-069, and other applications using the curl library use the DEFAULT cipher string. The DEFAULT cipher string contains the following cipher suites that the phone support:

DEFAULT Cipher Suites (28 suites):
ECDHE_ECDSA_WITH_AES_256_GCM_SHA384
ECDHE_RSA_WITH_AES_256_GCM_SHA384
DHE_RSA_WITH_AES_256_GCM_SHA384
ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256
ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256
DHE_RSA_WITH_CHACHA20_POLY1305_SHA256
ECDHE_ECDSA_WITH_AES_128_GCM_SHA256
ECDHE_RSA_WITH_AES_128_GCM_SHA256
DHE_RSA_WITH_AES_128_GCM_SHA256
ECDHE_ECDSA_WITH_AES_256_CBC_SHA384
ECDHE_RSA_WITH_AES_256_CBC_SHA384
DHE_RSA_WITH_AES_256_CBC_SHA256
ECDHE_ECDSA_WITH_AES_128_CBC_SHA256
ECDHE_RSA_WITH_AES_128_CBC_SHA256
DHE_RSA_WITH_AES_128_CBC_SHA256
ECDHE_ECDSA_WITH_AES_256_CBC_SHA
ECDHE_RSA_WITH_AES_256_CBC_SHA
DHE_RSA_WITH_AES_256_CBC_SHA
ECDHE_ECDSA_WITH_AES_128_CBC_SHA
ECDHE_RSA_WITH_AES_128_CBC_SHA
DHE_RSA_WITH_AES_128_CBC_SHA
RSA_WITH_AES_256_GCM_SHA384
RSA_WITH_AES_128_GCM_SHA256
RSA_WITH_AES_256_CBC_SHA256
RSA_WITH_AES_128_CBC_SHA256
RSA_WITH_AES_256_CBC_SHA
RSA_WITH_AES_128_CBC_SHA
EMPTY_RENEGOTIATION_INFO_SCSV

Step 3

Click Submit All Changes.

- Supported Cipher Strings

Supported Cipher Strings

The supported cipher strings listed following is based on the OpenSSL 1.1.1d standards.

Supported Cipher Strings (OpenSSL 1.1.1d)

Strings

Strings

Strings

DEFAULT

kECDHE, kEECDH

CAMELLIA128, CAMELLIA256, CAMELLIA

COMPLEMENTOFDEFAULT

ECDHE, EECDH

CHACHA20

ALL

ECDH

SEED

COMPLEMENTOFALL

AECDH

MD5

HIGH

aRSA

SHA1, SHA

MEDIUM

aDSS, DSS

SHA256, SHA384

eNULL, NULL

aECDSA, ECDSA

SUITEB128, SUITEB128ONLY, SUITEB192

aNULL

TLSv1.2, TLSv1, SSLv3

kRSA, RSA

AES128, AES256, AES

kDHE, kEDH, DH

AESGCM

DHE, EDH

AESCCM, AESCCM8

ADH

ARIA128, ARIA256, ARIA

Enable Hostname Verification for SIP over TLS

You can enable increased phone security on a phone line if you use TLS. The phone line can verify the hostname to determine if the connection is secure.

Over a TLS connection, the phone can verify the hostname to check the server identity. The phone can check both the Subject Alternative Name (SAN) and the Subject Common Name (CN). If the hostname on the valid certificate matches the hostname that is used to communicate with the server, the TLS connection establishes. Otherwise, the TLS connection fails.

The phone always verifies the hostname for the following applications:

- LDAPS

- LDAP (StartTLS)

- XMPP

- Image upgrade over HTTPS

- XSI over HTTPS

- File download over HTTPS

- TR-069

When a phone line transports SIP messages over TLS, you can configure the line to enable or bypass the hostname verification with the TLS Name Validate field on the Ext(n) tab.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- On the Ext(n) tab, set SIP Transport to TLS.

Procedure

Step 1

Go to Voice > Ext(n).

Step 2

In the Proxy and Registration section, set the TLS Name Validate field to Yes to enable the hostname verification, or No to bypass the hostname verification.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<TLS_Name_Validate_1_ ua="na">Yes</TLS_Name_Validate_1_>

The allowed values are Yes or No. The default setting is Yes.

Step 3

Click Submit All Changes.

Enable Client-Initiated Mode for Media Plane Security Negotiations

To protect media sessions, you can configure the phone to initiate media plane security negotiations with the server. The security mechanism follows the standards stated in RFC 3329 and its extension draft Security Mechanism Names for Media (See https://tools.ietf.org/html/draft-dawes-sipcore-mediasec-parameter-08#ref-2). The transport of negotiations between the phone and the server can use SIP protocol over UDP, TCP, and TLS. You can limit that media plane security negotiation is applied only when the signaling transport protocol is TLS.

You can also configure the parameters in the configuration file (cfg.xml). To configure each parameter, see the syntax of the string in Parameters for Media Plane Security Negotiation.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext (n).

Step 2

In the SIP Settings section, set the MediaSec Request and MediaSec Over TLS Only field as defined in Parameters for Media Plane Security Negotiation

Step 3

Click Submit All Changes.

- Parameters for Media Plane Security Negotiation

Parameters for Media Plane Security Negotiation

The following table defines the function and usage of the parameters for media plane security negotiation in the SIP Settings section under the Voice> Ext (n) 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.

Parameters for Media Plane Security Negotiation

Parameter

Description

MediaSec Request

Specifies whether the phone initiates media plane security negotiations with the server.

Perform one of the following:

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

<MediaSec_Request_1_ ua="na">Yes</MediaSec_Request_1_>

- In the phone web interface, set this field to Yes or No as needed.

Allowed values: Yes|No

- Yes—Client-initiated Mode. The phone initiates media plane security negotiations.

- No—Server-initiated Mode. The server initiates media plane security negotiations. The phone doesn't initiate negotiations, but can handle negotiation requests from the server to establish secure calls.

Default: No

MediaSec Over TLS Only

Specifies the signaling transport protocol over which media plane security negotiation is applied.

Before setting this field to Yes, ensure that the signaling transport protocol is TLS.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No as needed.

Allowed values: Yes|No

- Yes—The phone initiates or handles media plane security negotiations only when the signaling transport protocol is TLS.

- No—The phone initiates and handles media plane security negotiations regardless of the signaling transport protocol.

Default: No

802.1X Authentication

Cisco IP Phones use Cisco Discovery Protocol (CDP) to identify the LAN switch and determine parameters such as VLAN allocation and inline power requirements. CDP does not identify locally attached workstations. Cisco IP Phones provide an EAPOL pass-through mechanism. This mechanism allows a workstation attached to the Cisco IP Phone to pass EAPOL messages to the 802.1X authenticator at the LAN switch. The pass-through mechanism ensures that the IP phone does not act as the LAN switch to authenticate a data endpoint before accessing the network.

Cisco IP Phones also provide a proxy EAPOL Logoff mechanism. In the event that the locally attached PC disconnects from the IP phone, the LAN switch does not see the physical link fail, because the link between the LAN switch and the IP phone is maintained. To avoid compromising network integrity, the IP phone sends an EAPOL-Logoff message to the switch on behalf of the downstream PC, which triggers the LAN switch to clear the authentication entry for the downstream PC.

Support for 802.1X authentication requires several components:

- Cisco IP Phone: The phone initiates the request to access the network. Cisco IP Phones contain an 802.1X supplicant. This supplicant allows network administrators to control the connectivity of IP phones to the LAN switch ports. The current release of the phone 802.1X supplicant uses the EAP-FAST and EAP-TLS options for network authentication.

- Cisco Secure Access Control Server (ACS) (or other third-party authentication server): The authentication server and the phone must both be configured with a shared secret that authenticates the phone.

- A LAN switch supporting 802.1X: The switch acts as the authenticator and pass the messages between the phone and the authentication server. After the exchange completes, the switch grants or denies the phone access to the network.

You must perform the following actions to configure 802.1X.

- Configure the other components before you enable 802.1X Authentication on the phone.

- Configure PC Port: The 802.1X standard does not consider VLANs and thus recommends that only a single device should be authenticated to a specific switch port. However, some switches support multidomain authentication. The switch configuration determines whether you can connect a PC to the PC port of the phone.

- Yes: If you are using a switch that supports multidomain authentication, you can enable the PC port and connect a PC to it. In this case, Cisco IP Phones support proxy EAPOL-Logoff to monitor the authentication exchanges between the switch and the attached PC.

- No: If the switch does not support multiple 802.1X-compliant devices on the same port, you should disable the PC Port when 802.1X authentication is enabled. If you do not disable this port and subsequently attempt to attach a PC to it, the switch denies network access to both the phone and the PC.

- Configure Voice VLAN: Because the 802.1X standard does not account for VLANs, you should configure this setting based on the switch support.

- Enabled: If you are using a switch that supports multidomain authentication, you can continue to use the voice VLAN.

- Disabled: If the switch does not support multidomain authentication, disable the Voice VLAN and consider assigning the port to the native VLAN.

- Enable 802.1X Authentication

Enable 802.1X Authentication

You can enable 802.1X authentication on the phone. When 802.1X authentication is enabled, the phone uses 802.1X authentication to request network access.When 802.1X authentication is turned off, the phone uses CDP to acquire VLAN and network access. You can also view the transaction status on the phone screen menu.

Procedure

Step 1

Perform one of the following actions to enable 802.1X authentication:

- In the phone web interface, select Voice > System and set the Enable 802.1X Authentication field to Yes. Then, click Submit All Changes.

- In the configuration file (cfg.xml), entering a string in this format:

<Enable_802.1X_Authentication ua="rw">Yes</Enable_802.1X_Authentication>

- On the phone, press Applications > Network configuration > Ethernet configuration > 802.1X authentication. Then, toggle the Device authentication field to On with the Select button and press Submit.

Step 2

(Optional)Select Transaction status to view the following:

- Transaction status: Displays the state of 802.1x authentication. The state can be

- Authenticating: Indicates that the authentication process is in progress.

- Authenticated: Indicates that the phone is authenticated.

- Disabled: Indicates that 802.1x authentication is disabled on the phone.

- Protocol: Displays the EAP method that is used for 802.1x authentication. The protocol can be EAP-FAST or EAP-TLS.

Step 3

Press Back to exit the menu.

Set Up a Proxy Server

You can configure the phone to use a proxy server to enhance security. A proxy server acts as a firewall between the phone and Internet. After successful configuration, the phone connects to Internet through the proxy server which protects the phone from cyber attack.
You can set up a proxy server by either using an automatic configuration script or manually configuring the host server (hostname or IP address) and port of the proxy server.

When configured, the HTTP proxy feature applies to all the applications that use the HTTP protocol. The applications include the following:

- GDS (Activation Code Onboarding)

- EDOS Device Activation

- Onboarding to Webex Cloud (via EDOS and GDS)

- Certificate Authentication

- Provisioning

- Firmware Upgrade (not supported by Cisco IP Phone 6821 Multiplatform Phones)

- Phone Status Report

- PRT Upload

- XSI Services

- Webex Services

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the section HTTP Proxy Settings, configure the parameter Proxy Mode and others according to your requirement. Detailed procedures are provided in the following steps.

Step 3

Do one of the following actions:

- Proxy Mode is Auto:

- If Use Auto Discovery (WPAD) is Yes, no further action is required. The phone will automatically retrieve a Proxy Auto-Configuration (PAC) file by the Web Proxy Auto-Discovery (WPAD) protocol.

- If Use Auto Discovery (WPAD) is No, enter a valid URL in PAC URL.

- Proxy Mode is Manual:

- If Proxy Server Requires Authentication is No, enter a proxy server in Proxy Host and a proxy port in Proxy Port.

- If Proxy Server Requires Authentication is Yes, enter a proxy server in Proxy Host and a proxy port in Proxy Port. And enter a username in Username and a password in Password.

- Proxy Mode is Off, the HTTP proxy feature is disabled on the phone.

You can also configure the parameters in the phone configuration file (cfg.xml). To configure each parameter, see the syntax of the string in the Parameters for HTTP Proxy Settings.

Step 4

Click Submit All Changes.

- Parameters for HTTP Proxy Settings

Parameters for HTTP Proxy Settings

The following table defines the function and usage of the HTTP proxy parameters in the HTTP Proxy Settings section under the Voice > System 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.

Parameters for HTTP Proxy Settings

Parameter

Description and Default Value

Proxy Mode

Specifies the HTTP proxy mode that the phone uses, or disables the HTTP proxy feature.

- Auto

The phone automatically retrieves a Proxy Auto-Configuration (PAC) file to select a proxy server. In this mode, you can determine whether to use Web Proxy Auto-Discovery (WPAD) protocol to retrieve a PAC file or manually enter a valid URL of the PAC file.

For details about the parameters, see Use Auto Discovery (WPAD) and PAC URL.

- Manual

You must manually specify a server (hostname or IP address) and a port of a proxy server.

For details about the parameters, see Proxy Host and Proxy Port.

- Off

You disable the HTTP proxy feature on the phone.

Perform one of the following:

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

<Proxy_Mode ua="rw">Off</Proxy_Mode>

- On the phone web interface, select a proxy mode or disable the feature.

Allowed values: Auto, Manual, and Off

Default: Off

Use Auto Discovery (WPAD)

Determines whether the phone uses the Web Proxy Auto-Discovery (WPAD) protocol to retrieve a PAC file.

WPAD protocol uses DHCP or DNS, or both network protocols to locate a Proxy Auto-Configuration (PAC) file automatically. PAC file is used to select a proxy server for a given URL. This file can be hosted locally or on a network.

- The parameter configuration takes effect when Proxy Mode is set to Auto.

- If you set the parameter to No, you must specify a PAC URL.

For details about the parameter, see PAC URL.

Perform one of the following:

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

<Use_Auto_Discovery__WPAD_ ua="rw">Yes</Use_Auto_Discovery__WPAD_>

- On the phone web interface, select Yes or No as needed.

Allowed values: Yes and No

Default: Yes

PAC URL

URL of a PAC file.

For example, http://proxy.department.branch.example.com

TFTP, HTTP, and HTTPS are supported.

If you set the Proxy Mode to Auto and Use Auto Discovery (WPAD) to No, you must configure this parameter.

Perform one of the following:

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

<PAC_URL ua="rw">http://proxy.department.branch.example.com/pac</PAC_URL>

- On the phone web interface, enter a valid URL that locates to a PAC file.

Default: Empty

Proxy Host

IP address or hostname of the proxy host server for the phone to access. For example:

proxy.example.com

The scheme (http:// or https://) is not required.

If you set the Proxy Mode to Manual, you must configure this parameter.

Perform one of the following:

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

<Proxy_Host ua="rw">proxy.example.com</Proxy_Host>

- On the phone web interface, enter an IP address or hostname of the proxy server.

Default: Empty

Proxy Port

Port number of the proxy host server.

If you set the Proxy Mode to Manual, you must configure this parameter.

Perform one of the following:

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

<Proxy_Port ua="rw">3128</Proxy_Port>

- On the phone web interface, enter a server port.

Default: 3128

Proxy Server Requires Authentication

Determines whether the user needs to provide the authentication credentials (username and password) that the proxy server requires. This parameter is configured according to the actual behaviour of the proxy server.

If you set the parameter to Yes, you must configure Username and Password.

For details about the parameters, see Username and Password.

The parameter configuration takes effect when Proxy Mode is set to Manual .

Perform one of the following:

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

<Proxy_Server_Requires_Authentication ua="rw">No</Proxy_Server_Requires_Authentication>

- On the phone web interface, set this field Yes or No as needed.

Allowed values: Yes and No

Default: No

Username

Username for a credential user on the proxy server.

If Proxy Mode is set to Manual and Proxy Server Requires Authentication is set to Yes, you must configure the parameter.

Perform one of the following:

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

<Proxy_Username ua="rw">Example</Proxy_Username>

- On the phone web interface, enter the username.

Default: Empty

Password

Password of the specified username for the proxy authentication purpose.

If Proxy Mode is set to Manual and Proxy Server Requires Authentication is set to Yes, you must configure the parameter.

Perform one of the following:

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

<Proxy_Password ua="rw">Example</Proxy_Password>

- On the phone web interface, enter a valid password for the proxy authentication of the user.

Default: Empty

Set Up a VPN Connection from the Phone

You can set up and enable the VPN connection from the phone.
Cisco IP Phone 6821 Multiplatform Phones doesn't support VPN connection.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > VPN settings.

Step 3

Enter the IP address or FQDN of a VPN server in VPN server.

Step 4

Enter the user credentials in Username and Password.

Step 5

(Optional) If needed, enter the name of a tunnel group in Tunnel group.

If the field is empty, this means no tunnel group is used for this VPN connection.

Step 6

Highlight Connect to VPN on bootup, press the Select button of the navigation cluster to select On.

Step 7

Press Set to save the settings.

Currently, the VPN settings are finished. You can manually reboot the phone to trigger the automatic connection to the VPN server. If you want to enable the VPN connection immediately, proceed with the next step.

Step 8

Highlight Enable VPN connection, select On to enable the VPN connection.

Note

Once you set Enable VPN connection to On, the phone immediately tries to connect to the VPN server. During the process, the phone reboots automatically.

The VPN connection takes about one minute.

After the phone reboots, the VPN connection icon on the upper-right corner of the phone screen indicates that the VPN connection is established successfully.

If the VPN connection fails, the value of Enable VPN connection remains Off.

Step 9

(Optional) View the details of the VPN connection. For example, the current VPN connection status and VPN IP address. For details, see View the VPN Status.

Step 10

(Optional) You can disable the VPN connection from the phone.

1. Press Applications .

2. Select Network configuration > VPN settings.

3. Highlight Connect to VPN on bootup, select Off.

4. Highlight Enable VPN connection, select Off to disable the VPN connection. This causes an immediate reboot of the phone.

- View the VPN Status

View the VPN Status

You can check the details of the VPN connection. For example, the current VPN status and VPN IP address of your phone.
You can also view the status from the phone web page by selecting Info > Status > VPN Status.

Procedure

Step 1

Press Applications .

Step 2

Select Status > VPN status.

You can view the following information:

- VPN connection—Indicates whether the phone connects to the VPN server. The status can be either Connected or Disconnected.

- VPN IP address—VPN IP address assigned from the VPN server.

- VPN subnet mask—VPN subnet mask assigned from the VPN server.

- Sent bytes—Total bytes the phone sent out to the network through the VPN server.

- Received bytes—Total bytes the phone received from the network through the VPN server.

Set Up a VPN Connection from the Phone Web Page

You can set up a VPN connection from the phone web page.
Cisco IP Phone 6821 Multiplatform Phones doesn't supprt VPN connection.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the section VPN Settings, configure the parameters as defined in Parameters for VPN Settings table.

Step 3

Click Submit All Changes to save the changes.

The changes don't take effect immediately. You need to manually reboot the phone or enable the VPN connection from the phone to trigger the VPN connection.

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 Parameters for VPN Settings table.

Step 4

(Optional) After the phone reboots successfully, you can view the status and other details of the VPN connection under the VPN Status section from Info > Status.

Step 5

(Optional) If you want to disable the VPN connection, set the parameter Connect on Bootup to No, and then manually reboot the phone. For details, see Reboot the Phone from the Phone Web Page.

- Parameters for VPN Settings

Parameters for VPN Settings

The following table defines the function and usage of the VPN connection parameters in the VPN Settings section under the Voice > System 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.

Parameters for VPN Settings

Parameter

Description and Default Value

VPN Server

IP address or FQDN of the VPN server for the phone to access. For example:

100.101.1.218 or vpn_server.example.com

Perform one of the following:

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

<VPN_Server ua="rw"><Server IP or FQDN></VPN_Server>

- On the phone web interface, enter an IP address or FQDN of the VPN server.

Default: Empty

VPN User Name

Username for a credential user on the VPN server.

Perform one of the following:

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

<VPN_User_Name ua="rw">Example</VPN_User_Name>

- On the phone web interface, enter the username.

Default: Empty

VPN Password

Password of the specified username to access the VPN server.

Perform one of the following:

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

<VPN_Password ua="rw">Example</VPN_Password>

- On the phone web interface, enter the password.

Default: Empty

VPN Tunnel Group

Tunnel group assigned to the VPN user.

Tunnel group is used to identify the group policy for the VPN connection.

Perform one of the following:

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

<VPN_Tunnel_Group ua="rw">Example</VPN_Tunnel_Group>

- On the phone web interface, enter the name of the tunnel group.

Default: Empty

Connect on Bootup

Enables or disables the automatic connection to the VPN server after the phone reboots.

Perform one of the following:

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

<Connect_on_Bootup ua="rw">No</Connect_on_Bootup>

- On the phone web interface, set this field Yes or No as needed.

Allowed values: Yes and No

Default: No

Cisco Product Security Overview

This product contains cryptographic features and is subject to U.S. and local country laws that govern import, export, transfer, and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute, or use encryption. Importers, exporters, distributors, and users are responsible for compliance with U.S. and local country laws. By using this product, you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately.

Further information regarding U.S. export regulations can be found at https://www.bis.doc.gov/policiesandregulations/ear/index.htm.

Chapter 10. Phone Features and Setup

- Phone Features and Setup Overview

- Cisco IP Phone User Support

- Telephony Features

- Feature Buttons and Softkeys

- Enable the User to Configure Features on Line Keys

- Configure Speed Dial on a Line Key

- Assign a Speed Dial Number

- DTMF Wait and Pause Parameters

- Phone Configuration for Monitoring Other Phones

- Enable Conference Button with a Star Code

- Configure Alphanumeric Dialing

- Call Park

- Set the Optional Network Configuration

- XML Services

- Shared Lines

- Assign a Ringtone to an Extension

- Enable Hoteling on a Phone

- Enable Flexible Seating on a Phone

- Enable Extension Mobility on a Phone

- Set the User Password

- Download Problem Reporting Tool Logs

- Configure Problem Report Tool

- Server-Configured Paging

- Configure Multicast Paging

- Configure a Phone to Accept Pages Automatically

- Manage Phones with TR-069

- View TR-069 Status

- Enable Electronic Hookswitch

- Set up a Secure Extension

- Configure the SIP Transport

- Block Non-Proxy SIP Messages to a Phone

- Configure a Privacy Header

- Enable P-Early-Media Support

- Enable Peer Firmware Sharing

- Specify the Profile Authentication Type

- Control the Authentication Requirement to Access the Phone Menus

- Silence an Incoming Call with Ignore Soft Key

- Move an Active Call from a Phone to Other Phones (Locations)

- Sync the Block Caller ID Feature with the Phone and the BroadWords XSI Server

- Enable Viewing BroadWorks XSI Call Logs on a Line

- Enable Feature Key Sync

- DND and Call Forward Status Sync

- Enable Synchronization of Anonymous Call Rejection via XSI Service

- Enable Synchronization of Call Waiting via XSI Service

- Executives and Assistants

- Enable End-of-Call Statistics Reports in SIP Messages

- SIP Session ID

- Select the Line Key LED Behavior

- Set Up a Phone for Remote SDK

- Hide a Menu Item from Being Displayed on the Phone Screen

- Display Caller Number Instead of Unresolved Caller Name

- Add a Menu Shortcut to a Line Key

- Menu Shortcuts Mapping on PLK and PSK

- Add an Extended Feature to a Line Key

- Configurable Features on Line Keys

- Add a Menu Shortcut to a Programmable Softkey

- Enable LDAP Unified Search

- Shut Down a Line Key

Phone Features and Setup Overview

After you install Cisco IP Phones in your network, configure their network settings, and add them to Third-Party Call Control System, you must use the Third-Party Call Control System to configure telephony features, optionally modify phone templates, set up services, and assign users.

You can modify additional settings for the Cisco IP Phone from Third-Party Call Control Configuration Utility. Use this web-based application to set up phone registration criteria and calling search spaces, to configure corporate directories and services, and to modify phone button templates, among other tasks.

Cisco IP Phone User Support

If you are a system administrator, you are likely the primary source of information for Cisco IP Phone users in your network or company. It is important to provide current and thorough information to end users.

To successfully use some of the features on the Cisco IP Phone (including Services and voice message system options), users must receive information from you or from your network team or must be able to contact you for assistance. Make sure to provide users with the names of people to contact for assistance and with instructions for contacting those people.

We recommend that you create a web page on your internal support site that provides end users with important information about their Cisco IP Phones.

Consider including the following types of information on this site:

- User guides for all Cisco IP Phone models that you support

- Information on how to access the Cisco Unified Communications Self Care Portal

- List of features supported

- User guide or quick reference for your voicemail system

Telephony Features

After you add Cisco IP Phones to Third-Party Call Control system, you can add functionality to the phones. The following table includes a list of supported telephony features, many of which you can configure by using Third-Party Call Control system.

Note

The Third-Party Call Control system also provides several service parameters that you can use to configure various telephony functions.

Feature

Description and More Information

AES 256 Encryption Support for Phones

Enhances security by supporting TLS 1.2 and new ciphers.

Alphanumeric Dialing

Allows users to place a call with alphanumeric characters. You can use these characters for alphanumeric dialing: a-z, A-Z, 0-9, -, _, ., and +.

Any Call Pickup

Allows users to pick up a call on any line in their call pickup group, regardless of how the call was routed to the phone.

Assisted Directed Call Park

Enables users to park a call by pressing only one button using the Direct Park feature. Administrators must configure a Busy Lamp Field (BLF) Assisted Directed Call Park button. When users press an idle BLF Assisted Directed Call Park button for an active call, the active call is parked at the Direct Park slot associated with the Assisted Directed Call Park button.

Audio Settings

Configures audio settings for the phone speaker, the handset, and the headsets that are connected to the phone.

Auto Answer

Connects incoming calls automatically after a ring or two.

Auto Answer works with either the speakerphone or the headset.

Blind Transfer

Blind Transfer: This transfer joins two established calls (call is in hold or in connected state) into one call and drops the feature initiator from the call. Blind Transfer does not initiate a consultation call and does not put the active call on hold.

Some JTAPI/TAPI applications are not compatible with the Join and Blind Transfer feature implementation on the Cisco IP Phone and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or possibly across lines.

Busy Lamp Field (BLF)

Allows user to monitor call state of a directory number.

Busy Lamp Field (BLF) Pickup

Allows user to pick up incoming calls to the directory number monitored through BLF.

Call Back

Provides users with an audio and visual alert on the phone when a busy or unavailable party becomes available.

Call Display Restrictions

Determines the information that will display for calling or connected lines, depending on the parties who are involved in the call. RPID and PAID caller id handling are supported.

Call Forward

Allows users to redirect incoming calls to another number. Call Forward services include Call Forward All, Call Forward Busy, Call Forward No Answer.

Call Forward Destination Override

Allows you to override Call Forward All (CFA) in cases where the CFA target places a call to the CFA initiator. This feature allows the CFA target to reach the CFA initiator for important calls. The override works whether the CFA target phone number is internal or external.

Call Forward Notification

Allows you to configure the information that the user sees when receiving a forwarded call.

Call History for Shared Line

Allows you to view shared line activity in the phone Call History. This feature:

- Logs missed calls for a shared line.

- Logs all answered and placed calls for a shared line.

Call Park

Allows users to park (temporarily store) a call and then retrieve the call by using another phone.

Call Pickup

Allows users to redirect a call that is ringing on another phone within their pickup group to their phone.

You can configure an audio and visual alert for the primary line on the phone. This alert notifies the users that a call is ringing in their pickup group.

Call Waiting

Indicates (and allows users to answer) an incoming call that rings while on another call. Incoming call information appears on the phone display.

Caller ID

Caller identification such as a phone number, name, or other descriptive text appear on the phone display.

Caller ID Blocking

Allows a user to block their phone number or name from phones that have caller identification enabled.

Calling Party Normalization

Calling party normalization presents phone calls to the user with a dialable phone number. Any escape codes are added to the number so that the user can easily connect to the caller again. The dialable number is saved in the call history and can be saved in the Personal Address Book.

Cisco Extension Mobility

Allows users to temporarily access their Cisco IP Phone configuration such as line appearances, services, and speed dials from shared Cisco IP Phone by logging into the Cisco Extension Mobility service on that phone when they log into the Cisco Extension Mobility service on that phone.

Cisco Extension Mobility can be useful if users work from a variety of locations within your company or if they share a workspace with coworkers.

Cisco Extension Mobility Cross Cluster (EMCC)

Enables a user configured in one cluster to log into a Cisco IP Phone in another cluster. Users from a home cluster log into a Cisco IP Phone at a visiting cluster.

Note

Configure Cisco Extension Mobility on Cisco IP Phones before you configure EMCC.

Cisco WebDialer

Allows users to make calls from web and desktop applications.

Classic Ringtone

Supports narrowband and wideband ringtones. The feature makes the available ringtones common with other Cisco IP Phones.

Client Matter Code (CMC)

Enables a user to specify that a call relates to a specific client matter.

Conference

Allows a user to talk simultaneously with multiple parties by calling each participant individually.

Allows a noninitiator in a standard (adhoc) conference to add or remove participants; also allows any conference participant to join together two standard conferences on the same line.

Note

Be sure to inform your users whether these features are activated.

Configurable RTP/sRTP Port Range

Provides a configurable port range (Port Min to Port Max) for Real-Time Transport Protocol (RTP) and secure Real-Time Transport Protocol (sRTP).

The value range for the Port Min and Port Max is 2048 to 49151.

The default RTP and sRTP port range is 16384 to 16482.

Note

If the value range (Port Max - Port Min) is less than 16 or you use an incorrect port range, the port range (16382 to 32766) is used instead.

You configure the RTP and sRTP port range in the SIP Profile.

Contacts Management of the BroadSoft Personal Directory on the Phone

Provides the user with the ability to add, edit, and delete in the BroadSoft Personal directory. Allows the user to add contacts from recent calls or any types of directories (if enabled).

In addition administrator can set the BroadSoft Personal directory as the target directory to store new contacts.

CTI Applications

A computer telephony integration (CTI) route point can designate a virtual device to receive multiple, simultaneous calls for application-controlled redirection.

Device Invoked Recording

Provides end users with the ability to record their telephone calls via a softkey.

In addition administrators may continue to record telephone calls via the CTI User Interface.

Directed Call Park

Allows a user to transfer an active call to an available directed call park number that the user dials or speed dials. A Call Park BLF button indicates whether a directed call park number is occupied and provides speed-dial access to the directed call park number.

Note

If you implement Directed Call Park, avoid configuring the Park softkey. This prevents users from confusing the two Call Park features.

Directed Call Pickup

Allows a user to pick up a ringing call on a DN directly by pressing the GPickUp softkey and entering the directory number of the device that is ringing.

Divert

Allows a user to transfer a ringing, connected, or held call directly to a voice-messaging system. When a call is diverted, the line becomes available to make or receive new calls.

Do Not Disturb (DND)

When DND is turned on, either no audible rings occur during the ringing-in state of a call, or no audible or visual notifications of any type occur.

DND and Call Forward Indication on Non-selected Line Key

Displays the DND and call forward icons next the to the line key label. The line key should be enabled with feature key sync. The line key should also be enabled with DND or call forward.

Emergency Calls

Enables users to make emergency calls. The emergency services receive the phone's location and a call-back number, to use when the emergency call unexpectedly disconnects.

EnergyWise

Enables an IP Phone to sleep (power down) and wake (power up) at predetermined times, to promote energy savings.

Enhanced Secure Extension Mobility Cross Cluster (EMCC)

Improves the Secure Extension Mobility Cross Cluster (EMCC) feature by preserving the network and security configurations on the login phone. By so doing, security policies are maintained, network bandwidth is preserved and network failure is avoided within the visiting cluster (VC).

Extension Mobility Size Safe and Feature Safe

With Feature Safe, your phone can use any phone button template that has the same number of line buttons that the phone model supports.

Size Safe allows your phone to use any phone button template that is configured on the system.

Executive-Assistant (Cisco IP Phone 6871 Multiplatform Phones only)

Indicates shared call control for executives and their assistants.

Executive-Assistant Setting Enhancements (Cisco IP Phone 6871 Multiplatform Phones only)

Allows you to show or hide the Call filter menu item on the phone for the users of the assistant role.

Enables the executive to set the call filtering criteria and view the associated assistants.

Enables the assistant to view the associated executives and choose to opt in to or opt out of the executive's assistants pool. Allows the assistant to activate or deactivate call diversion and call filtering.

Forced Authorization Code (FAC)

Controls the types of calls that certain users can place.

Feature Activation Code

Allows a user to enable, disable, or configure the Call Forward All service.

Headset Sidetone Control

Allows an administrator to set the sidetone level of a wired headset.

Group Call Pickup

Allows a user to answer a call that is ringing on a directory number in another group.

Hold Status

Enables phones with a shared line to distinguish between the local and remote lines that placed a call on hold.

Hold/Resume

Allows the user to move a connected call from an active state to a held state.

- No configurations are required unless you want to use Music On Hold. See "Music On Hold" in this table.

- See "Hold Reversion" in this table.

HTTP Download

Enhances the file download process to the phone to use HTTP by default. If the HTTP download fails, the phone reverts to using the TFTP download.

HTTP Proxy

Allows you to set up a proxy server for the phone.

HTTPS for Phone Services

Increases security by requiring communication using HTTPS.

Note

When the web is in HTTPS mode, the phone is an HTTPS server.

Improve Caller Name and Number Display

Improves the display of caller names and numbers. If the Caller Name is known, then the Caller Number is displayed instead of Unknown.

IPv6 Support

Provides support for expanded IP addressing on Cisco IP Phones. IPv6 support is provided in standalone or in dual-stack configurations. In dual-stack mode, the phone is able to communicate using IPv4 and IPv6 simultaneously, independent of the content.

Jitter Buffer

The Jitter Buffer feature handles jitter from 10 milliseconds (ms) to 1000 ms for both audio and video streams.

Join Across Lines

Allows users to combine calls that are on multiple phone lines to create a conference call.

Some JTAPI/TAPI applications are not compatible with the Join and Direct Transfer feature implementation on the Cisco IP Phone and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or possibly across lines.

Join

Allows users to combine two calls that are on one line to create a conference call and remain on thecall.

Line Display Enhancement

Improves Call Display by removing the central dividing line when it is not required. This feature applies to the Cisco IP Phone 7841 only.

Log out of hunt groups

Allows users to log out of a hunt group and temporarily block calls from ringing their phone when they are not available to take calls. Logging out of hunt groups does not prevent nonhunt group calls from ringing their phone.

Malicious Caller Identification (MCID)

Allows users to notify the system administrator about suspicious calls that are received.

Meet Me Conference

Allows a user to host a Meet Me conference in which other participants call a predetermined number at a scheduled time.

Message Waiting

Defines directory numbers for message waiting on and off indicators. A directly-connected voice-message system uses the specified directory number to set or to clear a message waiting indication for a particular Cisco IP Phone.

Message Waiting Indicator

A light on the handset that indicates that a user has one or more new voice messages.

A line key LED or a KEM key LED that indicates that a monitored voicemail user or group has one or more new voice messages.

Minimum Ring Volume

Sets a minimum ringer volume level for an IP phone.

Missed Call Logging

Allows a user to specify whether missed calls will be logged in the missed calls directory for a given line appearance.

Mobile Connect

Enables users to manage business calls using a single phone number and pick up in-progress calls on the desk phone and a remote device such as a mobile phone. Users can restrict the group of callers according to phone number and time of day.

Mobile Voice Access

Extends Mobile Connect capabilities by allowing users to access an interactive voice response (IVR) system to originate a call from a remote device such as a cellular phone.

Monitoring and Recording

Allows a supervisor to silently monitor an active call. The supervisor cannot be heard by either party on the call. The user might hear a monitoring audible alert tone during a call when it is being monitored.

When a call is secured, the security status of the call is displayed as a lock icon on Cisco IP Phones. The connected parties might also hear an audible alert tone that indicates the call is secured and is being monitored.

Note

When an active call is being monitored or recorded, the use can receive or place intercom calls; however, if the user place an intercom call, the active call will be put on hold, which causes the recording session to terminate and the monitoring session to suspend. To resume the monitoring session, the party whose call is being monitored must resume the call.

Multicasting Paging

Enables users to page some or all phones. If the phone is on an active call while a group page starts, the incoming page is ignored.

Multiple Calls Per Line Appearance

Each line can support multiple calls. By default, the phone supports two active calls per line, and a maximum of ten active calls per line. Only one call can be connected at any time; other calls are automatically placed on hold.

The system allows you to configure maximum calls/busy trigger not more than 10/6. Any configuration more than 10/6 is not officially supported.

Music On Hold

Plays music while callers are on hold.

Mute

Mutes the handset or headset microphone.

No Alert Name

Makes it easier for end users to identify transferred calls by displaying the original caller’s phone number. The call appears as an Alert Call followed by the caller’s telephone number.

Onhook Dialing

Allows a user to dial a number without going off hook. The user can then either pick up the handset or press Dial.

Other Group Pickup

Allows a user to answer a call ringing on a phone in another group that is associated with the user's group.

Pause in Speed Dial

Users can set up the speed-dial feature to reach destinations that require Forced Authorization Code (FAC) or Client Matter Code (CMC), dialing pauses, and additional digits (such as a user extension, a meeting access code, or a voicemail PIN) without manual intervention. When the user presses the speed dial, the phone establishes the call to the specified DN and sends the specified FAC, CMC, and DTMF digits to the destination and inserts the necessary dialing pauses.

Peer Firmware Sharing (PFS)

Allows IP Phones located at remote sites to share the firmware files amongst them, which saves bandwidth when the upgrade process takes place. This feature uses Cisco Peer-to-Peer-Distribution Protocol (CPPDP) which is a Cisco proprietary protocol used to form a peer-to-peer hierarchy of devices. CPPDP is also used to copy firmware or other files from peer devices to the neighbouring devices.

PFS aids in firmware upgrades in branch/remote office deployment scenarios that run over bandwidth-limited WAN links.

Provides the following advantages over the traditional upgrade method:

- Limits congestion on TFTP transfers to centralized remote TFTP servers

- Eliminates the need to manually control firmware upgrades

- Reduces phone downtime during upgrades when large numbers of devices are reset simultaneously

The more the number of IP phones, the better it's performance compared to the traditional firmware upgrade method.

PLK Support for Queue Statistics

The PLK Support for Queue Statistics feature enables the users to query the call queue statistics for hunt pilots and the information appears on phone screen.

Plus Dialing

Allows the user to dial E.164 numbers prefixed with a plus (+) sign.

To dial the + sign, the user needs to press and hold the star (*) key for at least 1 second. This applies to dialing the first digit for an on-hook (including edit mode) or off-hook call.

Power Negotiation over LLDP

Allows the phone to negotiate power using Link Level Endpoint Discovery Protocol (LLDP) and Cisco Discovery Protocol (CDP).

Problem Reporting Tool

Submits phone logs or reports problems to an administrator.

Programmable Feature Buttons

You can assign features, such as New Call, Call Back, and Call Forward All to line buttons.

Quality Reporting Tool (QRT)

Allows users to submit information about problem phone calls by pressing a button. QRT can be configured for either of two user modes, depending upon the amount of user interaction desired with QRT.

Redial

Allows users to call the most recently dialed phone number by pressing a button or the Redial softkey.

Remote Customization (RC)

Allows a service provider to customize the phone remotely. There is no need for either the service provider to physically touch the phone or a user to configure the phone. The service provider can work with a sales engineer at the time of ordering to set this up.

Ringtone Setting

Identifies ring type used for a line when a phone has another active call.

Reverse Name Lookup

Identifies the caller name using the incoming or outgoing call number. You must configure either the LDAP Directory or the XML directory. You can enable or disable the reverse name lookup using the phone administration web page.

RTCP Hold For SIP

Ensures that held calls are not dropped by the gateway. The gateway checks the status of the RTCP port to determine if a call is active or not. By keeping the phone port open, the gateway will not end held calls.

Secure Conference

Allows secure phones to place conference calls using a secured conference bridge. As new participants are added by using Confrn, Join, cBarge softkeys or MeetMe conferencing, the secure call icon displays as long as all participants use secure phones.

The Conference List displays the security level of each conference participant. Initiators can remove nonsecure participants from the Conference List. Noninitiators can add or remove conference participants if the Advanced Adhoc Conference Enabled parameter is set.

Serviceability for SIP Endpoints

Enables administrators to quickly and easily gather debug information from phones.

This feature uses SSH to remotely access each IP phone. SSH must be enabled on each phone for this feature to function.

Shared Line

Allows a user with multiple phones to share the same phone number or allows a user to share a phone number with a coworker.

Show Caller Name and Caller Number

The phones can display both the caller name and caller number for incoming calls. The phone screen size limits the length of the caller name and the caller number that display.

If boxes are displayed in the caller name, follow the procedure in Display Caller Number Instead of Unresolved Caller Name.

This feature applies to the incoming call alert only and doesn’t change the Call Forward and Hunt Group features.

See “Caller ID” in this table.

Show Product Configuration Version

Allows you to customize the product configuration version that shows on the phone screen Product information.

Show Duration for Call History

Displays the time duration of placed and received calls in the Call History details.

If the duration is greater than or equal to one hour, the time is displayed in the Hour, Minute, Second (HH:MM:SS) format.

If the duration is less than one hour, the time is displayed in the Minute, Second (MM:SS) format.

If the duration is less than one minute, the time is displayed in the Second (SS) format.

Silence Incoming Call

Allows you to silence an incoming call by pressing Ignore softkey or by pressing the volume button down.

Speed Dial

Dials a specified number that has been previously stored.

Support Executive and Assistant Roles for a User (Cisco IP Phone 6871 Multiplatform Phones only)

Allows you to set the preference for the executive-assistant role. The phone can select the role when it retrieves both roles from the BroadWorks server.

Synchronization of Call Waiting and Anonymous Call Rejection

Allows you to enable or disable synchronization of the Call Waiting and Anonymous Call Rejection functions between a specific line and a BroadSoft XSI server.

Time Zone Update

Updates the Cisco IP Phone with time zone changes.

Transfer

Allows users to redirect connected calls from their phones to another number.

Some JTAPI/TAPI applications are not compatible with the Join and Direct Transfer feature implementation on the Cisco IP Phone and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or possibly across lines.

Voice Message System

Enables callers to leave messages if calls are unanswered.

VPN Connection

Allows you to set up a VPN connection for the phone.

Web Access Enable by Default

Web services are enabled by default.

XSI call logs display

Allows you to configure a phone to display recent call logs from either the BroadWorks server or the local phone. After you enable the feature, the Recents screen has a Display recents from menu and the user can choose the XSI call logs or the local call logs.

Feature Buttons and Softkeys

The following table provides information about features that are available on softkeys, features that are available on dedicated feature buttons, and features that you need to configure as programmable feature buttons. A "Supported" entry in the table indicates that the feature is supported for the corresponding button type or softkey. Of the two button types and softkeys, only programmable feature buttons require configuration in the web interface or in the configuration file (cfg.xml).

Note

The Cisco IP Phone 6821 Multiplatform Phones does not have programmable feature buttons. Supported features display on softkeys.

Features with Corresponding Buttons and Softkeys

Feature Name

Dedicated Feature Button

Programmable Feature Button

Softkey

Answer

Not supported

Supported

Supported

Call Forward All

Not supported

Supported

Supported

Call Forward Busy

Not supported

Supported

Supported

Call Forward No Answer

Not supported

Supported

Supported

Call Park

Not supported

Supported

Supported

Call Park Line Status

Not supported

Supported

Not supported

Call Pickup (Pick Up)

Not supported

Supported

Supported

Call Pickup Line Status

Not supported

Supported

Not supported

Category

Not supported

Supported

Supported

Conference

Supported

Not supported

Supported (only displayed during connected call conference scenario)

Divert

Not supported

Not supported

Supported

Do Not Disturb

Not supported

Supported

Supported

Hold

Supported

Not supported

Supported

Intercom

Not supported

Supported

Not supported

Mute

Supported

Not supported

Not supported

Redial

Not supported

Supported

Supported

Speed Dial

Not supported

Supported

Supported

Speed Dial Line Status

Not supported

Supported

Not supported

Transfer

Supported

Not supported

Supported (only displayed during connected call transfer scenario)

Cisco IP Phone 6871 Multiplatform Phones have additional buttons and softkeys:

Additional Features with Corresponding Buttons and Softkeys for Cisco IP Phone 6871

Feature Name

Dedicated Feature Button

Programmable Feature Button

Softkey

Executive - Join ongoing call

Supported

Supported

Executive - Call filtering activation and deactivation

Supported

Executive - Call filtering status

Supported

Executive - Call transfer to self

Supported

Supported

Executive - Access to Settings > Executive menu

Supported

Executive Assistant - Call diversion activation and deactivation

Supported

Executive Assistant - Call filtering activation and deactivation

Supported

Executive Assistant - Call initiation on behalf of executive

Supported

Supported

Executive Assistant - Call transfer to executive

Supported

Supported

Executive Assistant - Access to Settings > Assistant menu

Supported

Enable the User to Configure Features on Line Keys

You can enable the user to configure features on line keys. The user can then add any of the configured features to the dedicated line keys. For the supported features, see Configurable Features on Line Keys.

For the phones with a key expansion module, the user can configure the features on the key expansion module keys. For more information, see Enable the User to Configure Features on Key Expansion Module Line Keys.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Ensure that the line keys are not in the Inert mode.

Procedure

Step 1

(Optional) Disable the extension of a line key on which you allow users to configure the features.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to allow users to configure the features on the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

1. Select Voice > Phone.

2. Select a line key.

3. Set the Extension to Disabled.

You can also disable the line key in the configuration file (cfg.xml):

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

where n is the extension number.

Step 2

Select Voice > Att Console.

Step 3

In the General section, configure the Customizable PLK Options parameter with the codes of your desired features, as defined in Configurable Features on Line Keys.

Example: You configure this parameter with blf;sd;mwi;shortcut;dnd;. The user does a long-press on an unassigned line key to see the feature list. The feature list looks like:

1 None

2 Speed dial

3 BLF presence

4 BLF + Speed dial

5 MWI

6 MWI + Speed dial

7 Menu shortcut

8 Do not disturb

The user can then select a feature or a menu shortcut to add to the line key.

If you don't enter any value in the Customizable PLK Options parameter, that is when the parameter is empty, the Select softkey doesn't appear on the Select feature screen of the phone.

You can also configure this parameter in the configuration file (cfg.xml) with a string in this format:

<Customizable_PLK_Options ua="na">mwi;sd;blf;shortcut;dnd;</Customizable_PLK_Options>

Step 4

Click Submit All Changes.

- Parameters for Features on Line Keys

Parameters for Features on Line Keys

The following table defines the function and usage of the speed dial parameters in the Line Key (n) sections under the Voice > Phone 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.

Parameters for Features on Line Keys

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. You can assign the line key with extended functions, for example, speed dial, Busy Lamp Field, and call pickup.

By default, you don't need to disable the extension to assign the line key with extended functions. However, if the Direct PLK Configuration feature is disabled, you must disable the extension to achieve the assignment. For more information about how to enable the feature, see Enable Direct PLK Configuration.

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

<Extension_1_ ua="na">1</Extension_1_>

<Extension_2_ ua="na">2</Extension_2_>

<Extension_3_ ua="na">3</Extension_3_>

<Extension_4_ ua="na">Disabled</Extension_4_>

- On the phone web interface, select a number or Disabled from the options.

Allowed values: Disabled|1|2|3|4, the allowed values vary with phones.

Default: n, where n is the line key number.

Extended Function

Used to assign extended functions to a line key on the phone. The supported functions are:

- Busy Lamp Field (BLF)

Example: fnc=blf;sub=BLF_List_URI@$PROXY;usr=user_ID@$PROXY

- BLF with Call Pickup

Example: fnc=blf+cp;sub=BLF_List_URI@$PROXY;usr=user_ID@$PROXY

- Speed Dial

Example: fnc=sd;usr=user_ID@$PROXY

- BLF with Speed Dial

Example: fnc=blf+sd;sub=BLF_List_URI@$PROXY;usr=user_ID@$PROXY

- BLF with Speed Dial and Call Pickup

Example: fnc=blf+sd+cp;sub=BLF_List_URI@$PROXY;usr=user_ID@$PROXY

Note

If the Direct PLK Configuration feature is disabled, the parameter is available only when the Extension of the line key is set to Disabled.

Perform one of the following:

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

<Extended_Function_1_>fnc=sd;ext=user_ID@$PROXY</Extended_Function_1_>

- On the phone web interface, specify the functions that apply to the line key.

Default: Empty

Configure Speed Dial on a Line Key

You can configure speed dial on an idle line of a user phone. The user can then use that line key to speed-dial a number. When you enable the speed dial on the line key, the user sees the speed-dial icon, a name, and an extension number for the speed dial line key. The user presses the line key to dial the assigned extension.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key on which to configure speed-dial.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to configure speed dial on the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Extension_2_ ua="na">Disabled</Extension_2_>

Step 4

In the Extended Function parameter, enter a string in this format:

fnc=sd;ext=9999@$PROXY;nme=xxxx

If you configure a phone with alphanumeric dialing feature in which the phone can place a call with alphanumeric characters instead of the traditional digits, you can enter a string in this format:

fnc=sd;ext=xxxx.yyyy@$PROXY;vid=n;nme=xxxx

where:

- fnc= sd means function=speed dial

- ext= 9999 is the phone that the line key calls. Replace 9999 with appropriate phone number.

ext= xxxx.yyyy is the phone that the line key calls. Replace xxxx.yyyy with alphanumeric characters. You can use these characters for alphanumeric dialing: a-z, A-Z, 0-9, -, _, ., and +.

- vid=n is the line index of the phone.

- nme= XXXX is the name displayed on the phone for the speed-dial line key. Replace XXXX with a name.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Extended_Function_2_ ua="na">fnc=sd;ext=9999@$PROXY;nme=xxxx</Extended_Function_2_>

Step 5

Click Submit All Changes.

Assign a Speed Dial Number

You can configure speed dials on the phone with the web interface. The user can see the configured speed dials on the phone and can use the speed dial number to call the corresponding contact.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

In the Speed Dial section, enter a name in Speed Dial (n) Name and the number in Speed Dial (n) Number that corresponds to the speed dial entry.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. The speed dial parameters are line-specific. Enter a string in the format

<Speed_Dial_1_Name ua="rw">John Wood</Speed_Dial_1_Name>
<Speed_Dial_1_Number ua="rw">12345678</Speed_Dial_1_Number>

Step 3

Click Submit All Changes.

Related Tasks

Access the Phone Web Interface

DTMF Wait and Pause Parameters

Speed dial, directory, extended function, and other strings configured in the phone can include wait (X) and pause (,) characters. These characters that allow manual and automatic DTMF (Dual-Tone Multi-Frequency) signal transmission.

You can add the wait and pause character with speed-dial, extended function, or directory strings in the format:

{Dial_String}[ ][,|X][DTMF_string][,|X][DTMF_string]
where:

- Dial_String —is the number that the user is trying to reach. For example, 8537777 or 14088537777.

- [ ](space)—is a dial termination character that defines or delimits the end of the dial string. The space is mandatory. If the phone encounters an X or a comma (,) before the space, the characters are treated as part of dial string.

- , (comma)—is a 2-second pause that is inserted for each comma in the string.

- X (wait)—indicates that the phone is waits for user input and acknowledgement.

When the user manually enters the DTMF signal with the key pad, the user sees a message to acknowledge that the transmission of the manual entry is complete. On confirmation, the phone sends any DTMF signals defined by the DTMF_string. The phone executes the next parameter. If there are no more parameters in the dial string to execute, the phone exits to the main screen.

The wait prompt window does not disappear until the user confirms the wait prompt or the call is ended either by the user or ended by the remote device.

- DTMF_string—is the DTMF signals that a user sends to a remote device after the call is connected. The phone cannot send signals other than valid DTMF signals.

Example:

18887225555,,5552X2222

A speed dial entry triggers the phone to dial 18887225555. The space indicates the end of the dial string. The phone waits 4 seconds (2 commas), and then sends the DTMF signals 5552.

A message is displayed, prompting the user to manually enter digits. When the user finishes dialing the digits, the user presses OK to confirm the manual input is complete. The phone sends the DTMF signals 2222.

Usage Guidelines

A user can transmit digits any time, as long as the call is connected.

The maximum length of the string, including the Xs or commas (,), is limited to the length of a speed-dial entry, dial screen entry, directory entry, and other dialed strings.

When a wait is initiated, the phone displays the home screen and prompts the user to input more digits with the key pad. If this action occurs while the user is editing an entry, the edits might be lost.

If only the first part of a dial string matches a dial plan when the call is dialed, the portion of the dial string that does not match the dial string is ignored. For example:

85377776666,,1,23

If 8537777 matches a dial plan, the characters 6666 are ignored. The phone waits 4 seconds before sending DTMF 1. It then wait 2 seconds and sends DTMF 23.

When logging the call, the phone only logs the dial string; the DTMF strings are not logged.

Valid DTMF signals are 0-9, *, or #. All other characters are ignored.

Limitations

When the call is connected and immediately transferred, the phone might not be able to process the DTMF signals. This depends on the length of time that the call is connected before it is transferred.

Phone Configuration for Monitoring Other Phones

You can configure the phone to monitor the status of lines on other phones. This feature is useful if users routinely handle calls for colleagues and need to see if they are available to answer calls. The phone monitors each line on a separate line key. The monitoring line keys function as Busy Lamp Field (BLF) keys. A BLF is an LED that changes color to indicate the status of the monitored line:

BLF Key LED Status

LED Color

Meaning

Green

The monitored line is available.

Red

The monitored line is busy.

Blinking red

The monitored line is ringing.

Amber

Error in BLF key configuration.

If the phone is registered to a BroadSoft server, you can set up the phone to monitor multiple users, with a single set of configurations.

- Configure the Phone to Monitor Multiple Users' Lines

- Configure a Line Key on the Phone to Monitor a Single User's Line

Configure the Phone to Monitor Multiple Users' Lines

If the phone is registered to a BroadSoft 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) code. To configure each parameter, see the syntax of the string in Parameters for Monitoring Multiple Users' Lines.

Before you begin

- Ensure that the phone is registered to a BroadSoft server.

- You set up a BLF list for a user of the phone on the BroadSoft server.

- Access the administration web interface. See Access the Phone Web Interface.

- Ensure that the monitored lines on the BLF keys are not in the Inert mode.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Configure BLF List URI, Use Line Keys For BLF List, BLF List, and BLF Label Display Mode as described in Parameters for Monitoring Multiple Users' Lines.

If you allow users to configure individual BLF keys (see Enable the User to Configure Features on Line Keys), we recommend setting BLF List to Hide.

Step 3

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 Voice > Att Console 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.

Parameters for Monitoring Multiple Users' Lines

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 BroadSoft server.

This field is only applicable if the phone is registered to a BroadSoft server. The BLF list is the list of users whose lines the phone is allowed to monitor. See Phone Configuration for Monitoring Other Phones for details.

The BLF List URI must be specified in the format <URI_name>@<server>. The BLF List URI specified must be the same as the value configured for the List URI: sip parameter on the BroadSoft server.

Perform one of the following:

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

<BLF_List_URI ua="na">MonitoredUsersList@sipurash22.com</BLF_List_URI>

- On the phone web interface, specify the BLF list that is defined on the BroadSoft server.

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.

When set to No, the phone uses only the Key Expansion Module keys to monitor the BLF list.

This setting only has significance when BLF List is set to Show.

Perform one of the following:

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

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

- On the phone web interface, set this field to Yes to use the unregistered line keys to monitor the BLF list entries. Set it to No to prevent the line keys from being used for monitoring the BLF list entries.

Default: No

BLF List

Activates or deactivates monitoring of the BLF list.

When set to Show, the phone assigns available line keys or the Key Expansion Module 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:

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

<BLF_List ua="rw">Show</BLF_List>

- On the phone web interface, set this field to Show or Hide to activate or deactivate the BLF monitoring feature.

Allowed values: Show|Hide

Default: Show

BLF Label Display Mode

Specifies how the BLF entries are displayed on the line keys or on the Key Expansion Module keys . The options are: Name, Ext (extention number), and Both.

Perform one of the following:

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

<BLF_Label_Display_Mode ua="na">Name</BLF_Label_Display_Mode>

- On the phone web interface, select an option from the list.

Allowed values: Name|Ext|Both

Default: Name

Configure a Line Key on the Phone to Monitor a Single User's Line

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 alone, busy lamp field and speed dial, busy lamp field and call pickup, or busy lamp field, speed dial, and call pickup can all be configured to work together. But speed dial alone requires a different configuration.

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 Monitoring a Single Line.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Ensure that the line key on which to configure a busy lamp field is not in the Inert mode.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key on which to configure a busy lamp field.

Step 3

Configure the Extension, Extended Function, fields as definined in Parameters for Monitoring a Single Line.

Step 4

Click Submit All Changes.

- Parameters for Monitoring a Single Line

Parameters for Monitoring a Single Line

The following table defines the function and usage of the Busy Lamp Field (BLF) parameters in the Line Key (n) sections under the Voice > Phone 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.

Parameters for Monitoring a Single Line

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. You can assign the line key with extended functions, for example, speed dial, Busy Lamp Field, and call pickup.

By default, you don't need to disable the extension to assign the line key with extended functions. However, if the Direct PLK Configuration feature is disabled, you must disable the extension to achieve the assignment. For more information about how to enable the feature, see Enable Direct PLK Configuration.

Perform one of the following:

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

<Extension_1_ ua="na">1</Extension_1_>

<Extension_2_ ua="na">Disabled</Extension_2_>

<Extension_3_ ua="na">Disabled</Extension_3_>

<Extension_4_ ua="na">Disabled</Extension_4_>

- On the phone web interface, set the parameter to Disabled to monitor another line on the line key.

Allowed values: Disabled|1|2|3|4, the allowed values vary with phones.

Default: n, where n is the line key number.

Extended Function

Used to assign extended functions to a line key on the phone. The supported functions are:

- Busy Lamp Field (BLF)

Example: fnc=blf;sub=BLF_List_URI@$PROXY;ext=user_ID@$PROXY

- BLF with Call Pickup

Example: fnc=blf+cp;sub=BLF_List_URI@$PROXY;ext=user_ID@$PROXY

- BLF with Speed Dial

Example: fnc=blf+sd;sub=BLF_List_URI@$PROXY;ext=user_ID@$PROXY

- BLF with Speed Dial and Call Pickup

Example: fnc=blf+sd+cp;sub=BLF_List_URI@$PROXY;ext=user_ID@$PROXY

Note

If the Direct PLK Configuration feature is disabled, the parameter is available only when the Extension of the line key is set to Disabled.

Perform one of the following:

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

<Extended_Function_1_>fnc=blf;sub=BLF_List_URI@$PROXY;ext=user_ID@$PROXY</Extended_Function_1_>

- In the phone web interface, configure the parameter with a valid syntax to enable monitoring another user or extension using the line key.

Default: Empty

Enable Conference Button with a Star Code

You can add a star code to the Conference button so that your user can press the button only once to add many active calls to a conference. You can enable this feature from the phone web page.

Before you begin

- The phone server must suppport this feature.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the Call Features Settings section, configure the Conference Single Hardkey and Conference Bridge URL fields as definined in Conference Button Parameters.

You can also enable the conference button with a xml file. Enter a string in this format:

<Conference_Bridge_URL_1_ ua="na">*55</Conference_Bridge_URL_1_>

<Conference_Single_Hardkey_1_ ua="na">Yes</Conference_Single_Hardkey_1_>

Step 3

Click Submit All Changes.

- Conference Button Parameters

Conference Button Parameters

The following table defines the function and usage of the conference button parameters in the Call Features Settings section under the Voice > Ext (n) 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.

Conference Button Parameters

Parameter

Description and default value

Conference Single Hardkey

You can use this field to specify whether to use only the Conferenc button on the key to initiate a conference call. When set to Yes, the user can use only the Conference button to initiate a conference call. The Conf softkey is deactivated. When set to No, the user can use both the Conference button and the Conf softkey.

Perform one of the following:

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

<Conference_Single_Hardkey_1_ ua="na">Yes</Conference_Single_Hardkey_1_>

- In the phone web interface, set this field to Yes or No to enable or disable this feature.

Allowed values: Yes|No

Default: No

Conference Bridge URL

URL used to join a conference call, generally in the form of a dialable number or a URI in this format user@IPaddress:port.

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

<Conference_Bridge_URL_1_ ua="na">*55</Conference_Bridge_URL_1_>

- in the phone web interface, specify the URI or a number as the conference bridge.

Default: Empty

Configure Alphanumeric Dialing

You can configure a phone so that the user of the phone can make a call by dialing alphanumeric characters instead of dialing only digits. In the phone web page, you can configure alphanumeric dialing with speed-dial, blf, and call pickup.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext (n).

Step 2

In the Dial Plan section, set Enable URI Dialing to Yes to enable alphanumeric dialing.

You can also configure the parameter in the configuration file (cfg.xml). The parameter is line-specific.

<Enable_URI_Dialing_1_ ua="na">Yes</Enable_URI_Dialing_1_>

Step 3

Select Voice > Phone, you can add a string on a line key in this format to enable speed dial with alphanumeric dialing capability:

fnc=sd;ext=xxxx.yyyy@$PROXY;nme=yyyy,xxxx

For example:

fnc=sd;ext=first.last@$PROXY;nme=Last,First

The above example will enable the user to dial "first.last" to make a call.

Note

The supported characters that you can use for alphanumeric dialing are a-z, A-Z, 0-9, -, _, ., and +.

Step 4

Click Submit All Changes.

Call Park

With call park, a call can be parked and then retrieved either from your phone or from another phone. The following LED colors display on the line key when this feature is configured:

- Green LED—Call park is successfully configured.

- Amber LED—Call park is not configured.

- Red slow blinking LED—A call is parked.

- Configure Call Park with Star Codes

- Configure One-Button Call Park

- Add Call Park to a Line Key

Configure Call Park with Star Codes

You can configure call park so that the user can put a call on hold and then retrieve the call from either the user's phone or another phone.

When configuring call park, the Call Park Code and the Call Unpark Code must match the Feature Access Code configured on the server.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

Enter *68 in the Call Park Code field.

You can also configure the parameter in the configuration file (cfg.xml) by entering a string in the following format:

<Call_Park_Code ua="na">*68</Call_Park_Code>

Step 3

Enter *88 in the Call Unpark Code field.

You can also configure the parameter in the configuration file (cfg.xml) by entering a string in the following format:

<Call_Unpark_Code ua="na">*88</Call_Unpark_Code>

Step 4

Click Submit All Changes.

Configure One-Button Call Park

With One-Button Call Park, there is no need to enter a combination of key strokes for parking and unparking a call.

You can also configure parking calls in a dedicated call park extension

Before you begin

Access the Phone Web Interface, and Configure the Phone to Monitor Multiple Users' Lines

Procedure

Step 1

Go to Voice > Att Console.

Step 2

In the BLF List URI field, enter uri_name@server.

The BLF List URI field must have the same value as that configured for the List URI:sip parameter on the BroadSoft server.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<BLF_List_URI ua="na">uri_name@server</BLF_List_URI>

Step 3

From the BLF List Feature Options drop down list, select prk.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<BLF_List_Feature_Options ua="na">prk</BLF_List_Feature_Options>

The allowed values are blf+sd+cp|prk. The default value is blf+sd+cp.

Example:

- When the BLF List Feature Options field is set to blf+sd+cp, the auto-assigned keys can be used for blf+sd+cp function or a call park function. If type="park" is included in the resource element, then the auto-assigned keys are used for call park function, else it is used for blf+sd+cp function when the type is not included.

In the server, there are three resource elements in NOTIFY requests. Enter the strings in the resource elements in the format and add new attribute type="park" in the call park extension resource URI:

<resource uri="sip:test01@as1bsoft.sipurash.com"><name>test 01</name><instance id="cROdMctbQE" state="active" cid="BQQ7uR@broadworks"/></resource>

<resource uri="sip:2345@as1bsoft.sipurash.com" type="park"><name>Park Location1</name><instance id="cROdMctbQE" state="active" cid="BQQ8uR@broadworks"/></resource>

<resource uri="sip:test02@as1bsoft.sipurash.com"><name>test 02</name><instance id="cROdMctbQE" state="active" cid="BQQ9uR@broadworks"/></resource>

After successful configuration, the phone monitors test 01, Park Location1, and test 02. If call park extensions and users are subscribed successfully, linekeys will be assigned for callpark or blf function respectively, and get the monitored extension and user status.

- When the BLF List Feature Options field is set to prk, the BLF List URI monitors call park extensions and users.

In the server, there are three resource elements in NOTIFY requests. Enter the strings in the resource elements in the format and add new attribute type="park" in the call park extension resource URI:

<resource uri="sip:test01@as1bsoft.sipurash.com"><name>test 01</name><instance id="cROdMctbQE" state="active" cid="BQQ7uR@broadworks"/></resource>

<resource uri="sip:2345@as1bsoft.sipurash.com" type="park"><name>Park Location1</name><instance id="cROdMctbQE" state="active" cid="BQQ8uR@broadworks"/></resource>

<resource uri="sip:test02@as1bsoft.sipurash.com"><name>test 02</name><instance id="cROdMctbQE" state="active" cid="BQQ9uR@broadworks"/></resource>

After successful configuration, the phone monitors test 01, Park Location1, and test 02. If call park extensions and users are subscribed successfully, linekeys will be assigned for call park function and get the monitored extension and user status.

Step 4

Click Submit All Changes.

Add Call Park to a Line Key

You can add call park to a line key to enable the user to temporarily store and retrieve calls. Call park is supported on private lines and shared lines.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to add call park to the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Extension_2_ ua="na">Disabled</Extension_2_>

Step 4

In the Extended Function parameter, enter a string in this format:

For a private line, enter

fnc=prk;sub=$USER@$PROXY;nme=CallPark-Slot1;vid=1

For a shared line, enter

fnc=prk;sub=$USER@$PROXY;nme=CallPark-Slot1;orbit=<DN of shared line>;vid=1

where:

- fnc=prk means function=call park

- sub is the SIP URI of the monitored park slot.

- nme is the name displayed on the phone for the call park line key.

- orbit is the DN of the shared line.

- vid is the extension ID. The values of $USER and $PROXY are retrieved from the specified extension. If vid is missing in the function string, the values of $USER and $PROXY are retrieved from extension 1.

You can also configure the line-specific parameter in the configuration file (cfg.xml). Enter a string in the following format:

<Extended_Function_2_ ua="na">fnc=prk;sub=$USER@$PROXY;nme=CallPark-Slot1;vid=1;</Extended_Function_2_>

Step 5

Click Submit All Changes.

Set the Optional Network Configuration

Optional network servers provide resources such as DNS lookup, network time, logging, and device discovery. It also enables you to add PC port mirroring on the user phone. Your user can also enable or disable this service from the phone.

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 Optional Network Configuration.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the Optional Network Configuration section, set up the fields as described in Parameters for Optional Network Configuration.

Step 3

Click Submit All Changes.

- Parameters for Optional Network Configuration

Parameters for Optional Network Configuration

The following table defines the function and usage of the access control parameters in the Optional Network Configuration section under the Voice > System 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.

Parameters for Optional Network Configuration

Parameter

Description and Default Value

Host Name

The hostname of the server that the phone uses.

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

<Host_Name ua="rw">serverhost.com</Host_Name>

- On the phone web interface, enter the host name of the server to use.

Default: Empty

Domain

The network domain of the Phone.

If you’re using LDAP, see LDAP Configuration.

Perform one of the following:

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

<Domain ua="rw">domainexample.com</Domain>

- In the phone web interface, enter the domain of the phone.

Default: Empty

DNS Server Order

Specifies the sequence for selecting the DNS server.

Perform one of the following:

- Manual, DHCP

- Manual

- DHCP, Manual

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

<DNS_Server_Order ua="na">Manual,DHCP</DNS_Server_Order>

- In the phone web interface, specify the order that the phone follows to select the DNS server.

Allowed values: Manual,DHCP|Manual|DHCP,Manual

Default: Manual, DHCP

DNS Query Mode

Specifies the mode of DNS query.

Perform one of the following:

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

<DNS_Query_Mode ua="na">Parallel</DNS_Query_Mode>

- In the phone web interface, select the mode of DNS query.

Allowed values: Parallel|Sequential

Default: Parallel

DNS Caching Enable

Enables or disables DNS caching. When enabled, the DNS query results are cached. The phone retrieves the local DNS cache until the local cache is expired. When disabled, the phone always performs DNS queries.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No to enable or disable DNS caching.

Allowed values: Yes|No

Default: Yes

Switch Port Config

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

- Auto

- 10 HALF

- 10 FULL

- 100 HALF

- 100 FULL

Perform one of the following:

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

<Switch_Port_Config ua="na">AUTO</Switch_Port_Config>

- On the phone web interface, select the speed for the port or select Auto to allow the system to select the speed.

Default: Auto

PC Port Config

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

- Auto

- 10 HALF

- 10 FULL

- 100 HALF

- 100 FULL

Perform one of the following:

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

<PC_Port_Config ua="na">AUTO</PC_Port_Config>

- In the phone web interface, select the speed for the port or select Auto to allow the system to select the speed.

Default: Auto

PC PORT Enable

Enables or disables the PC port on the phone.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No to enable or disable the PC port on the phone.

Allowed values: Yes|No

Default: Yes

Enable PC Port Mirror

Enables or disables PC Port mirroring on the phone. When set to Yes, you can see the packets on the phone.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No to enable or disable PC port mirroring on the phone.

Allowed values: Yes|No

Default: No

Syslog Server

See System Log Parameters.

Syslog identifier

See System Log Parameters.

Primary NTP Server

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

You can set primary NTP server for both IPv4 and IPv6.

Perform one of the following:

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

<Primary_NTP_Server ua="rw">192.168.1.10</Primary_NTP_Server>

- In the phone web interface, specify the IP address or host name of the NTP server.

Default: Blank

Secondary NTP Server

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

You can set primary NTP server for both IPv4 and IPv6.

Perform one of the following:

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

<Secondary_NTP_Server ua="rw">192.168.1.11</Secondary_NTP_Server>

- In the phone web interface, specify the IP address or host name of the NTP server.

Default: Blank

Use Config TOS

This field controls whether the phone uses the Time of Service (TOS) parameters on the Ext (n) tab. Set this field to Yes when you want the phones to use the TOS configuration specified on the Ext (n) tab. Otherwise, set this field to No.

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

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

- In the phone web interface, select Yes or No as needed.

Allowed values: Yes|No

Default: No

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 is 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 Multiplatform Phones, located here:

https://www.cisco.com/c/en/us/support/collaboration-endpoints/ip-phone-6800-series-multiplatform-firmware/products-programming-reference-guides-list.html

- XML Directory Service

- Configure a Phone to Connect to an XML Application

- Macro Variables

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.

Configure a Phone to Connect to an XML Application

You can also configure the parameters in the configuration file (cfg.xml) as defined in Parameters for XML Applications.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the XML Service section, configure the XML Application Service Name and XML Application Service URL fields as defined in Parameters for XML Applications.

Step 3

(Optional) Specify the username and password to authenticate XML service in the XML User Name and XML Password fields as defined in Parameters for XML Applications.

Step 4

(Optional) Enable and configure authentication for CGI/Execute URL via Post from an external application (for example, a web application) to the phones.

Configure the CISCO XML EXE Enable and CISCO XML EXE Auth Mode fields as defined in Parameters for XML Applications.

Step 5

Click Submit All Changes.

- Parameters for XML Applications

Parameters for XML Applications

The following table defines the function and usage of the XML application parameters in the XML Service section under the Voice > Phone 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.

Parameters for XML Applications

Parameter

Description

XML Application Service Name

Name of the XML application. The name displays on the phone as a web application choice.

Perform one of the following:

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

<XML_Application_Service_Name ua="na">XML_APP</XML_Application_Service_Name>

- In the phone web interface, enter a name for the XML application.

Default: Empty

XML Application Service URL

The URL where the XML application is located.

Macro variables are supported in XML URLs. For the valid macro variables, see Macro Variables.

Perform one of the following:

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

<XML_Application_Service_URL ua="na">XML_APP</XML_Application_Service_URL>

- In the phone web interface, enter the URL for the XML application.

Default: Empty

XML User Name

XML service username for authentication purposes.

Perform one of the following:

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

<XML_User_Name ua="na">username</XML_User_Name>

- In the phone web interface, enter the usename used for authenticating XML service.

Default: Empty

XML Password

XML service password for the specified XML User Name. The password you entered in this field shows in the configuration file (cfg.xml) as

<!--
<XML_Password ua="na">*************</XML_Password>
-->

Default: Empty

CISCO XML EXE Enable

Specifies whether authentication is required to access the XML application server.

Perform one of the following:

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

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

- In the phone web interface, set it to Yes or No to enable or disable authentication.

Allowed values: No

Default: No

CISCO XML EXE Auth Mode

Specifies the authentication mode for Cisco XML EXE. The available options are:

- Trusted—No authentication is performed regardless of the local credential.

- Local Credential—Authentication is based on the digest authentication using the local credential, if set. If the local credential is not set, then no authentication is performed.

- Remote Credential—Authentication is based on the digest authentication using the remote credential as set in the XML application on the web page (to access an XML application server).

Perform one of the following:

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

<CISCO_XML_EXE_Auth_Mode ua="na">Local Credential</CISCO_XML_EXE_Auth_Mode>

- In the phone web interface, select an authentication mode.

Allowed values: Trusted|Local Credential|Remote Credential

Default: Local Credential

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 lowercase hex digits—MA

- Product Name—PN

- Product Series Number—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.

Note

$SA through $SD are recognized as arguments to the optional resync URL qualifier, --key.

MA

MAC address using lowercase hex digits (000e08aabbcc).

MAU

MAC address using uppercase hex digits (000E08AABBCC).

MAC

MAC address using lowercase hex digits with a colon to separate hex digit pairs (00:0e:08:aa:bb:cc).

PN

Product Name; for example, IP Phone 6841.

PSN

Product Series Number; for example, 6841.

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. Use the software version string to compare against the current phone's firmware load.

Follow the format below:

- For Firmware Release 11.3(1)SR1 and previous:

sipyyyy.11-0-1MPP-376

where yyyy indicates the phone model or phone series; 11 is the major version; 0 is the minor version; 1MPP is the micro version; and 376 is the build number.

- For Firmware Release 11.3(2) and later:

sipyyyy.11-3-2MPP0001-609

where yyyy indicates the phone model or phone series; 11 is the major version; 3 is the minor version; 2MPP0001 is the micro version; and 609 is the build number.

There are two methods to compare firmware loads:

- With quotes, "$SWVER"–Variable acts as a string in firmware load name comparisons. For "$SWVER" eq "sipyyyy.11-2-1MPP-312.loads" or "$SWVER" eq "sipyyyy.11-3-2MPP0001-609.loads", the phone model number and the version numbers in the load name are part of the comparison.

- Without quotes, $SWVER–Variable is parsed to determine a build number, plus major, minor, and micro revision numbers. For example, when the sip88xx.11-3-2MPP0001-598.loads and sip8845_65.11-3-2MMP0001-598.loads firmware names are parsed, the result ignores the model number and load number. The result for both firmware names yields a major revision=11, minor revision=3, micro revision=2MPP0001, and build number=598.

See more information about firmware version comparison, see Macro Expansion Variables.

HWVER

Hardware version string; for example, 1.88.1.

PRVST

Provisioning State (a numeric string):

- -1 = explicit resync request

- 0 = power-up resync

- 1 = periodic resync

- 2 = resync failed, retry attempted

UPGST

Upgrade State (a numeric string):

- 1 = first upgrade attempt

- 2 = upgrade failed, retry attempt

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 hostname.

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.

Note

Customization status viewable on Web UI Info page.

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 is provided.

DISPLAYNAMEn

The contents of the Line N Display Name configuration parameter.

AUTHIDn

The contents of the Line N auth ID configuration parameter.

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 various 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 shared with the line can resume the call by pressing or the Resume softkey.

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

- Add Dialog-Based Shared Line Appearance

Configure a Shared Line

You can create a shared line by assigning the same directory number to more than one phone on the phone web page.

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 Configuring a Shared Line.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where (n) is the number of an extension to share.

Step 2

In the General section, set the Line Enable parameter as described in the Parameters for Configuring a Shared Line table.

Step 3

In the Share Line Appearance section, set Share Ext, Shared User ID field, Subscription Expires, and Restrict MWI parameters as described in the Parameters for Configuring a Shared Line table.

Step 4

In the Proxy and Registration section, enter the IP address of the proxy server in the Proxy field.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Proxy_1_ ua="na">as1bsoft.sipurash.com</Proxy_1_>

Example for proxy server address: as1bsoft.sipurash.com

Step 5

In the Subscriber Information section, enter the Display Name and User ID (extension number) for the shared extension.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Display_Name_1_ ua="na">name</Display_Name_1_>
<User_ID_1_ ua="na">4085273251</User_ID_1_>

Step 6

In the Miscellaneous Line Key Settings section, set SCA Barge-In Enable parameter as described in the Parameters for Configuring a Shared Line table.

Step 7

Click Submit All Changes.

- Parameters for Configuring a Shared Line

Parameters for Configuring a Shared Line

The following table describes the parameters in the Voice > Ext(n) tab of the phone web page.

The following table defines the function and usage of Shared Line parameters in the General and Share Line Appearance sections under the Ext(n) 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.

Parameters for Shared Lines

Parameter

Description

Line Enable

Enables a line for the service.

Perform one of the following:

- In the phone web interface, select yes to enable. Otherwise, select No.

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

<Line_Enable_1_ ua="na">Yes</Line_Enable_1_>

Valid values: Yes|No

Default: Yes

Share Ext

Indicates whether other Cisco IP phones share this extension is, or the extension is private.

Perform one of the following:

- In the phone web interface, select yes to enable. Otherwise, select No.

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

<Share_Ext_1_ ua="na">No</Share_Ext_1_>

If you set Share Ext to No, this extension is private and doesn't share calls, regardless of the Share Line Appearance setting. If you set this extension to Yes, calls follow the Share Line Appearance setting.

Valid values: Yes|No

Default: Yes

Shared User ID

The user identified assigned to the shared line appearance.

Perform one of the following:

- In the phone web interface, enter the user ID.

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

<Shared_User_ID_1_ ua="na">Shared UserID</Shared_User_ID_1>

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.

Perform one of the following:

- In the phone web interface, enter the value in seconds.

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

<Subscription_Expires_1_ ua="na">3600</Subscription_Expires_1_>

Valid values: An integer from 10 through 65535

Default: 3600 seconds

Restrict MWI (Message Waiting Indicator)

Indicates the message waiting indicator lights only for messages on private.

Perform one of the following:

- In the phone web interface, select Yes to enable. When enabled the message waiting indicator lights only for messages on private. Otherwise, select No.

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

<Restrict_MWI_1_ ua="na">No</Restrict_MWI_1_>

Valid values: Yes|No

Default: No

The following table describes the parameters in the Voice > Phone tab of the phone web page.

Miscellaneous Line Key Settings

Parameter

Description

SCA Barge-In Enable

Enables the SCA Barge-In.

Perform one of the following:

- In the phone web interface, select Yes to enable. Otherwise, select No.

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

<SCA_Barge-In-Enable ua="na">No</SCA_Barge-In-Enable>

Valid values: Yes|No

Default: No

Add Dialog-Based Shared Line Appearance

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

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Parameters section, set the Share Line Event Package Type parameter to Dialog to subscribe the phone to the dialog event package.

You can also set the parameter to Call-Info and the phone retains the legacy behavior.

Default value: Call-Info

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

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

Step 3

Click Submit All Changes.

Assign a Ringtone to an Extension

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 Ringtone.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where (n) is the number of a phone extension.

Step 2

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:

<Default_Ring_3_ ua="rw">1</Default_Ring_3_>

Step 3

Select Voice > Phone.

Step 4

In the Ringtone section, set the Ring(n) and Silent Ring Duration parameters as described in the Parameters for Ringtone table.

Step 5

Click Submit All Changes.

- Parameters for Ringtone

- Add Distinctive Ringtone

- Restrict User from Controlling the Ringer Volume

Parameters for Ringtone

The following table describes the parameters for Ringtone.

Parameters for Ringtone

Parameter

Description

Ring1 to Ring12

Ring tone scripts for various ringtones.

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

<!-- Ringtone -->
<Ring1 ua="na">n=Sunrise;w=file://Sunrise.rwb;c=1</Ring1>
<Ring2 ua="na">n=Chirp 1;w=file://chirp1.raw;c=1</Ring2>
<Ring3 ua="na">n=Chirp 2;w=file://chirp2.raw;c=1</Ring3>
<Ring4 ua="na">n=Delight;w=file://Delight.rwb;c=1</Ring4>
<Ring5 ua="na">n=Evolve;w=file://Evolve.rwb;c=1</Ring5>
<Ring6 ua="na">n=Mellow;w=file://Mellow.rwb;c=1</Ring6>
<Ring7 ua="na">n=Mischief;w=file://Mischief.rwb;c=1</Ring7>
<Ring8 ua="na">n=Reflections;w=file://Reflections.rwb;c=1</Ring8>
<Ring9 ua="na">n=Ringer;w=file://Ringer.rwb;c=1</Ring9>
<Ring10 ua="na">n=Ascent;w=file://Ascent.rwb;c=1</Ring10>
<Ring11 ua="na">n=Are you there;w=file://AreYouThereF.raw;c=1</Ring11>
<Ring12 ua="na">n=Chime;w=file://Chime.raw;c=1</Ring12>
<Silent_Ring_Duration ua="na">60</Silent_Ring_Duration>

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: <Ring1 ua="na">n=Sunrise;w=file://Sunrise.rwb;c=1</Ring1>

<Silent_Ring_Duration ua="na">60</Silent_Ring_Duration>

Add Distinctive Ringtone

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.

Procedure

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:

- 1 = Classic phone with mechanical bell

- 2 = Typical phone ring

- 3 = Classic ring tone

- 4 = Wide-band frequency sweep signal

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:

<!-- Ringtone -->
<Ring1 ua="na">n=Sunrise;w=file://Sunrise.rwb;c=1</Ring1>
<Ring2 ua="na">n=Chirp 1;w=file://chirp1.raw;c=1</Ring2>
<Ring3 ua="na">n=Chirp 2;w=file://chirp2.raw;c=1</Ring3>
<Ring4 ua="na">n=Delight;w=file://Delight.rwb;c=1</Ring4>
<Ring5 ua="na">n=Evolve;w=file://Evolve.rwb;c=1</Ring5>
<Ring6 ua="na">n=Mellow;w=file://Mellow.rwb;c=1</Ring6>
<Ring7 ua="na">n=Mischief;w=file://Mischief.rwb;c=1</Ring7>
<Ring8 ua="na">n=Reflections;w=file://Reflections.rwb;c=1</Ring8>
<Ring9 ua="na">n=Ringer;w=file://Ringer.rwb;c=1</Ring9>
<Ring10 ua="na">n=Ascent;w=file://Ascent.rwb;c=1</Ring10>
<Ring11 ua="na">n=Are you there;w=file://AreYouThereF.raw;c=1</Ring11>
<Ring12 ua="na">n=Chime;w=file://Chime.raw;c=1</Ring12>
<Silent_Ring_Duration ua="na">60</Silent_Ring_Duration>

Restrict User from Controlling the Ringer Volume

Some users tend to lower the ringer volume when they don't want to answer a call. Hence, they miss important calls. To avoid such problem, you can disable the ability of the users to control the ringer volume.

Note

When you configure the restriction on the ringing volume control, this configuration doesn't restrict the user's ability of controlling other volumes, such as speaker volume and headset volume.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

In the Audio Volume section, set the Ringer Volume Control parameter to No.

The default value is Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Step 3

Select Submit All Changes

When you set the parameter to No you see the following results:

- The user presses the volume button on the phone, a message appears indicating that the user has no permission to change the ringer volume.

- On the phone administration web page, under the User Login > Advanced access level, the Ringer Volume parameter doesn't appear under the Audio Volume section. Hence, the user doesn't get any option to change the ringer volume.

Enable Hoteling on a Phone

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.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 2

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:

<Enable_Broadsoft_Hoteling_1_ua="na">Yes</Enable_Broadsoft_Hoteling_1>

Options: Yes and No

Default: No

Step 3

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:

<Hoteling_Subscription_Expires_1_ua="na">3600</Hoteling_Subscription_Expires_1>

Valid values: An integer from 10 through 86400

Default: 3600

Step 4

Click Submit All Changes.

Enable Flexible Seating on a Phone

With the Flexible Seating feature of BroadSoft, the phone downloads and is reconfigured with Flexible Seating Guest's device files when the guest is associated with the host. The phone is treated as an alternate device of the guest. The call originations from guest's primary device are also allowed. The guest’s primary device is also alerted on incoming calls to the guest. For more information, see the BroadSoft documentation.

In addition, with the feature enabled on the phone, the phone can cache the user credentials for the LDAP directory. If the cache contains the user credentials, the guest user can bypass the sign-in procedure to access the LDAP directory. The cache can store up to 50 user credentials. The phone removes the least-used credentials when the cache size limit is reached.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 2

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:

<Enable_Broadsoft_Hoteling_1_ua="na">Yes</Enable_Broadsoft_Hoteling_1>

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Enable Extension Mobility on a Phone

With the Extension Mobility (EM) feature enabled on the phone, any user can sign in to the phone other than their own in the same network. In this scenario, the phone can be shared with other users. After the users sign in, they can see their own line number displayed on the phone screen, and their contacts in the personal address directory.

In addition, the phone can cache the user credentials for the LDAP directory when the user signs into the phone with the feature. If the cache contains the user credentials, the user can bypass the sign-in procedure to access the LDAP directory. The cache can store up to 50 user credentials. The phone removes the least-used credentials when the cache size limit is reached.

You can also configure the parameters in the phone configuration file with XML (cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Extension Mobility section, set EM Enable to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: No

Step 3

Set the amount of time (in minutes) that the user can be signed in on the phone in Session Timer(m).

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Session_Timer_m_ ua="na">480</Session_Timer_m_>

Default: 480

Step 4

Click Submit All Changes.

Set the User Password

Configure a password so the phone is protected and secured. Both administrators and users can configure a password and control access to the phone.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

Under the section System Configuration, locate the parameter User Password, and click Change Password next to the parameter.

Step 3

Enter the current user password in the Old Password field.

If you don't have a password, keep the field empty.

Step 4

Enter a new password in the New Password field.

Step 5

Click Submit.

The message Password has been changed successfully. will display in the web page. The web page will refresh in several seconds.

After you set the user password, this parameter displays the following in the phone configuration XML file (cfg.xml):

<!--
<User_Password ua="rw">*************</User_Password>
-->

Download Problem Reporting Tool Logs

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 Configuration Utility page.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info > Debug Info > Device Logs.

Step 2

In the Problem Reports area, click the problem report file to download.

Step 3

Save the file to your local system and open the file to access the problem reporting logs.

Configure Problem Report Tool

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 Parameters for Configure Problem Report Tool table.

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.");
}

?>

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

In the Problem Report Tool section, set the fields as described in the Parameters for Configure Problem Report Tool table.

Step 3

Click Submit All Changes.

- Parameters for Configure Problem Report Tool

Parameters for Configure Problem Report Tool

The following table defines the function and usage of Configure Problem Report Tool parameters in the Problem Report Tool section under the Voice > Provisioning 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.

Parameters for Configure 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:

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

<PRT_Upload_Rule ua="na">https://proxy.example.com/prt_upload.php</PRT_Upload_Rule>

- In the phone web page, enter the path in the format:

https://proxy.example.com/prt_upload.php

or

http://proxy.example.com/prt_upload.php

Default: Empty

PRT Upload Method

Determines the method used to upload PRT logs to the remote server.

Perform one of the following:

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

<PRT_Upload_Method ua="na">POST</PRT_Upload_Method>

- In the phone web page, select POST or PUT methods to upload the logs to the remote server.

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:

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

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

- In the phone web page, enter the interval duration in minutes.

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:

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

<PRT_Name ua="na">prt-string1-$MACRO</PRT_Name>
Enter the name in the format:

prt-string1-$MACRO

- In the phone web page, enter the name in the format:

prt-string1-$MACRO

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:

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

<PRT_HTTP_Header ua="na">x-cisco-spark-canary-opts</PRT_HTTP_Header>

- In the phone web page, enter the HTTP header in the format:

x-cisco-spark-canary-opts

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:

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

<PRT_HTTP_Header_Value ua="na">always</PRT_HTTP_Header_Value>

- In the phone web page, enter the value in the format:

always

Valid value range: a-z, A-Z, 0-9, underscore (_), comma (,), semicolon (;), equal (=), and hyphen (-)

Note

Except for the underscore (_), the first character must not be a special character.

Default: Empty

Server-Configured Paging

You can configure a paging group on a server so that users can page a group of phones. For more details, refer to your server documentation.

Configure Multicast Paging

You can set up Multicast paging to allow users to page to 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.

You can add a phone to up to 10 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) is enabled, 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 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.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Go to the Multiple Paging Group Parameters section.

Step 3

Enter multicast paging scripts as defined in Parameters for Multiple Paging Group.

Step 4

Click Submit All Changes.

- Parameters for Multiple Paging Group

Parameters for Multiple Paging Group

The following table defines the function and usage of the multiple paging group parameters in the Voice > Phone 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.

Multiple Paging Group Parameters

Feature

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:

pggrp=<multicast-address>:<port>;<name=group_name>;<num=multicast_number>;
<listen=boolean_value>;<pri=priority_level>;<codec=codec_name>;

Example script:

pggrp=224.168.168.168:34560;name=GroupA;num=500;listen=yes;pri=1;codec=g711a;

- Multicast IP address (multicast-address) and port (port)—Enter the multicast IP address and the port specified on your paging server. The port number must be unique for each group and an even number within 1000 and 65534.

Make sure that you set the same multicast IP address and port for all the phones within a paging group. Otherwise, the phones can't receive paging.

- Paging group name (name)—Optionally enter the name of the paging group. The name helps you identify the paging group the phone is in when you have multiple paging groups.

- Multicast number (num)—Specify the number for the phone to listen for multicast paging and initiate a multicast paging session. Assign the same multicast number to all the phones within the group. The number must comply to the dial plan specified for the line to initiate a multicast.

- Listen status (listen)—Specify whether the phone listens for paging from this group. Set this parameter to yes to make the phone listen for the paging. Otherwise, set it to no, or don't include this parameter in the script.

- Priority (pri)—Specify priority between paging and phone call. If you don't specify the priority or don't include this parameter in the script, the phone uses priority 1. The four priority levels are:

- 0: Paging takes precedent over phone call. When the phone is on an active call, an incoming paging places the active call on hold. The call resumes when the paging ends.

- 1: When the phone receives an incoming paging on an active call, the user hears the mix of the paging and the call.

- 2: The user is alerted with the paging tone when receiving an incoming paging on an active line. The incoming paging isn't answered unless the active call is put on hold or ends.

- 3: The phone ignores the incoming paging without any alert when the phone is on an active call.

- Audio codec (codec)—Optionally specify the audio codec for the multicast paging to use. The supported codecs are G711a, G711u, G722, and G729. If you don't specify the codec or don't include the codec parameter in the script, the phone uses G711u codec.

Perform one of the following:

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

<Group_1_Paging_Script ua="na">pggrp=224.168.168.168:34560;name=Group_1;
num=800;listen=yes;pri=1;codec=g722</Group_1_Paging_Script>

- In the phone web interface, configure this field with a valid string.

Default: Empty

Configure a Phone to Accept Pages Automatically

The Single Paging or Intercom feature enables a user to directly contact another user by phone. If the phone of the person being paged has been configured to accept 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

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

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:

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

Options: Yes and No

Default: Yes

Step 3

Click Submit All Changes.

Manage Phones with TR-069

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.

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 Parameters for TR-069 Configuration table.

As a bidirectional SOAP/HTTP-based protocol, TR-069 provides the communication between CPEs and Auto Configuration Servers (ACS).

For TR-069 Enhancements, see TR-069 Parameter Comparison.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > TR-069.

Step 2

Set up the fields as described in Parameters for TR-069 Configuration table.

Step 3

Click Submit All Changes.

View TR-069 Status

When you enable TR-069 on a user phone, you can view status of TR-069 parameters on the phone web interface.

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 Parameters for TR-069 Configuration table.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Select Info > Status > TR-069 Status.

You can view status of TR-069 parameters in Parameters for TR-069 Configuration table.

- Parameters for TR-069 Configuration

Parameters for TR-069 Configuration

The following table defines the function and usage of Call Center Agent Setup parameters in the ACD Settings section under the Ext(n) 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.

Parameters for TR-069 Configuration

Parameter

Description

Enable TR-069

Settings that enables or disables the TR-069 function.

Perform one of the following:

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

<Enable_TR-069 ua="na">No</Enable_TR-069>

- In the phone web page, select Yes to enable this feature and select No to disable it.

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:

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

<ACS_URL ua="na">https://acs.url.com</ACS_URL>

- In the phone web page, enter a valid HTTP or HTTPS URL of the ACS.

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:

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

<ACS_Username ua="na">acs username</ACS_Username>

- In the phone web page, enter a valid username for HTTPS-based authentication of the CPE.

Default: admin

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:

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

<ACS_Password ua="na"/>

- In the phone web page, enter a valid password for HTTPS-based authentication of the CPE.

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:

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

<Connection_Request_Password ua="na"/>

- In the phone web page, enter a valid username that authenticates the ACS.

Connection Request Password

Password used to authenticate the ACS that makes a connection request to the CPE.

Perform one of the following:

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

<Connection_Request_Password ua="na"/>

- In the phone web page, enter a valid password that authenticates the ACS.

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:

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

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

- In the phone web page, enter a valid duration in seconds.

Default: 20

Periodic Inform Enable

Settings that enables or disables the CPE connection requests.

Perform one of the following:

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

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

- In the phone web page, select Yes to enable this feature and select No to disable it.

Valid values: Yes|No

Default: Yes

TR-069 Traceability

Settings that enables or disables TR-069 transaction logs.

Perform one of the following:

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

<TR-069_Traceability ua="na">Yes</TR-069_Traceability>

- In the phone web page, select Yes to enable this feature and select No to disable it.

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:

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

<CWMP_V1.2_Support ua="na">Yes</CWMP_V1.2_Support>

- In the phone web page, select Yes to enable this feature and select No to disable it.

Valid values: Yes|No

Default: Yes

TR-069 VoiceObject Init

Settings to modify voice objects.

Perform one of the following:

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

<TR-069_VoiceObject_Init ua="na">Yes</TR-069_VoiceObject_Init>

- In the phone web page, select Yes to initialize all voice objects to factory default values or select No to retain the current values.

Valid values: Yes|No

Default: Yes

TR-069 DHCPOption Init

Settings to modify DHCP settings.

Perform one of the following:

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

<TR-069_DHCPOption_Init ua="na">Yes</TR-069_DHCPOption_Init>

- In the phone web page, select Yes to initialize the DHCP settings from the ACS or select No to retain the current DHCP settings.

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:

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

<BACKUP_ACS_URL ua="na">https://acs.url.com</BACKUP_ACS_URL>

- In the phone web page, enter a valid URL that uses the CPE WAN Management Protocol.

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:

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

<BACKUP_ACS_User ua="na">backup username</BACKUP_ACS_User>

- In the phone web page, enter a valid username that authenticates the CPE to the ACS when ACS uses the CPE WAN Management Protocol.

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:

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

<BACKUP_ACS_Password ua="na"/>

- In the phone web page, enter a valid password that authenticates the CPE to the ACS when ACS uses the CPE WAN Management Protocol.

Default: Blank

Note

If you do not configure the above parameters, you can also fetch them through DHCP options 60,43, and 125.

Enable Electronic Hookswitch

The Electronic Hookswitch feature enables users to use headsets that electronically connect a wireless headset to a phone. Typically, the headset requires a base that plugs into the phone and communicates with the headset. Here are the supported headsets:

- Plantronics Savi 740

- Jabra PRO920

- Jabra PRO9400

- Sennheiser DW Pro1

Only the Cisco IP Phone 6851 Multiplatform Phones and Cisco IP Phone 6861 Multiplatform Phones support the Electronic Hookswitch feature.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

In the Audio Volume section, set the Electronic HookSwitch Control parameter to Yes to enable the feature.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Set up a Secure Extension

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

- Make sure that Secure Call Serv is enabled (set to Yes) in the Supplementary Services area on the Voice > Phone tab.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

- 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.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

In the Call Feature Settings section, in the Secure Call Option field, choose 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

- Optional－Retains the current secure call option for the phone.

- Required－Rejects nonsecure calls from other phones.

- Strict－Allows SRTP only when SIP transport is set to TLS. Allows RTP only when SIP transport is UDP/TCP.

Default: Optional

Step 3

Click Submit All Changes.

Configure the SIP Transport

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.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

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:

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

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

Step 3

Click Submit All Changes.

Block Non-Proxy SIP Messages to a Phone

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

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the System Configuration section, set the Block Nonproxy SIP paramter 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:

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

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Configure a Privacy Header

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.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Extension.

Step 2

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:

<Privacy_Header_2_ ua="na">header</Privacy_Header_2_>

Options:

- Disabled (default)

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

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

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

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

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

Default: Disabled

Step 3

Click Submit All Changes.

Enable P-Early-Media Support

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.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext (n).

Step 2

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:

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

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Enable Peer Firmware Sharing

Peer Firmware Sharing (PFS) is a firmware distribution model which allows a Cisco IP phone to find other phones of the same model or series on the subnet and share updated firmware files when you need to upgrade multiple phones all at the same time. PFS uses Cisco Peer-to-Peer-Distribution Protocol (CPPDP) which is a Cisco proprietary protocol. With CPPDP, all the devices in the subnet form a peer-to-peer hierarchy, and then copy the firmware or the other files from peer devices to the neighboring devices. To optimize firmware upgrades, a root phone downloads the firmware image from the load server and then transfers the firmware to other phones on the subnet using TCP connections.

Peer firmware sharing:

- Limits congestion on TFTP transfers to centralized remove load servers.

- Eliminates the need to manually control firmware upgrades.

- Reduces phone downtime during upgrades when large numbers of phones are reset simultaneously.

Note

- Peer firmware sharing does not function unless multiple phones are set to upgrade at the same time. When a NOTIFY is sent with Event:resync, it initiates a resync on the phone. Example of an xml that can contain the configurations to initiate the upgrade:

“Event:resync;profile="http://10.77.10.141/profile.xml

- When you set the Peer Firmware Sharing Log server to an IP address and port, the PFS specific logs are sent to that server as UDP messages. This setting must be done on each phone. You can then use the log messages when troubleshooting issues related to PFS.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

In the Firmware Upgrade section, set the parameters:

1. Set the Peer Firmware Sharing parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: Yes

2. Set the Peer Firmware Sharing Log Server paramter to indicate the IP address and the port to which the UDP message is sent.

For example: 10.98.76.123:514 where, 10.98.76.123 is the IP address and 514 is the port number.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Peer_Firmware_Sharing_Log_Server>192.168.5.5</ Peer_Firmware_Sharing_Log_Server>

Peer_Firmware_Sharing_Log_Server specifies UDP Remote syslog server hostname and the port. The port defaults to the default syslog 514.

Step 3

Click Submit All Changes.

Specify the Profile Authentication Type

Profile Authentication allows phone users to resynchronize the provisioning profile onto the phone. Authentication information is required while the phone tries to resynchronize and download configuration file for the first time and gets an HTTP or HTTPS 401 authentication error. When you enable this feature, the Profile account setup screen is displayed on the phone for the following situations:

- When the HTTP or HTTPs 401 authentication error occurs during first-time provisioning after the phone reboots

- When the profile account username and password are empty

- When there are no username and password in the Profile Rule

If the Profile account setup screen is missed or ignored, the user can also access the setup screen through the phone screen menu, or the Setup softkey, which displays only when no line on the phone is registered.

When you disable the feature, the Profile account setup screen doesn't display on the phone.

The username and password in the Profile Rule field have a higher priority than the profile account.

- When you provide a correct URL in the Profile Rule field without a username and password, the phone requires authentication or digest to resynchronize the profile. With the correct profile account, authentication passes. With an incorrect profile account, authentication fails.

- When you provide a correct URL in the Profile Rule field with a correct username and password, the phone requires authentication or digest to resynchronize the profile. The profile account is not used for phone resynchronization. Sign-in is successful.

- When you provide a correct URL in the Profile Rule field with an incorrect username and password, the phone requires authentication or digest to resynchronize the profile. The profile account isn't used for phone resynchronization. Sign-in always fails.

- When you provide an incorrect URL in the Profile Rule field, sign-in always fails.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

You can specify the profile authentication type from the phone administration web page.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Provisioning.

Step 2

In the Configuration Profile section, set the Profile Authentication Type parameter to specify the credentials to use for profile account authentication.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options:

- Disabled: Disables the profile account feature.When this feature is disabled, the Profile account setup menu doesn't display on the phone screen.

- Basic HTTP Authentication: The HTTP login credentials are used to authenticate the profile account.

- XSI Authentication: XSI login credentials or XSI SIP credentials are used to authenticate the profile account. The authentication credentials depend on the XSI Authentication Type for the phone:

When the XSI Authentication Type for the phone is set to Login Credentials, the XSI login credentials are used.

When the XSI Authentication Type for the phone is set to SIP Credentials, the XSI SIP credentials are used.

Default: Basic HTTP Authentication

Step 3

Click Submit All Changes.

Control the Authentication Requirement to Access the Phone Menus

You can control if authentication is required to access phone menus.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Set the LCD Authentication and LCD Authentication Customization sections as described in the Parameters for User Authentication Control table.

- Parameters for User Authentication Control

Parameters for User Authentication Control

The following table defines the function and usage of the parameters for user authentication control feature in the LCD Authentication and LCD Authentication Customization section under the Voice > Phone 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.

Parameters for User Authentication Control

Parameter

Description

Require Authentication for LCD Menu Access

Controls whether the user requires authentication to access phone menus.

Perform one of the following:

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

<Require_Authentication_for_LCD_Menu_Access ua="na">Default</Require_Authentication_for_LCD_Menu_Access>

- On the phone web interface, select the required value.

Allowed values: Default|Customized|No

- Default—When selected, user needs to provide password and then sign in to access the phone menus that requires authentication. Phone continues to support all the functionalities that are supported in the releases prior to 11.3(2). Phone displays lock screen icon.

To access any phone menus that require authentication, user needs to provide the password and press Sign in. The lock icon remains locked. After the user signs in, the lock icon is unlocked.

- Customized—When selected, user requires authentication only to access Profile rule and Factory reset menus on the phone. Authenticaion control of these two menus also depends on the settings of the Factory Reset Menu menu and the Profile Rule Menu menu. User will not require any authentication to access other phone menus.

- No—When selected, the Sign in menu, the Sign out menu, the lock icon, and the Set password menus are not available on the phone. User can access phone menus without any authentication.

Default value: Default

Factory Reset Menu

Specifies if the user requires authentication to access Factory reset menu on the phone.

You can customize this parameter to Yes or No only when you set the Require Authentication for LCD Menu Access parameter to Customized.

Perform one of the following:

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

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

- On the phone web interface, set this parameter to Yes or No as needed.

Allowed values: Yes|No

Default value: Yes

Profile Rule Menu

Specifies if the user requires authentication to access Profile rule menu on the phone.

You can customize this parameter to Yes or No only when you set the Require Authentication for LCD Menu Access parameter to Customized.

Perform one of the following:

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

- On the phone web interface, set this parameter to Yes or No as needed.

Allowed values: Yes|No

Default value: Yes

Silence an Incoming Call with Ignore Soft Key

You can add the Ignore softkey on the phone. User can press this softkey to silence an incoming call when busy and don't want to be disturbed. When the user presses the softkey, the phone stops ringing, but the user gets a visual alert, and, can answer the phone call.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Programmable Softkeys section, set the Programmable Softkey Enable to Yes.

Step 3

Enter the following values in the Ringing Key List field:

answer|1;ignore|2;ignoresilent|3;

Step 4

Click Submit All Changes.

Move an Active Call from a Phone to Other Phones (Locations)

You can configure a phone to allow a call to seamlessly be moved from one desk phone(location) to another mobile phone or desk phone(location).

When you enable this feature, the Anywhere menu is added into the phone screen. The user can use this menu to add multiple phones as locations to the extension. When there is an incoming call in that extension, all the added phones will ring and the user can answer the incoming call from any location. The locations list also gets saved to the BroadWorks XSI server.

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 Parameters for Moving Active Call to Other Locations table.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

In the XSI Line Service section, set the XSI Host Server, XSI Authentication Type, Login User ID, Login Password, and Anywhere Enable parameters as described in the Parameters for Moving Active Call to Other Locations table.

If you select SIP Credentials for XSI Authentication Type, you need to enter subsriber Auth ID and Password in the Subscriber Information section.

Step 3

Click Submit All Changes.

- Parameters for Moving Active Call to Other Locations

Parameters for Moving Active Call to Other Locations

The following table defines the function and usage of Moving Active Call to Locations parameters in the XSI Line Service section under the Ext(n) 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.

Parameters for Moving Active Call to Locations

Parameter

Description

XSI Host Server

Enter the name of the server. For example:

xsi.iop1.broadworks.net

Note

XSI Host Server uses http protocol by default. To enable XSI over HTTPS, you can specify https:// in the server.

Perform one of the following:

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

<XSI_Host_Server ua="na">https://xsi.iop1.broadworks.net</XSI_Host_Server>

- In the phone web page, enter the server.

For example:

https://xsi.iop1.broadworks.net

You can also specify a port for the server. For example:

https://xsi.iop1.broadworks.net:5061

If you don't specify a port. The default port for the specified protocol is used.

Default: Blank

XSI Authentication Type

Determines the XSI authentication type.

Perform one of the following:

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

<XSI_Authentication_Type ua="na">SIP Credentials</XSI_Authentication_Type>

- In the phone web page, select an authentication type.

Options:

Login Credentials－authenticates access with Login User ID and Login Password.

SIP Credentials－authenticates access with the register Auth ID and Password of the SIP account registered on the phone.

If you select SIP Credentials for XSI Authentication Type, you need to enter subscriber Auth ID and Password in the Subscriber Information section.

Default: Login Credentials

Login User ID

BroadSoft User ID of the phone user.

For example:

johndoe@xdp.broadsoft.com.

Perform one of the following:

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

<Login_User_ID ua="na">4081005300@as1bsoft22.sipurash.com</Login_User_ID>

- In the phone web page, enter a valid user ID.

For any XSI Authentication Type, you must enter Login User ID. The BroadWorks Anywhere feature does not work without this parameter.

Default: admin

Login Password

Alphanumeric password associated with the Login User ID.

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

After you enter the password, this parameter shows the following in the configuration file (cfg.xml): <ACS_Password ua="na">************</ACS_Password>

Default: Blank

Anywhere Enable

Enables BroadWorks Anywhere feature on an extension.

Perform one of the following:

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

<Anywhere_Enable_1_ ua="na">Yes</Anywhere_Enable_1_>

- In the phone web page, select Yes, Anywhere is enabled on this line, and the user can use the phone menu to add multiple locations to this specific line.

Valid values: Yes|No

Default: Yes

Sync the Block Caller ID Feature with the Phone and the BroadWords XSI Server

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.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

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:

<Block_CID_Enable_1_ ua="na">No</Block_CID_Enable_1_>

Note

- When Feature Key Sync is set to Yes, FKS takes precedent over XSI synchronization.

- If XSI host server and credentials are not entered and the CFWD Enable field is set to Yes, the phone user can't forward calls on the phone.

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Enable Viewing BroadWorks XSI Call Logs on a Line

You can configure a phone to display recent call logs from either the BroadWorks server or the local phone. After you enable the feature, the Recents screen has a Display recents from menu and the user can choose the XSI call logs or the local call logs.

You can set up a feature to do a reverse name lookup against local contacts for BroadWorks server call logs. For example, on server you set up a user 3280 (4085273280) with name “cx400 liu” and another user 3281(4085273281) with name “cx401 liu”. User 3280 is registered on phone A and user 3281 is registered on phone B. From phone A you make a missed call, a received call, or a placed call on phone B. The display of the broadsoft call logs on phone B appears as follows:

- If the personal directory doesn't have a contact that matches with the caller name, the BroadWorks call logs on phone B displays the original name "cx400 liu" saved in the server as the caller name.

- If the personal directory has a contact with "Name" = "B3280" and "Work" = "3280" that matches with the calling number, the BroadWorks call logs on phone B displays the contact name "B3280" as the caller name.

- If the personal directory has a contact with "Name" = "C3280" and "Work" = "03280", and the user configures a caller id map rule (<3:03>x.), the BroadWorks call logs on the phone B displays "C3280" using the mapped phone number 03280. If there is a matched contact of the unmapped phone number, the mapped phone number will not be used for reverse name lookup.

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 Parameters for BroadWorks XSI Call Logs on a Line table.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

CallLog Enable field is enabled.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the XSI Phone Service section, set the XSI Host Server, XSI Authentication Type, Login User ID, Login Password, and Directory Enable fields as described in Parameters for BroadWorks XSI Call Logs on a Line.

If you select SIP Credentials for XSI Authentication Type, you need to enter SIP Auth ID and SIP Password in this section.

Step 3

Set the CallLog Associated Line and Display Recents From fields as described in Parameters for BroadWorks XSI Call Logs on a Line.

Note

The Display recents from menu doesn't appear in the Recents phone screen when you set the value of the CallLog Enable field to No,

Step 4

Click Submit All Changes.

- Parameters for BroadWorks XSI Call Logs on a Line

Parameters for BroadWorks XSI Call Logs on a Line

The following table defines the function and usage of XSI Call Logs on a Line parameters in the XSI Phone Service section under the Phone 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.

Parameters for XSI Call Logs on a Line

Parameter

Description

XSI Host Server

Enter the name of the server; for example,

xsi.iop1.broadworks.net
.

Note

XSI Host Server uses http protocol by default. To enable XSI over HTTPS, you can specify https:// in the server.

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

<XSI_Host_Server ua="na">https://xsi.iop1.broadworks.net</XSI_Host_Server>

- In the phone web interface, enter the XSI server to use.

Default: Empty

XSI Authentication Type

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

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

<XSI_Authentication_Type ua="na">SIP Credentials</XSI_Authentication_Type>

- In the phone web interface, specify the authentication type for XSI service.

Options: SIP Credentials and Login Credentials

Default: Login Credentials

Login User ID

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

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

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

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

<Login_User_ID ua="na">username</Login_User_ID>

- In the phone web interface, enter the username used to authenticate the access to the XSI server.

Default: Empty

Login Password

Alphanumeric password associated with the User ID.

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

Default: Empty

Directory Enable

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

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the BroadSoft directory.

Option: Yes and No

Default: No

CallLog Associated Line

Allows you to select a phone line for which you want to display the recent call logs.

Perform one of the following:

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

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

- In the phone web interface, Select a phone line.

Valid values: 1 to 10

Default: 1

Display Recents From

Allows you to set which type of recent call logs the phone will display.

Perform one of the following:

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

<Display_Recents_From ua="na">Phone</Display_Recents_From>

- In the phone web interface, Choose Server to display BroadSoft XSI recent call logs and select Phone to display local recent call logs.

Option: Phone and Server

Default: Phone

Note

The Display recents from is added to the Recents screen of the phone only when you set CallLog Enable to Yes and Display Recents From type to Server.

Enable Feature Key Sync

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.

For the Cisco IP Phone 6871 Multiplatform Phones, if the FKS is enabled, executives can access the Settings > Executive menu on the phone. Similarly, assistants can access the Settings > Assistant menu.

Note

For the Cisco IP Phone 6871 Multiplatform Phones that supports the executive-assistant feature, make sure that the Feature Key Sync is enabled for the executive and assistant users.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext [n] (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.

Related Tasks

Enable Call Forward Status Sync via XSI Service

Enable DND Status Sync via XSI Service

Related References

DND and Call Forward Status Sync

DND and Call Forward Status Sync

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

Note

For the Cisco IP Phone 6871 Multiplatform Phones, make sure that Feature Key Sync is enabled for all the executive and assistant users. Only the Cisco IP Phone 6871 Multiplatform Phones supports the executive-assistant feature.

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.

Interaction Between FKS and XSI Synchronization

Feature Key Sync

DND Enabled

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)

Yes

No

No

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

If a line key is configured with FKS or XSI synchronization and is also enabled with DND or call forward, the respective DND icon or the call forward icon is displayed next to the line key label. If the line key has a missed call, a voice message, or an urgent voicemail alert, the DND icon or the call forward icon is also displayed with the alert notification.

- Enable Call Forward Status Sync via XSI Service

- Enable DND Status Sync via XSI Service

Related Tasks

Enable Feature Key Sync

Enable Call Forward Status Sync via XSI Service

Enable DND Status Sync via XSI Service

Enable Call Forward Status Sync via XSI Service

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.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Configure the XSI host server and the corresponding credentials on the Voice > Ext (n) 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 Voice > Ext (n) .

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 2

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:

<CFWD_Enable_1_ ua="na">Yes</CFWD_Enable_1_>

Options: Yes and No

Default: Yes

Note

If XSI sync for call forward is enabled and the XSI host server or XSI account is not configured correctly, the phone user can't forward calls on the phone.

If XSI sync for call forward is enabled but the XSI host server or XSI account isn’t configured correctly, the Forward softkey isn’t available (doesn't appear). The user can't forward calls on the phone until the XSI account is configured correctly.

Step 3

Click Submit All Changes.

Related Tasks

Enable Feature Key Sync

Related References

DND and Call Forward Status Sync

Enable DND Status Sync via XSI Service

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.

Note

If XSI sync for DND is enabled but the XSI host server or XSI account is not configured correctly, the DND softeky isn't not available (doesn't appear) on the phone. The user can't see the softkey and turn on DND mode on the phone until the XSI account is configured correctly.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Configure the XSI host server and the corresponding credentials on the Voice > Ext (n) 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 Voice > Ext (n).

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 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:

<DND_Enable_1_ ua="na">Yes</DND_Enable_1_>

Options: Yes and No

Default: Yes

Step 3

Click Submit All Changes.

Related Tasks

Enable Feature Key Sync

Related References

DND and Call Forward Status Sync

Enable Synchronization of Anonymous Call Rejection via XSI Service

You can enable synchronization of Anonymous Call Rejection for each line via the XSI service. The function can be used to reject calls from callers who have blocked the display of their number.
Except for the setting for each line, you can also use the Block ANC Setting field under the Supplementary Services section from Voice > User to directly enable or disable the function for all lines.

The priority of the setting: Block Anonymous Call Enable > Block ANC Setting.

For example, if you set Block Anonymous Call Enable to Yes for a specific line, the setting in the Block ANC Setting doesn't take effect for the line, it takes effect for other lines on which Block Anonymous Call Enable is No.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Configure the XSI host server and the corresponding credentials on the Voice > Ext (n) 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.

- Ensure that Anonymous Call Rejection is enabled on the line or in the XSI service. Otherwise, your user still receives anonymous calls.

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 2

In the XSI Line Service section, set the Block Anonymous Call Enable parameter to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Where n is the extension number.

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

After the change takes effect, the XSI service takes over the phone to provide the function. The function doesn't work in the following scenarios even though Block Anonymous Call Enable is set to Yes:

- The function is disabled in the XSI service.

- The function is disabled on the line.

Because the function status is synchronized between the XSI service and the line.

- Set Feature Activation Code for Anonymous Call Rejection

Set Feature Activation Code for Anonymous Call Rejection

You can set activation code to block or remove blocking of anonymous calls for all lines on which synchronization of Anonymous Call Rejection is disabled.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

In the Vertical Service Activation Codes section, ensure that the Block ANC Act Code field is set to the value defined by the server. The default value is *77.

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

<Block_ANC_Act_Code ua="na">*77</Block_ANC_Act_Code>

Step 3

In the Vertical Service Activation Codes section, ensure that the Block ANC Deact Code field is set to the value defined by the server. The default value is *87.

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

<Block_ANC_Deact_Code ua="na">*87</Block_ANC_Deact_Code>

Step 4

Click Submit All Changes.

Your user can dial *77 or *87 and press the Call softkey to block all anonymous calls or remove the blocking.

This operation is identical to the setting on the Block ANC Setting field under the Supplementary Services section from Voice > User. It takes effect for the lines on which the Block Anonymous Call Enable (under the XSI Line Service section from Voice > Ext) is set to No.

Enable Synchronization of Call Waiting via XSI Service

You can enable synchronization of Call Waiting for each line via the XSI service. The function allows the user to receive incoming calls while on another call.
Except for the setting, you can also use the CW Setting field under the Supplementary Services section from Voice > User to directly enable or disable the function for all lines.

The priority of the setting: Call Waiting Enable > CW Setting.

For example, if you set Call Waiting Enable to Yes for a specific line, the setting in the CW Setting doesn't take effect for the line, it only takes effect for other lines on which Call Waiting Enable is set to No.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Configure the XSI host server and the corresponding credentials on the Voice > Ext (n) 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.

- Ensure that Call Waiting is enabled on the line or in the XSI service. Otherwise, your user doesn't receive any incoming calls while on a call.

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 2

In the XSI Line Service section, set the Call Waiting Enable parameter to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Where n is the extension number.

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

After the change takes effect, the XSI service takes over the phone to provide the function. The function doesn't work in the following scenarios even though Call Waiting Enable is set to Yes:

- The function is disabled in the XSI service.

- The function is disabled on the line.

Because the function status is synchronized between the XSI service and the line.

- Set Feature Activation Code for Call Waiting

Set Feature Activation Code for Call Waiting

You can set activation code (star code) that can be used to activate or deactivate Call Waiting for all lines.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

In the Vertical Service Activation Codes section, ensure that the CW Act Code field is set to the value defined by the server. The default value is *56.

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

<CW_Act_Code ua="na">*56</CW_Act_Code>

Step 3

In the Vertical Service Activation Codes section, ensure that the CW_Deact_Code field is set to the value defined by the server. The default value is *57.

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

<CW_Deact_Code ua="na">*57</CW_Deact_Code>

Step 4

In the Vertical Service Activation Codes section, ensure that the CW_Per_Call_Act_Code field is set to the value defined by the server. The default value is *71.

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

<CW_Per_Call_Act_Code ua="na">*71</CW_Per_Call_Act_Code>

Step 5

In the Vertical Service Activation Codes section, ensure that the CW_Per_Call_Deact_Code field is set to the value defined by the server. The default value is *70.

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

<CW_Per_Call_Deact_Code ua="na">*70</CW_Per_Call_Deact_Code>

Step 6

Click Submit All Changes.

Your user can dial *56 or *57 and press the Call softkey to activate or deactivate Call Waiting for all incoming calls. This operation is identical to the setting on the CW Setting field under the Supplementary Services section from Voice > User. These activation codes don't take effect for the lines where synchronization of Call Waiting via the XSI service is enabled.

Your user can dial *71 or *70 and press the Call softkey to temporarily activate or deactivate Call Waiting for the next incoming call on an active call. These activation codes still take effect for the lines where synchronization of Call Waiting via the XSI service is enabled. If Call Waiting is disabled in the XSI service, the server blocks all incoming calls, therefore these activation codes don't take effect.

Executives and Assistants

Note

Only the Cisco IP Phone 6871 Multiplatform Phones support the executive-assistant features.

You can set up executives and their assistants to share control of calls.

You configure users as executives and assistants in BroadWorks. The BroadWorks configuration also sets up the relationships between the executives and assistants. For more information, see the BroadWorks documentation.

After the BroadWorks configuration, you configure the following phone settings.

- Enable the synchronization of executive-assistant settings between the phone and the server.

- Update the dial plan to enable users to dial service activation codes.

- Configure a line key for the Executive / Assistant menu access if necessary.

- Change the service activation codes if necessary.

- Change the programmable softkeys if necessary.

- Set up preference of the executive-assistant role for the phone.

Important

- Executives and assistants cannot share phones. Do not configure extensions on the same phone for an executive and an assistant.

- The executive-assistant feature is recommended for private lines.

- The number of calls that an assistant can initiate in parallel is limited to the Call Appearances Per Line setting under Miscellaneous Line Key Settings on the phone web page > Voice > Phone tab.

- Set Up Preference for the Executive-Assistant Role

- Executive-Assistant Setting Synchronization

- Dial Plan for Executives and Assistants

- Enable Alphanumeric Dialing for Executives

- Configure Executive and Assistant Menu Access on a Line Key

- Service Activation Codes for Executives and Assistants

- Programmable Softkeys for Executives and Assistants

Set Up Preference for the Executive-Assistant Role

Perform this task if you want to set the preference for the executive-assistant role that an extension selects. The preference setting can't directly determine the executive-assistant role of the phone. The preference setting on the phone and the role relationship setting on the BroadWorks can finally determine the actual role for the phone. For more information about the logic of the role selections, see Selection Logic of the Executive-Assistant Role.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Make sure that the Feature Key Synchronization (FKS) is enabled. See Enable Feature Key Sync.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Executive Assistant section, set the Executive Assistant Role field to Default, Executive, or Assistant.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is phone-specific. Enter a string in this format:

<Executive_Assistant_Role ua="na">Default</Executive_Assistant_Role>

Step 3

Click Submit All Changes.

- Selection Logic of the Executive-Assistant Role

Selection Logic of the Executive-Assistant Role

This section describes the selection logic of the executive-assistant role for an extension and a phone respectively.

Selection Logic of the Executive-Assistant Role for an Extension

The executive-assistant role for an extension is determined by the following criteria:

- The preferred role ("Default", "Executive", or "Assistant") that is set up on the phone web page or by the phone configuration file (cfg.xml)

- The relationship setting between executives and assistants from the BroadWorks server

The following information shows the selection logic of the different preferred executive-assistant role:

- First priority: If an assistant doesn't have an executive list, this means that the assistant has no relationship with any executive. In this case, none of the phone's extension acts as the assistant role.

- If Executive Assistant Role is set to Default:

- An extension that has only the assistant role acts as the assistant role.

If the assistant role of the extension doesn't have an executive list, then the extension doesn't act as any role.

- An extension that has only the executive role acts as the executive role.

- An extension that has both the assistant and the executive roles acts as the assistant role.

Note

An extension's assistant role has higher priority than its executive role. If the assistant role of the extension doesn't have an executive list, according to the First priority, the extension will act as the executive role.

- If Executive Assistant Role is set to Executive:

An extension acts as the executive role after it retrieves the role from the BroadWorks server. Before that, it doesn't act as any role.

When the BroadWorks server assigns both roles to an extension, the extension acts as the executive role according to the setting on the phone.

- If Executive Assistant Role is set to Assistant:

An extension acts as the assistant role after it retrieves the role from the BroadWorks server. Before that, it doesn't act as any role.

When the BroadWorks server assigns both roles to an extension, the extension acts as the assistant role according to the setting on the phone.

If the assistant role doesn't have an executive list, the extension will not act as any role.

Selection Logic of the Executive-Assistant Role for a Phone

The executive-assistant role for a phone is determined by the first available executive or assistant extension:

1. The phone searches all its extensions from the first one (extension 1).

2. The phone selects the first extension that acts as the executive or assistant role as the executive-assistant extension of the phone.

3. The phone acts as the same role of the selected extension.

The following table shows the examples of which role a phone will act as in different scenarios:

Examples of Role Selection for a Phone

Executive-Assistant Role

Extension 1

Extension 2

Role of the Phone

Executive-Assistant Extension

Get Executive Role

Get Assistant Role

Assistant Has Executive List

Get Executive Role

Get Assistant Role

Assistant Has Executive List

Default

No

Yes

No

No

Yes

No

None

N/A

Default

No

Yes

No

No

Yes

Yes

Assistant

Ext 2

Default

Yes

No

No

Yes

Yes

Yes

Executive

Ext 1

Assistant

Yes

No

No

No

Yes

No

None

N/A

Assistant

No

No

No

Yes

Yes

Yes

Assistant

Ext 2

Executive

No

Yes

Yes

No

Yes

Yes

None

N/A

Executive

Yes

No

No

Yes

Yes

Yes

Executive

Ext 1

Executive-Assistant Setting Synchronization

Executive and assistant functions require settings to be synchronized between the phones and the server via Feature Key Synchronization (FKS). Once you enable FKS, executives can access the Settings > Executive menu on the phone. Similarly, assistants can access the Settings > Assistant menu.

To enable FKS, see Enable Feature Key Sync.

Dial Plan for Executives and Assistants

The dial plan for executives and assistants must include the following digit sequences:

- #xx or a variant that includes the expression, to enable users to dial # codes.

- *xx or a variant that includes the expression, to enable users to dial * codes.

- #xx+xxxxxxxxxxx*xxxxxxxxxx to enable users to dial # codes followed by numbers.

See Dial Plan Overview for information on the dial plan.

See Edit the Dial Plan on the IP Phone for details on how to add entries to the dial plan.

Enable Alphanumeric Dialing for Executives

The executives can directly make calls to their assistants when they check the assistant list.

The target phone number might include alphanumeric characters. To achieve this function, you must set Enable URI Dialing to Yes in the phone web page. For more information, see Configure Alphanumeric Dialing.

Configure Executive and Assistant Menu Access on a Line Key

You can configure a line key for access to the Settings > Executive menu on an executive's phone, and the Settings > Assistant menu on an assistant's phone.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select an available Line Key on which to access the executive or assistant menu.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to configure the executive and assistant menu access on the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Extension_2_ ua="na">Disabled</Extension_2_>

Step 4

Set the Share Call Appearance parameter to Private.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Share_Call_Appearance_2_ ua="na">private</Share_Call_Appearance_2_>

Step 5

In the Extended Function parameter, enter a string in this format:

fnc=bw-exec-assist

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a sting in this format:

<Extended_Function_2_ ua="na">fnc=bw-exec-assist</Extended_Function_2_>

Step 6

Click Submit All Changes.

Service Activation Codes for Executives and Assistants

Executives and assistants can access most functions through a service activation code or a programmable softkey.

- For executives:

- Activating call filtering

- Deactivating call filtering

- Joining ongoing calls

- Transferring ongoing calls to themselves

- For assistants:

- Joining ongoing calls

- Transferring ongoing calls to themselves

The following functions can be accessed only through service activation codes.

Important

You will need to inform the users of the service activation codes for these functions in order to enable them to perform these actions.

All service activation codes for the feature are set up by default.

You can change the service activation codes for specific functions according to your organization's requirement. See Vertical Service Activation Codes for details.

Important

If you change a service activation code through the phone website, make sure to update the corresponding setting in the BroadWorks server, and vice versa.

Executive-Assistant Service Activation Codes in the XML Configuration File

Important

If you change a service activation code in the XML Configuration file, make sure to update the corresponding setting in the BroadWorks server.

The following sample XML Configuration file section shows the parameters (XML tags) and values for the executive-assistant feature service activation codes.

<!-- Vertical Service Activation Codes -->
<Exec_Assistant_Call_Initiate_Code ua="na">#64</Exec_Assistant_Call_Initiate_Code>
<Exec_Call_Filter_Act_Code ua="na">#61</Exec_Call_Filter_Act_Code>
<Exec_Call_Filter_Deact_Code ua="na">#62</Exec_Call_Filter_Deact_Code>
<Exec_Assistant_Call_Push_Code ua="na">#63</Exec_Assistant_Call_Push_Code>
<Exec_Call_Retrieve_Code ua="na">*11</Exec_Call_Retrieve_Code>
<Exec_Call_Bridge_Code ua="na">*15</Exec_Call_Bridge_Code>

The following table describes these parameters and values.

Parameter

Value

Description

Exec_Assistant_Call_Initiate_Code

The # or * code that you want to be used for the function

For assistants to initiate calls on behalf of executives

Exec_Call_Filter_Act_Code

The # or * code that you want to be used for the function

For executives to activate call filtering

Exec_Call_Filter_Deact_Code

The # or * code that you want to be used for the function

For executives to deactivate call filtering

Exec_Assistant_Call_Push_Code

The # or * code that you want to be used for the function

For assistants to transfer an ongoing call to an executive

Exec_Call_Bridge_Code

The # or * code that you want to be used for the function

For executives or assistants to join an ongoing call

Exec_Call_Retrieve_Code

The # or * code that you want to be used for the function

For executives or assistants to transfer an ongoing call to themselves

Programmable Softkeys for Executives and Assistants

All programmable softkeys for executives and assistants are set up by default.

See Programmable Softkeys for details on programmable softkeys.

Executive-Assistant Programmable Softkeys in the XML Configuration File

The following sample XML Configuration file section shows the parameters (XML tags) and values for the programmable softkeys for executives and assistants.

<!-- Programmable Softkeys -->
<Programmable_Softkey_Enable ua="na">No</Programmable_Softkey_Enable>
<Idle_Key_List ua="na">em_login;acd_login;acd_logout;astate;avail;unavail;redial;
recents;cfwd;dnd;lcr;pickup;gpickup;unpark;em_logout;guestin;guestout;callretrieve;
bridgein;</Idle_Key_List>
<Hold_Key_List ua="na">resume|1;endcall|2;newcall|3;redial;dir;cfwd;dnd;
callpush;</Hold_Key_List>
<Shared_Active_Key_List ua="na">newcall|1;barge|2;bargesilent|3;cfwd|4;dnd|5;
callretrieve;bridgein</Shared_Active_Key_List>
<Shared_Held_Key_List ua="na">resume|1;barge|2;cfwd|3;dnd|4;</Shared_Held_Key_List>
<Exec_Assistant_Key_List ua="na">proxycall|2;divert|3;</Exec_Assistant_Key_List>

The following table describes these parameters and values.

Parameters

Value

Description

Idle_Key_List, Shared_Active_Key_List

bridgein

Enables executives to join an ongoing call

Idle_Key_List, Shared_Active_Key_List

callretrieve

Enables executives to transfer an ongoing call to themselves

Hold_Key_List

callpush

Enables assistants to transfer an ongoing call to an executive, after putting the call on hold

Exec_Assistant_Key_List

proxycall

Enables assistants to initiate calls on behalf of executives, from the Settings > Assistant menu

Exec_Assistant_Key_List

divert

Enables assistants to activate or deactivate call diversion, from the Settings > Assistant menu

Enable End-of-Call Statistics Reports in SIP Messages

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.

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

For description of the attributes in call statistics, see 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>

Before you begin

Access the phone administration web page, see Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

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:

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

The allowed values are Yes|No. The defaut value is No.

Step 3

Click Submit All Changes.

- Attributes for Call Statistics in SIP Messages

Attributes for Call Statistics in SIP Messages

Audio: RTP-RxStat Payload

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

Audio: RTP-TxStat Payload

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

SIP Session ID

The Multiplatform phones now support “Session Identifier”. This feature helps to overcome the limitations with the existing call-identifiers and 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 Example 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=298da61300105000a00000ebd5cbd5c1

- Enable SIP Session ID

- Session ID Parameters

Enable SIP Session ID

You 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.

Before you begin

Access the Phone Web Interface

Procedure

Step 1

Select Voice > Ext(n).

Step 2

Go to the SIP Settings section.

Step 3

Set the SIP SessionID Support field as described in the Session ID Parameters table.

Step 4

Click Submit All Changes.

Session ID Parameters

The following table defines the function and usage of each parameter in the SIP Settings section in the Voice > Ext(n) 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

SIP SessioID Support

Controls the SIP session ID support.

Perform one of the following

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

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

- In the phone web page select Yes to enable the feature.

Allowed values: Yes/No

Default: Yes.

Select the Line Key LED Behavior

You can choose the Line Key LED Behavior in your multiplatform phone.

In the phone web page, you can choose one of the following options:

- Default

- Preset 1

- Custom

Before you begin

- Access the administration web interface. See Access the Phone Web Interface.

- Ensure that the line keys are not in the Inert mode.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Line Key LED Pattern section, choose an option from the Custom LED Type drop down list.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Custom_LED_Type ua="na">Default</Custom_LED_Type>

The allowed values are Default|Preset 1|Custom. The default value is Default.

Step 3

Click Submit All Changes.

When you submit your selection in the Custom LED Type drop down list, changes also occur in the Att Console tab. This only happens when you configure simultaneous LED behavior for phone line keys and key expansion module line keys.

To view the key expansion module LED behavior:

1. Select Voice > Att Console.

2. View the Att Console Key LED Pattern section.

3. Change the LED pattern and color from the fields MWI None LED, MWI New LED, and MWI Urgent LED.

- Line Key LED Behaviour Customization

Line Key LED Behaviour Customization

The line key LED behaviour in your multiplatform phones has three optional settings. In the phone web page, you can choose one of the following options:

- Default

- Preset 1

- Custom

The following table compares the basic Default and Preset 1 line key LED behaviour settings.

Basic LED Behaviour on Line Keys

Field and Description

Default Pattern and Color

Preset 1 Pattern and Color

Idle—Line is idle.

Solid green

Off

Registration Failed—Line failed to register with the server.

Solid amber

Off

Registering—Line is trying to register with the server.

Slow blinking amber

Off

Disabled—Line is disabled or not in use.

Off

Off

Basic LED Behaviour on the Voicemail PLK

State

Default Pattern and Color

Preset 1 Pattern and Color

New Message

Solid red

Solid red

Registration Failed—Line failed to register with the server.

Solid amber

Off

No new messages

Solid green

Off

Urgent messages

Solid red

Solid red

For the Custom option, use the following color-pattern key:

- COLOR (c)

- g = GREEN

- r = RED

- a = AMBER

- o = OFF

When you set COLOR to OFF, PATTERN is ignored even if you set it.

- PATTERN (p)

- b = Blink with a Color

This is equivalent to the system default of Slow Blink.

- n = No Blink, Solid Color

Note

If you misconfigure any field, the system applies the system Default settings to that field.

The table below contains three sample custom LED behaviour settings.

Sample LED Configurations

LED

User Input

Solid Red

c=r;p=n

c=r

Blinking Amber

c=a;p=b

OFF

c=o

Each of the following features has LED behaviour settings. The tables that follow describe the LED behaviour for each feature.

- Basic Line and Bluetooth Hands-Free Line

- Shared Line

The following table describes the LED behaviour for the Shared Line feature. The table compares the Default and Preset 1 settings.

Shared Line

Field and Description

Default Pattern and Color

Preset 1 Pattern and Color

Remote Active LED—Shared line on the local device is receiving status that another device is in a connected call.

Pulsing (double blinking) red

Solid red

Remote Held LED—Shared line on the local device is receiving status that another device has put a call on hold.

Slow blinking red

Blinking red

Remote Progressing LED—Shared line on the local device is receiving status that another device is sending outgoing call.

Pulsing (double blinking) red

Solid red

Remote Ringing LED—Shared line on the local device is receiving status that another device is ringing on incoming call.

Pulsing (double blinking) red

Blinking amber

Remote Seized LED—Shared line on the local device is receiving status that another device is going off-hook.

Pulsing (double blinking) red

Solid red

Remote Undefined LED—Shared line on the local device is not in operational state.

Solid red

Off

You can configure the LED behaviour that applies to both phone line keys and key expansion module buttons.

For the Cisco 6800 Key Expansion Module, the custom LED settings apply only to the phone line keys. If you configure custom LED behaviour on the phone line keys, the expansion module buttons take the expansion module default behaviour, instead.

The following table describes the basic subscription line key LED behaviour for the simultaneous settings of the phone and key expansion module buttons. The table compares the Default and Preset 1 settings.

Basic Line Key and Key Expansion Module Buttons Settings

Field and Description

Default Pattern and Color

Preset 1 Pattern and Color

Serv Subscribe Failed LED—The service subscription failed.

Solid amber

Off

Serv Subscribing LED—The service subscription is proceeding.

Slow blinking amber

Off

The following table describes the LED behaviour settings for the Speed Dial, XML Applications, and the PLK part of the Executive-Assistant feature (only for the Cisco IP Phone 6871 Multiplatform Phones). The table compares the Default and Preset 1 settings.

Speed Dial and XML Application Feature Settings

Field and Description

Default Pattern and Color

Preset 1 Pattern and Color

Application LED—Extended service application is operational.

Solid green

Off

The following table describes the LED behaviour for the Call Park feature. The table compares the Default and Preset 1 settings.

Call Park Feature Settings

Field and Description

Default Pattern and Color

Preset 1 Pattern and Color

Parking Lot Idle LED—The monitored line has no parked call.

Solid green

Off

Parking Lot Busy LED—The monitored line has a parked call.

Slower blinking red

Blinking red

The following table describes the LED behaviour for the Busy Lamp Field (BLF) feature. The table compares the Default and Preset 1 settings.

BLF Feature Settings

Field and Description

Default Pattern and Color

Preset 1 Pattern and Color

BLF Idle LED—The monitored line is idle.

Solid green

Off

BLF Ringing LED—The monitored line is ringing.

Fast blinking red

For the key expansion module, it is Slow blinking red because the key expansion module is limited to:

- ON

- OFF

- Slow Blink

Blinking amber

BLF Busy LED—The monitored line is busy in a call.

Solid red

Solid red

BLF Held LED—The monitored line has put a call on hold.
Applies to phone line keys, only:

BLF Busy LED applies to key expansion module buttons.

Slow blinking red

Solid red

After you configure a phone with custom LED behaviour, and you downgrade it to an older Firmware version, the custom LED settings remain. If you wish to remove the custom LED settings, then set the LED behaviour to the system default before you perform the downgrade.

Set Up a Phone for Remote SDK

You can configure remote SDK for a multiplatform phone. The remote SDK provides a WebSocket based protocol through which the phone can be controlled.

Before you begin

- Access the Phone Web Interface

- A WebSocket server must be running with an address and port reachable from the phone.

Procedure

Step 1

Select Voice > Phone.

Step 2

Go to the WebSocket API section.

Step 3

Set the Control Server URL and the Allowed APIs fields as described in the WebSocket API Parameters table.

Step 4

Click Submit All Changes.

- WebSocket API Parameters

WebSocket API Parameters

The following table defines the function and usage of each parameter in the WebSocket API section in the Voice > Phone 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.

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

<Control_Server_URL ua="na"/>

- In the phone web page enter the URL of a WebSocket server.

For example:

<Control_Server_URL>wss://my-server.com
/ws-server-path</Control_Server_URL>

The URL should be in one of the following formats:

- For a nonsecure HTTP connection:

ws://your-server-name/path

- For a secure HTTPS connection:

wss://your-server-name/some-path

We recommend a secure connection.

Default: Empty.

Allowed APIs

A regular expression that can be used to limit the API calls that are allowed from the controlling server.

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

<Allowed_APIs ua="na">.*</Allowed_APIs>

- In the phone web page enter an appropriate regular expression.

The regular expression provided is matched with the Request-URI path provided in the API request from the controlling server. If the entire path is not matched by the given regular expression, the API call is rejected.

Allowed values are:

- .*: All APIs are allowed

- /api/Call/v1/.*: All v1 Call interface calls are allowed.

- /api/Call/v1/(Dial|Hangup): Only the v1 Call interface calls Dial and Hangup are allowed.

Default: .*

Hide a Menu Item from Being Displayed on the Phone Screen

By default, all the menu items on the phone screen Information and settings are visible to users. You can configure the phone to hide or show specific menu items. When hidden, the items don't display on the phone screen.

You can hide any of the following menu items as needed:

- Speed dials

- Executive assistant (Cisco IP Phone 6871 Multiplatform Phones only)

- Assistant call filter (Cisco IP Phone 6871 Multiplatform Phones only)

- User preferences

- Network configuration

- Device administration

- Status

- Report problem

You can also configure the visibility of the menu items in the configuration file (cfg.xml) with strings in this format:

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

See the parameter syntax and valid values in Parameters for Menu Visibility.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Menu Visibility section, set the menu items that you want to hide to No.

Step 3

Click Submit All Changes.

- Parameters for Menu Visibility

Parameters for Menu Visibility

The following table defines the function and usage of each parameter in the Menu Visibility section of the Voice > Phone tab.

Parameters for Menu Visibility

Parameter Name

Description and Default Value

Speed Dials

Controls whether to show the Speed dials menu on the phone screen. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Executive Assistant

This field is only available on the Cisco IP Phone 6871 Multiplatform Phones.

Controls whether to show the Executive menu on the executive's phone and the Assistant menu on the assistant's phone. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Assistant Call Filter

This field is only available on the Cisco IP Phone 6871 Multiplatform Phones.

Controls whether to show the Call filter menu on the phone screen for the assistant role. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

User Preferences

Controls whether to show the User preferences menu on the phone screen. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Network Configuration

Controls whether to show the Network configuration menu on the phone screen. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Device Administration

Controls whether to show the Device administration menu on the phone screen. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Status

Controls whether to show the Status menu on the phone screen. Set this field to Yes to show the menu. Otherwise, set it to No.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Report Problem

Controls whether to show the Report problem menu under the Status menu on the phone screen. Set this field to Yes to show the menu. Otherwise, set it to No.

When the Status menu is invisible, the Report problem menu is invisible as well.

Perform one of the following:

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

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

- In the phone web interface, select Yes or No to show or hide the menu.

Valid values: Yes and No

Default: Yes

Display Caller Number Instead of Unresolved Caller Name

By default, the phone displays both the caller name and the caller number in an incoming call alert. When the phone can't resolve the characters in caller name, the user sees boxes instead of the caller name. You can configure the phone to display only the number when any unresolved characters are detected in the caller name.

Procedure

Step 1

Select Voice > Regional.

Step 2

In the Language section, set the Replace Unresolved Caller Name with Number field to Yes.

You can also configure this parameter in the configuration file (cfg.xml) with a string in this format:

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

The valid values are Yes and No. The default setting is No.

Step 3

Click Submit All Changes.

Add a Menu Shortcut to a Line Key

You can configure a line key as a phone menu shortcut

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to add a menu shortcut to the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

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

where n is the extension number.

Step 4

In the Extended Function parameter, enter a string in this format:

fnc=shortcut;url=userpref;nme=User Preferences

where:

- fnc= shortcut means function=phone menu shortcut.

- url= userpref is the menu to open with this line key. It's the User preferences menu in this example. For more shortcut mappings, see Menu Shortcuts Mapping on PLK and PSK.

- nme= XXXX is the menu shortcut name displayed on the phone. If you don't specify a display name, the line key displays the target menu item. In the example, the line key displays User preferences.

You can also configure this parameter in the configuration file (cfg.xml). Enter a string in this format:

<Extended_Function_n_ ua="na">fnc=shortcut;url=userpref;nme=User Preferences</Extended_Function_n_>

where n is the extension number.

Step 5

Click Submit All Changes.

Menu Shortcuts Mapping on PLK and PSK

Menu Shortcuts Mapping

Function

(fnc=)

URL String

(url=)

Target Menu

shortcut

settings

Settings

shortcut

accessibility

Settings > Accessibility

shortcut

recents

Settings > Recents

shortcut

allcalls

Settings > Recents > All calls

shortcut

missedcalls

Settings > Recents > Missed calls

shortcut

receivedcalls

Settings > Recents > Received calls

shortcut

placedcalls

Settings > Recents > Placed calls

shortcut

speeddials

Settings > Speed dials

shortcut

Cisco IP Phone 6871 only

executive

Settings > Executive

shortcut

Cisco IP Phone 6871 only

assistant

Settings > Assistant

shortcut

userpref

Settings > User preferences

shortcut

callpref

Settings > User preferences > Call preferences

shortcut

cfwsetting

Settings > User preferences > Call preferences > Call forwarding

shortcut

anywhere

Settings > User preferences > Call preferences > Anywhere

shortcut

audiopref

Settings > User preferences > Audio preferences

shortcut

screenpref

Settings > User preferences > Screen preferences

shortcut

screensaver

Settings > User preferences > Screen preferences > Screen saver

shortcut

attconsole

Settings > User preferences > Attendant console preferences

shortcut

ringtone

Settings > User preferences > Ringtone

shortcut

bluetooth

Settings > Bluetooth

shortcut

networkconf

Settings > Network configuration

shortcut

ethernetconf

Settings > Network configuration > Ethernet configuration

shortcut

Cisco IP Phone 6861 only

wificonf

Settings > Network configuration > Wi-Fi configuration

shortcut

Cisco IP Phone 6861 only

wifiprofile

Settings > Network configuration > Wi-Fi configuration > Wi-Fi profile

shortcut

Cisco IP Phone 6861 only

wpspbc

Allows the phone to connect to a wireless router with WPS button.

shortcut

Cisco IP Phone 6861 only

wpspin

Allows the phone to connect to a wireless router with a WPS PIN code.

shortcut

Cisco IP Phone 6861 only

wifistatus

Settings > Network configuration > Wi-Fi configuration > Wi-Fi status

shortcut

ipv4setting

Settings > Network configuration > IPv4 address settings

shortcut

ipv6setting

Settings > Network configuration > IPv6 address settings

shortcut

adminsetting

Settings > Device administration

shortcut

setpassword

Settings > Device administration > Set password

shortcut

usersignin

Settings > Device administration > Sign in

shortcut

usersignout

Settings > Device administration > Sign out

shortcut

datetime

Settings > Device administration > Date/Time

shortcut

language

Settings > Device administration > Language

shortcut

restart

Settings > Device administration > Restart

shortcut

powersave

Settings > Device administration > Power save

shortcut

factoryreset

Settings > Device administration > Factory reset

shortcut

profilerule

Settings > Device administration > Profile rule

shortcut

profileaccount

Settings > Device administration > Profile account setup

shortcut

status

Settings > Status

shortcut

productinfo

Settings > Status > Product information

shortcut

networkstatus

Settings > Status > Network status

shortcut

ipv4status

Settings > Status > Network status > IPv4 status

shortcut

ipv6status

Settings > Status > Network status > IPv6 status

shortcut

phonestatus

Settings > Status > Phone status

shortcut

phonestat

Settings > Status > Phone status > Phone status

shortcut

linestatus

Settings > Status > Phone status > Line status

shortcut

provstatus

Settings > Status > Phone status > Provisioning

shortcut

callstat

Settings > Status > Phone status > Call statistics

shortcut

reportproblem

Settings > Status > Report problem

shortcut

reboothistory

Settings > Status > Reboot history

shortcut

accessories

Settings > Status > Accessories

shortcut

statusmessage

Settings > Status > Status messages

shortcut

Cisco IP Phone 6861 only

wifimessage

Settings > Status > Wi-Fi messages

shortcut

directories

Directories

shortcut

personaldir

Directories > Personal address book

shortcut

alldir

Directories > All

shortcut

ldapdir

Directories > Corporate directory (LDAP)

The LDAP directory name is customizable.

shortcut

broadsoftdir

Directories > BroadSoft directory

The BroadSoft directory name is customizable.

shortcut

bsdirpers

Directories > BroadSoft directory > Personal

The BroadSoft directory name is customizable.

shortcut

bsdirgrp

Directories > BroadSoft directory > Group

The BroadSoft directory name is customizable.

shortcut

bsdirent

Directories > BroadSoft directory > Enterprise

The BroadSoft directory name is customizable.

shortcut

bsdirgrpcom

Directories > BroadSoft directory > Group common

The BroadSoft directory name is customizable.

shortcut

bsdirentcom

Directories > BroadSoft directory > Enterprise common

The BroadSoft directory name is customizable.

shortcut

xmppdir

Directories > IM&P contacts

The XMPP directory name is customizable.

shortcut

xmlapp

Settings > Cisco XML services

The XML application name is customizable.

shortcut

xmldir

Directories > Corporate directory (XML)

The XML directory name is customizable.

shortcut

webexdir

Directories > Webex directory

The Webex directory name is customizable. By default, the softkey displays the directory name as Webex Dir.

shortcut

proxyset

Settings > Network configuration > HTTP proxy settings

shortcut

vpnstatus

Settings > Status > VPN status

shortcut

vpnsetting

Settings > Network configuration > VPN settings

Add an Extended Feature to a Line Key

You can add a feature to a line key. Then, the user can press the configured line key to access the feature. For the supported features, see Configurable Features on Line Keys.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to add an extended function to the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

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

where n is the extension number.

Step 4

In the Extended Function parameter, enter a string in this format:

fnc=cfwd

where fnc= cfwd means function=Call forward.

You can also configure this parameter in the configuration file (cfg.xml). Enter a string in this format:

<Extended_Function_n_ ua="na">fnc=cfwd</Extended_Function_n_>

where n is the extension number.

Step 5

Click Submit All Changes.

Configurable Features on Line Keys

The following table lists the supported features on the line keys.

Configurable Features on Line Keys

Feature

Configuration Code

(fnc=)

Description and String Syntax

Agent Sign-in and Sign-out

acd

Signs the user into or out of Automatic Call Distribution (ACD).

The ACD status can be Available automatically when the user signs into ACD.

For more information, see the description for Auto Available After Sign-In in Parameters for Call Center Agent Setup.

String Syntax:

fnc=acd[;nme=<softkey_display_name>]

Bridge in

Cisco IP Phone 6871 only

bridgein

Joins the user (executive) to an ongoing call with an assistant.

Only available to executives who have assistants.

String Syntax:

fnc=bridgein[;nme=<softkey_display_name>]

Busy lamp field

blf

Monitors a coworker's line status.

String Syntax:

fnc=blf[;nme=<softkey_display_name>]

Call pickup

blf;cp

Answers incoming calls for the monitored line.

String Syntax:

fnc=blf;cp[;nme=<softkey_display_name>]

BLF, Call pickup, and Speed dial

blf;cp;sd

Calls the monitored line or answers incoming calls for the monitored line.

String Syntax:

fnc=blf;cp;sd[;nme=<softkey_display_name>]

Call information

callinfo

Shows the information of the ongoing call.

String Syntax:

fnc=callinfo[;nme=<softkey_display_name>]

Call list

calllist

Provides access to the call list while on a connected video call.

String Syntax:

fnc=calllist[;nme=<softkey_display_name>]

Call push

Cisco IP Phone 6871 only

callpush

Only available to assistant lines.

Transfers an ongoing call from the user (assistant) to the executive.

String Syntax:

fnc=callpush[;nme=<softkey_display_name>]

Call retrieve

Cisco IP Phone 6871 only

callretrieve

Only available to executive lines.

Transfers an ongoing call from the assistant to the user (executive).

String Syntax:

fnc=callretrieve[;nme=<softkey_display_name>]

Call forward

cfwd

Turns off call forward or opens the Call forward settings screen.

String Syntax:

fnc=cfwd[;nme=<softkey_display_name>]

Call return

lcr

Redials the last missed call or the last received call (if there's no missed calls). If there's no missed or received calls in the history, it opens the dialing screen.

String Syntax:

fnc=lcr[;nme=<softkey_display_name>]

Divert

Cisco IP Phone 6871 only

divert

Only available to executive assistants.

Activates call diversion for the user (assistant).

String Syntax:

fnc=divert[;nme=<softkey_display_name>]

Do not disturb

dnd

Turn do not disturb (DND) on or off.

String Syntax:

fnc=dnd[;nme=<softkey_display_name>]

Executive-assistant

Cisco IP Phone 6871 only

bw-exec-assist

For executives, show the call filtering status, and open the Executive phone screen.

For assistants, show if the call diversion is activated, and open the Assistant phone screen.

String Syntax:

fnc=bw-exec-assist[;nme=<softkey_display_name>]

Menu shortcut

shortcut

Opens the specified menu item.

String Syntax:

fnc=shortcut;url=<menu_shortcut_code>[;nme=<softkey_display_name>]

where,

- fnc=shortcut means function=phone menu shortcut.

- url is the menu to open with this line key. For more shortcut mappings, see Menu Shortcuts Mapping on PLK and PSK.

- nme is the menu shortcut name displayed on the phone. If you don't specify a display name, the line key displays the target menu item.

Message Waiting Indicator

mwi

Monitors the voicemail of a user or group.

String syntax:

fnc=mwi;sub=<group_vm>@<domain>[;vid=<n>][;nme=<softkey_display_name>]

- fnc=mwi means function=Message Waiting Indicator

- sub=group_vm@domain is the SIP URI of a voicemail account that MWI monitors.

- vid is the extension ID with which the MWI associates. It's an optional string.

- nme is the name displayed on the phone for MWI line key. It's an optional string.

For more information, see String Syntax for Voicemail PLK.

Message Waiting Indicator + Speed dial

mwi+sd

Monitors and enables speed dial for voicemail of a user or group.

String syntax:

fnc=mwi+sd;ext=<number> ,<id>#,<pin>#@<domain>;
sub=<group_vm>@<domain>[;vid=<n>][;nme=<softkey_display_name>]

- fnc=mwi+sd means function=Message Waiting Indicator and speed dial

- ext is made up of:

- ext is the speed dial number.

If the SIP proxy requires a domain, add a domain for the speed dial number. For example, 8000@doman.

- ,id#,pin# consists of DTMF characters, where id and pin are the voicemail account ID and PIN respectively. You need a space between the speed dial number and the voicemail ID. This part of string is optional. We don't recommend that you add the PIN in the string.

For more information about the speed dial string, see DTMF Wait and Pause Parameters.

- sub=group_vm@domain is the SIP URI of a voicemail account that MWI monitors.

- vid is the extension ID with which the PLK associates. It's an optional string.

- nme is the name displayed on the phone for MWI+speed dial line key. It's an optional string.

For more information, see String Syntax for Voicemail PLK.

Proxy call

Cisco IP Phone 6871 only

proxycall

Only available to assistant lines.

Initiates a call on behalf of the selected executive.

String Syntax:

fnc=proxycall;ext=<number>[;vid=<n>][;nme=<softkey_display_name>]

where,

- fnc=proxycall means function = proxy call.

- ext is the executive's phone number or extension number for which the line serves as the proxy.

- vid is the line index associated with the executive's extension number. It's an optional string.

- nme is the display name of the proxy line. It's an optional string.

Speed dial

sd

Dials the specified speed dial number.

String syntax:

fnc=sd;ext=<number>@$PROXY[;vid=<n>][;nme=<softkey_display_name>]
where,

- fnc=sd means function=speed dial

- ext is the phone that the line key calls.

- vid is the line index of the phone. It's an optional string.

- nme is the name displayed on the phone for the speed-dial line key. It's an optional string.

Inert

inert

Shuts down the line key to completely disable it. The line key is unavailable when it is in the Inert mode.

String syntax:

fnc=inert;

Add a Menu Shortcut to a Programmable Softkey

You can configure a softkey as a phone menu shortcut.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Programmable Softkeys section, set the Programmable Softkey Enable field to Yes.

You can also configure the parameter in the configuration file (cfg.xml) with a string in this format:

<Programmable_Softkey_Enable ua="rw">Yes</Programmable_Softkey_Enable>

Step 3

Configure a PSK field from PSK 1 through PSK 16 with a string in this format:

fnc=shortcut;url=userpref;nme=User preferences

where:

- fnc= shortcut means function=phone menu shortcut.

- url= userpref is the menu to open with this line key. It's the User preferences menu in this example. For more shortcut mapping, see Menu Shortcuts Mapping on PLK and PSK.

- nme= XXXX is the menu shortcut name displayed on the phone. In the example, the softkey displays User preferences.

You can also configure this parameter in the configuration file (cfg.xml). Enter a string in this format:

<PSK_n ua="rw">fnc=shortcut;url=userpref;nme=User preferences</PSK_n>

where n is the PSK number.

Step 4

Add the configured PSK to the desired key list.

Example: Add the configured PSK 2 to Idle Key List. Do any of these actions:

- Add psk2 to the Idle Key List field.

psk2;em_login;acd_login;acd_logout;astate;redial;cfwd;dnd;lcr;

- In the configuration file (cfg.xml), enter a string in this format:

<Idle_Key_List ua="rw">psk2;em_login;acd_login;acd_logout;astate;redial;cfwd;dnd;lcr;</Idle_Key_List>

Step 5

Click Submit All Changes.

Enable LDAP Unified Search

You can enable the unified search in the LDAP directory. The search allows you to enter any value as filters. For example, first name, last name, extension, or phone number. The phone transfers the request as a single search request.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Browse Mode Enable parameter set to Yes or No.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the LDAP section, set the parameter Unified Search Enable to Yes to enable the LDAP unified search. If the parameter is set to Yes, the phone transfers requests with OR filter.

If you set the value to No, the phone uses simple or advanced search and transfers requests with AND filter.

Default value is No.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<LDAP_Unified_Search_Enable>Yes</LDAP_Unified_Search_Enable>

Conditions based on Browse Mode Enable and Unified Search Enable parameter values:

- Browse Mode Enable parameter is No and Unified Search Enable parameter is No－when the user chooses the LDAP directory on the phone, the Query LDAP server screen displays Simple search and Advanced search menus.

- Browse Mode Enable parameter is No and Unified Search Enable parameter is Yes－when the user chooses the LDAP directory, the phone navigates to the LDAP query form (unified search screen) directly. If there is no value in the search box, the search displays all contacts in the directory.

- Browse Mode Enable parameter is Yes and Unified Search Enable parameter is No－when the user navigates to the LDAP directory and clicks the Option softkey, the phone displays the Simple search and the Advanced search menus.

- Browse Mode Enable parameter is Yes and Unified Search Enable parameter is Yes－when the user navigates to the LDAP directory and clicks the Option softkey, the phone displays only one Search menu. After clicking the Search menu, the unified search screen LDAP query form appears.

Step 3

Click Submit All Changes.

Shut Down a Line Key

You can shut down a line key by setting the Inert mode for it from the phone web page. When the line key is in the Inert mode, it is disabled thoroughly. For example, the line key LED is disabled, no icon or text displays next to the line key, and the line key button is not responsive. In a word, it is completely unavailable.

Before you begin

Access the phone administration web interface. See Access the Phone Web Interface

Procedure

Step 1

Disable the line key that will use the Inert mode. Do any of the following actions on the phone web page:

- Set Extension to Disabled for the specific line key on the Voice > Phone tab.

You can also disable the line key in the configuration file (cfg.xml):

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

where n is the extension number.

- Set Line Enable to No for the specific line key on the Voice > Ext(n) tab.

You can also disable the line key in the configuration file (cfg.xml):

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

where n is the extension number.

Step 2

(Optional) If you don't want to manually disable the line key as described in Step 1, you can enable the Direct PLK Configuration feature. For more information, see Enable Direct PLK Configuration.

Step 3

Set the Inert mode for the specific line key.

1. Select Voice > Phone.

2. Select the target line key.

3. In the Extended Function parameter, enter a string in this format:

fnc=inert;

where fnc=inert means function=inert.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Extended_Function_n_ ua="na">fnc=inert;</Extended_Function_n_>

where n is the extension number.

4. Click Submit All Changes.

Chapter 11. Phone Information and Display Configuration

- Phone Information and Display Settings

- Configure the Phone Name

- Customize the Startup Screen

- Customize Wallpaper for the Phone Display

- Configure the Screen Saver with the Phone Web Interface

- Adjust Backlight Timer from the Phone Web Interface

- Customize the Product Configuration Version

- Keep Focus on the Active Call

- Report Headset Inventory

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

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Under General, enter the phone name in the Station Display Name field.

This name displays on the phone screen. You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Station_Display_Name ua="na">Recetion Desk</Station_Display_Name>

Step 3

Click Submit All Changes.

Customize the Startup Screen

You can create a text or an 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.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Click Voice > User.

Step 2

In the Screen section, select any option from the Boot Display field.

- Default: Displays a blank screen or existing screen as the startup screen.

- Download Picture: Displays a picture as the startup screen. Enter the path in the Picture Download URL field.

- Logo: Displays a logo as the startup screen. Enter the path in the Logo URL field.

- Text: Displays a text as the startup screen. Enter text in the Text Display field.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Boot_Display ua="na">Logo</Boot_Display>

The allowed values are Default|Download Picture|Logo|Text. The default option is Default.

Step 3

To display a picture or a logo, enter the path in the Picture Download URL or Logo URL field.

For example:

http://10.64.84.147/pictures/image04.png

When you enter an incorrect URL to download the image, the phone fails to upgrade to the new image and displays the existing image. If the phone does not have any image downloaded earlier, it displays a gray screen.

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. The logo display area is at the mid-center of the phone screen. The display area sizes are:

- 6821 and 6861: 48x48 pixels

- 6841 and 6851: 64x64 pixels

- 6871: 74x40 pixels

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Picture_Download_URL ua="na">http://10.64.84.147/pictures/bootimage1.jpg</Picture_Download_URL>

<Logo_URL ua="na">http://10.64.84.147/pictures/logo_image.jpg</Logo_URL>

Step 4

To display text at bootup, enter the text to display in the Text Display field following the requirements:

- Enter up to two lines of text with less than 32 characters for each line.

- Insert a new line character (\n) and escape code (%0a) between the two lines.

For example,

Super\n%0aTelecom
displays:

Super
Telecom

- Use the + character to add spaces for formatting. You can add multiple + characters before and after the text to center it.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Text_Display ua="na">Super\n%0aTelecom</Text_Display>

Step 5

Click Submit All Changes.

The phone reboots, retrieves the image file, and displays the picture, logo, or text when it boots next time.

Customize Wallpaper for the Phone Display

You can set the phone to display a custom logo or picture as the background on the phone screen.

Procedure

Step 1

On the phone web interface, select Voice > User.

User can also change the wallpaper in the phone web interface.

Step 2

In the Screen section, choose one of the options for the Phone Background field:

- Default—Keeps the system default background.

- Download Picture—Displays a picture downloaded from a TFTP, FTP, or HTTPS server. When select this option, enter the URL for the picture in the Picture Download URL field.

Only Cisco IP Phone 6871 Multiplatform Phones support displaying custom picture as the background.

- Logo—Displays a logo downloaded from a TFTP, FTP, or HTTPS server. When select this option, enter the URL for the logo image in the Logo URL field.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Phone_Background ua="na">Logo</Phone_Background>

Step 3

Upload the custom wallpaper to a TFTP, HTTP, or HTTPS server.

The image is a .jpg or .png file. Preferred dimension is 480x272 pixels for Cisco IP Phone 6871. If the image is not the preferred size, user still can upload it but it will resize to fit the screen.

Step 4

In the Picture Download URL field, enter the path where the wallpaper image has been uploaded.

The URL must include the TFTP, HTTP, or HTTPS server name (or IP address), directory, and file name. Don't exceed 255 characters for the URL.

Example:

http://10.64.84.147/pictures/image04.jpg

When you enter an incorrect URL to download a new wallpaper, the phone fails to upgrade to the new wallpaper and displays the existing downloaded wallpaper. If the phone does not have any wallpaper downloaded earlier, it displays a gray screen.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Picture_Download_URL ua="na">http://10.64.84.147/pictures/image04.jpg</Picture_Download_URL>

Step 5

Upload the logo image to a TFTP, HTTP, or HTTPS server.

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. The logo display area is at the mid-center of the phone screen. The display area sizes are:

- 6821 and 6861: 48x48 pixels

- 6841 and 6851: 64x64 pixels

- 6871: 74x40 pixels

Step 6

In the Logo URL field, enter the path where the logo image has been uploaded.

The URL must include the TFTP, HTTP, or HTTPS server name (or IP address), directory, and file name. Don't exceed 255 characters for the URL.

Example:

http://10.64.84.147/pictures/logo_image.jpg

When you enter an incorrect URL to download a new logo, the phone fails to upgrade to the newer logo and displays the existing downloaded logo. If the phone does not have any logo downloaded earlier, it displays a gray screen.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Logo_URL ua="na">http://10.64.84.147/pictures/logo_image.jpg</Logo_URL>

Step 7

Click Submit All Changes.

The phone reboots after you change the background image URL.

Configure the Screen Saver with the Phone Web Interface

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.

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 Screen Saver.

Before you begin

Access the phone administration web interface. See Access the Phone Web Interface.

Procedure

Step 1

On the phone web page, select Voice > User.

The user can select User Login > Voice > User to add screen saver to the phone.

Step 2

In the Screen section, set up the fields as described in Parameters for Screen Saver.

Step 3

Click Submit All Changes.

- Parameters for Screen Saver

Parameters for Screen Saver

The following table defines the function and usage of the screen saver parameters in the Screen section under theVoice> User tab in the phone web interface. It also definesthe syntax of the stringthat is added in thephone configuration file (cfg.xml) with XML code to configure a parameter.

Parameters for Screen Saver

Parameter

Description

Screen Saver Enable

Select Yes to enable a screen saver on the phone. When the phone is idle for a specified time, it enters screen saver mode.

Perform one of the following:

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

<Screen_Saver_Enable ua="rw">Yes</Screen_Saver_Enable>

- In the phone web interface, set this field to Yes to enable screen saver.

Allowed values: Yes|No

Default: No

Screen Saver Type

Types of screen saver. Options you can choose:

- Clock—Displays a digital clock on a plain background.

- Download Picture—Displays a picture pushed from the phone webpage. Enter the image path in the Picture Download URL field.

- Logo: Displays a logo on the phone screen. Add a logo image in the Logo URL field.

Perform one of the following:

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

<Screen_Saver_Type ua="rw">Clock</Screen_Saver_Type>

- In the phone web interface, select a screen saver.

Allowed values: Clock|Download Picture|Logo

Default: Clock

Screen Saver Wait

Amount of idle time before screen saver displays.

Enter the number of seconds of idle time to elapse before the screen saver starts.

Perform one of the following:

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

<Screen_Saver_Wait ua="rw">300</Screen_Saver_Wait>

- In the phone web interface, set the time in seconds.

Allowed values: An integer from 30 through 65000

Default: 300

Picture Download URL

URL locating the (.png) file to display on the phone screen background. The image can display as the screen background, the screensaver, or at bootup depending on the settings of the Phone Background, Screen Saver Type, or Boot Display field.

When you enter an incorrect URL to download a new image, the phone fails to update to the new image and displays the existing downloaded image. If the phone does not have any image downloaded earlier, it displays a gray screen.

Perform one of the following:

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

<Picture_Download_URL ua="rw">http://10.74.3.52/images/screensaver1.png</Picture_Download_URL>

- In the phone web interface, specify the URL where the picture is located.

Allowed values: A valid URL not exceeding 255 characters

Default: Empty

Logo URL

Enter a URL or path for the location where the logo image is saved. The logo image can display as the screen background, the screensaver, or at bootup depending on the settings of the Screen Saver Type, Boot Display, or Phone Background field.

Perform one of the following:

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

<Logo_URL ua="rw">http://10.74.3.52/images/Logo1.png</Logo_URL>

- In the phone web interface, specify the URL where the logo image is located.

Allowed values: A valid URL not exceeding 255 characters

Default: Empty

Adjust Backlight Timer from the Phone Web Interface

You can save energy by disabling the backlight on each phone at a preset time.

Procedure

Step 1

Select Voice > User.

Step 2

In the Screen section, select a duration for the Back Light Timer parameter.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Back_Light_Timer ua="rw">30s</Back_Light_Timer>

- For 6821, 6841, 6851, and 6861: The allowed values are Off|10s|20s|30s|Always On. The default value is 30s (30 seconds).

- For 6871: The allowed values are 1m|5m|30m|Always On. The default value is 5m (5 minutes).

Step 3

Click Submit All Changes.

Customize the Product Configuration Version

You can customize the configuration version of the product in the phone configuration file (cfg.xml). After the change takes effect, the user can view the configuration version of the product information on the phone.

Procedure

Step 1

Edit the phone configuration file (cfg.xml) in a text or XML editor.

Step 2

Add a value for the element <Device_Config_Version> in the cfg.xml file.

For example:

<Device_Config_Version ua="na">2021-01-05-v1</Device_Config_Version>

Default: Empty

Value range: 0 to 64 characters

If the tag doesn't exist in the cfg.xml file or the parameter value is empty, then the Configuration version menu item doesn't display on the phone screen Product information.

Note

For the Cisco IP Phone 6871, if the length of the assigned characters exceeds the width of the phone screen, the exceeded characters are truncated and represented as an ellipsis (...) on the phone screen.

Step 3

Save the changes to the cfg.xml file.

Keep Focus on the Active Call

You can configure the phone to ensure that the active call is still in focus when the user has an incoming call.
By default, the focus on the phone screen automatically moves from the active call to the incoming call. However, you can configure the phone to ensure that the active call always remains in focus, even when the user has an incoming call.

The focus still moves to an incoming call in the following situations:

- The user places an active call on hold and then receives one or more incoming calls, the focus automatically moves to the first incoming call.

- The user is on an active call and receives one or more incoming calls, if the user places the active call on hold, then the focus automatically moves to the first incoming call.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

In the Supplementary Services section, set the parameter Keep Focus On Active Call to Yes.

You can also configure this parameter in the configuration file:

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

Allowed values: Yes and No

Default: No

Step 3

Click Submit All Changes.

Report Headset Inventory

You can configure a phone to report the connected or disconnected peripheral information to the server. The peripherals that the Cisco IP Phone Multiplatform Phones support are Key Expansion Module (KEM) and Cisco headset.

The supported Cisco headsets are Cisco Headset 500 Series and Cisco Headset 700 Series.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the Peripheral section, set the Peripheral Inventory Enable parameter to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

When the parameter is set to Yes, the peripheral inventory headers are included in the SIP Register message. When set to No, the headers are not included in the SIP message. Default value of the parameter is No.
When one peripheral is connected or disconnected to the phone, next scheduled Register provides the peripheral information in the Peripheral-Data header. All subsequent Registers do not carry peripheral information. The Peripheral-Data header is included for each peripheral, for example, if there are two headsets present, the header appears twice.

Step 3

Click Submit All Changes.

Chapter 12. Call Features Configuration

The phone web user interface and the xml configuration files allows you to customize calling features of your phone such as call transfer, call park, conferencing, and speed-dial.

- Enable Call Transfer

- Call Forward

- Enable Feature Activation Code Synchronization for Forward All Calls

- Enable Conferencing

- Enable Remote Call Recording with SIP REC

- Enable Remote Call Recording with SIP INFO

- Configure Missed Call Indication

- Enable Do Not Disturb

- Enable Synchronization of Settings Between the Phone and the Server

- Enable Webex Contacts on the Phone

- Configure Webex Contacts on a Line Key

- Add a Softkey for Webex Contacts

- Enable Webex Call Logs on the Phone

- Configure Star Codes for DND

- Set Up a Call Center Agent Phone

- Set Up a Phone for Presence

- Configure the Number of Call Appearances Per Line

- Enable Reverse Name Lookup

- Emergency Calls

- PLK Configuration

- Spam Indication for Incoming Webex Calls

- Programmable Softkeys Configuration

Enable Call Transfer

You can enable attended call transfer and blind call transfer services for your user.

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 Enable Call Transfer table.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Under Supplementary Services, configure the parameters as defined in the Parameters for Enable Call Transfer table.

Step 3

Click Submit All Changes.

- Parameters for Enable Call Transfer

Parameters for Enable Call Transfer

The following table defines the function and usage of Enable Call Transfer parameters in the Supplementary Services section under the Phone 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.

Parameters for Enable Call Transfer

Parameter

Description

Attn Transfer Serv

Attended call transfer service. The user answers the call before transferring it.

Perform one of the following:

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

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

- In the phone web page, select Yes to enable the transfer service. Select No to disable it.

Options: Yes and No

Default: Yes

Blind Transfer Serv

Blind call transfer service. The user transfers the call without speaking to the caller.

Perform one of the following:

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

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

- In the phone web page, select Yes to enable the transfer service. Select No to disable it.

Options: Yes and No

Default: Yes

Call Forward

To enable call forward, you can enable the feature in two places: on the Voice tab and the User tab of the phone web page.

- Enable Call Forward on Voice Tab

- Enable Call Forward on User Tab

Enable Call Forward on Voice Tab

Perform this task if you want to enable call forward for a user.

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 Parameters for Enable Call Forward on Voice Tab table.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Under Supplementary Services, configure the parameters as described in the Parameters for Enable Call Forward on Voice Tab table.

Step 3

Click Submit All Changes.

- Parameters for Enable Call Forward on Voice Tab

Related Tasks

Enable Feature Key Sync

Enable Call Forward Status Sync via XSI Service

Related References

DND and Call Forward Status Sync

Parameters for Enable Call Forward on Voice Tab

The following table defines the function and usage of Enable Call Forward on Voice Tab parameters in the Supplementary Services section under the Phone 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.

Parameters for Enable Call Forward on Voice Tab

Parameter

Description

Cfwd All Serv

Forwards all calls.

Perform one of the following:

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

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

- In the phone web page, select Yes to forward all calls. Select No to disable it.

Options: Yes and No

Default: Yes

Cfwd Busy Serv

Forwards calls only if the line is busy.

Perform one of the following:

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

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

- In the phone web page, select Yes to forward calls when the line is busy. Select No to disable it.

Options: Yes and No

Default: Yes

Cfwd No Ans Serv

Forwards calls only if the line is not answered.

Perform one of the following:

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

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

- In the phone web page, select Yes to forward calls if the line is not answered. Select No to disable it.

Options: Yes and No

Default: Yes

Enable Call Forward on User Tab

Perform the following task to change the call forward settings from the phone web page.

The settings of call forward are synchronized between the phone and the server when one of the following ways is enabled:

- Feature key synchronization (FKS)

- BroadSoft's Extended Services Interface (XSI) Synchronization

To ensure the settings of call forward on the local phone take effect, you need to disable FKS and XSI first. See Enable Feature Key Sync and Enable Call Forward Status Sync via XSI Service.

The priority of taking effect for call forward setting in the supported modes is: FKS > XSI > Local.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Ensure that the call forward setting is enabled on the Voice tab. See Enable Call Forward on Voice Tab.

Procedure

Step 1

Select Voice > User.

Step 2

In the Call Forward section, configure the parameters as described in the Parameters for Enable Call Forward on User Tab table.

Step 3

Click Submit All Changes.

- Parameters for Enable Call Forward on User Tab

Parameters for Enable Call Forward on User Tab

The following table defines the function and usage of Voice > User > Call Forward in 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.

Parameters for Enable Call Forward on User Tab

Parameter

Description

Cfwd All

Forwards all calls. The setting of this parameter takes precedence over Cfwd Busy and Cfwd No Answer.

Perform one of the following:

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

<Cfwd_All ua="rw">No</Cfwd_All>

- In the phone web page, select Yes to forward all calls. Select No to disable it.

Options: Yes and No

Default: No

Cfwd All Dest

Specifies the destination to which all calls are forwarded. The destination can be an alphanumeric input, a phone number, or a SIP URI.

Perform one of the following:

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

<Cfwd_All_Dest ua="rw">DestinationNumber</Cfwd_All_Dest>

- In the phone web page, enter the destination number in the field.

When you select Yes for Cfwd All, make sure to configure the parameter.

Default: Empty

Cfwd Busy

Forwards calls only if the line is busy.

Perform one of the following:

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

<Cfwd_Busy ua="rw">No</Cfwd_Busy>

- In the phone web page, select Yes to forward calls when the line is busy. Select No to disable it.

Options: Yes and No

Default: No

Cfwd Busy Dest

Specifies the destination to which calls are forwarded if the line is busy. The destination can be an alphanumeric input, a phone number, or a SIP URI.

Perform one of the following:

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

<Cfwd_Busy_Dest ua="rw">DestinationNumber</Cfwd_Busy_Dest>

- In the phone web page, enter the destination number in the field.

When you select Yes for Cfwd Busy, make sure to configure the parameter.

Default: Empty

Cfwd No Answer

Forwards the incoming call only if the call isn’t answered.

Perform one of the following:

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

<Cfwd_No_Answer ua="rw">No</Cfwd_No_Answer>

- In the phone web page, select Yes to forward the incoming call if the call isn't answered. Select No to disable it.

Options: Yes and No

Default: No

Cfwd No Ans Dest

Specifies the phone number of destination to which the incoming call is forwarded if the call isn’t answered. The destination can be an alphanumeric input, a phone number, or a SIP URI.

Perform one of the following:

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

<Cfwd_No_Answer_Dest ua="rw">DestinationNumber</Cfwd_No_Answer_Dest>

- In the phone web page, enter the destination number in the field.

When you select Yes for Cfwd No Answer, make sure to configure the parameter.

Default: Empty

Cfwd No Ans Delay

Assigns a response delay time (in seconds) for the no answer scenario.

Perform one of the following:

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

<Cfwd_No_Answer_Delay ua="rw">20</Cfwd_No_Answer_Delay>

- In the phone web page, enter the delay time in the field.

Default: 20

Forward Softkey

Controls the scope of the call forward services that the user can set up by a dedicated softkey. Options are:

- All Cfwds: Allows the user to set up all call forward services, including Call Forward All, Call Forward Busy, and Call Forward No Answer by pressing the Forward softkey.

In this setting, the softkey name is Forward for activation and Clr fwd for deactivation.

- Only the Cfwd All: Allows the user to directly set up the Call Forward All service by pressing the softkey Forward all.

The user can still set up all call forward services from the Settings > User preferences > Call preferences > Call forwarding > Call forward settings screen.

In this setting, the softkey name is Forward all for activation and Clr fwd all for deactivation.

Perform one of the following:

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

<Forward_Softkey ua="na">All Cfwds</Forward_Softkey>

- In the phone web page, select the value that determines the scope of the call forward services for the users.

Note

The parameter takes effect even though FKS, XSI, or FAC is enabled.

Default: All Cfwds

Enable Feature Activation Code Synchronization for Forward All Calls

You can synchronize forwarding all calls feature to the server with a Feature Activation Code (FAC). When you enable this feature, the FAC sends out the star code and the destination number with INVITE to the server.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext (n).

Step 2

In the Feature Activation Code Sync field, select Yes to enable the feature.

After you enable this feature, your user can press the Forward or Forward all softkey on the phone and enter the destination contact number. When the user presses the Call softkey, a voice message plays to confirm the call forward setting status. After successful configuration, a call forward icon displays at the top of the phone screen.

The softkey name is different based on the value of the parameter Forward Softkey, see Parameters for Enable Call Forward on User Tab.

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

<Feature_Activation_Code_Sync_n_ ua="na">Yes</Feature_Activation_Code_Sync_n_>
where, n is the extension number.

Default value: No

Allowed values: Yes or No

Step 3

Click Submit All Changes.

- Set Feature Activation Code for the Call Forward All Service

Set Feature Activation Code for the Call Forward All Service

You can set activation code (star code) that can be used to activate or deactivate the call forward all service.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

In the Vertical Service Activation Codes section, ensure that the Cfwd All Act Code field is set to the value defined by the server. The default value is *72.

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

<Cfwd_All_Act_Code ua="na">*72</Cfwd_All_Act_Code>

Step 3

In the Vertical Service Activation Codes section, ensure that the Cfwd All Deact Code field is set to the value defined by the server. The default value is *73.

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

<Cfwd_All_Deact_Code ua="na">*73</Cfwd_All_Deact_Code>

Step 4

Click Submit All Changes.

Your user can dial *72 in combination with the destination number and press the Call softkey to activate the call forward all service.

Your user can dial *73 and press the Call softkey to deactivate the call forward all service.

Enable Conferencing

You can enable your user to talk to several people in a single call. When you enable this feature, your user dials several people and add them to the call.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Under Supplementary Services, choose Yes for the Conference Serv parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: Yes

Step 3

Click Submit All Changes.

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 and Recording Softkeys

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.

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.

Recording Icons

Icon

Meaning

Recording in progress.

Recording paused

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Supplementary Services section, click Yes or click No to enable or to disable the Call Recording Serv parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: No

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

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.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Call_Recording_Protocol_3_ ua="na">SIPREC</Call_Recording_Protocol_3_>

Options: SIPREC and SIPINFO

Default: SIPREC

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.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

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.

Your user presses the following softkeys to control the phone recording:

- Record

- StopRec

The recording only starts when your user presses the Record softkey. Your user sees a message when recording state changes and the recording icon displays on the call screen.

Once a phone recording starts, the StopRec softkey can work. The recording stops when your user presses the StopRec softkey. Your user sees a message when the recording state changes.

Recording Icons

Icon

Meaning

Recording in progress.

Before you begin

- You need to set up call recording on the call control system.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Supplementary Services section, click Yes or click No to enable or to disable call recording in the Call Recording Serv parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: No

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

Click the Ext(n) tab that requires call recording.

Step 5

In the SIP Settings section, for the Call Recording Protocol parameter, select SIPINFO as the call recording protocol.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Call_Recording_Protocol_1_ ua="na">SIPINFO</Call_Recording_Protocol_1_>

Options: SIPREC and SIPINFO

Default: SIPREC

Step 6

Click Submit All Changes.

Configure Missed Call Indication

You can configure a missed call alert on the phone handset LED.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

The user can select User Login > Voice > User.

Step 2

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:

<Handset_LED_Alert ua="rw">Voicemail,Missed Call</Handset_LED_Alert>

Options: Voicemail and Voicemail, Missed Call

Default: Voicemail

Step 3

Click Submit All Changes.

Enable Do Not Disturb

You can allow persons to turn the Do not disturb feature on or off. The caller receives a message that the person is unavailable. A person can press the Ignore softkey on the phone to divert an incoming call to another destination.

If the feature is enabled for the phone, users can turn the feature on or off with the DND softkey.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

In the Supplementary Services area, for the DND Setting parameter, select Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<DND_Setting ua="rw">Yes</DND_Setting>

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

When you select a line (multiline phone), a Do Not Disturb banner displays at the top of the phone screen.

What to do next

Change another setting to ensure that multiline phones correctly display the Do not disturb (currently, a steady, green color) status for each selected or unselected line. See DND and Call Forward Status Sync.

Users can enable or turn off the DND feature for each phone line if you configure star codes for DND. See Configure Star Codes for DND.

Related Tasks

Enable Feature Key Sync

Enable DND Status Sync via XSI Service

Related References

DND and Call Forward Status Sync

Enable Synchronization of Settings Between the Phone and the Server

Enable the synchronization of settings between the phone and the server.

This setting must be enabled for the following features and types of users:

- Call forward all

- DND

- Executives and assistants (Cisco IP Phone 6871 only)

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

If a line key is configured with feature key sync and is also enabled with DND or call forward feature, the respective DND icon or the call forward icon is displayed next to the line key label. If the line key has a missed call, a voice message, or an urgent voicemail alert, the DND icon or the call forward icon is also displayed with the alert notification.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext [n] (where [n] is the extension number).

Step 2

In the Call Feature Settings section, set the Feature Key Sync parameter to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<!-- Call Feature Settings -->
<Feature_Key_Sync_1_ ua="na">Yes</Feature_Key_Sync_1_>

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Enable Webex Contacts on the Phone

When you onboard a phone to Webex cloud successfully, you can enable the phone to support Webex contacts. When you enable this feature on the phone, your user can see the Webex directory under the phone directory list.

When you configure Max Display Records parameter value more than 100, the query result displays only hundred contacts for a search in Webex directory and All directory. When search result has count more than the allowed display record value, user see a message: Too many matches found.Refine your search. For more information on Max Display Records parameter, see Parameters for Directory Services.

Before you begin

- Phone onboards to Cisco Webex cloud successfully. For more information on phone onboarding to Webex Cloud, see Webex for Cisco BroadWorks Solution Guide.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Webex section, set the Directory Enable to Yes.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Webex_Directory_Enable ua="na”>Yes</Webex_Directory_Enable>

Default value: No

Step 3

In the Directory Name field, enter a name for the Webex directory.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Webex_Directory_Name ua="na”>wcdir</Webex_Directory_Name>

Default value: Empty

The name you enter (for example, wcdir ), displays as the Webex directory name on the phone under the directory list. You can modify this name from the phone administration web page or from the configuration XML file string. When required, your user can also modify this name from the phone. When the Directory Name field is empty, by default, the Webex directory name on the phone appears as Webex directory.

When the phone is not onboarded to Cisco Webex cloud successfully, the Webex directory doesn't appear under the directory list.

Step 4

Click Submit All Changes.

Configure Webex Contacts on a Line Key

You can configure Webex contacts on a line key. This line key becomes a shortcut to the Webex directory.

Before you begin

- Phone onboards to Cisco Webex cloud successfully.

- Access the phone administration web page. See Access the Phone Web Interface.

- Directory Enable on the phone administration web page is set to Yes.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to configure Webex contacts on the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

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

where n is the extension number.

Step 4

In the Extended Function parameter, enter a string in this format:

fnc=shortcut;url=webexdir;nme=cloudplk

where fnc=shortcut means function=shortcut, url is the menu to open this line key, and nme is the name for the Webex directory.

In the string, when nme is empty or you don't include nme in the string, by default, the line key displays the directory name as Webex directory.

You can also configure this parameter in the configuration file (cfg.xml). Enter a string in this format:

<Extended_Function_n_ ua="na">fnc=shortcut;url=webexdir;nme=cloudplk</Extended_Function_n_>
where n is the extension number.

The line key is configured with the feature. For example, if you assign the feature in line key number nine, the user sees cloudplk appears in the line number nine as a shortcut to the Webex directory. By pressing this configured line key, user can access the Search Webex directory screen and can search the Webex contacts.

If Directory Enable on the phone administration web page is set to No, the line key doesn't work.

If the phone is not onboarded to the Webex cloud successfully, the line key doesn't work.

Step 5

Click Submit All Changes.

Add a Softkey for Webex Contacts

You can configure Webex contacts to a softkey. This softkey becomes a shortcut to the Webex directory.

Before you begin

- Phone onboards to Cisco Webex cloud successfully.

- Access the phone administration web page. See Access the Phone Web Interface.

- Directory Enable on the phone administration web page is set to Yes.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Programmable Softkeys section, set the Programmable Softkey Enable to Yes.

Step 3

Configure a PSK field from PSK 1 through PSK 16 with a string in this format:

fnc=shortcut;url=webexdir;nme=cloudplk
You can also configure this parameter in the configuration file (cfg.xml). Enter a string in this format:

<PSK_n ua=na>fnc=shortcut;url=webexdir;nme=cloudplk</PSK_n>

A softkey is configured with the feature and appears on the phone. For example, cloudplk appears as a softkey and acts as a shortcut to the Webex directory. By pressing this softkey, user can access the Search Webex directory screen and can search the Webex contacts.

In the string, when nme is empty or you don't include nme in the string, by default, the softkey displays the directory name as Webex Dir.

If Directory Enable on the phone administration web page is set to No, the softkey doesn't work.

If the phone is not onboarded to Cisco Webex cloud successfully, the softkey doesn't work.

Enable Webex Call Logs on the Phone

You can now enable a phone to support Webex call logs. When you enable this feature, the Display recents from menu under the Recents screen includes the Webex option in the calls list. The user then can set the option Webex to see the list of recent Webex calls.

Before you begin

- Phone onboards to Webex cloud successfully. For more information on phone onboarding to Webex cloud, see Webex for Cisco BroadWorks Solution Guide.

- Access the phone administration web page. See Access the Phone Web Interface.

- Under the Call Log section, enable the CallLog Enable parameter and select a phone line from CallLog Associated Line for which you want to display the Webex recent call logs.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Call Log section, set the CallLog Enable parameter to Yes and Display Recents From parameter to Webex.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

<Display_Recents_From ua="na”>Webex</Display_Recents_From>

Default value of Display Recents From : Phone

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.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

In the Vertical Service Activation Codes section, enter *78 for the DND Act Code parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<DND_Act_Code ua="na">*78</DND_Act_Code>

Step 3

In the Vertical Service Activation Codes section, enter *79 for the DND Deact Code parameter.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<DND_Deact_Code ua="na">*79</DND_Deact_Code>

Step 4

Click Submit All Changes.

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.

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 Parameters for Call Center Agent Setup table.

Before you begin

- Set up the phone as a call center phone on the BroadSoft server.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

In the ACD Settings section, set up the fields as described in Parameters for Call Center Agent Setup table.

Step 3

Click Submit All Changes.

- Parameters for Call Center Agent Setup

- Restore the ACD Status

- Display or Hide Unavailable Menu Text Box of Agent Status on the Phone

Parameters for Call Center Agent Setup

The following table defines the function and usage of Call Center Agent Setup parameters in the ACD Settings section under the Ext(n) 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.

Parameters for Call Center Agent Setup

Parameter

Description

Broadsoft ACD

Enables the phone for Automatic Call Distribution (ACD).

Perform one of the following:

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

<Broadsoft_ACD_1_ ua="na">Yes</Broadsoft_ACD_1_>

- In the phone web page, select Yes to enable this feature and select No to disable it.

Options: Yes and No

Default: No

Call Information Enable

Enables the phone to display details of a call center call.

Perform one of the following:

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

<Call_Information_Enable_1_ ua="na">Yes</Call_Information_Enable_1_>

- In the phone web page, select Yes to enable this feature. Select No to disable it.

Options: Yes and No

Default: Yes

Disposition Code Enable

Enables the user to add a disposition code.

Perform one of the following:

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

<Disposition_Code_Enable_1_ ua="na">Yes</Disposition_Code_Enable_1_>

- In the phone web page, select Yes to enable this feature. Select No to disable it.

Options: Yes and No

Default: Yes

Trace Enable

Enables the user to trace the last incoming call.

Perform one of the following:

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

<Trace_Enable_1_ ua="na">Yes</Trace_Enable_1_>

- In the phone web page, select Yes to enable this feature. Select No to disable it.

Options: Yes and No

Default: Yes

Emergency Escalation Enable

Enables the user to escalate a call to a supervisor in case of emergency.

Perform one of the following:

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

<Emergency_Escalation_Enable_1_ ua="na">Yes</Emergency_Escalation_Enable_1_>

- In the phone web page, select Yes to enable this feature. Select No to disable it.

Options: Yes and No

Default: Yes

Queue Status Notification Enable

Displays the call center status and the agent status.

Perform one of the following:

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

<Queue_Status_Notification_Enable_1_ ua="na">Yes</Queue_Status_Notification_Enable_1_>

- In the phone web page, select Yes to enable this feature. Select No to disable it.

Options: Yes and No

Default: Yes

Auto Available After Sign-In

Sets the agent status to Available automatically when the user signs into the phone as a call center agent.

Perform one of the following:

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

<Auto_Available_After_Sign-In_1_ ua="na">Yes</Auto_Available_After_Sign-In_1_>

- In the phone web page, select Yes to enable this feature and select No to disable it.

Options: Yes and No

Default: No

Restore the ACD Status

You can enable the phone to automatically set the ACD status to the last local value in one of the following situations:

- Phone is powered on.

- Phone status is changed to “Registered” from “Unregistered” or “Registration failed” status.

- Registration destination server IP address is changed when failover happens, a fallback happens, or a DNS response is changed.

Before you begin

- Set up the phone as a call center phone on the BroadSoft server.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

In the ACD Settings section, set BraodSoft ACD to Yes.

Step 3

From ACD Status field, select one of the options:

- Sync From Local: Select this option to restore the last local status as ACD status when the phone boots up, status is changed to "Registered" from "Unregistered" or "Registration failed", or registration destination ip address is changed due to failover, fallback or DNS response is changed.

When the intial ACD status is configured to sync from local, and the last local status is unavailable with a reason code, after the phone boots up, the reason code will not be restored.

- Sync From Server: Select this option to get ACD intial status from the server. This is the default value.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<ACD_Status_n_ ua="na">Sync From Local</ACD_Status_n_>

where n = 1 to 16

Step 4

Click Submit All Changes.

Display or Hide Unavailable Menu Text Box of Agent Status on the Phone

You can control if your user wants to hide the Unavailable menu text box of the Set agent status screen on the phone.

Before you begin

- Set up the phone as a call center phone on the BroadSoft server.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

In the ACD Settings section, set the Unavailable Reason Code Enable parameter to No to hide the Unavailable text box on the phone.

To display the text box, select Yes. This is the default value.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

<Unavailable_Reason_Code_Enable_1_ ua="na">Yes</Unavailable_Reason_Code_Enable_1_>

Step 3

Click Submit All Changes.

Set Up a Phone for Presence

You can enable the BroadSoft XMPP directory for the phone user.

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 Parameters for Set Up Presence table.

Before you begin

- Set up the BroadSoft server for XMPP.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Broadsoft XMPP section, set the fields as described in the Parameters for Set Up Presence.

Step 3

Click Submit All Changes.

- Parameters for Set Up Presence

- Use DNS SRV for XMPP

Parameters for Set Up Presence

The following table defines the function and usage of Set Up Presence parameters in the Broadsoft XMPP section under the Phone 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.

Parameters for Set Up Presence

Parameter

Description

XMPP Enable

Enables the BroadSoft XMPP directory for the phone user.

Perform one of the following:

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

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

- In the phone web page, select Yes to forward all calls. Select No to disable it.

Options: Yes and No

Default: No

Server

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

Perform one of the following:

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

<XMPP_Server ua="na">xsi.iop1.broadworks.net</XMPP_Server>

- In the phone web page, enter a name for the server.

Default: Empty

Port

Server port for the XMPP server.

Perform one of the following:

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

<XMPP_Port ua="na">5222</XMPP_Port>

- In the phone web page, enter the server port.

Allowed values: An integer from 0 through 65535

If the value is set to 0, the phone first sends a DNS SRV query for the domain (specified in Server or User ID) to obtain the XMPP server IP address. If there is no A record in the DNS SRV response, then the phone sends as fallback an A record lookup for the same domain to obtain the IP address. In this scenario, the actual port number is 5222.

Note

When both Server and User ID contain the domain names, the domain name in Server is preferred.

If the value isn't set to 0, the phone directly sends an A record lookup for the domain (specified in Server or User ID) to obtain the XMPP server IP address.

Default: 5222

User ID

BroadSoft User ID of the phone user; for example, username1@xdp.broadsoft.com or username1.

Perform one of the following:

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

<XMPP_User_ID ua="na">username1</XMPP_User_ID>

- In the phone web page, enter the user ID.

If the value doesn't contain the domain name, the phone first generates a new user ID by combining the values of this parameter and Server. For example, the server is xsi.iop1.broadworks.net and user ID is username1, the generated user ID is username1@xsi.iop1.broadworks.net.

Then, the phone sends A record lookup or DNS SRV query for the domain xsi.iop1.broadworks.net to obtain the XMPP server IP address.

Default: Empty

Password

Alphanumeric password associated with the User ID.

Perform one of the following:

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

<XMPP_Password ua="na"></XMPP_Password>

- In the phone web page, enter a supported password.

Default: Empty

Login Invisible

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

Perform one of the following:

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

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

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

Options: Yes and No

Default: No

Retry Intvl

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

Perform one of the following:

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

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

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

Options: Yes and No

Default: No

Use DNS SRV for XMPP

You can configure the phone to use DNS SRV to obtain the IP address of the BroadSoft XMPP server.

Before you begin

- Set up the BroadSoft server for XMPP.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Broadsoft XMPP section, set XMPP Enable to Yes.

Step 3

Set the field Port to 0.

Step 4

Set the fields Server, User ID, and Password as described in the Parameters for Set Up Presence table.

Step 5

Click Submit All Changes.

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.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

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:

<Call_Appearances_Per_Line ua="na">2</Call_Appearances_Per_Line>

The allowed values range from 2 to 10. The default value is 2.

Step 3

Click Submit All Changes.

Enable Reverse Name Lookup

Reverse name lookup searches for the name of a number in an incoming, outgoing, conference, or transferred call. The reverse name lookup acts when the phone cannot find a name using the service provider directory, Call History, or your contacts. Reverse name lookup needs a valid BroadSoft (XSI) Directory, LDAP Directory, or XML Directory configuration.

The reverse name lookup searches the phone's external directories. When a search succeeds, the name is placed in the call session and in the call history. For simultaneous, multiple phone calls, reverse name lookup searches for a name to match the first call number. When the second call connects or is placed on hold, reverse name lookup searches for a name to match the second call. The reverse lookup searches the external directories for 8 secs, if in 8secs there are no results found, there will be no display of the name. If results are found in 8secs, the name is diplayed on the phone. The external directory search priority order is : BroadSoft (XSI) > LDAP > XML.

While searching if the lower priority name is received before the higher priority name, the search shows the lower prioirty name first and then replaced it with the higher priority name if the higher priority name is found within 8 secs.

The precedence of the phone list lookup in BroadSoft (XSI) Directory is:

1. Personal phone list

2. Group common phone list

3. Enterprise common phone list

Reverse name lookup is enabled by default.

Reverse name lookup searches the directories in the following order:

1. Personal Address Book

2. SIP Header

3. Call History

4. BroadSoft (XSI) Directory

5. LDAP Directory

6. XML Directory

Note

The phone searches the XML directory using this format: directory_url?n=incoming_call_number.

Example: For a multiplatform phone using a third-party service, the phone number (1234) search query has this format, http://your-service.com/dir.xml?n=1234.

Before you begin

- Configure one of these directories before you can enable or disable the reverse name lookup:

- BroadSoft (XSI) Directory

- LDAP Corporate Directory

- XML Directory

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Supplementary Services area, set the Reverse Phone Lookup Serv parameter to Yes to enable this feature.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

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

The allowed values are Yes|No. The default value is Yes.

Step 3

Click Submit All Changes.

Emergency Calls

- Emergency Call Support Background

- Emergency Call Support Terminology

- Configure a Phone to Make Emergency Calls

Emergency Call Support Background

Emergency call service providers can register a phone's location for each IP-based phone in a company. The location information server (LIS) transfers the emergency response location (ERL) to the phone. The phone stores its location during registration, after the phone restarts, and when a person signs in to the phone. The location entry can specify the street address, building number, floor, room, and other office location information.

When you place an emergency call, the phone transfers the location to the call server. The call server forwards the call and the location to the emergency call service provider. The emergency call service provider forwards the call and a unique call-back number (ELIN) to the emergency services. The emergency service or public safety answering point (PSAP) receives the phone location. The PSAP also receives a number to call you back, if the call disconnects.

See Emergency Call Support Terminology for the terms used to describe emergency calls from the phone.

You insert the following parameters to obtain the phone's location for any phone extension number:

- Company Identifier–A Unique number (UUID) assigned to your company by the NG9-1-1 service provider.

- Primary Request URL–The HTTPS address of the primary server used to obtain the phone location.

- Secondary Request URL–The HTTPS address of a secondary server (backup) used to obtain the phone location.

- Emergency Number–A sequence of digits that identify an emergency call. You can specify multiple emergency numbers, by separating each emergency number with a comma.

Common emergency service numbers include:

- North America–911

- European countries–112

- Hong Kong–999

The phone requests new location information for the following activities:

- You register the phone with the call server.

- A person restarts the phone and the phone was previously registered with the call server.

- A guest signs in to the phone.

- You change the network interface used in the SIP registration. For example, change Wi-Fi to Ethernet.

- You change the IP address of the phone.

If all of the location servers do not send a location response, the phone re-sends the location request every two minutes.

Emergency Call Support Terminology

The following terms describe emergency call support for the Cisco Multiplatform Phones.

- Emergency Location ID Number (ELIN)–A number used to represent one or more phone extensions that locate the person who dialed emergency services.

- Emergency Response Location (ERL)–A logical location that groups a set of phone extensions.

- HTTP Enabled Location Delivery (HELD)–An encrypted protocol that obtains the PIDF-LO location for a phone from a location information server (LIS).

- Location Information Server (LIS)–A server that responds to a SIP-based phone HELD request and provides the phone location using a HELD XML response.

- Emergency Call Service Provider–The company that responds to a phone HELD request with the phone's location. When you make an emergency call (which carries the phone's location), a call server routes the call to this company The emergency call service provider adds an ELIN and routes the call to the emergency services (PSAP). If the call is disconnected, the PSAP uses the ELIN to reconnect with the phone used to make the emergency call.

- Public Safety Answering Point (PSAP)–Any emergency service (for example, fire, police, or ambulance) joined to the Emergency Services IP Network.

- Universally Unique Identifier (UUID)–A 128-bit number used to uniquely identify a company using emergency call support.

Configure a Phone to Make Emergency Calls

Before you begin

- Obtain the E911 Geolocation Configuration URLs and the company identifier for the phone from your emergency call services provider. You can use the same Geolocation URLs and company identifier for multiple phone extensions in the same office area.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext n, where n is the phone extension number (1-10) of the phone web dialog.

Step 2

In the Dial Plan section, set the Emergency Number parameter

Step 3

In the E911 Geolocation Configuration section, set the Company UUID, Primary Request URL, and Secondary Request URL parameters as described in the Parameters to Make an Emergency Call.

Step 4

Click Submit All Changes.

- Parameters to Make an Emergency Call

Parameters to Make an Emergency Call

The following table defines the function and usage of Making of Emergency Calls parameters in the Dial Plan and E911 Geolocation Configuration sections under the Ext(n) 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.

Parameters to Make an Emergency Call

Parameter

Description

Section: Dial Plan

Emergency Number

Enter a comma-separated list of emergency numbers.

To specify multiple emergency numbers, separate each emergency number with a comma.

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.

Perform one of the following: to the digits that correspond to the customer emergency service numbers.

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

<Emergency_Number_1_ ua="na"/>

- In the phone web page, set Emergency Number parameter to the digits that correspond to the customer emergency service numbers.

Valid Values: Maximum number length is 63 characters

Default: Blank (no emergency number)

Section: E911 Geolocation Configuration

Company UUID

The Universally Unique Identifier (UUID) assigned to the customer by the emergency call services provider.

For example:

07072db6-2dd5-4aa1-b2ff-6d588822dd46

Perform one of the following:

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

<Company_UUID_1_ ua="na"/>

- In the phone web page, enter a valid identifier assigned by the call services provider.

Valid Values: Maximum identifier length is 128 characters.

Default: Blank

Primary Request URL

Encrypted HTTPS phone location request. The request uses the phone IP addresses, MAC address, Network Access Identifier (NAI), and chassis ID and port ID assigned by the network switch manufacturer. The request also includes the location server name and the customer identifier.

The server used by the emergency call services provider responds with an Emergency Response Location (ERL) that has a location Uniform Resource Identifier (URI) tied to the user phone IP address.

Perform one of the following:

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

<Primary_Request_URL_1_ ua="na"/>

- In the phone web page, enter encrypted HTTPS phone location request.

For example:

https://prod.blueearth.com/e911Locate/held/held_request.action

Default: Blank

Secondary Request URL

Encrypted HTTPS request sent to the emergency call services provider's backup server to obtain the user's phone location.

Perform one of the following:

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

<Secondary_Request_URL_1_ ua="na"/>

- In the phone web page, enter the encrypted for the backup server that can return location information.

For example:

https://prod2.blueearth.com/e911Locate/held/held_request.action

Default: Blank

PLK Configuration

- Programmable Line Keys

- Enable the Extension of a Line Key

- Enable Direct PLK Configuration

- Phone Configuration for Monitoring Other Phones

- Add an XML Service to a Line Key

Programmable Line Keys

The Programmable Line Key (PLK) feature allows you to program feature buttons or services URL buttons on the line key buttons. You can configure the line keys with:

- Line buttons—see Enable the Extension of a Line Key

- Speed dials—see Configure Speed Dial on a Line Key

- Call Park—see Add Call Park to a Line Key

- Busy Lamp Field (BLF)—see Phone Configuration for Monitoring Other Phones

- Executive and Assistant—

see Configure Executive and Assistant Menu Access on a Line Key

Note

Only the Cisco IP Phone 6871 Multiplatform Phones support the feature.

- XML Services—see Add an XML Service to a Line Key

- Voicemail—see Configure the Voicemail PLK on a Line Key

Enable the Extension of a Line Key

You can use the buttons on both sides of the phone screen as line keys when you enable the extensions of the line keys.

You can also configure the parameter in the phone configuration file with XML (cfg.xml) code. The parameter for the extension is line-specific.

<Extension_n_ ua="rw">1</Extension_n_>

where n is the extension number.

Before you begin

Access the phone administration web interface. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Choose a line key and assign an extension number in the Extension parameter to enable it.

When Extension is set to Disabled, the user can't use the line key as a telephony extension.

Step 3

Click Submit All Changes.

Enable Direct PLK Configuration

You can directly perform the Programmable Line Key (PLK) configuration on a line key, which means you don't need to disable the extension function of a line key. Before the firmware release 11.3(7), you must disable the line extension to achieve the PLK configuration.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Miscellaneous Line Key Settings section, set the Enable Direct PLK Configuration parameter to Yes.

You can also configure the parameter in the phone configuration file (cfg.xml) with the following XML string:

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

Allowed values: Yes and No

Default value: Yes

Note

If you set the parameter to No, the extension of a line key must be disabled for the PLK configuration.

Step 3

Select Voice > Ext [n] (where [n] is the extension number).

Step 4

In the Proxy and Registration section, ensure that the Proxy parameter is empty.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Proxy_n_ ua="na"></Proxy_n_>

where n is the extension number.

Step 5

In the Subscriber Information section, ensure that the User ID parameter is empty.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<User_ID_n_ ua="na"></User_ID_n_>

where n is the extension number.

Step 6

Click Submit All Changes.

Phone Configuration for Monitoring Other Phones

You can configure the phone to monitor the status of lines on other phones. This feature is useful if users routinely handle calls for colleagues and need to see if they are available to answer calls. The phone monitors each line on a separate line key. The monitoring line keys function as Busy Lamp Field (BLF) keys. A BLF is an LED that changes color to indicate the status of the monitored line:

BLF Key LED Status

LED Color

Meaning

Green

The monitored line is available.

Red

The monitored line is busy.

Blinking red

The monitored line is ringing.

Amber

Error in BLF key configuration.

If the phone is registered to a BroadSoft server, you can set up the phone to monitor multiple users, with a single set of configurations.

Add an XML Service to a Line Key

You can add an XML service to a line key to enable the user to access the XML application or directory.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a line key.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to add an XML service to the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

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

where n is the extension number.

Step 4

In the Extended Function parameter, enter a string in this format:

fnc=xml;url=http://xml.service.url;nme=name

where:

- fnc= prk means function=call park.

- url= http://xml.service.url is the URL for the xml application or directory.

- nme= XXXX is the name displayed on the phone for the xml service. Replace XXXX with a name.

You can also configure the line-specific parameter in the configuration file (cfg.xml). Enter a string in the following format:

<Extended_Function_2_ ua="na">fnc=xml;url=http://xml.service.url;nme=name</Extended_Function_2_>

Step 5

Click Submit All Changes.

Spam Indication for Incoming Webex Calls

To support a spam indication for the incoming calls in Webex environment, server sends the X-Cisco-CallerId-Disposition disposition information to the phone. The phone translates this information as authentication icons. Based on the caller's STIR/SHAKEN verification result, the phone displays three types of icons. The icons are displayed next to the caller ID for call session, local call logs, Webex cloud call logs.

- Validated call－The server sends the disposition information, X-Cisco-CallerId-Disposition=valid, to the phone. The phone displays an extra icon next to the caller ID with a color screen indicating a validated caller. For a phone with grayscale screen, an extra icon next to the caller ID is displayed.

- Invalidated or Spam call－The server sends the disposition information, X-Cisco-CallerId-Disposition=invalid, to the phone. The phone displays an extra icon next to the caller id indicating an illegitimate caller.

- Unverified call－The server sends the disposition information, X-Cisco-CallerId-Disposition=unverified, to the phone. The phone displays an extra icon next to the caller id indicating an unverified call.

When there is no disposition information, the phone displays the same icons as before.

Programmable Softkeys Configuration

- Customize Display of the Softkeys

- Customize a Programmable Softkey

- Configure Speed Dial on a Programmable Softkey

- Configure a PSK with DTMF Support

- Enable Softkeys to Calls History List Menu

- Spam Indication for Incoming Calls

- Programmable Softkeys

Customize Display of the Softkeys

You can customize display of the softkeys on the phone screen during a specific state.

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 Programmable Softkeys.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Programmable Softkeys section, edit the softkeys depending on the call state that you want the softkey to display. For more information, see Parameters for Programmable Softkeys and .

Step 3

Click Submit All Changes.

- Parameters for Programmable Softkeys

Parameters for Programmable Softkeys

The following table defines the function and usage of the programmable softkeys parameters in the Programmable Softkeys section under the Voice > Phone 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 the XML code to configure a parameter.

Parameters for Programmable Softkeys

Parameter

Description and default value

Programmable Softkey Enable

Enables or disables the programmable softkeys. Set this field to Yes to enable the programmable softkeys.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No to enable or disable the programmable softkeys.

Allowed values: Yes | No

Default: No

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.

Configure the PSKs in this format:

- Speed Dial:

fnc=sd;ext=extension_number@$PROXY;vid=n;nme=display_name

- Vertical Service Activation Code:

fnc=sd;ext=star_code@$PROXY;vid=n;nme=display_name

See Vertical Service Activation Codes.

- XML Service:

fnc=xml;url=http://server_IP/services.xml;vid=n;nme=display_name

- Menu shortcut:

fnc=shortcut;url=userpref;nme=User preferences

For more information, see Menu Shortcuts Mapping on PLK and PSK.

When you add a programmable softkey to a softkey list, such as Idle Key List, Missed Call Key List, and so on, the programmable softkey displays on the phone screen.

Perform one of the following:

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

<PSK_1 ua="na">fnc=xml;url=http://server_IP/services.xml;vid=n;
nme=display_name</PSK_1>

- In the phone web interface, set the PSKs in the valid format.

Default: Empty

Customize a Programmable Softkey

The phone provides sixteen programmable softkeys (fields PSK1 through PSK16). You can define the fields by a speed-dial script.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Programmable Softkeys section, set the Programmable Softkey Enable to Yes.

Step 3

Select a programmable softkey number field on which to configure a phone feature.

Step 4

Enter the string for the programmable soft key. See the different types of programmable softkeys described in Configure Speed Dial on a Programmable Softkey.

Step 5

Click Submit All Changes.

Configure Speed Dial on a Programmable Softkey

You can configure programmable softkeys as speed dials. The speed dials can be extensions or phone numbers. You can also configure programmable softkeys with speed dials that perform an action that a vertical service activation code (or a star [*] code) defines. For example, if you configure a programmable softkey with a speed dial for *67, the call is placed on hold.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Programmable Softkeys section, set the Programmable Softkey Enable to Yes.

Step 3

To configure a speed dial PSK, enter the following in the PSK number field:

fnc=sd;ext=extensionname/starcode@$PROXY;vid=n;nme=name

Where:

- fnc= function of the key (speed dial)

- extensionname=extension being dialed or the star code action to perform

- vid= n is the extension that the speed dial will dial out

- name is the name of the speed dial being configured

Note

The name field displays on the softkey on the IP phone screen. We recommend a maximum of 10 characters for a phone. If more characters are used, the label might be truncated on the phone screen.

Step 4

Edit the following:

- Idle Key List: Edit the field as described in the following example:

redial|1;newcall|2;dnd;psk1

If the user incorrectly configures the programmable softkey list features on the phone, the key list on the phone LCD does not update. For example:

- If a user enters rdeial;newcall;cfwd (redial has been misspelt), the key list is not updated and the user does not see any change on the LCD.

- If a user enters redial;newcall;cfwd;delchar, the user will not see a change on the LCD, as the delchar softkey is not allowed in the Idle Key List. Hence, this is an incorrect configuration of the programmable softkey list.

- PSK1:

fnc=sd;ext=5014@$PROXY;nme=sktest1

Note

In this example, we are configuring a softkey on a phone as a speed dial number for extension 5014 (sktest1).

You can also configure an XML service on the programmable softkey. Enter the string in this format:

<PSK_1 ua="na">fnc=xml;url=http://xml.service.url;nme=name</PSK_1>

Step 5

Click Submit All Changes.

Configure a PSK with DTMF Support

You can configure programmable softkeys (PSK) with dual tone multifrequency (DTMF). This configuration enables the phone to send digital pulses inband (or out-of-band via SIP INFO) to the server during an active call. When you enable a function on a PSK, the user sees the softkey name, and presses it to perform the named function. The applied actions to the DTMF digit string are similar to those applied to Speed Dial, such as the following:

- Pause represented by ,

- Wait represented by X

For example, ext=<DTMF_DIGITS>[[,|X][<DTMF_DIGITS>]], where the valid DTMF digits are 0-9,*, #, a, b, c, d, and where the parts in [ ] brackets are optional.

This feature applies only to programmable softkeys. It doesn’t apply to the programmable line keys (PLK) on the desk phones. If you configure any PLK for this feature, the display will present the Circled X icon Ⓧ, and nothing will happen if you press the key.

This feature supports only Connected Key List and Connected Video Key List.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone > Programmable Softkeys.

Step 2

Set the Programmable Softkey Enable field to yes.

Step 3

From the PSK list (PSK#1 - PSK#16), select a PSK to configure.

Step 4

In the PSK(n) field, where n is a programmable softkey number, enter a string in this format:

fnc=dtmf;ext=<dtmf_digits_to_be_outpulsed>;nme=<softkey_display_name>;
vid=<extension_n_to_be_associated>

When a phone has more than one registered line, you must include the vid= that is asociated with the particular line/extension in order for the softkey to appear. Otherwise, the softkey will not display.

Step 5

(Optional) To configure the PSK softkey to toggle within a pair (outpulse-display) each time you press it, enter a string in this format:

fnc=dtmf;ext=<dtmf_digits_to_be_outpulsed>;nme=<softkey_display_name>;
ext2=<second_set_of_dtmf_digits_to_be_outpulsed>;nme2=<second_softkey_display_name_after_first_press>;
vid=<extension_n_to_be_associated>

The PSK softkey toggle always starts with the ext/nme for each new call.

Step 6

In the Connected Key List field or Connected Video Key List field, enter the configured PSK keywords according to where on the phone screen you wish the softkey name to appear.

For example, in the following entry, the hold softkey name appears in the first position. The softkey name that is listed in the psk1 field appears in the second position, and so on.

hold;psk1;endcall;xfer;conf;xferLx;confLx;bxfer;phold;redial;dir;park

Step 7

Select Voice > Ext(n), where n is the extension number you wish to configure.

Step 8

In the Audio Configuration section, set the DTMF Tx Method to one of the following methods from the drop-down list.

- InBand

- AVT

- INFO

- Auto

- InBand+INFO

- AVT+INFO

Step 9

Click Submit All Changes.

Use these examples to help you understand how to configure PSK with DTMF Support options:

Example: PSK toggles when pressed.

- Voice > Phone > Programmable Softkeys > Programmable Softkey Enable: Yes

- Connected Key List: psk1|1 ;endcall|2;conf|3;xfer|4;

- PSK 1: fnc=dtmf;ext=#1;nme=PressStart;ext2=*2;nme2=PressStop;vid=1

- Voice > Ext 1 > DTMF Tx Method: Auto

Example: Phone sends DTMF digits inband via a PSK softkey.

- Voice > Phone > Programmable Softkeys

- Programmable Softkey Enable: yes.

- Connected Key List: psk1|1;endcall|2;conf|3;xfer|4;

- PSK 1: fnc=dtmf;ext=#1;nme=PressMe;vid=1

- Voice > Ext 1 > DTMF Tx Method: Auto

Example: The PSK softkey pauses between digits.

- Voice > Phone > Programmable Softkeys > Programmable Softkey Enable: Yes

- Connected Key List: psk1|1;endcall|2;conf|3;xfer|4;

- PSK 1: fnc=dtmf;ext=#1,1006;nme=PressMe;vid=1

- Voice > Ext 1 > DTMF Tx Method: Auto

Example: The PSK softkey waits for the user's input between digits.

- Voice > Phone > Programmable Softkeys > Programmable Softkey Enable: Yes

- Connected Key List: psk1|1;endcall|2;conf|3;xfer|4;

- PSK 1: fnc=dtmf;ext=#1X1006;nme=PressMe;vid=1

- Voice > Ext 1 > DTMF Tx Method: Auto

Enable Softkeys to Calls History List Menu

You can configure the Option, Call, Edit call, Filter, and Back softkeys on the screen for All, Placed, Received, and Missed calls list. When you press the Recents softkey on the phone, you can directly access the All calls screen and see the list of all types of recents calls.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

.

Procedure

Step 1

Select Voice > Phone.

Step 2

Configure the XSI account information by providing values in the XSI Host Server, XSI Authentication Type, Login User ID, Login Password, and CallLog Associated Line parameters.

For more information about configuring XSI account, see Configure BroadSoft Settings.

Step 3

Set the CallLog Enable parameter to Yes.

Step 4

Set Display Recents From to Server.

Step 5

In the Programmable Softkeys section,

1. Set the Programmable Softkey Enable parameter to Yes.

2. In the Broadsoft Call History Key List field, the default string is: option|1;call|2;editcall|3;back|4;

Supported strings are option, call, editcall, filter, and back. This parameter doesn't support psk string.

Availability of all these softkeys under the All, Placed, Received, and Missed calls list or the Option menu in these calls list depends on the following conditions:

- Programmable Softkey Enable = Yes and Broadsoft Call History Key List = option|1;call|2;editcall|3;back|4;－Option, Call, Edit call, Back softkeys appear on the All, Placed, Received, and Missed calls list. Filter appears in the Option menu of calls list.

- Programmable Softkey Enable = Yes and Broadsoft Call History Key List = option|1;call|2;back|4;－Option, Call, Back softkeys appear on the All, Placed, Received, and Missed calls list. Edit call and Filter appears in the Option menu of calls list.

- Programmable Softkey Enable = Yes and Broadsoft Call History Key List = option|1;call|2;editcall|3;filter|4;－Option, Call, Edit call, Filter softkeys appear on the All, Placed, Received, and Missed calls list.

- Programmable Softkey Enable = Yes, PSK 1 = fnc=shortcut;url=missedcalls, and Broadsoft Call History Key List = option|1;call|2;psk1|3;filter222|4;－only Option and Call softkeys appear on the All, Placed, Received, and Missed calls list because strings psk and filter222 are invalid values. Edit call and Filter appears in the Option menu of calls list.

- Programmable Softkey Enable = Yes, and Broadsoft Call History Key List = blank－The softkeys appears as default setting option|1;call|2;editcall|3.Option, Call, Edit call softkeys appear on the All, Placed, Received, and Missed calls list. Filter appears in the Option menu of calls list.

Note

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

<Broadsoft_Call_History_Key_List ua="na">option|1;call|2;editcall|3</Broadsoft_Call_History_Key_List>

Step 6

Click Submit All Changes.

Spam Indication for Incoming Calls

New technology standard Secure Telephony Identity Revisited (STIR) and Signature-based Handling of Asserted information using toKENs (SHAKEN). These standards define procedures to authenticate and verify caller identification for calls carried over the IP network. The STIR-SHAKEN framework is developed to provide the end user with a great degree of identification and control over the type of calls they receive. These sets of standards are intended to provide a basis for verifying calls, classifying calls, and facilitating the ability to trust caller identity end to end. Illegitimate callers can easily be identified.

When STIR/SHAKEN support is implemented on the server, the phone displays an extra icon next to the caller ID based on the caller's STIR/SHAKEN verification result. Based on the verification result, the phone displays three types of icons. This helps reduce wasted time from answering calls from robocallers, and the security risk from callers with spoofed or tampered Caller ID.

Note

- Validated call－When the caller carries verstat=TN-Validation-Passed in the SIP Header PAID or FROM, an extra icon next to the caller id is displayed on the phone with a color screen indicating a validated caller. For a phone with grayscale screen, an extra icon next to the caller id is displayed.

- Spam call－When the caller carries verstat=TN-Validation-Failed in the SIP Header PAID or FROM, an extra icon next to the caller id is displayed on the phone indicating an illegitimate caller.

- Unverified call－When the caller carries verstat=NO-TN-Validation in the SIP Header PAID or FROM, an extra icon next to the caller id is displayed on the phone indicating an unverified call.

For detailed spam notifications for calls in Webex environment, see Spam Indication for Incoming Webex Calls.

Programmable Softkeys

Keyword

Key Label

Definition

Available Phone Status

acd_login

Agt signin

Logs user in to Automatic Call Distribution (ACD).

Idle

acd_logout

AgtSignOut

Logs user out of ACD.

Idle

answer

Answer

Answers an incoming call.

Ringing

astate

Agt Status

Checks the ACD status.

Idle

avail

Avail

Denotes that a user who is logged in to an ACD server has set his status as available.

Idle

barge

Barge

Allows another user to interrupt a shared call.

Shared-Active, Shared-Held

bargesilent

BargeSilent or BargeSlnt

Allows another user to interrupt a shared call with the mic disabled.

Shared-Active

bxfer

BlindXfer

Performs a blind call transfer (transfers a call without speaking to the party to whom the call is transferred). Requires that Blind Xfer Serv is enabled.

Connected

call (or dial)

Call

Calls the selected item in a list.

Dialing Input

call info

Call Info

Show call information

Progressing

cancel

Cancel

Cancels a call (for example, when conferencing a call and the second party is not answering.

Off-Hook

cfwd

Forward / Clr fwd

Forwards all calls to a specified number.

Idle, Off-Hook, Shared-Active, Hold, Shared-Held

crdpause

PauseRec

Pause recording

Connected, Conferencing

crdresume

ResumeRec

Resume recording

Connected, Conferencing

crdstart

Record

Start a recording

Connected, Conferencing

crdstop

StopRec

Stop recording

Connected, Conferencing

conf

Conference

Initiates a conference call. Requires that Conf Server is enabled and there are two or more calls that are active or on hold.

Connected

confLx

Conf line

Conferences active lines on the phone. Requires that Conf Serv is enabled and there are two or more calls that are active or on hold.

Connected

delchar

delChar - backspace Icon

Deletes a character when entering text.

Dialing Input

dir

Dir

Provides access to phone directories.

Idle, Miss, Off-Hook (no input), Connected, Start-Xfer, Start-Conf, Conferencing, Hold, Ringing, Shared-Active, Shared-Held

disp_code

DispCode

Enter Disposition Code

Idle, Connected, Conferencing, Hold

dnd

DND / Clr Dnd

Sets Do Not Disturb to prevent calls from ringing the phone.

Idle, Off-Hook, Hold, Shared-Active, Shared-Held, Conferencing, Start-Conf, Start-Xfer

emergency

Emergency

Enter emergency number

Connected

em_login (or signin)

Sign in

Logs user in to Extension Mobility.

Idle

em_logout (or signout)

Sign out

Logs user out of Extension Mobility.

Idle

endcall

End call

Ends a call.

Connected, Start-Xfer, Start-Conf, Conferencing, Hold

favorites

Favorites

Provides access to "Speed Dials".

Idle, Miss, Off-Hook (no input), Connected, Start-Xfer, Start-Conf, Conferencing, Hold, Ringing, Shared-Active, Shared-Held

gpickup

GrPickup

Allows user to answer a call ringing on an extension by discovering the number of the ringing extension.

Idle, Off-Hook

hold

Hold

Put a call on Hold.

Connected, Start-Xfer, Start-Conf, Conferencing

ignore

Decline

Ignores an incoming call.

Ringing

ignoresilent

Ignore

Silences an incoming call

Ringing

join

Join

Connects a conference call. If the conference host is user A and users B & C are participants, when A presses "Join", A will drop off and users B & C will be connected.

Conferencing

lcr

Call Rtn/lcr

Returns the last missed call.

Idle, Missed-Call,Off-Hook (no input)

left

Left arrow icon

Moves the cursor to the left.

Dialing Input

messages

Messages

Provides access to voicemail.

Idle, Miss, Off-Hook (no input), Connected, Start-Xfer, Start-Conf, Conferencing, Hold, Ringing, Shared-Active, Shared-Held

miss

Miss

Displays the list of missed calls.

Missed-Call

newcall

New Call

Begins a new call.

Idle, Hold, Shared-Active, Shared-Held

option

Option

Opens a menu of input options.

Off-Hook

park

Park

Puts a call on hold at a designated "park" number.

Connected

phold

PrivHold

Puts a call on hold on an active shared line.

Connected

pickup

PickUp

Allows a user to answer a call ringing on another extension by entering the extension number.

Idle, Off-Hook

recents

Recents

Displays the All calls list from call history.

Idle, Off-Hook, Shared-Active, Shared-Held

redial

Redial

Displays the redial list.

Idle, Connected, Start-Conf, Start-Xfer, Off-Hook (no input), Hold

resume

Resume

Resumes a call that is on hold.

Hold, Shared-Held

right

Right arrow icon

Moves the cursor to the right.

Dialing (input)

settings

Settings

Provides access to "Information and Settings".

All

starcode

Input Star Code/*code

Displays a list of star codes that can be selected.

Off-Hook, Dialing (input)

trace

Trace

Trigger trace

Idle, Connected, Conferencing, Hold

unavail

Unavail

Denotes that a user who is logged in to an ACD server has set his status as unavailable.

Idle

unpark

Unpark

Resumes a parked call.

Idle, Off-Hook, Connected, Shared-Active

xfer

Transfer

Performs a call transfer. Requires that Attn Xfer Serv is enabled and there is at least one connected call and one idle call.

Connected, Start-Xfer, Start-Conf

xferlx

Xfer line

Transfers an active line on the phone to a called number. Requires that Attn Xfer Serv is enabled and there are two or more calls that are active or on hold.

Connected

Programmable Softkeys for Executives and Assistants

Note

Only the Cisco IP Phone 6871 Multiplatform Phones support the executive and assistant programmable softkeys.

Keyword

Key Label

Definition

Available Phone Status

bridgein

Bridge in

Only available to executives who have assistants. Joins the user (executive) to an ongoing call with an assistant.

Idle, Shared-Active

callpush

Call push

Only available to executive assistants. Transfers an ongoing call from the user (assistant) to the executive.

Hold, Shared-Active

callretrieve

Retrieve

Only available to executives who have assistants. Transfers an ongoing call from the assistant to the user (executive).

Idle, Shared-Active

divert

Clr divert

Only available to executive assistants. Deactivates call diversion for the user (assistant).

Available only when call diversion is active and you navigate to the menu Settings > Assistant.

It is also available when you press the line key that is configured as Assistant.

divert

Divert

Only available to executive assistants. Activates call diversion for the user (assistant). All incoming calls to the user, for executives that the user handles, are diverted to the specified destination.

Available when you navigate the menu Settings > Assistant.

It is also available when you press the line key that is configured as Assistant.

proxycall

Proxy call

Only available to executive assistants. Initiates a call on behalf of the selected executive.

Available when you navigate the menu Settings > Assistant.

It is also available when you press the line key that is configured as Assistant.

Chapter 13. Audio Configuration

- Configure Different Audio Volume

- Configure the Voice Codecs

- Voice Quality Reporting

Configure Different Audio Volume

You can configure the volume settings in the phone web interface.

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 Parameters for Audio Volume table in Parameters for Audio Volume.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > User.

Step 2

In the Audio Volume section, configure the volume level for audio parameters as described in the Parameters for Audio Volume table in Parameters for Audio Volume.

Step 3

Click Submit All Changes.

- Parameters for Audio Volume

Parameters for Audio Volume

The following two tables describes the acoustic and audio settings.

The following table defines the function and usage of Audio Volume parameters in the Audio Volume section under the User 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.

Parameters for Audio Volume

Parameter

Description

Ringer Volume

Sets the default volume for the ringer.

Perform one of the following:

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

<Ringer_Volume ua="rw">8</Ringer_Volume>

- In the phone web page, enter a valid value as the ringer volume.

Allowed values: an integer ranging between 0 and 15

Default: 9

Speaker Volume

Sets the default volume for the speakerphone.

Perform one of the following:

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

<Speaker_Volume ua="rw">11</Speaker_Volume>

- In the phone web page, enter a valid value as the speaker volume.

Allowed values: an integer ranging between 0 and 15

Default: 11

Handset Volume

Sets the default volume for the handset.

Perform one of the following:

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

<Handset_Volume ua="rw">9</Handset_Volume>

- In the phone web page, enter a valid value as the handset volume.

Allowed values: an integer ranging between 0 and 15

Default: 10

Headset Volume

Sets the default volume for the headset.

Perform one of the following:

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

<Headset_Volume ua="rw">9</Headset_Volume>

- In the phone web page, enter a valid value as the headset volume.

Allowed values: an integer ranging between 0 and 15

Default: 10

Electronic Hookswitch Control

Enables or disables the Electronic HookSwitch (EHS) feature. After EHS is enabled, the AUX port does not output phone logs.

Perform one of the following:

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

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

- In the phone web page, enter a valid value as the EHS volume.

Allowed values: Yes|No

Default: No

Configure the Voice Codecs

A codec resource is considered allocated if it has been included in the SDP codec list of an active call, even though it eventually might not be chosen for the connection. Negotiation of the optimal voice codec sometimes depends on the ability of the Cisco IP Phone to match a codec name with the far-end device or gateway codec name. The phone allows the network administrator to individually name the various codecs that are supported such that the correct codec successfully negotiates with the far-end equipment.

The Cisco IP Phone supports voice codec priority. You can select up to three preferred codecs. The administrator can select the low-bit-rate codec that is used for each line. G.711a and G.711u are always enabled.

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 Audio Codec Parameters.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the Audio Configuration section, configure the parameters as defined in the Audio Codec Parameters table.

Step 3

Click Submit All Changes.

- Audio Codec Parameters

Audio Codec Parameters

The following table defines the function and usage of the voice codec parameters in the Audio Configuration section under the Voice > Ext (n) 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.

Audio Codec Parameters

Parameter

Description

Preferred Codec

Preferred codec for all calls. The actual codec used in a call still depends on the outcome of the codec negotiation protocol.

Perform one of the following:

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

<Preferred_Codec_1_ ua="rw">G711u</Preferred_Codec_1_>

- In the phone web interface, select your preferred codec from the list.

Allowed values: G711u|G711a|G729a|G722|G722.2|iLBC|OPUS

Default: G711u

Use Pref Codec Only

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.

Perform one of the following:

- In the phone configuration file with XML(cfg.xml), enter a string in this format: <Use_Pref_Codec_Only_1_ ua="rw">No</Use_Pref_Codec_Only_1_>

- In the phone web interface, set this field to Yes or No as needed.

Allowed values: Yes|No

Default: No

Second Preferred Codec

Codec to use if the codec specified in Preferred Codec fails.

Perform one of the following:

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

<Second_Preferred_Codec_1_ ua="rw">Unspecified</Second_Preferred_Codec_1_>

- In the phone web interface, select your preferred codec from the list.

Allowed values: Unspecified|G711u|G711a|G729a|G722|G722.2|iLBC|OPUS

Default: Unspecified

Third Preferred Codec

Codec to use if the codecs specified in Preferred Codec and Second Preferred Codec fail.

Perform one of the following:

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

<Third_Preferred_Codec_1_ ua="rw">Unspecified</Third_Preferred_Codec_1_>

- In the phone web interface, select your preferred codec from the list.

Allowed values: Unspecified|G711u|G711a|G729a|G722|G722.2|iLBC|OPUS

Default: Unspecified

G711u Enable

G711a Enable

G729a Enable

G722 Enable

G722.2 Enable

iLBC Enable

Enables the use of a specific codec.

Perform one of the following:

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

<G711u_Enable_1_ ua="rw">Yes</G711u_Enable_1_>

<G711a_Enable_1_ ua="rw">Yes</G711a_Enable_1_>

<G729a_Enable_1_ ua="rw">Yes</G729a_Enable_1_>

<G722_Enable_1_ ua="rw">Yes</G722_Enable_1_>

<G722_Enable_1_ ua="rw">Yes</G722_Enable_1_>

<G722.2_Enable_1_ ua="rw">No</G722.2_Enable_1_>

<iLBC_Enable_1_ ua="rw">No</iLBC_Enable_1_>

<OPUS_Enable_1_ ua="rw">Yes</OPUS_Enable_1_>

- In the phone web interface, set the corresponding field to Yes to enable the use of a specific codec, or No to disable it.

Note

The transmit rate for the G.729a codec is at 8 kbps.

Silence Supp Enable

Enables or disables silence suppression. When set Yes, silent audio frames are not transmitted.

Perform one of the following:

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

<Silence_Supp_Enable_1_ ua="rw">No</Silence_Supp_Enable_1_>

- In the phone web interface, set this field to Yes to enable silence suppression, or No to disable it.

Allowed values: Yes|No

Default: No

DTMF Tx Method

The method for transmitting DTMF signals to the far end. The options are:

- AVT—Audio video transport. Sends DTMF as AVT events.

- InBand—Sends DTMF by using the audio path.

- Auto—Uses InBand or AVT based on the outcome of codec negotiation.

- INFO—Uses the SIP INFO method.

- InBand+INFO—Uses both the audio path and the SIP INFO method.

- AVT+INFO—Uses both the AVT and the SIP INFO method.

Perform one of the following:

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

<DTMF_Tx_Method_1_ ua="rw">Auto</DTMF_Tx_Method_1_>

- In the phone web interface, select your preferred transmitting method from the list.

Default: Auto

Codec Negotiation

When set to Default, the phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the phone responds listing all the codecs that the phone supports.

Perform one of the following:

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

<Codec_Negotiation_1_ ua="na">Default</Codec_Negotiation_1_>

- In the phone web interface, select the desired option from the list.

Allowed values: Default|List All

Default: Default

Encryption Method

Encryption method to be used during secured call. Options are AES 128 and AES 256 GCM

Perform one of the following:

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

<Encryption_Method_1_ ua="na">AES 128</Encryption_Method_1_>

- In the phone web interface, select your preferred encryption method from the list.

Allowed values: AES 128 |AES 256 GCM

Default: AES 128.

Voice Quality Reporting

You can capture voice quality metrics for Voice over Internet Protocol (VoIP) sessions with a Session Initiation Protocol (SIP) event package. Voice call quality information derived from RTP and call information from SIP is conveyed from a User Agent (UA) in a session (reporter) to a third party (collector).

The Cisco IP phone uses User Datagram Protocol (UDP) to send a SIP PUBLISH message to a collector server.

- Supported Scenarios for Voice Quality Reporting

- Mean Opinion Scores and Codecs

- Configure Voice Quality Reporting

Supported Scenarios for Voice Quality Reporting

Currently, only the basic call scenario supports voice quality reporting. A basic call can be a peer to peer incoming or outgoing call. The phone supports periodic SIP publish message.

Mean Opinion Scores and Codecs

The voice quality metrics use Mean Opinion Score (MOS) to rate the quality. A MOS rating of 1 is the lowest quality; a MOS rating of 5 is the highest quality. The following table gives a description of some of the codecs and MOS scores. The phone supports all codecs. For all codecs, the phone sends the SIP Publish message.

Codec

Complexity and Description

MOS

Minimum Call Duration for Valid MOS Value

G.711 (A-law and u-law)

Very low complexity. Supports uncompressed 64 kbps digitized voice transmission at one to ten 5 ms voice frames-per-packet. This codec provides the highest voice quality and uses the most bandwidth of any of the available codecs.

A minimum value of 4.1 indicates good voice quality.

10 seconds

G.729A

Low to medium complexity.

A minimum value of 3.5 indicates good voice quality.

30 seconds

G.729AB

Contains the same reduced complexity modifications present in the G.729A.

A minimum value of 3.5 indicates good voice quality.

30 seconds

Configure Voice Quality Reporting

You can generate a voice quality report for each extension on the phone. The parameters for the Voice Quality Metrics (VQM) SIP Publish Message help you to:

- Generate voice quality reports.

- Name your reports.

- Determine when your phone sends SIP Publish messages.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. See VQM SIP Publish Message Parameters

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where (n) is the extension number.

Step 2

In SIP Settings, enter a value for the Voice Quality Report Address parameter. You can enter either a domain name or an IP address.

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

Step 3

Enter your report name for the Voice Quality Report Group parameter.

Your report name can't begin with a hyphen (-), semicolon (;), or a space.

Step 4

Enter an interval, in seconds, for the Voice Quality Report Interval parameter. Example: 20 for 20-second interval reporting.

Step 5

Click Submit All Changes.

- VQM SIP Publish Message Parameters

VQM SIP Publish Message Parameters

The following table defines the Voice Quality Metrics (VQM) SIP Publish Message parameters in the Sip Settings section under Voice > Ext(n) 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.

VQM SIP Publish Message Parameters

Parameter Name

Description

Voice Quality Report Address

Allows you to enter one of the following options:

- Domain Name

- IP address

- The SIP UDP port number along with the domain name

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

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

Default parameter = empty (no report)

Default SIP UDP Port = 5060

Voice Quality Report Group

Allows you to enter a voice quality report name.

Your report name cannot begin with a:

- hyphen (-)

- semicolon (;)

- space

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

<Voice_Quality_Report_Group_1_ ua="na">test-group-1</Voice_Quality_Report_Group_1_>

Default parameter = empty (The report will use the canonical name in the form of identifier@ipAddress.)

Voice Quality Report Interval

Allows you to determine when the phones send SIP Publish messages.

If you have properly configured the Voice Quality Report Address, the SIP Publish messages can be sent:

- When the call has ended or is placed on hold.

- Periodically, when you enter an interval in seconds for this parameter. Example: 20 for 20-second intervals.

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

<VQ_Report_Interval_1_ ua="na">20</VQ_Report_Interval_1_>

Default parameter = 0 (no periodic SIP Publish Message)

Chapter 14. Voicemail Configuration

- Configure Voicemail

Configure Voicemail

You can configure the internal or external phone number or URL for the voicemail system. If you use an external voicemail service, the number must include any digits required to dial out and any required area code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the General section, enter the Voice Mail Number that is a phone number or URL to check the voicemail.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

<Voice_Mail_Number ua="na">123</Voice_Mail_Number>

Default: Empty

Step 3

Click Submit All Changes.

The phone reboots.

- Configure Voicemail for An Extension

- Configure the Message Waiting Indicator

- Configure the Voicemail PLK on a Line Key

- Configure the Voicemail PLK on the Phone

Configure Voicemail for An Extension

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where (n) is the number of an extension.

Step 2

In the Call Feature Settings section, configure the parameters Voice Mail Server, Voice Mail Subscribe Interval (optional), and Voice Mail Enable as described in Parameters for Voicemail Server and Message Waiting.

Step 3

Click Submit All Changes.

The phone reboots.

Configure the Message Waiting Indicator

You can configure the Message Waiting Indicator for a specific extension 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 one or more voicemails are left. This light can show if a message is waiting.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where (n) is the number of an extension.

Step 2

In the Call Feature Settings section, configure the parameter Message Waiting and relevant parameters as described in Parameters for Voicemail Server and Message Waiting.

Step 3

Click Submit All Changes.

The phone reboots.

- Parameters for Voicemail Server and Message Waiting

Parameters for Voicemail Server and Message Waiting

The following table describes the Call Feature Settings for Voicemail and Message Waiting.

Parameters for Voicemail and Message Waiting

Parameter

Description

Voice Mail Server

Identifies the SpecVM server for the phone, generally the IP address, and port number of the VM server.

Perform one of the following:

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

<Voice_Mail_Server_1_ ua="na"></Voice_Mail_Server_1_>

- In the phone web page, enter the IP address of the voicemail server.

Default: Empty

Voice Mail Subscribe Interval

The expiration time, in seconds, of a subscription to a voicemail server.

Perform one of the following:

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

<Voice_Mail_Subscribe_Interval_1_ ua="na">86400</Voice_Mail_Subscribe_Interval_1_>

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

Allowed values: An integer from 0 through 86400

If the value is set to 0, then the phone uses the default value instead.

Default: 86400

Voice Mail Enable

Enables or disables the subscription to the voicemail server for the specific extension.

Perform one of the following:

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

<Voice_Mail_Enable_1_ ua="na">Yes</Voice_Mail_Enable_1_>

- In the phone web interface, set this field to Yes or No to enable or disable the function.

Allowed values: Yes and No

Default: Yes

Message Waiting

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.

This parameter is valid when the parameters Voice Mail Server, Voice Mail Subscribe Interval, and Voice Mail Enable are configured.

Perform one of the following:

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

<Message_Waiting_1_ ua="na">Yes</Message_Waiting_1_>

- In the phone web interface, set this field to Yes or No to enable or disable the function.

Allowed values: Yes and No

Default: Yes

Configure the Voicemail PLK on a Line Key

You can configure the voicemail PLK on a line key for the users to monitor a specified voicemail account of a user or a group.

The voicemail PLK can monitor both the voicemail of an extension and the voicemail account of another user or a group. Monitoring the voicemail of another user or a group requires the support from the SIP proxy.

For example, if the users belong to a customer service group, this feature allows the users to monitor both their voicemails and their group's voicemails.

If you add speed dial for the same line key, the users can press the line key to make a speed dial to the assigned extension.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

Select a Line Key on which to configure the voicemail PLK.

Step 3

(Optional) Set the Extension parameter to Disabled to disable the extension.

Note

If the Direct PLK Configuration feature is disabled, you must disable the extension to configure the voicemail PLK on the line key. If the feature is enabled, you can skip this step. For details, see Enable Direct PLK Configuration.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

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

where n is the extension number.

Step 4

In the Extended Function parameter, enter a string in this format:

- For MWI only:

fnc=mwi;sub=group_vm@domain;vid=1;nme=Group;

- For MWI + Speed Dial:

fnc=mwi+sd;ext=8000@domain;sub=group_vm@domain;vid=1;nme=Group;

- For MWI + speed dial + DTMF:

fnc=mwi+sd;ext=8000 ,4085283300#,123456#@domain;sub=group_vm@domain;vid=1;nme=Group;

For more information about the string syntax, see String Syntax for Voicemail PLK.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Extended_Function_2_ ua="na">mwi+sd;ext=8000 ,4085283300#,123456#@domain;
sub=group_vm@domain;vid=1;nme=Group;</Extended_Function_2_>

Step 5

In the General section, add mwi or mwi;sd in the parameter Customizable PLK Options.

Parameter in the configuration file (cfg.xml):

<Customizable_PLK_Options ua="na">mwi;sd</Customizable_PLK_Options>

After the configuration, users can configure the corresponding features on the line key.

Step 6

Click Submit All Changes.

- String Syntax for Voicemail PLK

String Syntax for Voicemail PLK

The following table describes the string syntax associated with the voicemail Programmable Line Key (PLK) configured in the phone web interface.

String Syntax for PLK of Message Waiting Indicator

String

Description

fnc

Specifies the function of the key. You can use the PLK for only MWI or the combination of MWI and speed dial.

Valid values: mwi|mwi+sd

- mwi: Enables voicemail account monitoring.

- mwi+sd: Enables voicemail account monitoring and speed dial. If used, you must configure "ext". Otherwise, speed dial doesn't function.

Example: fnc=mwi+sd;

Type: Mandatory

sub

Specifies the SIP URI of a voicemail account that the PLK monitors.

The voicemail account can be the voicemail account of the user of an extension on the phone or a voicemail account of a group.

For example, the user ID of extension 1 is 4085289931. The related voicemail account is 4085289931@example.com. The user belongs to a customer group whose voicemail account is 4085283300@example.com.

In this example, the value can be 4085289931@example.com. If the SIP proxy allows the group member to monitor the group's voicemail, the value can be 4085283300@example.com.

Examples:

- sub=4085283300@example.com;

- sub=4085283300@$PROXY;

Type: Mandatory

vid

The extension ID with which the voicemail PLK associates.

The voicemail PLK associates with an extension of the phone to generate the SIP messages based on the extension user ID and proxy.

Specifically, the voicemail PLK generates the From and Contact headers based on the associated extension user ID and proxy. Then it sends a SUBSCRIBE message to the specified SIP URI.

If the string is missing, the PLK associates with extension 1.

Example: vid=2;

Type: Optional

ext

Specifies a speed dial number or a SIP URI if the key uses both MWI and speed dial functions (fnc=mwi+sd).

The speed dial number is used to place a call for the voicemail messages.

Example: ext=8000;

To bypass the voicemail login session during a speed dial from the key, you can input the DTMF characters (including voicemail account ID and PIN) in the string.

Example: ext=8000 ,4085283300#,123456#@$PROXY;

Where: "4085283300" is the voicemail account ID, "123456" is the PIN.

Note

We don't recommend that you add the PIN to the speed dial string.

A space is required between the speed dial number (8000) and the DTMF characters (,4085283300#,123456#).

A comma (,) in the speed dial characters means a 2-second pause.

For more information about the speed dial string, see DTMF Wait and Pause Parameters.

Type: Optional

nme

Name displayed on the phone for the key.

If this string is missing, the value will be the user part of the "sub" field. For example, "4085283300".

Example: nme=Group

Type: Optional

Related Tasks

Configure the Voicemail PLK on a Line Key

Configure the Voicemail PLK on a Key Expansion Module Button

Configure the Voicemail PLK on the Phone

Configure the Voicemail PLK on the Phone

You can configure the voicemail Programmable Line Key (PLK) on the phone. The maximum number of voicemail PLKs is ten. The voicemail PLK can monitor the voicemail account of a phone or monitor a voicemail account that isn't configured on the phone.

Before you begin

Make sure that one of the following situations are met:

- The Extension parameter under the Line Key (n) section from Voice > Phone is set to Disabled.

- The Direct PLK Configuration feature is enabled. In this case, you don't need to disable the extension of a line key. For more information about how to enable the feature, see Enable Direct PLK Configuration.

Procedure

Step 1

On the phone, press the line key that you want to configure as an voicemail PLK for 2 seconds.

Step 2

Click MWI or MWI + Speed dial in the Select feature screen.

Step 3

In the Define MWI screen, set up the parameters as described in the following table.

Parameter Name

Description and Default Value

Label

The label of the PLK. For example, VM 3300. If this parameter is missing, the key displays the name part of the User ID parameter.

This parameter is optional.

User ID

The SIP address of a voicemail account. For example, 4085283300@$PROXY.

This parameter is mandatory.

Number

The speed dial number or the SIP URI. For example, 8000 ,3300#,123456#

Step 4

Click Save.

Related Tasks

Enable Direct PLK Configuration

Related References

String Syntax for Voicemail PLK

Chapter 15. Corporate and Personal Directory Setup

- Configure Directory Services

- LDAP Configuration

- Configure BroadSoft Settings

- Set up Personal Directory

- Enable Reverse Name Lookup

Configure Directory Services

With the Directory Services, you control the display of the directories:

- Personal address book

- All enabled directories

Also, you control the directory browse mode and the maximum number of contacts displayed on the phone.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Directory Services, set up the fields as described in Parameters for Directory Services.

Step 3

Click Submit All Changes.

- Parameters for Directory Services

- Disable Contact Search in All Directories

- Disable Personal Directory

Parameters for Directory Services

The following table defines the function and usage of the parameters in the Directory Services section under the Voice > Phone 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.

Parameters for Directory Services

Parameter

Description

Personal Directory Enable

Enables the personal address book directory for the phone user.

Select Yes to enable the directory and select No to disable it.

If you disable the directory:

- users can't search contacts from their personal address book

- users can't add a contact in their personal address book

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the personal address book directory.

Valid values: Yes|No

Default: Yes

Search All Enable

Determines whether the phone user can search for contacts in the All directories.

Select Yes to enable the search operation and select No to disable it.

The All directories contain the following directories with the priority from highest to lowest:

1. Personal address book

2. BroadSoft directory

3. LDAP directory

4. Bluetooth phone directory

The All directories only contain the enabled directories.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the search operation.

Valid values: Yes|No

Default: Yes

Browse Mode Enable

Determines whether to trigger an auto preload operation to show the contacts when you enter a directory in the phone.

Select Yes to enable the browse mode for any directories and select No to disable it.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the browse mode.

Valid values: Yes|No

Default: No

Max Display Records

Sets up the maximum number of contacts that display in a directory.

Perform one of the following:

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

<Max_Display_Records ua="na">50</Max_Display_Records>

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

The phone only displays distinct contacts. If there're duplicated contacts in the directories, the number of displayed contacts might be less than the configured value.

Value range: 50 to 999

Default: 50

Disable Contact Search in All Directories

By default, the user can search for contacts in all the directories on the phone. You can configure the phone to disable this feature. Then, the user can only search for a contact in a single directory each time.

When you complete this procedure, the All directories option doesn't display under the Directories menu on the phone screen.

You can also configure this parameter in the configuration file (cfg.xml) with a string in this format:

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

The valid values are Yes and No. The default setting is Yes.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Directory Services section, set the Search All Enable field to No.

Step 3

Click Submit All Changes.

Disable Personal Directory

By default, the personal directory is enabled on the phone. You can disable the personal directory from the phone web interface. When you disable the personal directory:

- the Personal Directory tab doesn't display in the phone web interface.

- the Personal address book option doesn't display on the Directories phone screen.

- the user can't add contacts to the personal directory from the call history or other directories.

- the phone skips the personal directory when the user searches for a contact in all the directories.

- as the user dials a number with the keypad or there is an incoming call, the phone skips the personal directory when it searches for a matching number in directories.

You can also configure the parameter in the configuration file (cfg.xml) with a string in this format:

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

The valid values are Yes and No. The default setting is Yes.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Directory Services section, set the Personal Directory Enable field to No.

By default, this field is set to Yes.

Step 3

Click Submit All Changes.

LDAP Configuration

The Cisco IP Phone supports Lightweight Directory Access Protocol (LDAP) v3. LDAP Corporate Directory Search allows a user to search a specified LDAP directory for a name, phone number, or both. LDAP-based directories, such as Microsoft Active Directory 2003 and OpenLDAP-based databases, are supported.

Users access LDAP from the Directory menu on their IP phone. An LDAP search returns up to 20 records.

The instructions in this section assume that you have installed an LDAP server, such as OpenLDAP or Microsoft Active Directory Server 2003.

- Prepare the LDAP Corporate Directory Search

- Overview of LDAP Directory Access

Prepare the LDAP Corporate Directory Search

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the IPv4 Settings section, enter the IP address of the DNS server in the Primary DNS field.

This step is required only if you are using Active Directory with authentication set to MD5.

You can configure this parameter in the configuration file by entering a string in this format:

<Primary_DNS ua="na">10.74.2.7</Primary_DNS>

Step 3

In the Optional Network Configuration section, in the Domain field, enter the LDAP domain.

This step is required only if you are using Active Directory with authentication set to MD5.

Some sites might not deploy DNS internally and instead use Active Directory 2003. In this case, it is not necessary to enter a Primary DNS address and an LDAP Domain. However, with Active Directory 2003, the authentication method is restricted to Simple.

You can configure this parameter in the configuration file by entering a string in this format:

<Domain ua="na">LDAPdomainname.com</Domain>

Step 4

Click the Phone tab.

Step 5

Configure the LDAP fields as described in Parameters for LDAP Directory.

Step 6

Click Submit All Changes.

- Parameters for LDAP Directory

Parameters for LDAP Directory

The following table defines the function and usage of the LDAP directory parameters in the LDAP section under the Voice > Phone 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.

Parameters for LDAP Directory

Parameter

Description

LDAP Dir Enable

Enables or disables the LDAP directory.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No to enable or disable LDAP directory.

Valid values: Yes and No

Default: No

Corp Dir Name

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

Perform one of the following:

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

<LDAP_Corp_Dir_Name ua="na">Coprorate Directory</LDAP_Corp_Dir_Name>

- In the phone web interface, enter the name of the corporate directory.

Valid values: Text string with no more than 63 characters

Default: Empty

Server

Enter a fully qualified domain name or IP address of an LDAP server.

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

Perform one of the following:

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

<LDAP_Server ua="na">ldapserver.com</LDAP_Server>

- In the phone web interface, enter IP address or host name of the LDAP server.

Default: Empty

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

Perform one of the following:

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

<LDAP_Search_Base ua="na">dc=cv2bu,dc=com</LDAP_Search_Base>

- In the phone web interface, enter the search base.

Default: Empty

Client DN

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

dc=cv2bu,dc=com

If you’re 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

cn=”David Lee”,dc=cv2bu,dc=com

username@domain is the client DN format for a Windows server

For example, DavidLee@cv2bu.com

This parameter is available when Auth Method is set to Simple.

Perform one of the following:

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

<LDAP_Client_DN ua="na">dc=cv2bu,dc=com</LDAP_Client_DN>

- In the phone web interface, enter the client domain name.

Default: Empty

User Name

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

This parameter is available when Auth Method is set to DIGEST-MD5.

Perform one of the following:

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

<LDAP_User_Name ua="na">dc=cv2bu,dc=com</LDAP_User_Name>

- In the phone web interface, enter the user name.

Default: Empty

Password

If you allow the user to access the LDAP directory without entering the credentials, enter the password for the user in this field. If you allow access of specific users, leave this field empty. The phone prompts for credentials to access the LDAP directory.

User entry of credentials on the phone updates this field and the configuration file.

The password entered in this field shows as the following in the configuration file (cfg.xml).

<!-- <LDAP_Password ua="na">**********</LDAP_Password >-->

Default: Empty

Auth Method

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

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

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

If selected, the phone prompts the Client DN and Password credentials to access the LDAP directory.

If either or both of the credentials are empty, the operation used to authenticate the clients is the anonymous simple bind. The success of the operation depends on whether the LDAP server supports it.

Users can access the LDAP directory without the need to enter the user credentials when the one of the following situations is satisfied:

- The user credentials are cached on the phone.

- The LDAP server allows the anonymous simple bind operation, and the operation succeeds. And the parameter LDAP Prompt For Empty Credentials is set to No.

- 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.

If selected, the phone prompts the Username and Password credentials to access the LDAP directory.

Users can access the LDAP directory without the need to enter the user credentials when the credentials are cached on the phone.

For more information, see Overview of LDAP Directory Access.

Perform one of the following:

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

<LDAP_Auth_Method ua="na">Simple</LDAP_Auth_Method>

- In the phone web interface, choose a authentication method.

Default: None

LDAP Prompt For Empty Credentials

Enable or disable the LDAP sign-in prompt when there’s no user credential on the phone. This function is used only for the simple authentication method that involves the anonymous simple bind operation.

- If the value is Yes, the phone always prompts for LDAP credentials. If the LDAP server supports the anonymous simple bind, users can either enter the credentials or leave them empty to access the LDAP directory.

- If the value is No, users can directly access the LDAP directory when the anonymous simple bind operation succeeds.

If the LDAP server doesn't support the anonymous simple bind (empty credentials), then the users must enter the client DN and password to access the LDAP directory.

This parameter doesn't display on the phone administration web page. To configure the parameter, perform the following:

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

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

Valid values: Yes and No

Default: No

StartTLS Enable

Enable or disable the Start Transport Layer Security (StartTLS) operation. It provides the ability to establish TLS in an LDAP session.

When StartTLS Enable is set to Yes, the phone behaviour vary depending on the LDAP server setting:

- If the LDAP server is defined as "ldap://server:port", then the phone sends the StartTLS request to the LDAP server.

- If the LDAP server is defined as "ldaps://server:port", then the phone directly performs the LDAP over TLS (LDAPS) operation.

When StartTLS Enable is set to No, the phone behaviour vary depending on the LDAP server setting:

- If the LDAP server is defined as "ldap://server:port", then the phone performs the LDAP operation.

- If the LDAP server is defined as "ldaps://server:port", then the phone performs the LDAPS operation.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes or No to enable or disable the StartTLS operation.

Valid values: Yes and No

Default: No

Last Name Filter

Use this field to specify how the phone must perform searches based on the last name or surname (sn), when users search for contacts.

Examples:

sn:(sn=$VALUE*) instructs the phone to find all last names that begin with the entered search string.

sn:(sn=*$VALUE*) instructs the phone to find all last names in which the entered search string appears anywhere in the last name. This method is more inclusive and retrieves more search results. This method is consistent with the search method in other directories such as the BroadSoft directories and the user's personal address book on the phone.

Perform one of the following:

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

<LDAP_Last_Name_Filter ua="na">sn:(sn=L*)</LDAP_Last_Name_Filter>

- In the phone web interface, enter the filter.

Default: Empty

First Name Filter

Use this field to specify how the phone must perform searches based on the first name or common name (cn), when users search for contacts.

Examples:

cn:(cn=$VALUE*) instructs the phone to find all first names that begin with the entered search string.

cn:(cn=*$VALUE*) instructs the phone to find all first names in which the entered search string appears anywhere in the first name. This method is more inclusive and retrieves more search results. This method is consistent with the search method in other directories such as the BroadSoft directories and the user's personal address book on the phone.

Perform one of the following:

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

<LDAP_First_Name_Filter ua="na">cn:(cn=John*)</LDAP_First_Name_Filter>

- In the phone web interface, enter the filter.

Default: Empty

Search Item 3

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

This parameter is used only for the reserve name lookup feature for the LDAP directory. For more information about the feature, see Enable Reverse Name Lookup.

Perform one of the following:

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

<LDAP_Search_Item_3 ua="na">search_item</LDAP_Search_Item_3>

- In the phone web interface, enter a name for the additional item to search.

Default: Empty

Search Item 3 Filter

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

This parameter is used only for the reserve name lookup feature for the LDAP directory. For more information about the feature, see Enable Reverse Name Lookup.

Perform one of the following:

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

<LDAP_Item_3_Filter ua="na">cn:(cn=John*)</LDAP_Item_3_Filter>

- In the phone web interface, enter the filter.

Default: Empty

Search Item 4

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

Perform one of the following:

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

<LDAP_Search_Item_4 ua="na">search_item</LDAP_Search_Item_4>

- In the phone web interface, enter a name for the additional item to search.

Default: Empty

Search Item 4 Filter

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

Perform one of the following:

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

<LDAP_Item_4_Filter ua="na">cn:(cn=John*)</LDAP_Item_4_Filter>

- In the phone web interface, enter the filter.

Default: Empty

Display Attrs

Format of LDAP results displayed on phone, where:

- a—Attribute name

For example, a=telephoneNumber means that the attribute name is used for a phone number.

Other typical values: facsimileTelephoneNumber, mobile, mobiletelephonenumber, ipphone, homephone, otherphone, and pagertelephonenumber

- cn—Common name

- sn—Surname (last name)

- n—Display name

For example, n=Phone causes "Phone" to be displayed in front of the phone number of an LDAP query result when the details softkey is pressed.

- t—type

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

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

- p—phone number

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

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

Perform one of the following:

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

<LDAP_Display_Attrs ua="na">a=givenName,n=firstname;a=sn,n=lastname;a=cn,n=cn;a=telephoneNumber,n=tele,t=p;a=mail,n=mail;a=postalAddress,n=address;a=postalCode,n=postalcode;a=uid,n=uid;a=facsimileTelephoneNumber,n=fax;a=title,n=title</LDAP_Display_Attrs>

- In the phone web interface, enter the attributes to display.

Default: Empty

Number Mapping

With the LDAP number mapping, you can manipulate the number that was retrieved from the LDAP server. For example, you can append 9 to the number if your dial plan requires a user to enter 9 before dialing. Add the 9 prefix by adding (<:9xx.>) to the LDAP Number Mapping field. For example, 555 1212 would become 9555 1212.

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

Leave this field blank if not needed.

Perform one of the following:

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

<LDAP_Number_Mapping ua="na"><:9xx.></LDAP_Number_Mapping>

- In the phone web interface, enter the mapping number.

Default: Empty

Overview of LDAP Directory Access

The following diagram shows the logic of the LDAP directory access in different authentication methods:

Configure BroadSoft Settings

The BroadSoft directory service enables users to search and view their personal, group, or enterprise contacts. This application feature uses BroadSoft's Extended Services Interface (XSI).

To improve security, the phone firmware places access restrictions on the host server and directory name entry fields.

The phone uses two types of XSI authentication methods:

- User login credentials: The phone uses the XSI user id and password.

- SIP credentials: The register name and password of the SIP account registered on the phone. For this method, the phone can use the XSI user ID along with the SIP authentication credentials for the authentication.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the XSI Service section, choose Yes from the Directory Enable drop down list box.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

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

Step 3

Set up the fields as described in Parameters for XSI Phone Service.

Step 4

Click Submit All Changes.

- Parameters for XSI Phone Service

Parameters for XSI Phone Service

The following table defines the function and usage of the XSI directory parameters in the XSI Phone Service section under the Voice > Phone 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.

Parameters for XSI Phone Service

Parameter

Description

XSI Host Server

Enter the name of the server; for example,

xsi.iop1.broadworks.net

Note

XSI Host Server uses http protocol by default. To enable XSI over HTTPS, you can specify https:// in the server.

Perform one of the following:

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

<XSI_Host_Server ua="na">https://xsi.iop1.broadworks.net</XSI_Host_Server>

- In the phone web interface, enter the XSI server to use.

Default: Empty

XSI Authentication Type

Determines the XSI authentication type.

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

Perform one of the following:

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

<XSI_Authentication_Type ua="na">SIP Credentials</XSI_Authentication_Type>

- In the phone web interface, specify the authentication type for XSI service.

Valid values: Login credentials|SIP Credentials

Default: Login Credentials

Login User ID

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

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

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

Perform one of the following:

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

<Login_User_ID ua="na">username</Login_User_ID>

- In the phone web interface, enter the username used to authenticate the access to the XSI server.

Default: Empty

Login Password

Alphanumeric password associated with the User ID.

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

Default: Empty

SIP Auth ID

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

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

Perform one of the following:

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

<SIP_Auth_ID ua="na">username</SIP_Auth_ID>

- In the phone web interface, enter the username used to authenticate the access to the XSI server.

Default: Empty

SIP Password

The password of the SIP account registered on the phone.

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

Default: Empty

Directory Enable

Enables BroadSoft directory for the phone user.

Select Yes to enable the directory and select No to disable it.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the BroadSoft directory.

Valid values: Yes and No

Default: No

Directory Individual Mode Enable

Enables the individual mode for the BroadSoft directories. The parameter is valid only when Directory Enable is set to Yes.

When this mode is enabled, the individual BroadSoft directories (such as, Enterprise, Group, Personal, and so on) display in the phone.

When this mode is disabled, only the BroadSoft directory displays in the phone.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the individual mode for the BroadSoft directories.

Valid values: Yes and No

Default: No

Directory Type

Select the type of BroadSoft directory:

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

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

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

- Enterprise Common: Allows users to search on name or number.

- Group Common: Allows users to search on name or number.

This parameter is valid only when "Directory Enable" is set to Yes and "Directory Individual Mode Enable" is set to No.

Perform one of the following:

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

<Directory_Type ua="na">Enterprise</Directory_Type>

- In the phone web interface, specify the type of BroadSoft directory.

Valid values: Enterprise, Group, Personal, Enterprise Common, and Group Common

Default: Enterprise

Directory Name

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

Perform one of the following:

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

<Directory_Name ua="na">DirName</Directory_Name>

- In the phone web interface, enter the name of the BroadSoft directory to display on the phone.

Default: Empty

If the value is empty, the phone displays "BoradSoft directory".

Directory Personal Enable

Enables the BroadSoft personal directory for the phone user.

Select Yes to enable the directory and select No to disable it.

The parameter is valid only when both Directory Enable and Directory Individual Mode Enable are set to Yes.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the directory.

Valid values: Yes and No

Default: No

Directory Personal Name

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

Perform one of the following:

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

<XsiDir_Personal_Name ua="na">DirPersonalName</XsiDir_Personal_Name>

- In the phone web interface, enter the name of the directory to display on the phone.

Default: Empty

If the value is empty, the phone displays "Personal".

Directory Group Enable

Enables the BroadSoft group directory for the phone user.

Select Yes to enable the directory and select No to disable it.

The parameter is valid only when both Directory Enable and Directory Individual Mode Enable are set to Yes.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the directory.

Valid values: Yes and No

Default: No

Directory Group Name

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

Perform one of the following:

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

<XsiDir_Group_Name ua="na">DirGroupName</XsiDir_Group_Name>

- In the phone web interface, enter the name of the directory to display on the phone.

Default: Empty

If the value is empty, the phone displays "Group".

Directory Enterprise Enable

Enables the BroadSoft enterprise directory for the phone user.

Select Yes to enable the directory and select No to disable it.

The parameter is valid only when both Directory Enable and Directory Individual Mode Enable are set to Yes.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the directory.

Valid values: Yes and No

Default: No

Directory Enterprise Name

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

Perform one of the following:

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

<XsiDir_Enterprise_Name ua="na">DirEnterpriseName</XsiDir_Enterprise_Name>

- In the phone web interface, enter the name of the directory to display on the phone.

Default: Empty

If the value is empty, the phone displays "Enterprise".

Directory GroupCommon Enable

Enables the BroadSoft GroupCommon directory for the phone user.

Select Yes to enable the directory and select No to disable it.

The parameter is valid only when both Directory Enable and Directory Individual Mode Enable are set to Yes.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the directory.

Valid values: Yes and No

Default: No

Directory GroupCommon Name

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

Perform one of the following:

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

<XsiDir_GroupCommon_Name ua="na">DirGroupCommon</XsiDir_GroupCommon_Name>

- In the phone web interface, enter the name of the directory to display on the phone.

Default: Empty

If the value is empty, the phone displays "Group Common".

Directory EnterpriseCommon Enable

Enables the BroadSoft EnterpriseCommon directory for the phone user.

Select Yes to enable the directory and select No to disable it.

The parameter is valid only when both Directory Enable and Directory Individual Mode Enable are set to Yes.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the directory.

Valid values: Yes and No

Default: No

Directory EnterpriseCommon Name

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

Perform one of the following:

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

<XsiDir_EnterpriseCommon_Name ua="na">DirEnterpriseCommon</XsiDir_EnterpriseCommon_Name>

- In the phone web interface, enter the name of the directory to display on the phone.

Default: Empty

If the value is empty, the phone displays "Enterprise Common".

Add Contacts to Directory Personal

Enables the user to add contacts to the BroadSoft personal directory instead of the local personal address book.

The parameter is valid only when Directory Personal Enable is set to Yes.

- If Directory Personal Enable is set to No and Personal Directory Enable is set to Yes, the contacts will be added to the local personal address book.

Personal Directory Enable is under the Directory Services section from Voice > Phone.

- If both parameters are set to No, the user can't add the contacts on the phone.

Perform one of the following:

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

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

- In the phone web interface, set this field to Yes to enable the feature.

Valid values: Yes and No

Default: No

Set up Personal Directory

Phone users can set up personal directory from either the web interface or the Contacts > Personal address book menu on the phone. The setup of personal directory is not available in the configuration file (cfg.xml)

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select the Personal Directory tab.

Step 2

You can do the following on this tab:

- Click Add to Personal Directory to add a contact to the personal address book.
Up to 3 phone numbers can be added to a contact entry.

- Click Edit on an existing contact entry to edit the contact information.

- Click Assign to assign a speed dial index to a phone number of the contact entry.

- Select an existing contact entry and click Delete Contact to delete it.

Step 3

Click Submit All Changes.

Enable Reverse Name Lookup

Reverse name lookup searches for the name of a number in an incoming, outgoing, conference, or transferred call. The reverse name lookup acts when the phone cannot find a name using the service provider directory, Call History, or your contacts. Reverse name lookup needs a valid BroadSoft (XSI) Directory, LDAP Directory, or XML Directory configuration.

The reverse name lookup searches the phone's external directories. When a search succeeds, the name is placed in the call session and in the call history. For simultaneous, multiple phone calls, reverse name lookup searches for a name to match the first call number. When the second call connects or is placed on hold, reverse name lookup searches for a name to match the second call. The reverse lookup searches the external directories for 8 secs, if in 8secs there are no results found, there will be no display of the name. If results are found in 8secs, the name is diplayed on the phone. The external directory search priority order is : BroadSoft (XSI) > LDAP > XML.

While searching if the lower priority name is received before the higher priority name, the search shows the lower prioirty name first and then replaced it with the higher priority name if the higher priority name is found within 8 secs.

The precedence of the phone list lookup in BroadSoft (XSI) Directory is:

1. Personal phone list

2. Group common phone list

3. Enterprise common phone list

Reverse name lookup is enabled by default.

Reverse name lookup searches the directories in the following order:

1. Personal Address Book

2. SIP Header

3. Call History

4. BroadSoft (XSI) Directory

5. LDAP Directory

6. XML Directory

Note

The phone searches the XML directory using this format: directory_url?n=incoming_call_number.

Example: For a multiplatform phone using a third-party service, the phone number (1234) search query has this format, http://your-service.com/dir.xml?n=1234.

Before you begin

- Configure one of these directories before you can enable or disable the reverse name lookup:

- BroadSoft (XSI) Directory

- LDAP Corporate Directory

- XML Directory

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Phone.

Step 2

In the Supplementary Services area, set the Reverse Phone Lookup Serv parameter to Yes to enable this feature.

You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:

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

The allowed values are Yes|No. The default value is Yes.

Step 3

Click Submit All Changes.

Chapter 16. Cisco IP Phone 6800 Key Expansion Module

- Cisco 6800 Series Key Expansion Module Setup Overview

- Configure the Key Expansion Module with the Phone Web Interface

- Configure a Speed Dial on a Key Expansion Module

- Configure Call Park on a Key Expansion Module line

- Configure the Busy Lamp Field on a Key Expansion Module

- Configure the Voicemail PLK on a Key Expansion Module Button

- Access Key Expansion Module Setup

- Adjust the Contrast of Key Expansion Module LCD from the Phone Web Page

- Change the Display Mode of Key Expansion Module from the Phone Web Page

- Enable the User to Configure Features on Key Expansion Module Line Keys

- Add a Menu Shortcut to a Key Expansion Module Line Key

- Add an Extended Feature to a Key Expansion Module Line Key

- Shut Down a Line Key on a Key Expansion Module

Cisco 6800 Series Key Expansion Module Setup Overview

The Cisco IP Phone 6800 Key Expansion Module add extra programmable buttons to the phone. The programmable buttons can be set up as speed-dial buttons, or phone feature buttons.

The Cisco IP Phone 6851 Multiplatform Phone supports only one key expansion module. The key expansion module provides 14 lines or programmable buttons, and two pages. Each page contains 14 lines or buttons.

Configure the Key Expansion Module with the Phone Web Interface

You can add number of supported key expansion modules from the phone web interface.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

From the Number of Units list, select 1 as the number of supported key expansion modules.

You can also configure the parameter in the configuration file (cfg.xml) by entering a string in the following format:

<Number_of_Units ua="na">2</Number_of_Units>

Default: 0

Step 3

Click Submit All Changes.

Configure a Speed Dial on a Key Expansion Module

You can configure speed dial on a key expansion module line. The user can then press the line key to call a frequently dialed number.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Select a key expansion module line key on which to enable the speed dial.

Step 3

Enter a string in this format:

fnc=sd;ext=9999@$PROXY;vid=n;nme=xxxx

where:

- fnc= sd means function=speed dial

- ext= 9999 is the phone that the line key calls. Replace 9999 with numbers.

- vid=n is the line index of the phone.

- nme= XXXX is the name displayed on the phone for the speed-dial line key. Replace XXXX with a name.

You can also configure the parameter in the configuration file (cfg.xml) by entering a string in the following format:

<Unit_n_Key_m>fnc=sd;ext=9999@$PROXY;vid=n;nme=xxxx

Step 4

Click Submit All Changes.

Configure Call Park on a Key Expansion Module line

You can enable call park on a key expansion module line. The user can then use the line to park a call. and then retrieve the call either from own phone or another phone.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Select a key expansion module line key on which to enable the call park.

Step 3

Enter a string in this format:

fnc=prk;ext=9999@$PROXY;vid=n;nme=xxxx

where:

- fnc= prk means function=call park

- ext= 9999 is the phone that the line key calls. Replace 9999 with numbers.

- vid=n is the line index of the phone.

- nme= XXXX is the name displayed on the phone for the call park line key. Replace XXXX with a name.

You can also configure an XML service on key expansion module key. Enter the string in this format:

fnc=xml;url=http://xml.service.url;nme=name

Configure the Busy Lamp Field on a Key Expansion Module

You can configure the busy lamp field on a key expansion module line so that the user can monitor a coworker's availability to receive a call.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Select a key expansion module line key.

Step 3

Enter a string in this format:

fnc=blf;sub=xxxx@$PROXY;usr=8888@$PROXY.

Where:

- fnc= blf means function=busy lamp field

- sub= the URI to which the SUBSCRIBE message is sent. This name must be identical to the name defined in the List URI: sip: parameter. xxxx is the name that is defined in List URI: sip: parameter. Replace xxxx with the exact defined name. $PROXY is the server. Replace $PROXY with the server address or name.

- usr= the BroadSoft user being monitored by BLF with 8888 as the phone being monitored. Replace 8888 with the exact number of the monitored phone. $PROXY is the server. Replace $PROXY with the server address or name.

Step 4

(Optional) To enable the busy lamp field to work with both speed dial and call pickup enter a string in the following format:

fnc=blf+sd+cp;sub=xxxx@$PROXY;usr=yyyy@$PROXY.

Where:

sd= speed dial

cp= call pickup

You can also enable busy lamp field only with call pickup or speed dial. Enter the string in the following format:

fnc=blf+cp;sub=xxxx@$PROXY;usr=yyyy@$PROXY

fnc=blf+sd;sub=xxxx@$PROXY;usr=yyyy@$PROXY

You can also configure the parameter in the configuration file (cfg.xml) by entering a string in the following format:

<Unit_1_Key_2_ ua="na">fnc=blf;ext=3252@$PROXY;nme=BLF_3252</Unit_1_Key_2_>

Step 5

Click Submit All Changes.

Configure the Voicemail PLK on a Key Expansion Module Button

You can configure the voicemail Programmable Line Key (PLK) on a Key Expansion Module button for the users to monitor a specified voicemail account of a user or a group.

The voicemail PLK can monitor both the voicemail of an extension and the voicemail account of another user or a group. Monitoring the voicemail of another user or a group requires the support from the SIP proxy.

For example, if the users belong to a customer service group. This feature allows the users to monitor both their voicemails and the group's voicemails.

If you configure speed dial for the same button, the users can press the button to make a speed dial to the assigned extension.

Before you begin

One or more expansion modules have been installed on the phone.

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Select an expansion module button on which to configure the voicemail PLK.

Step 3

Enter a string in this format:

- For MWI only:

fnc=mwi;sub=group_vm@domain;vid=1;nme=Group;

- For MWI + Speed Dial:

fnc=mwi+sd;ext=8000@domain;sub=group_vm@domain;vid=1;nme=Group;

- For MWI + speed dial + DTMF:

fnc=mwi+sd;ext=8000 ,4085283300#,123456#@domain;sub=group_vm@domain;vid=1;nme=Group;

For more information about the string syntax, see String Syntax for Voicemail PLK.

You can also configure this parameter in the phone configuration file (cfg.xml). The parameter is line-specific. Enter a string in this format:

<Unit_1_Key_1_ ua="na">fnc=mwi+sd;ext=8000 ,4085283300#,123456#@domain;
sub=group_vm@domain;vid=1;nme=Group;</Unit_1_Key_1_>

Step 4

In the General section, add mwi or mwi;sd in the field Customizable PLK Options.

Parameter in the configuration file (cfg.xml):

<Customizable_PLK_Options ua="na">mwi;sd</Customizable_PLK_Options>

After the configuration, users can configure the corresponding features on the expansion module button.

Step 5

Click Submit All Changes.

Access Key Expansion Module Setup

After you install a key expansion module on the phone and configure it on the phone web page, the phone automatically recognizes the key expansion module.

When the phone automatically recognizes the key expansion module, you can then choose the Details softkey for additional information about the selected key expansion module.

Procedure

Step 1

On the phone, press Applications .

Step 2

Press Status > Accessories.

Installed and configured key expansion module display in the list of accessories.

Adjust the Contrast of Key Expansion Module LCD from the Phone Web Page

You can adust the LCD contrast of the Key Expansion Module from the phone web page and the value gets updated on the phone. You can modify this value from the phone.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

In the General section, enter a value in the Attendant Console LCD Contrast field.

Valid values: 4 to 12

Step 3

Click Submit All Changes.

Change the Display Mode of Key Expansion Module from the Phone Web Page

You can modify the busy lamp field (BLF) label of Key Expansion Module line. The labels can be displayed by name, extension, or both. The change updates the phone. You can also modify the display mode from the phone.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

In the General section, select the type of display mode from the BLF Label Display Mode field.

Step 3

Click Submit All Changes.

Enable the User to Configure Features on Key Expansion Module Line Keys

You can enable the user to configure features on the line keys of the key expansion module. The user can then add any of the configured features to the dedicated line keys. For the supported features, see Configurable Features on Line Keys.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Ensure that the line keys of the key expansion module are not in the Inert mode.

Procedure

Step 1

Select Voice > Att Console.

Step 2

In the General section, configure the Customizable PLK Options parameter with the codes of your desired features defined in Configurable Features on Line Keys.

Example: You configure this parameter with blf;shortcut;dnd;. The user can call up the feature list with a long-press on a key expansion module line key. The feature list looks like:

1 None

2 BLF presence

4 Menu shortcut

3 Do not disturb

The user can then select a feature or a menu shortcut to add to the line key.

You can also configure this parameter in the configuration file (cfg.xml) with a string in this format:

<Customizable_PLK_Options ua="na">blf;shortcut;dnd;</Customizable_PLK_Options>

Step 3

Click Submit All Changes.

Add a Menu Shortcut to a Key Expansion Module Line Key

You can add a menu shortcut to a line key of the attached key expansion module. Then, the user can press the configured line key to access the menu.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Go to the Unit (n) section, where n is unit number of the key expansion module.

Step 3

Configure the Unit n Key m field, where n is unit number of the key expansion module, and m is the key number.

fnc=shortcut;url=userpref;nme=User preferences

where:

- fnc= shortcut means function=phone menu shortcut.

- url= userpref is the menu to open with this line key. It's the User preferences menu in this example. For more shortcut mappings, see Menu Shortcuts Mapping on PLK and PSK.

- nme= XXXX is the menu shortcut name displayed on the key expansion module screen. If you don't specify a display name, the line key displays the target menu item. In the example, the line key displays User preferences.

You can also configure the parameter in the configuration file (cfg.xml) with a string in this format:

<Unit_n_Key_m_ ua="na">fnc=shortcut;url=userpref;nme=User preferences</Unit_n_Key_m_>

where n is the unit number of the key expansion module, and m is the key number.

Step 4

Click Submit All Changes.

Add an Extended Feature to a Key Expansion Module Line Key

You can add a feature to a line key of the attached key expansion module. Then, the user can press the line key to access the feature. For the supported features, see Configurable Features on Line Keys.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Att Console.

Step 2

Go to the Unit (n) section, where n is unit number of the key expansion module.

Step 3

Configure the Unit n Key m field, where n is unit number of the key expansion module, and m is the key number.

fnc=dnd

The user can turn on or off Do not disturb mode with the line key. For more feature codes, see Configurable Features on Line Keys.

You can also configure the parameter in the configuration file (cfg.xml) with a string in this format:

<Unit_n_Key_m_ ua="na">fnc=dnd</Unit_n_Key_m_>

where n is the unit number of the key expansion module, and m is the key number.

Step 4

Click Submit All Changes.

Shut Down a Line Key on a Key Expansion Module

You can shut down a line key on a key expansion module by setting the Inert mode for it from the phone web page. When the key expansion module line key is in the Inert mode, it is disabled thoroughly. For example, the key expansion module LED is disabled (including LED behavior), no icon or text displays next to the key expansion module line key, and the key expansion module button is not responsive. In a word, it is completely unavailable.

Before you begin

Access the phone administration web interface. See Access the Phone Web Interface

Procedure

Step 1

Select Voice > Att Console.

Step 2

Go to the Unit (n) section, where n is unit number of the key expansion module.

Step 3

Configure the Unit n Key m field, where n is unit number of the key expansion module, and m is the key number.

fnc=inert;

where fnc=inert means function=inert.

You can also configure the parameter in the configuration file (cfg.xml) with a string in this format:

<Unit_n_Key_m_ ua="na">fnc=inert;</Unit_n_Key_m_>

where n is the unit number of the key expansion module, and m is the key number.

Step 4

Click Submit All Changes.

Part III. Cisco IP Phone Installation

- Cisco IP Phone Installation

Chapter 17. Cisco IP Phone Installation

- Verify the Network Setup

- Install the Cisco IP Phone

- Configure the Network from the Phone

- Wi-Fi Settings

- Verify Phone Startup

- Disable or Enable DF Bit

- Configure Internet Connection Type

- Configure VLAN Settings

- SIP Configuration

- NAT Transversal with Phones

- Dial Plan

- Regional Parameters Configuration

- Cisco IP Phone 6800 Series Documentation

Verify the Network Setup

For the phone to operate successfully as an endpoint in your network, your network must meet specific requirements.

Procedure

Step 1

Configure a VoIP Network to meet the following requirements:

- VoIP is configured on your routers and gateways.

Step 2

Set up the network to support one of the following:

- DHCP support

- Manual assignment of IP address, gateway, and subnet mask

Install the Cisco IP Phone

After the phone connects to the network, the phone startup process begins, and the phone registers with the Third-Party Call Control System. To finish phone installation, you need to configure the network settings on the phone either manually or with DHCP.

If you only have one LAN cable at your desk, you can plug your phone into the LAN with the SW port and then connect your computer into the PC port.

You can also daisy chain two phones together. Connect the PC port of the first phone to the SW port of the second phone.

Note

Cisco IP Phone 6861 Multiplatform Phones doesn't have a PC port. You can connect Cisco IP Phone 6861 to a wireless network through Wi-Fi.

Caution

Do not connect the SW and PC ports into the LAN.

Procedure

Step 1

Choose the power source for the phone:

- Power over Ethernet (PoE)—Cisco IP Phone 6821, 6851, and 6871 Multiplatform Phones

- External power supply

Step 2

Connect the handset to the handset port.

The Cisco IP Phone 6861 and 6871 Multiplatform Phones ship with a wideband handset. The Cisco IP Phone 6821, 6841, and 6851 Multiplatform Phones ship with a narrowband handset. You can buy a wideband-capable handset that is designd especially for use with the phone.

The handset includes a light strip that indicates incoming calls and waiting voice messages.

Step 3

Connect a headset to the headset port. You can add a headset later if you do not connect one now.

Step 4

Connect your phone to a network. Do one of the following actions:

- Connect a straight-through Ethernet cable from the switch to the network port on the phone. Each phone ships with one Ethernet cable in the box.

Use Category 3, 5, 5e, or 6 cabling for 10 Mbps connections; Category 5, 5e, or 6 for 100Mbps connections; and Category 5e or 6 for 1000 Mbps connections. For more information, see Network and Computer Port Pinouts.

- Connect a straight-through Ethernet cable from another network device, such as a desktop computer, to the computer port on the phone. You can connect another network device later if you do not connect one now.

Use Category 3, 5, 5e, or 6 cabling for 10 Mbps connections; Category 5, 5e, or 6 for 100Mbps connections; and Category 5e or 6 for 1000 Mbps connections. For more information, see Network and Computer Port Pinouts for guidelines.

- For Cisco IP Phone 6861 Multiplatform Phones, you can connect the phone to a wireless network through Wi-Fi. For more information, see Wi-Fi Settings.

Step 5

If the phone is wall-mounted, you might need to adjust the handset rest to ensure that the receiver cannot slip out of the cradle.

Step 6

Monitor the phone startup process. This step verifies that the phone is configured properly.

Step 7

Use DHCP or manually enter an IP address for the phone.

See Configure the Network from the Phone.

Step 8

Upgrade the phone to the current firmware image.

Firmware upgrades over the WLAN interface may take longer than upgrading over the wired interface, depending on the quality and bandwidth of the wireless connection. Some upgrades may take more than one hour.

Step 9

Make calls with the phone to verify that the phone and features work correctly.

See the Cisco IP Phone 6800 Series Multiplatform Phones User Guide.

Step 10

Provide information to end users about how to use their phones and how to configure their phone options. This step ensures that users have adequate information to successfully use their phones.

Configure the Network from the Phone

The phone includes many configurable network settings that you may need to modify before it is functional for your users. You can access these setting through the phone menus.

The Network configuration menu provides you with options to view and configure a variety of network settings.

You can configure settings that are display-only on the phone in your Third-Party Call Control system.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration.

Step 3

Use the navigation arrows to select the desired menu and edit.

Step 4

To display a submenu, repeat step 3.

Step 5

To exit a menu, press Back .

- Network Configuration Fields

- Text and Menu Entry From the Phone

Network Configuration Fields

Network Configurations Menu Options

Field

Field Type or Choices

Default

Description

Ethernet configuration

See the following Ethernet configuration submenu table.

IP mode

Dual mode

IPv4 only

IPv6 only

Dual mode

Select the Internet Protocol mode for which the phone operates.

In dual mode, the phone can have both IPv4 and IPv6 addresses.

Wi-Fi configuration

See Set Up a Wi-Fi Profile from the Phone

For Cisco IP Phone 6861 Multiplatform Phones only.

IPv4 address settings

DHCP

Static IP

Release DHCP IP

DHCP

See the IPv4 address submenu table in the following tables.

IPv6 address settings

DHCP

Static IP

DHCP

See the IPv6 address submenu table in the following tables.

DHCPv6 option to use

17, 160, 159

Indicates the order in which the phone uses the IPv6 addresses provided by DHCP server.

HTTP proxy settings

See the following HTTP proxy settings submenu table.

VPN settings

See the following VPN settings submenu table.

Web server

On

Off

On

Indicates whether the phone has web server enabled or disabled.

Ethernet Configuration Submenu

Field

Field Type

or Choices

Default

Description

802.1x authentication

Device authentication

Off

Enables or disables the 802.1x authentication. Valid options are:

- On

- Off

Transaction status

Disabled

- Transaction status—Indicates different authentication status when you turn on 802.1x in the Device authentication field.

- Connecting: Indicates that the authentication process is in progress.

- Authenticated: Indicates that the phone is authenticated.

- Disabled: Indicates that 802.1x authentication is disabled on the phone.

- Protocol—Displays the protocol of the server.

Switch port config

Auto

10MB half

10MB full

100MB half

100MB full

1000 full

Auto

Select speed and duplex of the network port.

If the phone is connected to a switch, configure the port on the switch to the same speed/duplex as the phone, or configure both to autonegotiate.

If you change the setting of this option, you must change the PC Port config option to the same setting.

PC port config

Auto

10MB half

10MB full

100 MB half

100MB full

100 half

1000 full (except for 6821)

Auto

Select Speed and duplex of the Computer (access) port.

If the phone is connected to a switch, configure the port on the switch to the same speed/duplex as the phone, or configure both to autonegotiate.

If you change the setting of this option, you must change the Switch Port config option to the same setting.

CDP

On

Off

On

Enable or disable Cisco Discovery Protocol (CDP).

CDP is a device-discovery protocol that runs on all Cisco manufactured equipment.

Using CDP, a device can advertise its existence to other devices and receive information about other devices in the network.

LLDP-MED

On

Off

On

Enable or disable LLDP-MED.

LLDP-MED enables the phone to advertise itself to devices that use the discovery protocol.

Startup delay

3 seconds

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

Default delay is 3 seconds.

VLAN

On

Off

Off

Enable or disable VLAN.

Permits you to enter a VLAN ID when you use VLAN without CDP or LLDP. When you use a VLAN with CDP or LLDP, that associated VLAN takes precedent over the VLAN ID you manually entered.

VLAN ID

1

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

PC port VLAN ID

1

Enter a value of the VLAN ID that is used to tag communications from the PC port on the phone.

The phone tags all the untagged frames coming from the PC (it does not tag any frames with an existing tag).

Valid values: 0 through 4095

Default: 0

PC port mirror

On

Off

Off

Adds the ability to port mirror on the PC port. When enabled, you can see the packets on the phone. Select On to enable PC port mirroring and select Off to disable it.

DHCP VLAN option

Enter a predefined DHCP VLAN option to learn the voice VLAN ID.

When you use a VLAN ID with CDP, LLDP, or manually select a VLAN ID, that VLAN ID takes precedent over the selected DHCP VLAN option.

Valid values are:

- Null

- 128 to 149

- 151 to 158

- 161 to 254

Default value is null.

Cisco recommends that you use DHCP Option 132.

IPv4 Address Settings Submenu

Field

Field Type

or Choices

Default

Description

Connection type

DHCP

Indicates whether the phone has DHCP enabled.

- DNS1—Identifies the primary Domain Name System (DNS) server that the phone uses.

- DNS2—Identifies the secondary Domain Name System (DNS) server that the phone uses.

- DHCP address released—Releases the IP address that DHCP assigned. You can edit this field if DHCP is enabled. To remove the phone from the VLAN and release the IP address for reassignment, set this field to Yes and press Set.

Static IP

When DHCP is disabled, you must set the Internet Protocol (IP) address of the phone.

- Static IP address—Identifies the IP that you assign to the phone. The phone uses this IP address instead of acquiring an IP from the DHCP server on the network.

- Subnet Mask—Identifies the subnet mask used by the phone. When DHCP is disabled, you must set the subnet mask.

- Gateway address—Identifies the default router used by the phone.

- DNS1—Identifies the primary Domain Name System (DNS) server that the phone uses. When DHCP is disabled, you must set this field manually.

- DNS2—Identifies the primary Domain Name System (DNS) server that the phone uses. When DHCP is disabled, you must set this field manually.

When you assign an IP address using this field, you must also assign a subnet mask and a gateway address. See the Subnet Mask and Default Router fields in this table.

IPv6 Address Settings Submenu

Field

Field Type

or Choices

Default

Description

Connection type

DHCP

Indicates whether the phone has Dynamic Host Configuration Protocol (DHCP) enabled.

- DNS1—Identifies the primary DNS server that the phone uses.

- DNS2—Identifies the secondary DNS server that the phone uses.

- Broadcast Echo—Identifies if the phone responses to multicast ICMPv6 message with destination address of ff02::1.

- Auto config— Identifies if the phone uses automatic configuration for the address.

Static IP

When DHCP is disabled, you must set the Internet Protocol (IP) address of the phone and must set the values of the fields:

- Static IP—Identifies the IP that you assign to the phone. The phone uses this IP address instead of acquiring an IP from the DHCP server on the network.

- Prefix length—Identifies how many bits of a Global Unicast IPv6 Address are there in the network part.

- Gateway—Identifies the default router used by the phone.

- Primary DNS—Identifies the primary DNS server that the phone uses. When DHCP is disabled, you must set this field manually.

- Secondary DNS—Identifies the primary DNS server that the phone uses. When DHCP is disabled, you must set this field manually.

- Broadcast Echo—Identifies if the phone responses to multicast ICMPv6 message with destination address of ff02::1.

VPN Settings Submenu

Field

Field Type or Choices

Description

VPN server

Enter an IP address or FQDN of the VPN server that the phone uses for the VPN connection.

Username

Enter a VPN username to access the VPN server.

Password

Enter a valid password of the username to access the VPN server.

Tunnel group

Enter a VPN tunnel group for the VPN connection.

Connect to VPN on bootup

On

Off

Determines whether the phone connects to the VPN server automatically after the phone reboots.

Default value is Off

Enable VPN connection

On

Off

Enables or disables the VPN connection.

When you enable or disable the VPN connection, the phone reboots automatically.

Default value is Off

HTTP Proxy Settings Submenu

Field

Field Type or Choices

Description

Proxy mode

Auto

Auto discovery (WPAD)—Enables or disables the Web Proxy Auto-Discovery protocol to retrieve a Proxy Auto-Configuration (PAC) file. Valid options are:

- On

- Off

If the value is set to Off, you need to further set the following field:

- PAC URL—Specifies the URL address for the PAC file that you want to retrieve. For example:

http://proxy.department.branch.example.com

The default value of Auto discovery (WPAD) is On.

Manual

- Proxy host—Specifies an IP address or hostname of the proxy server for the phone. The scheme (http:// or https://) is not required.

- Proxy port—Specifies a port number of the proxy server.

- Proxy authentication—Selects an option according to the actual situation of the proxy server. If the server requires authentication credentials to grant access to the phone, then select On. Otherwise, select Off. Options are:

- Off

- On

If the value is set to On, you need to further set the following fields:

- Username—Specifies the username of a credential user on the proxy server.

- Password—Provides the specified user's password to pass the authentication of the proxy server.

The default value of Proxy authentication is Off.

Off

Disables the HTTP proxy feature on the phone.

Text and Menu Entry From the Phone

When you edit the value of an option setting, follow these guidelines:

- Use the arrows on the navigation pad to highlight the field that you wish to edit. Press Select in the navigation pad to activate the field. After the field is activated, you can enter values.

- Use the keys on the keypad to enter numbers and letters.

- To enter letters by using the keypad, use a corresponding number key. Press the key one or more times to display a particular letter. For example, press the 2 key once for "a," twice quickly for "b," and three times quickly for "c." After you pause, the cursor automatically advances to allow you to enter the next letter.

- Press the softkey if you make a mistake. This softkey deletes the character to the left of the cursor.

- Press Back before pressing Set to discard any changes that you made.

- To enter a period (for example, in an IP address), press * on the keypad.

Note

The Cisco IP Phone provides several methods to reset or restore option settings, if necessary.

Wi-Fi Settings

Wi-Fi settings are available only on Cisco IP phone 6861 Multiplatform Phones.

You can customize Wi-Fi settings for your phone from the Wi-Fi configuration menu under the Network configuration menu on the phone. Some of the Wi-Fi settings are also available on the phone web page.

- Turn the Wi-Fi On or Off from Your phone

- Turn the Wi-Fi On or Off from the Phone Web Page

- Connect the Phone to a Wi-Fi Manually

- Connect Your Phone to a Wireless Network with WPS

- Set Up a Wi-Fi Profile from the Phone

- Set Up a Wi-Fi Profile

- Delete a Wi-Fi Profile

- Change the Order of a Wi-Fi Profile

- Scan and Save a Wi-Fi Network

- View the Wi-Fi Status

- View Wi-Fi Status Messages on the Phone

Turn the Wi-Fi On or Off from Your phone

You can enable or disable the wireless LAN of your phone from the Wi-Fi configuration menu. By default, the wireless LAN on your phone is enabled.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi .

Step 3

Press the Select button, to turn the Wi-Fi on or off.

Step 4

Press Set to save the changes.

Turn the Wi-Fi On or Off from the Phone Web Page

You can enable or disable the wireless LAN of your phone from the phone web page. You turn on the Wi-Fi so that the phone connects to a wireless network automatically or manually. By default, the wireless LAN on your phone is enabled.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

Set the Wi-Fi Settings fields as described in the Parameters for Wi-Fi Settings table.

Step 3

Click Submit All Changes.

- Parameters for Wi-Fi Settings

Parameters for Wi-Fi Settings

The following table defines the function and usage of each parameter in the Wi-Fi Settings section under the System tab in 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.

Wi-Fi Settings Parameters Table

Parameter

Description

Phone-wifi-on

Turns Wi-Fi on or off on your phone.

Perform one of the following:

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

<Phone-wifi-on ua="rw">No</Phone-wifi-on>

- In the phone web interface, set to Yes to turn on the Wi-Fi or set to No to turn it off.

Default: Yes

Phone-wifi-type

Only supported by Cisco IP Phone 6861

Controls which method the phone will connect to a wireless network.

Perform one of the following:

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

<Phone-wifi-type ua="na">WLAN</Phone-wifi-type>

- In the phone web interface, select one of the methods:

WLAN: This option requires the user to enter the credentials on the phone to connect to a protected wireless network.

WPS: The user can connect the phone to a wireless network using either the WPS button on the access point or the PIN code.

Allowed values: WLAN|WPS

Default: WLAN

Connect the Phone to a Wi-Fi Manually

When you set up a Wi-Fi profile, it provides you the options to connect the phone manually to a wireless network. You can establish the connection from the Wi-Fi profile screen or from the Setup Wi-Fi screen.

The top most Wi-Fi profile in the Wi-Fi profile screen gets connected automatically when the phone provisions.

Before you begin

- Turn on the Wi-Fi of your phone.

- Disconnect your phone with the wired network.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi profile.

Step 3

In the Wi-Fi profile screen, do any of the actions to connect to Wi-Fi.

- Select any of the configured Wi-Fi profile and click Connect.

- Press Scan and select one wireless in the Connect to Wi-Fi screen. In the Setup Wi-Fi screen, enter values in the fields and press Connect.

See the Profile Parameter table in the Set Up a Wi-Fi Profile from the Phone for the field values.

You can also configure the Wi-Fi settings using the configuration file.

<!--  Wi-Fi Settings  -->
<Phone-wifi-on ua="rw">Yes</Phone-wifi-on>
<Phone-wifi-type ua="na">WLAN</Phone-wifi-type>
<!--  available options: WLAN|WPS  -->
<!--  Wi-Fi Profile 1  -->
<Network_Name_1_ ua="rw">AP_SSID</Network_Name_1_>
<Security_Mode_1_ ua="rw">Auto</Security_Mode_1_>
<!--
available options: Auto|EAP-FAST|PEAP-GTC|PEAP-MSCHAPV2|PSK|WEP|None
-->
<Wi-Fi_User_ID_1_ ua="rw">User_ID</Wi-Fi_User_ID_1_>
<!--
<Wi-Fi_Password_1_ ua="rw">Password</Wi-Fi_Password_1_>
-->
<!--  <WEP_Key_1_ ua="rw"/>  -->
<!--  <PSK_Passphrase_1_ ua="rw"/>  -->
<Frequency_Band_1_ ua="rw">Auto</Frequency_Band_1_>
<!--  available options: Auto|2.4 GHz|5 GHz  -->
<Wi-Fi_Profile_Order_1_ ua="rw">1</Wi-Fi_Profile_Order_1_>
<!--  available options: 1|2|3|4  --><!--  Wi-Fi Profile 2  -->
<Network_Name_2_ ua="rw">AP_SSID</Network_Name_2_>
<Security_Mode_2_ ua="rw">PSK</Security_Mode_2_>
<!--
available options: Auto|EAP-FAST|PEAP-GTC|PEAP-MSCHAPV2|PSK|WEP|None
-->
<Wi-Fi_User_ID_2_ ua="rw"/>
<!--  <Wi-Fi_Password_2_ ua="rw"/>  -->
<!--  <WEP_Key_2_ ua="rw"/>  -->
<!--  <PSK_Passphrase_2_ ua="rw"/>  -->
<Frequency_Band_2_ ua="rw">Auto</Frequency_Band_2_>
<!--  available options: Auto|2.4 GHz|5 GHz  -->
<Wi-Fi_Profile_Order_2_ ua="rw">2</Wi-Fi_Profile_Order_2_>
<!--  available options: 1|2|3|4  -->
<!--  Wi-Fi Profile 3  -->
<Network_Name_3_ ua="rw"/>
<Security_Mode_3_ ua="rw">None</Security_Mode_3_>
<!--
available options: Auto|EAP-FAST|PEAP-GTC|PEAP-MSCHAPV2|PSK|WEP|None
-->
<Wi-Fi_User_ID_3_ ua="rw"/>
<!--  <Wi-Fi_Password_3_ ua="rw"/>  -->
<!--  <WEP_Key_3_ ua="rw"/>  -->
<!--  <PSK_Passphrase_3_ ua="rw"/>  -->
<Frequency_Band_3_ ua="rw">Auto</Frequency_Band_3_>
<!--  available options: Auto|2.4 GHz|5 GHz  -->
<Wi-Fi_Profile_Order_3_ ua="rw">3</Wi-Fi_Profile_Order_3_>
<!--  available options: 1|2|3|4  -->
<!--  Wi-Fi Profile 4  -->
<Network_Name_4_ ua="rw"/>
<Security_Mode_4_ ua="rw">PSK</Security_Mode_4_>
<!--
available options: Auto|EAP-FAST|PEAP-GTC|PEAP-MSCHAPV2|PSK|WEP|None
-->
<Wi-Fi_User_ID_4_ ua="rw"/>
<!--  <Wi-Fi_Password_4_ ua="rw"/>  -->
<!--  <WEP_Key_4_ ua="rw"/>  -->
<!--  <PSK_Passphrase_4_ ua="rw"/>  -->
<Frequency_Band_4_ ua="rw">Auto</Frequency_Band_4_>
<!--  available options: Auto|2.4 GHz|5 GHz  -->
<Wi-Fi_Profile_Order_4_ ua="rw">4</Wi-Fi_Profile_Order_4_>
<!--  available options: 1|2|3|4  -->

Connect Your Phone to a Wireless Network with WPS

Wi-Fi Protected Setup (WPS) provides an easier way to connect your phone to a wireless network. With WPS, you don't need to enter detailed settings for the access point to connect as you do with the connection through Wi-Fi profile. You can either use the WPS button on your access point or the PIN code to connect to the network through WPS.

The WPS option is available only in the menu on the phone screen. On the phone web page, you can only configure your phone to connect to a wireless network using Wi-Fi profile.

Before you begin

Enable WPS on your access point.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi type.

Step 3

Press the navigation key to switch the Wi-Fi type to WPS.

Step 4

Press Set.

Push-button configuration and PIN configuration display under Wi-Fi type. Follow either one of the following steps to connect to the network.

Step 5

Connect to the network using the WPS button on your access point.

1. Select Push-button configuration.

2. Press the WPS button on your access point.

The button name may vary on your access point.

3. Press Continue on your phone.

Step 6

Connect to the network using a PIN code.

1. Select PIN configuration.

An 8-digit PIN code displays on your phone screen.

2. Go to the web page of your access point and enter the PIN code.

The procedure of entering PIN code may vary on your access points. See the respective user guide of your access point for more details.

Upon either a successful connection or a connection failure, the connection status displays on your phone screen.

Set Up a Wi-Fi Profile from the Phone

You can add maximum of four Wi-Fi profiles. You can use this profile to connect your phone to a Wi-Fi network.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi profile.

Step 3

In the Wi-Fi profile screen, move to a row in the list on which you want to set up the profile.

Step 4

Press the Select button.

You can also press Options and then select Edit.

Step 5

In the Edit profile screen, set the parameters as mentioned in the Profile Parameters table.

Profile Parameters

Parameter

Description

Security mode

Allows you to select the authentication method that is used to secure access to the Wi-Fi network. Depending on the method you choose, a password, passphrase, or key field appears so that you can provide the credentials that are required to join this Wi-Fi network. Options are:

- Auto

- EAP-FAST

- PEAP-GTC

- PEAP-MSCHAPV2

- PSK

- WEP

- None

Default: PSK

Network name

Allows you to enter a name for the SSIDs. This name displays on the phone. Multiple profiles can have the same network name with different security mode.This name displays on the phone.

User ID

Allows you to enter a user ID for the network profile.

This field is available when you set the security mode to Auto, EAP-FAST, PEAP-GTC, PEAP-MSCHAPV2. This is a mandatory field and it allows maximum length of 32 alphanumeric characters.

Password

Allows you to enter password for the network profile that you create.

This field is available when you set the security mode to Auto, EAP-FAST, PEAP-GTC, PEAP-MSCHAPV2. This is a mandatory field and it allows maximum length of 64 alphanumeric characters.

WEP key

Allows you to enter password for the network profile that you create.

This field is available when you set the security mode to WEP. This is a mandatory field and it allows maximum length of 32 alphanumeric characters.

Passphrase

Allows you to enter password for the network profile that you create. You need to enter this value when the security mode is PSK.

Frequency band

Allows you to select the wireless signal frequency band that is used in the WLAN. Options are:

- Auto

- 2.4 GHz

- 5 GHz

Default: Auto

Step 6

Press Save.

Set Up a Wi-Fi Profile

You can configure a Wi-Fi profile from the phone web page or from remote device profile resync and then associate the profile to the available Wi-Fi networks. You can use this Wi-Fi profile to connect to a Wi-Fi. You can configure maximum of four profiles.

The profile contains the parameters required for phones to connect to the phone server with Wi-Fi. When you create and use a Wi-Fi profile, you or your users do not need to configure the wireless network for individual phones.

A Wi-Fi profile enables you to prevent or limit changes to the Wi-Fi configuration on the phone by the user.

We recommend that you use a secure profile with TFTP encryption enabled to protect keys and passwords when you use a Wi-Fi profile.

When you set up the phones to use EAP-FAST, PEAP-MSCHAPV, or PEAP-GTC authentication or security mode, your users need individual credentials to connect to an access point.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

Set the Wi-Fi Profile fields as described in the following table.

Field

Field Type or Choices

Default

Description

Security mode

Auto

EAP-FAST

PEP-GCT

PEAP-MSCHAPV2

PSK

WEP

None

PSK

Allows you to select the type of authentication that the phone uses to access the WLAN.

The security mode depends on the settings on your access point.

Network name

Allows you to enter a unique name for the Wi-Fi profile. This name displays on the phone.

User ID

Allows you to enter a user ID for the network profile.

Password

WEP Key

Passphrase

Allows you to enter password for the network profile that you create. The type of password depends on the security mode that you have selected.

- Password: Security mode is Auto.

- Passphrase: Security mode is PSK.

- WEP Key: Security mode is WEP.

Frequency band

- Auto

- 2.4 GHz

- 5 GHz

Auto

Allows you to select the wireless signal standard that the WLAN uses.

Step 3

Click Submit All Changes.

If the phone has an active call, you can not save the changes.

Delete a Wi-Fi Profile

You can remove a Wi-Fi profile from the list when the profile is no more required.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi profile.

Step 3

In the Wi-Fi profile screen, select the Wi-Fi profile that you want to remove.

Step 4

Press Options.

Step 5

Select Delete and then confirm the deletion.

Change the Order of a Wi-Fi Profile

You can determine the position of a Wi-Fi profile in the list. The Wi-Fi profile at the top of the list has the highest priority. When the Wi-Fi is turned on, the phone uses the Wi-Fi profile on the top of the list to connect automatically to a wirless network while provisioning.

Procedure

Step 1

If you change the Wi-Fi profile order from the phone, follow these steps:

1. Press Applications .

2. Select Network configuration > Wi-Fi configuration > Wi-Fi profile.

3. In the Wi-Fi profile screen, select a Wi-Fi of which you want to change the order.

4. Press Options.

5. Select Move up or Move down to move the Wi-Fi profile one level up or one level down respectively in the list.

Step 2

If you change the Wi-Fi profile order from the phone web page, follow these steps:

1. Select Voice > System.

2. In the Wi-Fi Profile (n) section, set the Wi-Fi Profile Order field to the desired order.

3. Click Submit All Changes.

Scan and Save a Wi-Fi Network

You can scan a Wi-Fi profile to get the list of available wireless networks (SSID). The security mode and the network name have the same value of the scanned SSID. You can then edit the fields of any of the wireless networks. When you save the changes, it saves as a Wi-Fi profile in the phone Wi-Fi profile list. You can then use this new Wi-Fi profile to connect the phone to a wireless network.

Note

- When the security mode of a wireless network is None, PSK, and WEP, you can't modify the security mode. On the Security mode screen, you only see the security mode that is set for the network. For example, if the security mode of a network is PSK, you see only PSK in the Security mode screen.

- When you scan a wireless network (SSID) which is the current connected wireless, you can't edit the Network name of this SSID.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi profile.

Step 3

In the Wi-Fi profile screen, press Scan to get all available wireless networks.

Step 4

(Optional) In the Connect to Wi-Fi screen, press Scan again to rescan the list.

Step 5

Select a wireless and press Select or the Select button.

Step 6

In the Setup Wi-Fi screen, set the parameters as mentioned in the Profile Parameters table.

Profile Parameters

Parameter

Description

Security mode

Allows you to select the authentication method that is used to secure access to the Wi-Fi network. Depending on the method you choose, a password, passphrase, or key field appears so that you can provide the credentials that are required to join this Wi-Fi network. Options are:

- Auto

- EAP-FAST

- PEAP-GTC

- PEAP-MSCHAPV2

- PSK

- WEP

- None

Default: PSK

Network name

Allows you to enter a name for the SSIDs. This name displays on the phone. Multiple profiles can have the same network name with different security mode.This name displays on the phone.

User ID

Allows you to enter a user ID for the network profile.

This field is available when you set the security mode to Auto, EAP-FAST, PEAP-GTC, PEAP-MSCHAPV2. This is a mandatory field and it allows maximum length of 32 alphanumeric characters.

Password

Allows you to enter password for the network profile that you create.

This field is available when you set the security mode to Auto, EAP-FAST, PEAP-GTC, PEAP-MSCHAPV2. This is a mandatory field and it allows maximum length of 64 alphanumeric characters.

WEP key

Allows you to enter password for the network profile that you create.

This field is available when you set the security mode to WEP. This is a mandatory field and it allows maximum length of 32 alphanumeric characters.

Passphrase

Allows you to enter password for the network profile that you create. You need to enter this value when the security mode is PSK.

Frequency band

Allows you to select the wireless signal frequency band that is used in the WLAN. Options are:

- Auto

- 2.4 GHz

- 5 GHz

Default: Auto

Step 7

Press Save.

View the Wi-Fi Status

You may experience issues related to Wi-Fi connection. You can gather information from the Wi-Fi status page to help you troubleshoot.

You can also view the status from the phone web page by selecting User Login > Advanced > Info > Status > System Information.

Procedure

Step 1

Press Applications .

Step 2

Select Network configuration > Wi-Fi configuration > Wi-Fi status.

You see the information:

- Wi-Fi status: Displays if the Wi-Fi is connected or disconnected.

- Network name: Indicates the name of the SSID.

- Signal strength: Indicates strength of the network signal.

- MAC address: Indicates MAC address of the phone.

- AP MAC address: Indicates MAC address of the access point (SSID).

- Channel: Indicated the channel on which the Wi-Fi network transmits and receives data.

- Frequency: Indicates the wireless signal frequency band that is used in the Wireless LAN.

- Security mode: Indicates the security mode that is set for the wireless LAN.

View Wi-Fi Status Messages on the Phone

You can view messages about the Wi-Fi connection status of your phone. The messages can help you diagnose Wi-Fi connection problems. The messages contain:

- connection time and MAC address of the AP

- disconnection time and diagnostic code

- connection failure time

- time that weak signal of the AP continues over 12 seconds

- the status of firmware memory when the free memory is less than 50K

- the status of losing AP beacon when the phone can't receive signal from the AP

- the status of no response for Wi-Fi authentication or association requests

- the status of TX failure

- the status of WPS connection failure

Procedure

Step 1

Press Applications .

Step 2

Select Status > Wi-Fi messages.

Step 3

Use the outer ring of the navigation cluster to scroll through the messages.

Step 4

Press Details to view more details of the selected message.

Step 5

(Optional) Press Clear to delete all the messages.

Verify Phone Startup

After the Cisco IP Phone has power connected to it, the phone automatically cycles through a startup diagnostic process.

Procedure

Step 1

If you are using Power over Ethernet, plug the LAN cable into the Network port.

Step 2

If you are using the power cube, connect the cube to the phone and plug the cube into an electrical outlet.

The buttons flash amber and then green in sequence during the various stages of bootup as the phone checks the hardware.

If the phone completes these stages successfully, it has started up properly.

Disable or Enable DF Bit

You can disable or enable Don't Fragment (DF) bit in the TCP, UDP, or ICMP messages to determine whether a packet is allowed to be fragmented.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the Network Settings section, configure the parameter Disable DF.

- If you set the Disable DF to Yes, the Don't Fragment (DF) bit is disabled. In this case, the network can fragment an IP packet. This is the default behaviour.

- If you set the Disable DF to No, the Don't Fragment (DF) bit is enabled. In this case, the network can't fragment an IP packet. This setting doesn't allow fragmentation in cases where the receiving host doesn't have sufficient resources to reassemble internet fragments.

Step 3

Click Submit All Changes.

You can also configure the parameter in the phone configuration file (cfg.xml) with the following XML string:

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

Allowed values: Yes and No

Default: Yes

Configure Internet Connection Type

You can choose how your phone receives an IP address. Set the connection type to one of the following:

- Static IP—A static IP address for the phone.

- Dynamic Host Configuration Protocol (DHCP)—Enables the phone to receive an IP address from the network DHCP server.

The Cisco IP phone typically operates in a network where a DHCP server assigns IP addresses to devices. Because IP addresses are a limited resource, the DHCP server periodically renews the phone lease on the IP address. If a phone loses the IP address, or if the IP address is assigned to another device on the network, the following occurs:

- Communication between the SIP proxy and the phone is severed or degraded.

The DHCP Timeout on Renewal parameter causes the phone to request renewal of its IP address if the following occurs:

- The phone doesn't receive an expected SIP response within programmable length of time after it sends a SIP command.

If the DHCP server returns the IP address that it originally assigned to the phone, the DHCP assignment is presumed to be operating correctly. Otherwise, the phone resets to try to fix the issue.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the IPv4 Settings section, use the Connection Type drop-down list to choose the connection type:

- Dynamic Host Configuration Protocol (DHCP)

- Static IP

Step 3

In the IPv6 Settings section, use the Connection Type drop-down list to choose the connection type:

- Dynamic Host Configuration Protocol (DHCP)

- Static IP

Step 4

If you choose Static IP, configure these settings in the Static IP Settings section:

- Static IP—Static IP address of the phone

- NetMask—Netmask of the phone (IPv4, only)

- Gateway—Gateway IP address

Step 5

Click Submit All Changes.

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

<Connection_Type ua="rw">DHCP</Connection_Type>
<!-- available options: DHCP|Static IP -->
<Static_IP ua="rw"/>
<NetMask ua="rw"/>
<Gateway ua="rw"/>

Configure VLAN Settings

The software tags your phone voice packets with the VLAN ID when you use a virtual LAN (VLAN).

In the VLAN Settings section of the Voice > System window, you can configure the different settings:

- LLDP-MED

- Cisco Discovery Protocol (CDP)

- Network Startup Delay

- VLAN ID (manual)

- DHCP VLAN Option

The multiplatform phones support these four methods to obtain VLAN ID information. The phone attempts to obtain the VLAN ID information in this order:

1. LLDP-MED

2. Cisco Discovery Protocol (CDP)

3. VLAN ID (manual)

4. DHCP VLAN Option

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- Disable CDP/LLDP and manual VLAN.

Procedure

Step 1

Select Voice > System.

Step 2

In the VLAN Settings section, configure the parameters as defined in the VLAN Settings Parameters table.

Step 3

Click Submit All Changes.

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 VLAN Settings Parameters table.

- VLAN Settings Parameters

VLAN Settings Parameters

The following table defines the function and usage of each parameter in the VLAN Settings Parameters section under the System tab in 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

Enable VLAN

Controls the VLAN feature.

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

<Enable_VLAN ua="rw">No</Enable_VLAN>

- In the phone web interface, set to Yes to enable VLAN.

The default value is Yes.

VLAN ID

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

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

<VLAN_ID ua="rw">1</VLAN_ID>

- In the phone web interface, enter an appropriate value.

Valid values: An integer ranging from 0 through 4095

Default: 1

PC Port VLAN ID

Allows you to enter a VLAN ID for the PC port.

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

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

- In the phone web interface, enter an appropriate value.

Valid values: An integer ranging from 0 through 4095

Default: 1

Enable CDP

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

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

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

- In the phone web page: set to Yes to enable CDP.

Valid values: Yes/No

Default: Yes

Enable LLDP-MED

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

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

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

<Enable_LLDP-MED ua="na">Yes</Enable_LLDP-MED>

- In the phone web interface, set to Yes to enable LLDP-MED.

Valid values: Yes/No

Default: Yes

Network Startup Delay

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

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

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

- In the phone web interface, enter the delay in seconds.

Valid values: An integer ranging from 1 through 300

Default: 3

DHCP VLAN Option

A predefined DHCP VLAN option to learn the voice VLAN ID. You can use the feature only when no voice VLAN information is available by CDP/LLDP and manual VLAN methods. CDP/LLDP and manual VLAN are all disabled.

Set the value to Null to disable DHCP VLAN option.

Cisco recommends that you use DHCP Option 132.

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

<DHCP_VLAN_Option ua="na">132</DHCP_VLAN_Option>

- In the phone web page: specify the DHCP VLAN option.

SIP Configuration

SIP settings for the Cisco IP Phone are configured for the phone in general and for the extensions.

- Configure the Basic SIP Parameters

- Configure the SIP Timer Values

- Configure the Response Status Code Handling

- Configure NTP Server

- Configure the RTP Parameters

- Control SIP and RTP Behaviour in Dual Mode

- Configure the SDP Payload Types

- Configure the SIP Settings for Extensions

- Configure the SIP Proxy Server

- Configure the Subscriber Information Parameters

- Set Up Your Phone to Use OPUS Codec Narrowband

Configure the Basic SIP Parameters

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Parameters section, set the parameters as described in the SIP Parameters table.

Step 3

Click Submit All Changes.

- SIP Parameters

SIP Parameters

Parameter

Description

Max Forward

Specifies SIP Max Forward value.

Perform one of the following.

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

<Max_Forward ua="na">70</Max_Forward>

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

Value range: 1 to 255

Default: 70

Max Redirection

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

Perform one of the following.

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

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

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

Default: 5

Max Auth

Specifies the maximum number of times (from 0 to 255) a request can be challenged.

Perform one of the following.

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

<Max_Auth ua="na">2</Max_Auth>

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

Allowed value: 0 to 255

Default: 2

SIP User Agent Name

Used in outbound requests.

Perform one of the following.

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

<SIP_User_Agent_Name ua="na">$VERSION</SIP_User_Agent_Name>

- In the phone web page, enter an appropriate name.

Default: $VERSION

If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed

SIP Server Name

Server header used in responses to inbound responses.

Perform one of the following.

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

<SIP_Server_Name ua="na">$VERSION</SIP_Server_Name>

- In the phone web page, enter an appropriate name.

Default: $VERSION

SIP Reg User Agent Name

User-Agent name to be used in a REGISTER request. If this is not specified, the SIP User Agent Name is also used for the REGISTER request.

Perform one of the following.

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

<SIP_Reg_User_Agent_Name ua="na">agent name</SIP_Reg_User_Agent_Name>

- In the phone web page, enter an appropriate name.

Default: Blank

SIP Accept Language

Accept-Language header used.

Perform one of the following.

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

<SIP_Accept_Language ua="na">en</SIP_Accept_Language>

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

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

DTMF Relay MIME Type

MIME Type used in a SIP INFO message to signal a DTMF event. This field must match that of the Service Provider.

Perform one of the following.

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

<DTMF_Relay_MIME_Type ua="na">application/dtmf-relay</DTMF_Relay_MIME_Type>

- In the phone web page, enter an appropriate MIME type.

Default: application/dtmf-relay

Hook Flash MIME Type

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

Perform one of the following.

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

<Hook_Flash_MIME_Type ua="na">application/hook-flash</Hook_Flash_MIME_Type>

- In the phone web page, enter an appropriate MIME type for a SIPINFO message.

Default:

Remove Last Reg

Enables you to remove the last registration before registering a new one if the value is different.

Set to Yes to remove the last registration.

Perform one of the following.

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

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

- In the phone web page, Select Yes or No.

Allowed values: Yes or No

Default: No

Use Compact Header

If set to yes, the phone uses compact SIP headers in outbound SIP messages. If inbound SIP requests contain normal headers, the phone substitutes incoming headers with compact headers. If set to no, the phones use normal SIP headers. If inbound SIP requests contain compact headers, the phones reuse the same compact headers when generating the response, regardless of this setting.

Perform one of the following.

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

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

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

Allowed values: Yes or No

Default: No

Escape Display Name

Enables you to keep the Display Name private.

Set to Yes if you want the IP phone to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages.

Perform one of the following.

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

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

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

Allowed values: Yes or No

Default: Yes.

Talk Package

Enables support for the BroadSoft Talk Package that lets users answer or resume a call by clicking a button in an external application.

Perform one of the following.

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

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

- In the phone web page, select Yes to enable the Talk Package.

Allowed values: Yes or No

Default: No

Hold Package

Enables support for the BroadSoft Hold Package, which lets users place a call on hold by clicking a button in an external application.

Perform one of the following.

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

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

- In the phone web page, select Yes to enable support for the Hold Package.

Allowed values: Yes or No

Default: No

Conference Package

Enables support for the BroadSoft Conference Package that enables users to start a conference call by clicking a button in an external application.

Perform one of the following.

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

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

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

Allowed values: Yes or No

Default: No

RFC 2543 Call Hold

If set to yes, unit includes c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either case.

Perform one of the following.

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

<RFC_2543_Call_Hold ua="na">Yes</RFC_2543_Call_Hold>

- In the phone web page, Yes or No.

Allowed values: Yes or No

Default: Yes

Random REG CID on Reboot

If set to yes, the phone uses a different random call-ID for registration after the next software reboot. If set to no, the Cisco IP phone tries to use the same call-ID for registration after the next software reboot. The Cisco IP phone always uses a new random Call-ID for registration after a power-cycle, regardless of this setting.

Perform one of the following.

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

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

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

Default: No.

SIP TCP Port Min

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

Perform one of the following.

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

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

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

Default: 5060

SIP TCP Port Max

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

Perform one of the following.

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

<SIP_TCP_Port_Max ua="na">5080</SIP_TCP_Port_Max>

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

Default: 5080

Caller ID Header

Provides the option to take the caller ID from PAID-RPID-FROM, PAID-FROM, RPID-PAID-FROM, RPID-FROM, or FROM header.

Perform one of the following.

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

<Caller_ID_Header ua="na">PAID-RPID-FROM</Caller_ID_Header>

- In the phone web page, select an option.

Allowed values: PAID-RPID-FROM, AID-FROM, RPID-PAID-FROM, RPID-FROM, and FROM

Default: PAID-RPID-FROM

Hold Target Before Refer

Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended call transfer (where the transfer target has answered).

Perform one of the following.

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

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

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

Default: No

Dialog SDP Enable

When enabled and the Notify message body is too big causing fragmentation, the Notify message xml dialog is simplified; Session Description Protocol (SDP) is not included in the dialog xml content.

Perform one of the following.

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

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

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

Allowed values: Yes or No

Default: No

Keep Referee When Refer Failed

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

Perform one of the following.

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

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

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

Allowed values: Yes or No

Default: No

Display Diversion Info

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

Perform one of the following.

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

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

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

Allowed values: Yes or No

Display Anonymous From Header

Show the caller ID from the SIP INVITE message “From” header when set to Yes, even if the call is an anonymous call. When the parameter is set to no, the phone displays "Anonymous Caller" as the caller ID.

Perform one of the following.

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

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

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

Allowed values: Yes or No

Default: No

Sip Accept Encoding

Supports the content-encoding gzip feature.

If gzip is selected, the SIP message header contains the string “Accept-Encoding: gzip”, and the phone is able to process the SIP message body, which is encoded with the gzip format.

Perform one of the following.

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

<Sip_Accept_Encoding ua="na">none</Sip_Accept_Encoding>

- In the phone web page, enter an appropriate MIME type for a SIPINFO message.

Allowed values: none and gzip

Default: none

SIP IP Preference

Sets if the phone uses IPv4 or IPv6.

Perform one of the following.

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

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

- In the phone web page, select IPv4 or IPv6.

Allowed values: IPv4/IPv6

Default: IPv4.

Disable Local Name To Header

Controls the display name in “Directory”, “Call History”, and in the “To” header during an outgoing call.

Perform one of the following.

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

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

- In the phone web page, select Yes to disable the display name.

Allowed values: Yes/No

Default: No

Configure the SIP Timer Values

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Timer Values section, set the SIP timer values in seconds as described in SIP Timer Values (sec).

Step 3

Click Submit All Changes.

- SIP Timer Values (sec)

SIP Timer Values (sec)

Parameter

Description

SIP T1

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

Default: 0.5 seconds

SIP T2

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

Default: 4 seconds

SIP T4

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

Default: 5 seconds.

SIP Timer B

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

Default: 16 seconds.

SIP Timer F

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

Default: 16 seconds.

SIP Timer H

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

Default: 16 seconds.

SIP Timer D

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

Default: 16 seconds.

SIP Timer J

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

Default: 16 seconds.

INVITE Expires

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

Default: 240 seconds

ReINVITE Expires

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

Default: 30

Reg Min Expires

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

Reg Max Expires

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

Reg Retry Intv

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

Default: 30

See the note below for additional details.

Reg Retry Long Intvl

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

Default: 1200

See the note below for additional details.

Reg Retry Random Delay

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

Default: 0

Reg Retry Long Random Delay

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

Default: 0

Reg Retry Intvl Cap

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

Default: 0

Sub Min Expires

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

Sub Max Expires

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

Default: 7200.

Sub Retry Intvl

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

Default: 10.

Note

The phone can use a RETRY-AFTER value when it is received from a SIP proxy server that is too busy to process a request (503 Service Unavailable message). If the response message includes a RETRY-AFTER header, the phone waits for the specified length of time before to REGISTER again. If a RETRY-AFTER header is not present, the phone waits for the value specified in the Reg Retry Interval or the Reg Retry Long Interval.

Configure the Response Status Code Handling

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the Response Status Code Handling section, set the values as specified in the Response Status Code Handling Paramters table.

Step 3

Click Submit All Changes.

- Response Status Code Handling Paramters

Response Status Code Handling Paramters

The following table defines the function and usage of the parameters in the Response Status Code Handling section under the SIP 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.

Response Status Code Handling Paramters

Parameter

Description

Try Backup RSC

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

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

Perform one of the following:

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

<Try_Backup_RSC ua="na"/>

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

Default: Blank

Retry Reg RSC

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

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

Perform one of the following:

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

<Retry_Reg_RSC ua="na"/>

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

Default: Blank

Configure NTP Server

You can configure NTP servers with IPv4 and IPv6. You can also configure NTP server with DHCPv4 option 42 or DHCPv6 option 56. Configuring NTP with Primary NTP Server and Secondary NTP server parameters has higher priority over configuring NTP with DHCPv4 option 42 or DHCPv6 option 56.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Systems.

Step 2

In the Optional Network Configuration section, set the IPv4 or IPv6 address as described in the NTP Server Parameters table.

Step 3

Click Submit All Changes.

- NTP Server Parameters

NTP Server Parameters

The following table defines the function and usage of NTP server parameters in the Optional Network Configuration section under the System 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.

NTP Server Parameters

Parameter

Description

Primary NTP Server

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

Perform one of the following:

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

<Primary_NTP_Server ua="rw"/>

- In the phone web page, enter the IP address of the priamry NTP server.

Default: Blank

Secondary NTP Server

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

Perform one of the following:

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

<Secondary_NTP_Server ua="rw"/>

- In the phone web page, enter the IP address of the secondary NTP server.

Default: Blank

Configure the RTP Parameters

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the RTP Parameters section, set the Real-Time Transport Protocol (RTP) parameter values as described in RTP Parameters.

Step 3

Click Submit All Changes.

- RTP Parameters

- Enable SSRC Reset for the New RTP and SRTP Sessions

RTP Parameters

The following table defines the function and usage of the parameters in the RTP Parameters section under the SIP 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.

RTP Parameters

Parameter

Description

RTP Port Min

Minimum port number for RTP transmission and reception.

Perform one of the following:

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

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

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

Allowed values: 2048 to 49151

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

Default: 16384

RTP Port Max

Maximum port number for RTP transmission and reception.

Perform one of the following:

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

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

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

Allowed values: 2048 to 49151

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

Default: 16482

RTP Packet Size

Specifies packet size in seconds.

Perform one of the following:

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

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

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

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

Default: 0.02

Max RTP ICMP Err

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

Perform one of the following:

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

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

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

Default: 0

RTCP Tx Interval

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

Perform one of the following:

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

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

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

Allowed values: 0 to 255 seconds

Default: 0

Call Statistics

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

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: No

SDP IP Preferences

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

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

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

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

Perform one of the following:

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

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

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

Allowed values: IPv4 and IPv6

Default: IPv4

RTP Before ACK

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

Perform one of the following:

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

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

- In the phone web page select:

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

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

Allowed values: Yes and No

Default: No

SSRC Reset on RE-INVITE

Controls whether to reset the Synchronization Source (SSRC) for the new RTP and SRTP sessions.

Perform one of the following:

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

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

- In the phone web page select:

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

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

Allowed values: Yes and No

Default: No

Enable SSRC Reset for the New RTP and SRTP Sessions

You can enable the SSRC Reset on RE-INVITE to avoid a call transfer error, where only one person on the call hears the audio. This error occurs on calls of 30 minutes or longer, and often on three-way calls.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the RTP Parameters section, set the parameter SSRC Reset on RE-INVITE to Yes.

You can also configure this parameter in the configuration file:

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

Allowed values: Yes and No.

Default: No

Note

If you set the parameter to No, the SSRC remains for the new RTP and SRTP sessions (SIP re-INVITEs). The call transfer error might occur during a long duration call.

Step 3

Click Submit All Changes.

Control SIP and RTP Behaviour in Dual Mode

You can control SIP and RTP parameters with SIP IP Preference and SDP IP Preference fields when phone is in dual mode.

SIP IP Preference parameter defines which IP address phone tries first when it is in dual mode.

SIP IP Preference and IP Mode

IP Mode

SIP IP Preference

Address List from DNS, Priority, Result

P1 - First Priority Address

P2 - Second Priority Address

Failover Sequence

Dual Mode

IPv4

P1- 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 1.1.1.1 first.

1.1.1.1 ->2009:1:1:1:1 ->

2.2.2.2 -> 2009:2:2:2:2

Dual Mode

IPv6

P1- 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 2009:1:1:1::1 first.

2009:1:1:1:1 ->

1.1.1.1 -> 2009:2:2:2:2 ->

2.2.2.2

Dual Mode

IPv4

P1- 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 2009:1:1:1::1 first.

2009:1:1:1:1 -> 2.2.2.2 -> 2009:2:2:2:2

Dual Mode

IPv6

P1- 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 1.1.1.1 first.

2009:1:1:1:1 -> 2009:2:2:2:2

->2.2.2.2

IPv4 Only

IPv4

or

IPv6

P1 - 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 1.1.1.1 first.

1.1.1.1 -> 2.2.2.2

IPv6 Only

IPv4

or

IPv6

P1 - 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 2009:1:1:1::1 first.

2009:1:1:1:1 -> 2009:2:2:2::2

SDP IP Preference - ALTC helps peers in dual-mode negotiate RTP address family.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Parameters section, select IPv4 or IPv6 in the SIP IP Preference field.

For details, see SDP IP Preference field in the SIP Parameters table.

Step 3

In the RTP Parameters section, select IPv4 or IPv6 in the SDP IP Preference field.

For details, see SDP IP Preference in the RTP Parameters table.

Configure the SDP Payload Types

Your Cisco IP Phone supports RFC4733. You can choose from three audio-video transport (AVT) options to send DTMF pulses to the server.

Configured dynamic payloads are used for outbound calls only when the Cisco IP Phone presents a Session Description Protocol (SDP) offer. For inbound calls with an SDP offer, the phone follows the caller’s assigned dynamic payload type.

The Cisco IP Phone uses the configured codec names in outbound SDP. For incoming SDP with standard payload types of 0-95, the phone ignores the codec names. For dynamic payload types, the phone identifies the codec by the configured codec names. The comparison is case-sensitive, so you need to set the name correctly.

You can also configure the parameters in the phone configuration file (cfg.xml). To configure each of the parameters, see the syntax of the string in SDP Payload Types.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SDP Payload Types section, set the value as specified in SDP Payload Types.

- AVT Dynamic Payload—is any nonstandard data. Both sender and receiver must agree on a number. The range is from 96 to 127. The default is 101.

- AVT 16kHz Dynamic Payload —is any nonstandard data. Both sender and receiver must agree on a number. The range is from 96 to 127. The default is 107.

- AVT 48kHz Dynamic Payload —is any nonstandard data. Both sender and receiver must agree on a number. The range is from 96 to 127. The default is 108.

Step 3

Click Submit All Changes.

- SDP Payload Types

SDP Payload Types

Parameter

Description

G722.2 Dynamic Payload

G722 Dynamic Payload type.

Perform one of the following:

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

<Try_Backup_RSC ua="na"/>

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

Allowed values:

Default: 96

iLBC Dynamic Payload

iLBC Dynamic Payload type.

Default: 97

OPUS Dynamic Payload

OPUS Dynamic Payload type.

Default: 99

AVT Dynamic Payload

AVT dynamic payload type. Ranges from 96-127.

Default: 101

INFOREQ Dynamic Payload

INFOREQ Dynamic Payload type.

H264 BP0 Dynamic Payload

H264 BPO Dynamic Payload type.

Default: 110

H264 HP Dynamic Payload

H264 HP Dynamic Payload type.

Default: 110

G711u Codec Name

G711u codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: PCMU

G711a Codec Name

G711a codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: PCMA

G729a Codec Name

G729a codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: G729a

G729b Codec Name

G729b codec name used in SDP.

Perform one of the following:

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

<G729b_Codec_Name ua="na">PCMU</G729b_Codec_Name>

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

Allowed values:

Default: G729b

G722 Codec Name

G722 codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: G722

G722.2 Codec Name

G722.2 codec name used in SDP.

Perform one of the following:

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

<G722.2_Codec_Name ua="na">PCMU</G722.2_Codec_Name>

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

Allowed values:

Default: G722.2

iLBC Codec Name

iLBC codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: iLBC

OPUS Codec Name

OPUS codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: OPUS

AVT Codec Name

AVT codec name used in SDP.

Perform one of the following:

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

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

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

Allowed values:

Default: telephone-event

AVT 16 kHz Dynamic Payload

AVT dynamic payload type for the 16 kHz clock rate.

Perform one of the following:

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

<AVT_16kHz_Dynamic_Payload ua="na">107</AVT_16kHz_Dynamic_Payload>

- In the phone web page, enter the payload.

Range: 96-127

Default: 107

AVT 48 kHz Dynamic Payload

AVT dynamic payload type for the 48 kHz clock rate.

Perform one of the following:

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

<AVT_48kHz_Dynamic_Payload ua="na">108</AVT_48kHz_Dynamic_Payload>

- In the phone web page, enter the payload.

Range: 96-127

Default: 108

Configure the SIP Settings for Extensions

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, set the parameter values as described in the Parameters for SIP Settings on Extensions table.

Step 3

Click Submit All Changes.

- Parameters for SIP Settings on Extensions

Parameters for SIP Settings on Extensions

The following table defines the function and usage of the parameters in the SIP Settings section under the Ext(n) 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.

SIP Settings in Extensions

Parameter

Description

SIP Transport

Specifies the transport protocol for SIP messages.

Perform one of the following:

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

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

- In the phone web page, select the transport protocol type.

- UDP

- TCP

- TLS

- AUTO

AUTO allows the phone to select the appropriate protocol automatically, based on the NAPTR records on the DNS server. See Configure the SIP Transport for more details.

Default: UDP

SIP Port

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

Note

Specify the port number here only when you are using UDP as the SIP transport protocol.

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

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

Perform one of the following:

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

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

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

Default: 5060

SIP 100REL Enable

Individually enables the SIP 100REL feature.

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

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: No

Precondition Support

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

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

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

Perform one of the following:

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

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

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

Allowed values: Disabled and Enabled

Default: Disabled

EXT SIP Port

The external SIP port number.

Perform one of the following:

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

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

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

Allowed values:

Default: 5060

Auth Resync-Reboot

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

- resync

- reboot

- report

- restart

- XML-service

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: Yes

SIP Proxy-Require

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

Perform one of the following:

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

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

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

Default: Blank

SIP Remote-Party-ID

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

Default: Yes

Referor Bye Delay

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

Perform one of the following:

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

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

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

Allowed values: An integer from 0 through 65535

Default: 4

Refer-To Target Contact

Indicates the refer-to target.

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: No

Referee Bye Delay

Specifies the Referee Bye Delay time in seconds.

Perform one of the following:

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

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

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

Allowed values: An integer from 0 through 65535

Default: 0

Refer Target Bye Delay

Specifies the Refer Target Bye Delay time in seconds.

Perform one of the following:

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

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

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

Allowed values: An integer from 0 through 65535

Default: 0

Sticky 183

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

Perform one of the following:

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

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

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

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

Allowed values: Yes and No

Default: No

Auth INVITE

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

Perform one of the following:

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

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

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

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

Allowed values: Yes and No

Default: No

Ntfy Refer On 1xx-To-Inv

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

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

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: Yes

Set G729 annexb

Configure G.729 Annex B settings.

Perform one of the following:

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

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

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

Allowed values:

- None

- No

- Yes

- Follow silence supp setting

Default: Yes

User Equal Phone

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

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

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: No

Call Recording Protocol

Determines the type of recording protocol that the phone uses. Options are:

- SIPINFO

- SIPREC

Perform one of the following:

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

<Call_Recording_Protocol_1_ ua="na">SIPREC</Call_Recording_Protocol_1_>

- In the phone web page, select a protocol from the list.

Allowed values: SIPREC|SIPINFO

Default: SIPREC

Privacy Header

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

The privacy header options are:

- Disabled (default)

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

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

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

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

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

Perform one of the following:

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

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

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

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

Default: Disabled

P-Early-Media Support

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

Perform one of the following:

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

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

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

Allowed values: Yes and No

Default: No

Configure the SIP Proxy Server

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the Proxy and Registration section, set the parameter values as described in the SIP Proxy and Registration for Extension Parameters table.

Step 3

Click Submit All Changes.

- SIP Proxy and Registration for Extension Parameters

SIP Proxy and Registration for Extension Parameters

The following table defines the function and usage of the parameters in the Proxy and Registration section under the Ext(n) 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.

SIP Proxy and Registration for Extension

Parameter

Description

Proxy

SIP proxy server and port number set by the service provider for all outbound requests. For example: 192.168.2.100:6060.

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

<Proxy_1_ ua="na">64.101.154.134</Proxy_1_>
<RTP_Port_Max ua="na">16482</RTP_Port_Max>

- In the phone web page, enter SIP proxy server and port number.

When you need to refer to this proxy in another setting, for example, the speed dial line key configuration, use the $PROXY macro variable.

Default: The port number is optional. If you don't specify a port, the default port 5060 is used for UDP, and the default port 5061 is used for TLS.

Outbound Proxy

Specifies an IP address or domain name. All outbound requests are sent as the first hop.

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

<Outbound_Proxy_1_ ua="na">10.79.78.45</Outbound_Proxy_1_>

- In the phone web page, enter an IP address and a domain name.

Default: Empty

Proxy

Outbound Proxy

For Survivable Remote Site Telephony (SRST) support

These parameters can be configured with an extension that includes a statically-configured DNS SRV record or DNS A record. This allows for failover and fallback functionality with a secondary proxy server.

The format for the parameter value is as follows:

FQDN format: hostname[:port][:SRV=host-list OR :A=ip-list]
Where:

- host-list: srv[|srv[|srv...]]

- srv: hostname[:port][:p=priority][:weight][:A=ip-list]

- ip-list: ip-addr[,ip-addr[,ip-addr...]]

Default:

- Priority is 0.

- Weight is 1.

- Port is 5060 and 5061 for UDP and TLS respectively.

Alternate Proxy

Alternate Outbound Proxy

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.

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

<Alternate_Proxy_1_ ua="na">10.74.23.43</Alternate_Proxy_1_><Alternate_Outbound_Proxy_1_ ua="na">10.74.23.44</<Alternate_Outbound_Proxy_1_>

- In the phone web page, 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.

Default: Empty

Use OB Proxy In Dialog

Determines whether to force SIP requests to be sent to the outbound proxy within a dialog.

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

<Use_OB_Proxy_In_Dialog_1_ ua="na">Yes</Use_OB_Proxy_In_Dialog_1_>

- In the phone web page, select Yes or No. The request is ignored if the Use Outbound Proxy field is set to No or if the Outbound Proxy field is empty.

Valid values: Yes and No

Default: Yes

Register

Enables periodic registration with the proxy. This parameter is ignored if a proxy is not specified.

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

<Register_1_ ua="na">Yes</Register_1_>

- In the phone web page, To enable this feature, select Yes.

Valid values: Yes and No

Default: Yes

Make Call Without Reg

Enables making outbound calls without successful (dynamic) registration by the phone.

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

<Make_Call_Without_Reg_1_ ua="na">No</Make_Call_Without_Reg_1_>

- In the phone web page, To enable this feature, select Yes. If set to No, the dial tone plays only when registration is successful.

Valid values: Yes and No

Default: No

Register Expires

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.

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

<Register_Expires_1_ ua="na">3600</Register_Expires_1_>

- In the phone web page, enter a value in seconds to define how often the phone renews registration with the proxy.

Valid values: Numeric. The range is from 32 seconds to 2000000 seconds.

Default: 3600 seconds

Ans Call Without Reg

If enabled, the user does not have to be registered with the proxy to answer calls.

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

<Ans_Call_Without_Reg_1_ ua="na">No</Ans_Call_Without_Reg_1_>

- In the phone web page, To enable this feature, select Yes.

Valid values: Yes and No

Default: No

Use DNS SRV

Enables DNS SRV lookup for the proxy and outbound proxy.

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

<Use_DNS_SRV_1_ ua="na">Yes</Use_DNS_SRV_1_>

- In the phone web page, To enable this feature, select Yes.

Valid values: Yes and No

Default: No

DNS SRV Auto Prefix

Enables the phone to automatically append a prefix to the proxy or outbound proxy name when performing a DNS SRV lookup on that name. The prefix to be appended varies with SIP transport protocols.

- _sip._udp. for UDP protocol

- _sip._tcp. for TCP protocol

- _sips._tcp. for TLS protocol

Perform one of the following:

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

<DNS_SRV_Auto_Prefix_1_ ua="na">Yes</DNS_SRV_Auto_Prefix_1_>

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

Valid values: Yes and No

Default: No

Proxy Fallback Intvl

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.

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

<Proxy_Fallback_Intvl_1_ ua="na">3600</Proxy_Fallback_Intvl_1_>

- In the phone web page, enter a value in seconds to set the duration in seconds after which the phone retries.

Valid values: Numeric. The range is from 0 seconds to 65535 seconds.

Default: 3600 seconds

Proxy Redundancy Method

The phone creates an internal list of proxies returned in the DNS SRV records.

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

<Proxy_Redundancy_Method_1_ ua="na">Normal</Proxy_Redundancy_Method_1_>

- In the phone web page, select Normal and Based on SRV Port.

If you set to Normal, the list contains proxies ranked by weight and priority.

If you set to Based on SRV Port, the phone uses normal, then inspects the port number based on the first-listed proxy port.

Valid values: Normal|Based on SRV Port

Default: Normal

Dual Registration

Controls both the dual registration and the fast fall back feature.

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

<Dual_Registration_1_ ua="na">No</Dual_Registration_1_>

- In the phone web page, 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.

Valid values: Yes and No

Default: No

Auto Register When Failover

Controls the fallback duration.

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

<Auto_Register_When_Failover_1_ ua="na">Yes</Auto_Register_When_Failover_1_>

- In the phone web page, 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.

Valid values: Yes and No

Default: Yes

TLS Name Validate

This field works only when SIP Transport is set to TLS for the phone line.

Specifies whether hostname verification is required when the phone line uses SIP over TLS. The options are:

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

<TLS_Name_Validate_1_ ua="na">Yes</TLS_Name_Validate_1_>

- In the phone web page, select Yes when hostname verification is required.

Select No to bypass the hostname verification.

Valid values: Yes and No

Default: Yes

Configure the Subscriber Information Parameters

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the Subscriber Information section, set the parameter values as described in the Subscriber Information Parameters table.

Step 3

Click Submit All Changes.

- Subscriber Information Parameters

Subscriber Information Parameters

The following table defines the function and usage of the parameters in the RTP Parameters section under the SIP 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.

Subscriber Information

Parameter

Description

Display Name

Name displayed as the caller ID.

Perform one of the following:

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

<Display_Name_1_ ua="na"/>

- In the phone web page, enter a name that represents the caller ID.

User ID

Extension number for this line.

When you need to refer to this user ID in another setting, for example, the short name for a line key, use the $USER macro variable.

Perform one of the following:

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

<User_ID_1_ ua="na">7001</User_ID_1_>

- In the phone web page, enter an extension number

Password

Password for this line.

Perform one of the following:

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

<Password_1_ ua="na">*************</Password_1_>

- In the phone web page, enter a value to add password for the line.

Default: Blank (no password required)

Auth ID

Authentication ID for SIP authentication.

Perform one of the following:

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

<Auth_ID_1_ ua="na"/>

- In the phone web page, enter a value for an authentication ID.

Default: Blank

Reversed Auth Realm

The IP address for an authentication realm other than the proxy IP address.

Perform one of the following:

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

<Reversed_Auth_Realm_1_ ua=”na”> </Reversed_Auth_Realm_1_>
The parameter for extension 1 appears as defined in the phone configuration file.

- In the phone web page, enter proxy IP address.

Default: Blank. The proxy IP address is used as the authentication realm.

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.

Set Up Your Phone to Use OPUS Codec Narrowband

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

Before you begin

Access the Phone Web Interface

Procedure

Step 1

Select Voice > Ext <n> where (n) is the number of the extension to configure.

Step 2

In the SIP Settings section, set Use low-bandwidth OPUS to Yes.

Step 3

Click Submit All Changes.

NAT Transversal with Phones

Network Address Translation (NAT) allows multiple devices to share a single, public, routable, IP address to establish connections over the Internet. NAT is present in many broadband access devices to translate public and private IP addresses. For VoIP to coexist with NAT, NAT traversal is required.

Not all service providers provide NAT traversal. If your service provider does not provide NAT traversal, you have several options:

- NAT Mapping with Session Border Controller: We recommend that you choose an service provider that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the service provider, you have more choices in selecting a router.

- NAT Mapping with SIP-ALG Router: NAT mapping can be achieved by using a router that has a SIP Application Layer Gateway (ALG). By using a SIP-ALG router, you have more choices in selecting an service provider.

- NAT Mapping with a Static IP Address: NAT mapping with an external (public) static IP address can be acheived to ensure interoperability with the service provider. The NAT mechanism used in the router must be symmetric. For more information, see Determine Symmetric or Asymmetric NAT.

Use NAT mapping only if the service provider network does not provide a Session Border Controller functionality. For more information on how to configure NAT mapping with a static IP, see Configure NAT Mapping with the Static IP Address.

- NAT Mapping with STUN: If the service provider network does not provide a Session Border Controller functionality and if the other requirements are met, it is possible to use Session Traversal Utilities for NAT (STUN) to discover the NAT mapping. For information on how to configure NAT mapping with STUN, see Configure NAT mapping with STUN.

- Enable NAT Mapping

- Configure NAT Mapping with the Static IP Address

- Configure NAT mapping with STUN

Enable NAT Mapping

You must enable NAT mapping to set NAT parameters.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n).

Step 2

Set up the fields as described in NAT Mapping Parameters.

Step 3

Click Submit All Changes.

- NAT Mapping Parameters

NAT Mapping Parameters

The following table defines the function and usage of NAT Mapping parameters in the NAT Settings section under the Voice>Ext(n) 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.

NAT Mapping Parameters

Parameter

Description

NAT Mapping Enable

To use externally mapped IP addresses and SIP/ RTP ports in SIP messages, select yes. Otherwise, select no.

Perform one of the following:

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

<NAT_Mapping_Enable_1_ ua="na">Yes</NAT_Mapping_Enable_1_>

- In the phone web page, set the parameter to Yes.

Allowed values: Yes|No

Default: No

NAT Keep Alive Enable

To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.

Perform one of the following:

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

<NAT_Keep_Alive_Enable_1_ ua="na">Yes</NAT_Keep_Alive_Enable_1_>

- In the phone web page, set the parameter to Yes.

Allowed values: Yes|No

Default: No

NAT Keep Alive Msg

Enter the keep alive message that should be sent periodically to maintain the current NAT mapping.

Perform one of the following:

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

<NAT_Keep_Alive_Msg_1_ ua="na">$NOTIFY</NAT_Keep_Alive_Msg_1_>

- In the phone web page, set the parameter to $NOTIFY or to $REGISTER.

If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent.

Allowed values: $NOTIFY and $REGISTER.

Default: $NOTIFY

NAT Keep Alive Dest

Destination that should receive NAT keep alive messages.

Perform one of the following:

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

<NAT_Keep_Alive_Dest_1_ ua="na">$PROXY</NAT_Keep_Alive_Dest_1_>

- In the phone web page, set the parameter to $PROXY or specify a proxy server.

If the value is $PROXY, the messages are sent to the current or outbound proxy.

Allowed values: $PROXY or a proxy server IP address

Default: $PROXY

Configure NAT Mapping with the Static IP Address

You can configure NAT mapping on the phone to ensure interoperability with the service provider.

Before you begin

- Access the phone administration web page. See Access the Phone Web Interface.

- You must have an external (public) IP address that is static.

- The NAT mechanism used in the router must be symmetric.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the NAT Support Parameters section, set the parameters as described in the NAT Mapping with Static IP Parameters table.

Step 3

Click the Ext(n) tab.

Step 4

In the NAT Settings section, set the parameters as described in the NAT Mapping from Ext Tab with Static IP Parameters table.

Step 5

Click Submit All Changes.

What to do next

Configure the firewall settings on your router to allow SIP traffic.

- NAT Mapping with Static IP Parameters

NAT Mapping with Static IP Parameters

The following table defines the function and usage of NAT mapping with Static IP parameters in the NAT Support Parameters section under the Voice>SIP 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.

NAT Mapping with Static IP Parameters

Parameter

Description

Handle VIA received

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

Perform one of the following:

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

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

- In the phone web page, set to Yes.

Default: No

Handle VIA rport

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

Perform one of the following:

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

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

- In the phone web page, set to Yes.

Default: No

Insert VIA received

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

Perform one of the following:

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

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

- In the phone web page, set to Yes.

Default: No

Insert VIA rport

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

Perform one of the following:

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

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

- In the phone web page, set to Yes.

Default: No

Substitute VIA Addr

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

Perform one of the following:

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

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

- In the phone web page, set to Yes.

Default: No

Send Resp To Src Port

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

Perform one of the following:

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

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

- In the phone web page, set to Yes.

Default: No

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Perform one of the following:

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

<NAT_Keep_Alive_Intvl ua="na">15</NAT_Keep_Alive_Intvl>

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

Allowed values: Numeric ranges from 0 through 65535

Default: 15

EXT IP

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

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

Perform one of the following:

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

<EXT_IP ua="na">10.23.31.43</EXT_IP>

- In the phone web page, enter an external static IP address.

Default: Blank

The following table defines the function and usage of NAT mapping with Static IP parameters in the NAT Support Parameters section under the Voice>Ext 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.

NAT Mapping from Ext Tab

Parameter

Description

NAT Mapping Enable

Controls the use of externally mapped IP addresses and SIP/ RTP ports in SIP messages.

Perform one of the following:

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

<NAT_Mapping_Enable_1_ ua="na">Yes</NAT_Mapping_Enable_1_>

- In the phone web page, set to Yes to use externally mapped IP addresses.

Allowed values: Yes and No.

Default: No

NAT Keep Alive Enable

(Optional)

Configured NAT keep alive message periodically.

Perform one of the following:

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

<NAT_Keep_Alive_Enable_1_ ua="na">Yes</NAT_Keep_Alive_Enable_1_>

- In the phone web page, set to Yes to configure periodic NAT keep alive messages.

Note

The service provider might require the phone to send NAT keep alive messages to keep the NAT ports open.

Check with your service provider to determine the requirements.

Allowed values: Yes and No.

Default: No

Configure NAT mapping with STUN

If the service provider network does not provide a Session Border Controller functionality and if the other requirements are met, it is possible to use Session Traversal Utilities for NAT (STUN) to discover the NAT mapping. The STUN protocol allows applications operating behind a network address translator (NAT) to discover the presence of the network address translator and to obtain the mapped (public) IP address (NAT addresses) and the port number that the NAT has allocated for the User Datagram Protocol (UDP) connections to remote hosts. The protocol requires assistance from a third-party network server (STUN server) located on the opposing (public) side of the NAT, usually the public Internet. This option is considered a last resort and should be used only if the other methods are not available. To use STUN:

- The router must use asymmetric NAT. See Determine Symmetric or Asymmetric NAT.

- A computer running STUN server software is available on the network. You can also use a public STUN server or set up your own STUN server.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the NAT Support Parameters section, set the Handle VIA received, Insert VIA received, Substitute VIA Addr, Handle VIA rport, Insert VIA rport, and Send Resp To Src Port parameters as described in the NAT Mapping with Static IP Parameters table.

Step 3

Set the parameters as described in the NAT Mapping with STUN Parameters table.

Step 4

Click the Ext(n) tab.

Step 5

In the NAT Settings section, set the parameters as described in the NAT Mapping from Ext Tab with Static IP Parameters table.

Step 6

Click Submit All Changes.

What to do next

Configure the firewall settings on your router to allow SIP traffic.

- NAT Mapping with STUN Parameters

- Determine Symmetric or Asymmetric NAT

NAT Mapping with STUN Parameters

The following table defines the function and usage of NAT mapping with STUN parameters in the NAT Support Parameters section under the Voice>SIP 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.

NAT Mapping with STUN Parameters

Parameter

Description

STUN Enable

Enables the use of STUN to discover NAT mapping.

Perform one of the following:

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

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

- In the phone web page, set to Yes to enable the feature.

Allowed values: Yes and No.

Default: No

STUN Server

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

Perform one of the following:

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

<STUN_Server ua="na"/>

- In the phone web page, enter an IP address or fully-qualified domain name of the STUN server.

Allowed values:

Default: Blank

Determine Symmetric or Asymmetric NAT

STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from one internal IP address and port to one external, routable destination IP address and port. If another packet is sent from the same source IP address and port to a different destination, a different IP address and port number combination is used. This method is restrictive because an external host can send a packet to a particular port on the internal host only if the internal host first sent a packet from that port to the external host.

This procedure assumes that a syslog server is configured and is ready to receive syslog messages.

To Determine Whether the Router Uses Symmetric or Asymmetric NAT:

Before you begin

- Verify that the firewall is not running on your PC. (It can block the syslog port.) By default, the syslog port is 514.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System and navigate to Optional Network Configuration section.

Step 2

Enter the IP address for the Syslog Server, if the port number is anything other than the default, 514. It is not necessary to include the port number if it is the default.

The address and port number must be reachable from the Cisco IP phone. The port number appears on the output log file name. The default output file is syslog.514.log (if port number was not specified).

Step 3

Set the Debug Level to Error, Notice, or Debug.

Step 4

To capture SIP signaling messages, click the Ext tab and navigate to SIP Settings. Set the SIP Debug Option to Full.

Step 5

To collect information about what type of NAT your router uses click the SIP tab and navigate to NAT Support Parameters.

Step 6

Click Voice > SIP and navigate to NAT Support Parameters.

Step 7

Set STUN Test Enable to Yes.

Step 8

Determine the type of NAT by viewing the debug messages in the log file. If the messages indicate that the device is using symmetric NAT, you cannot use STUN.

Step 9

Click Submit All Changes.

Dial Plan

- Dial Plan Overview

- Edit the Dial Plan on the IP Phone

Related Tasks

Edit the Dial Plan on the IP Phone

Dial Plan Overview

Dial plans determine how digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls such as long distance or international.

Use the phone web user interface to configure dial plans on the IP phone.

This section includes information that you must understand about dial plans, and procedures to configure your own dial plans.

The Cisco IP Phone has various levels of dial plans and processes the digits sequence.

When a user presses the speaker button on the phone, the following sequence of events begins:

1. The phone begins to collect the dialed digits. The interdigit timer starts to track the time that elapses between digits.

2. If the interdigit timer value is reached, or if another terminating event occurs, the phone compares the dialed digits with the IP phone dial plan. This dial plan is configured in the phone web user interface in Voice > Ext(n) under the Dial Plan section.

- Digit Sequences

- Digit Sequence Examples

- Acceptance and Transmission of the Dialed Digits

- Dial Plan Timer (Off-Hook Timer)

- Interdigit Long Timer (Incomplete Entry Timer)

- Interdigit Short Timer (Complete Entry Timer)

Digit Sequences

A dial plan contains a series of digit sequences, separated by the | character. The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan consists of a series of elements that are individually matched to the keys that the user presses.

White space is ignored, but can be used for readability.

Digit Sequence

Function

0 1 2 3 4 5 6 7 8 9 0 * #

Characters that represent a key that the user must press on the phone keypad.

x

Any character on the phone keypad.

[sequence]

Characters within square brackets create a list of accepted key presses. The user can press any one of the keys in the list.

A numeric range, for example, [2-9] allows a user to press any one digit from 2 through 9.

A numeric range can include other characters. For example, [35-8*] allows a user to press 3, 5, 6, 7, 8, or *.

.
(period)

A period indicates element repetition. The dial plan accepts 0 or more entries of the digit. For example, 01. allows users to enter 0, 01, 011, 0111, and so forth.

<dialed:substituted>

This format indicates that certain dialed digits are replaced by the substituted characters when the sequence is transmitted. The dialed digits can be zero to 9. For example:

<8:1650>xxxxxxx

When the user presses 8 followed by a seven-digit number, the system automatically replaces the dialed 8 with the sequence 1650. If the user dials 85550112, the system transmits 16505550112.

If the dialed parameter is empty and there is a value in the substituted field, no digits are replaced and the substituted value is always prepended to the transmitted string. For example:

<:1>xxxxxxxxxx

When the user dials 9725550112, the number 1 is added at the beginning of the sequence; the system transmits 19725550112.

,
(comma)

An intersequence tone played (and placed) between digits plays an outside line dial tone. For example:

9, 1xxxxxxxxxx

An outside line dial tone plays after the user presses 9. The tone continues until the user presses 1.

!
(exclamation point)

Prohibits a dial sequence pattern. For example:

1900xxxxxxx!

Rejects any 11-digit sequence that begins with 1900.

*xx

Allows a user to enter a 2-digit star code.

S0 or L0

For Interdigit Timer Master Override, enter S0 to reduce the short interdigit timer to 0 seconds, or enter L0 to reduce the long interdigit timer to 0 seconds.

P

To pause, enter P, the number of seconds to pause, and a space. This feature is typically used for implementation of a hotline and warm line, with a 0 delay for the hot line, and a nonzero delay for a warm line. For example:

P5

A pause of 5 seconds is introduced.

Digit Sequence Examples

The following examples show digit sequences that you can enter in a dial plan.

In a complete dial plan entry, sequences are separated by a pipe character (|), and the entire set of sequences is enclosed within parentheses:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

For the Cisco IP Phone 6871 MPP Phones:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! |
9, 011xxxxxx. | 0 | [49]11 ) | [*#]xx[*#] | #xx+xxxxxxxxxxx*xxxxxxxxxx

- Extensions on your system:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

[1-8]xx Allows a user to dial any three-digit number that starts with the digits 1 to 8. If your system uses four-digit extensions, enter the following string: [1-8]xxx

- Local dialing with seven-digit number:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)

9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can enter any seven-digit number, as in a local call.

- Local dialing with 3-digit area code and a 7-digit local number:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required. After a user presses 9, an external dial tone sounds. The user must enter a 10-digit number that begins with a digit 2 through 9. The system automatically inserts the 1 prefix before it transmits the number to the carrier.

- Local dialing with an automatically inserted 3-digit area code:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

8, <:1212>xxxxxxx This example is useful where a local area code is required by the carrier but most calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number. The system automatically inserts the 1 prefix and the 212 area code before it transmits the number to the carrier.

- U.S. long-distance dialing:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The user can enter any 11-digit number that starts with 1 and is followed by a digit 2 through 9.

- Blocked number:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S. After the user presses 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected.

- U.S. international dialing:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 011xxxxxx After the user presses 9, an external dial tone sounds. The user can enter any number that starts with 011, as in an international call from the U.S.

- Informational numbers:

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

0 | [49]11 This example includes two-digit sequences, separated by the pipe character. The first sequence allows a user to dial 0 for an operator. The second sequence allows the user to enter 411 for local information or 911 for emergency services.

- Service activation codes (Cisco IP Phone 6871 only):

[*#]xx[*#] Allows the user to dial # codes and * codes to access functions.

- Service activation codes with additional parameters (Cisco IP Phone 6871 only):

#xx+xxxxxxxxxxx*xxxxxxxxxx Allows the user to dial a # code, followed by two 10-digit numbers.

An executive assistant can use this pattern to initiate a call on behalf of an executive. The assistant dials the service activation code for call initiation, followed by the executive's number, then the number that he or she wants to call.

Acceptance and Transmission of the Dialed Digits

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As the user enters more digits, the set of candidates diminishes until only one or none is valid. When a terminating event occurs, the IP PBX either accepts the user-dialed sequence and initiates a call, or else rejects the sequence as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is invalid.

The following table explains how terminating events are processed.

Terminating Event

Processing

Dialed digits have not matched any sequence in the dial plan.

The number is rejected.

Dialed digits exactly match one sequence in the dial plan.

If the dial plan allows the sequence, the number is accepted and is transmitted according to the dial plan.

If the dial plan blocks the sequence, the number is rejected.

A timeout occurs.

The number is rejected if the dialed digits are not matched to a digit sequence in the dial plan within the time that the applicable interdigit timer specifies.

The Interdigit Long Timer applies when the dialed digits do not match any digit sequence in the dial plan.

Default: 10 seconds.

The Interdigit Short Timer applies when the dialed digits match one or more candidate sequences in the dial plan. Default: 3 seconds.

A user presses the # key or the dial softkey on the IP phone screen.

If the sequence is complete and is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan.

If the sequence is incomplete or is blocked by the dial plan, the number is rejected.

Dial Plan Timer (Off-Hook Timer)

You can think of the Dial Plan Timer as the off-hook timer. This timer starts when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated. Unless you have a special dial plan string to allow a null entry, the call is rejected.

Note

The timer before a number is dialed is whichever shorter of the dial plan default timer and the dial tone timer set in the Dial Tone field on the Regional tab.

- Syntax for the Dial Plan Timer

- Examples for the Dial Plan Timer

Syntax for the Dial Plan Timer

SYNTAX: (Ps<:n> | dial plan)

- s: The number of seconds; The timer before a number is dialed is whichever shorter of the dial plan default timer and the dial tone timer set in the Dial Tone field. With the timer set to 0 seconds, the call transmits automatically to the specified extension when the phone goes off hook.

- n: (optional): The number to transmit automatically when the timer expires; you can enter an extension number or a DID number. No wildcard characters are allowed because the number is transmitted as shown. If you omit the number substitution, <:n>, the user hears a reorder (fast busy) tone after the specified number of seconds.

Examples for the Dial Plan Timer

Note

The actual timer before a number is dialed is whichever shorter of the dial plan default timer and the dial tone timer set in the Dial Tone field. In the following examples, the dial tone timer is assumed to be longer than the dial plan timer.

Allow more time for users to start dialing after taking a phone off hook:

(P9 | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9 means that after taking a phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone. By setting a longer timer, you allow more time for users to enter digits.

To create a hotline for all sequences on the System Dial Plan:

(P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9<:23> means that after taking the phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23.

To create a hotline on a line button for an extension:

(P0 <:1000>)

With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook. Enter this sequence in the Phone Dial Plan for Ext 2 or higher on a client phone.

Interdigit Long Timer (Incomplete Entry Timer)

You can think of this timer as the incomplete entry timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. The default value is 10 seconds.

This section explains how to edit a timer as part of a dial plan. Alternatively, you can modify the Control Timer that controls the default interdigit timers for all calls.

- Syntax for the Interdigit Long Timer

- Example for the Interdigit Long Timer

Syntax for the Interdigit Long Timer

SYNTAX: L:s, (dial plan)

- s: The number of seconds; if no number is entered after L:, the default timer is 5 seconds. With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook.

- Note that the timer sequence appears to the left of the initial parenthesis for the dial plan.

Example for the Interdigit Long Timer

L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

L:15 means that this dial plan allows the user to pause for up to 15 seconds between digits before the Interdigit Long Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.

Interdigit Short Timer (Complete Entry Timer)

You can think of this timer as the complete entry timer. This timer measures the interval between dialed digits. The timer applies when the dialed digits match at least one digit sequence in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated. If the entry is valid, the call proceeds. If the entry is invalid, the call is rejected.

Default: 3 seconds.

- Syntax for the Interdigit Short Timer

- Examples for the Interdigit Short Timer

Syntax for the Interdigit Short Timer

SYNTAX 1: S:s, (dial plan)

Use this syntax to apply the new setting to the entire dial plan within the parentheses.

SYNTAX 2: sequence Ss

Use this syntax to apply the new setting to a particular dialing sequence.

s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies.

Examples for the Interdigit Short Timer

To set the timer for the entire dial plan:

S:6, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

S:6 means that while the user enters a number with the phone off hook, the user can pause for up to 15 seconds between digits before the Interdigit Short Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.

Set an instant timer for a particular sequence within the dial plan:

(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxxS0 | 9,8,011xx. | 9,8,xx.|[1-8]xx)

9,8,1[2-9]xxxxxxxxxS0 means that with the timer set to 0, the call is transmitted automatically when the user dials the final digit in the sequence.

Edit the Dial Plan on the IP Phone

Note

You can edit the dial plan in the XML configuration file. Locate the Dial_Plan_n_ parameter in the XML configuration file, where n denotes the extension number. Edit the value of this parameter. The value must be specified in the same format as in the Dial Plan field on the phone administration web page, described below.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

Scroll to the Dial Plan section.

Step 3

Enter the digit sequences in the Dial Plan field.

The default (US-based) systemwide dial plan appears automatically in the field.

Step 4

You can delete digit sequences, add digit sequences, or replace the entire dial plan with a new dial plan.

Separate each digit sequence with a pipe character, and enclose the entire set of digit sequences within parentheses. Example:

(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

Step 5

Click Submit All Changes.

The phone reboots.

Step 6

Verify that you can successfully complete a call with each digit sequence that you entered in the dial plan.

Note

If you hear a reorder (fast busy) tone, review your entries and modify the dial plan appropriately.

Related Information

Dial Plan

Regional Parameters Configuration

- Regional Parameters

- Set the Control Timer Values

- Localize Your Cisco IP Phone

- Vertical Service Activation Codes

Regional Parameters

In the phone web user interface, use the Regional tab to configure regional and local settings, such as control timer values, dictionary server script, language selection, and locale to change localization. The Regional tab includes these sections:

- Call Progress Tones—Displays values of all ringtones.

- Distinctive Ring Patterns—Ring cadence defines the ringing pattern that announces a telephone call.

- Control Timer Values—Displays all values in seconds.

- Vertical Service Activation Codes—Includes Call Back Act Code and Call Back Deact Code.

- Outbound Call Codec Selection Codes—Defines the voice quality.

- Time—Includes local date, local time, time zone, and Daylight Saving Time.

- Language—Includes Dictionary Server Script, Language Selection, and Locale.

Set the Control Timer Values

If you need to edit a timer setting only for a particular digit sequence or type of call, you can edit the dial plan.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

Set the Reorder Delay, Interdigit Long Timer, and Interdigit Short Timer parameters as described in the Control Timer Values (sec) table.

Step 3

Click Submit All Changes.

- Parameters for Control Timer Values (sec)

Parameters for Control Timer Values (sec)

The following table defines the function and usage of Control Timer Values parameters in the Control Timer Values(s) Parameters section under the Voice>Regional 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.

Parameters for Control Timer Values (sec)

Parameter

Description

Reorder Delay

Delay after far end hangs up before reorder (busy) tone is played.

Perform one of the following:

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

<Reorder_Delay ua="na">255</Reorder_Delay>

- In the phone web page, set a value in seconds ranges from 0-255 secs.

0 = plays immediately, inf = never plays. Set to 255 to return the phone immediately to on-hook status and to not play the tone.

Allowed values: 0–255 seconds

Default: 255

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.

Perform one of the following:

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

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

- In the phone web page, set a value in seconds ranges from 0-64 seconds.

Allowed values: 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.

Perform one of the following:

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

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

- In the phone web page, set a value in seconds ranges from 0-64 seconds.

Allowed values: 0–64 seconds

Default: 3

Localize Your Cisco IP Phone

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

Configure the values in the fields in the Time and Language sections.

Step 3

Click Submit All Changes.

- Configure Time and Date on Phone Web Page

- Configure Time and Date on the Phone

- Time and Date Settings

- Configure Daylight Saving Time

- Phone Display Language

Configure Time and Date on Phone Web Page

You can manually set the time and date on the phone web page.

Before you begin

Access the Phone Web Interface. Review Time and Date Settings.

Procedure

Step 1

Select Voice > Regional.

Step 2

In the Time section, enter the time and date information.

Step 3

Select Voice > User.

Step 4

In the Supplementary Services, choose 12h or 24hr from the Time Format drop down list.

Default: 12hr

Step 5

Choose the date format from the Date Format drop down list.

Step 6

Click Submit All Changes

Configure Time and Date on the Phone

You can set the time and date manually on the phone.

Before you begin

Review the Time and Date Settings.

Procedure

Step 1

Press Applications .

Step 2

Select Device administration > Date/Time.

Step 3

Select Set current time manually.

Step 4

Set the date and time in the format requested on the screen:

YYYY MM DD HH MM

Step 5

Select the OK softkey.

Step 6

Select the Save softkey.

Time and Date Settings

The Cisco IP Phone obtains the time settings in one of two ways:

- NTP Server— NTP 24-hour time format takes priority over the time you set using the menu options on the phone or web page.

When the phone boots up, it tries to contact the first Network Time Protocol (NTP) server to get and update the time. The phone periodically synchronizes its time with the NTP server, and between updates, it tracks time with its internal clock. The synchronization period is fixed at 64 seconds.

If you manually enter a time, this setting takes effect for now, but on the next NTP synchronization, the NTP time is displayed.

- Manual Setup—You can manually configure the local date and time by using one of the following methods:

- On the phone web interface

- On the phone itself

The default format is 12-hour which is overwritten with the 24-hour format as soon as the phone synchronizes with the NTP server.

Date and Time Parameters

Parameter

Description

Set Local Date (mm/dd/yyyy)

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

Default: Blank

Set Local Time (HH/mm)

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

Default: Blank

Time Zone

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

The time of the log messages and status messages are in UTC time and are not affected by the time zone setting.

Default: GMT-08:00

Time Offset (HH/mm)

This specifies the offset in 24-hour format from GMT to use for the local system time.

The NTP Server time is expressed in GMT time. The local time is obtained by offsetting the GMT according to the time zone of the region.

Default: 00/00

Ignore DHCP Time Offset

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

Default: Yes.

Daylight Saving Time Rule

Enter the rule for calculating daylight saving time. This rule is comprised of three fields. Each field is separated by a semicolon (;). Optional values inside brackets [ ] are assumed to be 0 if they are not specified. Midnight is represented by colons. For example, 0:0:0 of the given date.

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

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

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

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

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

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

Daylight Saving Time Rule (continued)

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

- HH stands for hours (0-23).

- mm stands for minutes (0-59).

- ss stands for seconds (0-59).

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

Daylight Saving Time Enable

Enables Daylight Saving Time.

Default: Yes

Time Format

Choose the time format for the phone (12-hour or 24-hour).

Default: 12hr

Date Format

Choose the date format for the phone (month/day or day/month).

Default: month/day

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

<!--  Time  -->
<Set_Local_Date__mm_dd_yyyy_ ua="na"/>
<Set_Local_Time__HH_mm_ ua="na"/>
<Time_Zone ua="na">GMT-08:00</Time_Zone>
<!--
available options: GMT-12:00|GMT-11:00|GMT-10:00|GMT-09:00|
GMT-08:00|GMT-07:00|GMT-06:00|GMT-05:00|GMT-04:00|GMT-03:30|
GMT-03:00|GMT-02:00|GMT-01:00|GMT|GMT+01:00|GMT+02:00|GMT+03:00|
GMT+03:30|GMT+04:00|GMT+04:30|GMT+05:00|GMT+05:30|GMT+05:45|
GMT+06:00|GMT+06:30|GMT+07:00|GMT+08:00|GMT+09:00|GMT+09:30|
GMT+10:00|GMT+11:00|GMT+12:00|GMT+13:00|GMT+14:00
-->
<Time_Offset__HH_mm_ ua="na"/>
<Ignore_DHCP_Time_Offset ua="na">Yes</Ignore_DHCP_Time_Offset>
<Daylight_Saving_Time_Rule ua="na">start=3/-1/7/2;end=10/-1/7/2;
save=1</Daylight_Saving_Time_Rule>
<Daylight_Saving_Time_Enable ua="na">Yes</Daylight_Saving_Time_Enable>
<Time_Format ua="na">12hr</Time_Format>
<!--  available options: 12hr|24hr  -->
<Date_Format ua="na">month/day</Date_Format>
<!--  available options: month/day|day/month  -->

Configure Daylight Saving Time

The phone supports automatic adjustment for daylight saving time.

Note

The time of the log messages and status messages are in UTC time. The time zone setting does not affect them.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Regional.

Step 2

Set the Daylight Saving Time Enable drop-down list box to Yes.

Step 3

In the Daylight Saving Time Rule field, enter the DST rule. This value affects the time stamp on the CallerID.

Step 4

Click Submit All Changes.

- Daylight Saving Time Examples

Daylight Saving Time Examples

The following example configures daylight saving time for the U.S, adding one hour starting at midnight on the second Sunday in March and ending at midnight on the first Sunday in November; add 1 hour (USA, North America):

start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1

The following example configures daylight saving time for Finland, starting at midnight on the last Sunday in March and ending at midnight on the last Sunday in October:

start=3/-1/7/03:0:0;end=10/-1/7/03:0:0;save=1 (Finland)

The following example configures daylight saving time for New Zealand (in version 7.5.1 and higher), starting at midnight on the last Sunday in September and ending at midnight on the first Sunday of April.

start=9/-1/7/02:0:0;end=4/1/7/02:0:0;save=1 (New Zealand)

The following example configures the daylight saving time starting on the last Monday (on or before April 8) and ending on the first Wednesday (on or after May 8).

start=4/-8/1;end=5/8/3;save=1

Phone Display Language

The Cisco IP Phone supports multiple languages for the phone display.

By default, the phone is set up for English. To enable the use of another language, you must set up the dictionary for the language. For some languages, you must also set up the font for the language.

After the setup is complete, you or your users can specify the desired language for the phone display.

- Supported Languages for the Phone Display

- Set Up Dictionaries and Fonts

- Setup for Latin and Cyrillic Languages

- Setup for an Asian Language

- Setup for RTL Languages

- Specify a Language for the Phone Display

Supported Languages for the Phone Display

On the phone administration web page, go to Admin Login > Advanced > Voice > Regional. In the Language section, click the Locale drop-down list box to see the supported languages for the phone display.

- ar-SA (Arabic)

- bg-BG (Bulgarian)

- ca-ES (Catalan)

- cs-CZ (Czech)

- da-DK (Danish)

- de-DE (German)

- el-GR (Greek)

- en-GB (English-Great Britain)

- en-US (English-United States)

- es-CO (Spanish-Colombia)

- es-ES (Spanish-Spain)

- fi-FI (Finnish)

- fr-CA (French-Canada)

- fr-FR (French)

- he-IL (Hebrew)

- hr-HR (Croatian)

- hu-HU (Hungarian)

- it-IT (Italian)

- ja-JP (Japanese)

- ko-KR (Korean)

- nl-NL (Dutch)

- nn-NO (Norwegian)

- pl-PL (Polish)

- pt-PT (Portuguese)

- ru-RU (Russian)

- sk-SK (Slovak)

- sl-SI (Slovenian)

- sv-SE (Swedish)

- tr-TR (Turkish)

- zh-CN (Chinese)

- zh-HK (Chinese-Hong Kong SAR)

Set Up Dictionaries and Fonts

Languages other than English require dictionaries. Some languages also require a font.

Note

To enable Latin and Cyrillic languages, you must not add a font file.

Procedure

Step 1

Download the locale zip file for your firmware version, from cisco.com. Place the file on your server, and unzip the file.

Dictionaries and fonts for all the supported languages are included in the zip file. Dictionaries are XML scripts. Fonts are standard TTF files.

Step 2

On the phone administration web page, go to Admin Login > Advanced > Voice > Regional. In the Language section, specify the necessary parameters and values in the Dictionary Server Script field as described below. Use a semicolon (;) to separate multiple parameter and value pairs.

- Specify the location of the dictionary and font files with the serv parameter.

For example: serv=http://server.example.com/Locales/

Make sure to include the IP address of the server, the path, and folder name.

Example: serv=http://10.74.128.101/Locales/

- For each language that you want to set up, specify a set of parameters as described below.

Note

In these parameter specifications, n denotes a serial number. This number determines the sequential order in which the language options are displayed in the Settings menu of the phone.

0 is reserved for US-English, which has a default dictionary. You can use it optionally, to specify your own dictionary.

Use numbers starting with 1 for other languages.

- Specify the language name with the dn parameter.

Example for language name for Asian language: d1=Chinese-Simplified

Example for language name for German (Latin and Cyrillic): d2=German

Example for language name for French (Latin and Cyrillic): d1=French

Example for language name for French (Canada) (Latin and Cyrillic) language: d1=French-Canada

Example for language name for Hebrew (RTL language): d1=Hebrew

Example for language name for Arabic (RTL language): d1=Arabic

This name is displayed as a language option in the Settings menu of the phone.

- Specify the name of the dictionary file with the xn parameter.

Example for Asian language: x1=zh-CN_78xx_68xx-11.2.1.1004.xml;

Example for French (Latin and Cyrillic) languages: x1=fr-FR_78xx_68xx-11.2.1.1004.xml;

Example for Arabic (RTL language) language: x1=ar-SA_78xx_68xx-11.2.1.1004.xml;

Example for French (Canada) language: x1=fr-CA_78xx_68xx-11.3.6.0006.xml;

Ensure to specify the correct file for the language and phone model that you use.

- If a font is required for the language, specify the name of the font file with the fn parameter.

For example: f1=zh-CN_78xx_68xx-11.2.1.1004.ttf;

Make sure to specify the correct file for the language and phone model that you use.

See Setup for Latin and Cyrillic Languages for specific details on setting up Latin languages.

See Setup for an Asian Language for specific details on setting up an Asian language.

See Setup for RTL Languages for specific details on setting up RTL languages.

Step 3

Click Submit All Changes.

Setup for Latin and Cyrillic Languages

If you use Latin and Cyrillic languages such as French or German, you can configure up to four language options for the phone. List of Latin and Cyrillic languages:

- Bulgarian

- Catalan

- Croatian

- Czech

- Danish

- Dutch

- English (UK)

- Finnish

- French (France)

- French (Canada)

- German

- Greek

- Hungarian

- Italian

- Portuguese (Portugal)

- Norwegian

- Polish

- Russian

- Slovak

- Slovenian

- Spanish (Columbia)

- Spanish (Spain)

- Swedish

- Turkish

- Ukraine

To enable the options, set up a dictionary for each language that you want to include. To enable the language, specify a pair of dn and xn parameters and values in the Dictionary Server Script field, for each language that you want to include.

Example for including French and German:

serv=http://10.74.128.101/Locales/;d1=French;x1=fr-FR_78xx_68xx-11.2.1.1004.xml;
d2=German;x2=de-DE_78xx_68xx-11.2.1.1004.xml

Example for including French (Canada):

serv=http://10.74.128.101/Locales/;d1=French-Canada;x1=fr-CA_78xx_68xx-11.3.6.0006xml;

serv=http://10.74.128.101/Locales/;d1=French-Canada;x1=fr-CA_88xx-11.3.6.0006xml;

Note

In the above examples http://10.74.128.101/Locales/ is a web folder. The dictionary files are extracted in this web folder and are used in the examples.

To configure this option in the phone configuration XML file (cfg.xml), enter a string in this format:

<!-- Language -->

<Dictionary_Server_Script ua="na">serv=http://10.74.10.215/lcdapi/resync_files/;d1=French-Canada;x1=fr-CA_88xx-11.3.6.0006.xml;</Dictionary_Server_Script>

<Language_Selection ua="na">French-Canada</Language_Selection>

<Locale ua="na">fr-CA</Locale>

Add values for:

- Language Selection Parameter as appropriate

For French: French

For French (Canada): French-Canada

For German: German

- Locale parameter list as appropriate

For French: fr-FR

For French (Canada): fr-CA

For German: de-DE

After the successful configuration, the user can see the configured language option on the phone under the Language menu. User can access the Language menu from Applications > Device administration.

Setup for an Asian Language

If you use an Asian language such as Chinese, Japanese, or Korean, you can only set up one language option for the phone.

You must set up the dictionary and the font for the language. To do this, specify the d1, x1 and f1 parameters and values in the Dictionary Server Script field.

Example for setting up Chinese-Simplified:

serv=http://10.74.128.101/Locales/;d1=Chinese-Simplified;
x1=zh-CN_78xx_68xx-11.2.1.1004.xml;f1=zh-CN_78xx_68xx-11.2.1.1004.ttf

Setup for RTL Languages

If you use a Right-to-Left (RTL) language such as Arabic and Hebrew, you can only set up one language option for the phone.

You must set up the dictionary and the font for the language. To do this, specify the d1, x1, and f1 parameters and values in the Dictionary Server Script field.

Example for Arabic:

serv=http://server.example.com/Locales;d1=Arabic;x1=ar-SA_88xx-11.3.4.xml;f1=ar-SA_88xx-11.3.4.ttf

Example for Hebrew:

serv=http://server.example.com/Locales;d1=Hebrew;x1=he-IL_88xx-11.3.4.xml;f1=he-IL_88xx-11.3.4.ttf

Values for Language Selection parameter must be Arabic or Hebrew as appropriate.

Values for Locale parameter must be ar-SA for Arabic and he-IL for Hebrew.

Specify a Language for the Phone Display

Note

Your users can select the language on the phone, from Settings > Device Administration > Language.

Before you begin

The dictionaries and fonts required for the language are set up. See Set Up Dictionaries and Fonts for details.

Procedure

Step 1

On the phone administration web page, go to Admin Login > Advanced > Voice > Regional, Language section. In the Language Selection field, specify the value of the appropriate dn parameter value from the Dictionary Server Script field, for the language of your choice.

Step 2

Click Submit All Changes.

Vertical Service Activation Codes

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 *95.

Cfwd All Act Code

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

Defaults to *72.

Cfwd All Deact Code

Cancels call forward 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 forward 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 forward 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 *61.

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

Defaults to *62.

Block CID Per Call Act Code

Removes caller ID blocking on the next inbound call.

Defaults to *81.

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

Defaults to *82.

Block ANC Act Code

Blocks all anonymous calls.

Defaults to *77.

Block ANC Deact Code

Removes blocking of all anonymous calls.

Defaults to *87.

DND Act Code

Enables the do not disturb feature.

Defaults to *78.

DND Deact Code

Disables the do not disturb feature.

Defaults to *79.

Secure All Call Act Code

Makes all outbound calls secure.

Defaults to *16.

Secure No Call Act Code

Makes all outbound calls not secure.

Defaults to *17.

Secure One Call Act Code

Makes a secure call.

Default: *18.

Secure One Call Deact Code

Disables secure call feature.

Default: *19.

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 *68.

Call Pickup Code

The star code used for picking up a ringing call.

Defaults to *97.

Call Unpark Code

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

Defaults to *88.

Group Call Pickup Code

The star code used for picking up a group call.

Defaults to *98.

Exec Assistant Call Initiate Code

For executive assistants: Initiates a call on behalf of an executive from the user's (assistant's) extension.

Default: #64

Applicable to Cisco IP Phone 6871 Multiplatform Phones only.

Exec Call Filter Act Code

For executives who have assistants: Activates call filtering. When call filtering is on, assistants receive incoming calls for executives.

Default: #61

Applicable to Cisco IP Phone 6871 Multiplatform Phones only.

Exec Call Filter Deact Code

For executives who have assistants: Deactivates call filtering.

Default: #62

Applicable to Cisco IP Phone 6871 Multiplatform Phones only.

Exec Assistant Call Push Code

For executive assistants: Transfers an ongoing call from the user (assistant) to the executive.

Default: #63

Applicable to Cisco IP Phone 6871 Multiplatform Phones only.

Exec Call Retrieve Code

For executives who have assistants: Transfers an ongoing call from an assistant to the user (executive).

For executive assistants: Transfers an ongoing call from the executive to the user (assistant).

Default: *11

Applicable to Cisco IP Phone 6871 Multiplatform Phones only.

Exec Call Bridge Code

For executives who have assistants: Joins the user (executive) to an ongoing call with an assistant.

For executive assistants: Joins the user (assistant) to an ongoing call with an executive.

Default: *15

Applicable to Cisco IP Phone 6871 Multiplatform Phones only.

Important

If you change any of the service activation codes used by executives or assistants, you must update the corresponding settings in BroadWorks.

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 forward, 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.

Cisco IP Phone 6800 Series Documentation

See the publications that are specific to your language, phone model, and multiplatform firmware release. Navigate from the following Uniform Resource Locator (URL):

https://www.cisco.com/c/en/us/support/collaboration-endpoints/ip-phone-6800-series-multiplatform-firmware/tsd-products-support-series-home.html

Part IV. Troubleshooting

- Troubleshooting

- Monitoring Phone Systems

- Maintenance

Chapter 18. Troubleshooting

- Feature Troubleshooting

- Phone Display Problems

- Report All Phone Issues from Phone Web Page

- Report Phone Issues from Webex Control Hub

- Factory Reset the Phone from Phone Web Page

- Reboot the Phone from the Phone Web Page

- Reboot the Phone from the Webex Control Hub

- Report a Phone Problem Remotely

- Capture Packets

- Voice Quality Troubleshooting Tips

- Where to Find Additional Information

Feature Troubleshooting

Here is troubleshooting information related to some of the phone features.

- ACD Call Information Missing

- Phone Doesn't Show ACD Softkeys

- Phone Doesn't Show ACD Agent Availability

- Call Doesn't Record

- An Emergency Call Doesn't Connect to Emergency Services

- Presence Status Doesn't Work

- Phone Presence Message: Disconnected from Server

- Phone Cannot Access BroadSoft Directory for XSI

- Executive or Assistant Menu Does Not Appear

- Phone Doesn't Show Contacts

- SIP Subscription Failure Message

- Number of Voicemail Messages Doesn't Display

- Unable to Place a Call with Speed Dial for Voicemail Messages

- Failed to Sign Into a Voicemail Account

- Voicemail PLK Options Don't Display on the Phone

- Phone Failed to Upload the PRT Logs to the Remote Server

- Saved Passwords Become Invalid after Downgrade

- Failed to Onboard the Phone to Webex

ACD Call Information Missing

Problem

A call center phone does not see call information during a call.

Solution

- Check the phone configuration to determine if Call Information Enable is set to yes.

- Check the Broadsoft server configuration to determine if the user's Device Profile is configured with "Support Call Center MIME Type".

Phone Doesn't Show ACD Softkeys

Problem

The phone doesn't display the Agent Sign In or Agent Sign Out softkeys.

Solution

- Check Broadsoft server configuration to determine if that user has been configured as a call center agent.

- Enable the programmable softkeys (PSK) and add the ACD softkeys to the softkey list. For more information, see Customize Display of the Softkeys.

- Check the phone configuration to determine if BroadSoft ACD is set to yes.

Phone Doesn't Show ACD Agent Availability

Problem

The phone doesn't display the Avail or Unavail softkeys for an agent.

Solution

1. Check Broadsoft server configuration to determine if that user has been configured as a call center agent.

2. Check the phone configuration to determine if BroadSoft ACD is set to yes.

3. Set up the Agt Status programmable softkey (PSK) and add the ACD softkey to the softkey list. For more information, see Customize Display of the Softkeys.

4. Instruct users to press the Agt Status key to display the Available, Unavailable, and Wrap-up possible states.

5. Select the desired agent state.

Call Doesn't Record

Problem

When a user tries to record a call, the recording doesn't takes place.

Cause

This is often due to configuration issues.

Solution

1. Set the phone to always record a call.

2. Make a call.

If the recording doesn't start, there are configuration problems. Check the configuration of the BroadWorks and third-party recorder.

If the recording does start:

1. Set the phone to record on demand.

2. Set up Wireshark to capture a trace of the network traffic between the phone and Broadworks when the problem occurs. When you have the trace, contact TAC for further assistance.

An Emergency Call Doesn't Connect to Emergency Services

Problem

A user tries to place an emergency call, but the call doesn't connect to the emergency services (fire, police, or emergency services operator).

Solution

Check the emergency call configuration:

- Company Identifier or location request URL setup is incorrect. See Configure a Phone to Make Emergency Calls.

- An incorrect or blank emergency number exists in the Dial Plan setup. See Edit the Dial Plan on the IP Phone.

The location request servers (emergency call service provider) did not respond with a phone location, after multiple attempts.

Presence Status Doesn't Work

Problem

The phone doesn't show presence information.

Solution

Use UC Communicator as a reference to verify that the account works.

Phone Presence Message: Disconnected from Server

Problem

Instead of presence information, the user sees the message Disconnected from server.

Solution

- Check the Broadsoft server configuration to determine if IM&P service is enabled and assigned to that user.

- Check the phone configuration to determine if the phone can connect to the internet and get the XMPP messages.

- Check the XMPP Incoming and Outgoing messages printed in the syslog to make sure it can login successfully.

Phone Cannot Access BroadSoft Directory for XSI

Problem

The phone displays XSI directory access error.

Solution

1. Check Broadsoft server configuration for the user login and SIP credentials.

2. Check error messages in syslog.

3. Check information on the error on the phone screen.

4. If HTTPS connection fails, check the error message on the phone screen and in the syslog.

5. Install custom CA for HTTPS connection if the BroadSoft certificate is not signed from phone built-in root CA.

Executive or Assistant Menu Does Not Appear

Problem

The Settings > Executive or Settings > Assistant menu item doesn't appear on an executive's or assistant's phone respectively.

Solution

- Ensure that synchronization of settings is enabled for the user's extension. See Executive-Assistant Setting Synchronization.

- Check if the phone has executives or assistants, or both configured on different extensions.

Phone Doesn't Show Contacts

Problem

The phone doesn't display any contacts in the All directories screen when Search All Enable and Browse Mode Enable are set to Yes.

Solution

1. Check that the personal address book is enabled in the phone.

2. Check that there are contacts in the local personal address book and the Bluetooth-paired phone.

SIP Subscription Failure Message

Problem

Subscription failure message displays on the phone screen.

Solution

- Ensure that the sub parameter is correct. The sub parameter needs a correct SIP URI.

For example, the following string has an incomplete URI because the domain part is missing:

fnc=mwi;sub=4085283300;vid=1;

- Check if the voicemail PLK monitors a voicemail account that is different from the associated extension’s user ID and the SIP proxy. If the associated extension's SIP proxy doesn't support this scenario, the subscription will fail.

For example, for extension1, the user ID is 4081009981. The PLK doesn't monitor 4081009981 but it monitors 4085283300 (a hunt group number or an ACD group number) although the PLK is associated with extension 1. In this case, the monitored voicemail user 4085283300 is different from the PLK's associated user 4081009981. If the SIP proxy of extension 1 doesn't support this scenario, the subscription will fail.

Number of Voicemail Messages Doesn't Display

Problem

The phone doesn't display the number of voicemail messages in the voicemail PLK.

Solution 1

Ensure there are new messages in the monitored voicemail account.

Solution 2

Ensure the SIP proxy sends message-summary events to the phone.

In the phone’s Problem Report Tool (PRT) report, check if the SIP proxy sends a message-summary event to the phone.

Find the NOTIFY message containing an message-summary event from the phone logs. If it is not found, the SIP proxy doesn’t send any message-summary event.

An example of a message-summary event:

6581 NOT May 20 19:54:04.162830 (31949:32029) voice- <===== Recv (UDP) [10.74.53.87]:5060 SIP MSG:: NOTIFY sip:4081009981@10.74.53.82:5065 SIP/2.0

Via: SIP/2.0/UDP 10.74.53.87:5060;branch=z9hG4bK-25824-1-2

From: "80000"<sip:8000@voicemail.sipurash.com>;tag=65737593823-1

To: <sip:4081009981@10.74.53.87>;tag=3855fbedd30b2464

Call-ID: 745bbebd-c35bc038@10.74.53.82

CSeq: 1001 NOTIFY

Max-Forwards: 20

Event: message-summary

Subscription-state: active;expires=3599

User-Agent: UMSIPVoicemail

Content-Length: 213

Content-Type: application/simple-message-summary

Messages-Waiting: yes

Message-Account: 4085283300@10.74.53.87

Voice-Message: 5/5 (2/3)

Fax-Message: 0/0 (0/0)

Pager-Message: 0/0 (0/0)

Multimedia-Message: 0/0 (0/0)

Text-Message: 0/0 (0/0)

None: 0/0 (0/0)

Unable to Place a Call with Speed Dial for Voicemail Messages

Problem

The phone is unable to place a call to the specified speed-dial number.

Solution

- Ensure that the sd parameter is included in the extension function script.

For example, the sd parameter is missing in this script: fnc=mwi;sub=4085283300@$PROXY;vid=1;ext=3000;

- Ensure that the ext parameter is set .

For example, the ext parameter is not set in this function script: fnc=mwi+sd;sub=4085283300@$PROXY;vid=1;

Failed to Sign Into a Voicemail Account

Problem

After the user press a voicemail PLK, the user cannot sign into the voicemail account automatically.

Solution

- Access the voicemail server’s Interactive Voice Response (IVR) and determine the proper delay for the voicemail ID and PIN input. Insert or delete one or more commas if needed.

For example, the delay between the voicemail user ID and the PIN is too short: fnc=mwi+sd;sub=4085283300@$PROXY;vid=1;ext=3000 ,3300#,123456#;

- Ensure that there is a space between the speed dial number and the DTMFs (containing the voicemail user ID and PIN).

For example, there is no space after "3000" in the string: fnc=mwi+sd;sub=4085283300@$PROXY;vid=1;ext=3000,3300#,123456#;

- Check if the monitored voicemail user ID is the same with the PLK's associated extension's user ID. If not, the voicemail server may take the caller ID (the associated extension's user ID) as the voicemail user ID. The voicemail server doesn't ask for the PIN only, it may ask for a star key (*) to switch the voicemail user.
For example:

Scenario:

Extension 1 User ID is "4081009981".

PLK function script is as follows:

fnc=mwi+sd;sub=4085283300@$PROXY;vid=1;ext=3000 ,3300#,123456#;

Result:

In this case, insert a star key (*) in the PLK function script, as shown below:

fnc=mwi+sd;sub=4085283300@$PROXY;vid=1;ext=3000 ,*,3300#,123456#;

Voicemail PLK Options Don't Display on the Phone

Problem

After you press and hold the voicemail PLK for 2 seconds, the MWI or MWI + Speed dial, or both options don't display in the Select feature screen.

Solution

Ensure that mwi; or mwi;sd; is added to the Customizable PLK Options field under the General section from Voice > Att Console.

Phone Failed to Upload the PRT Logs to the Remote Server

Problem

When you tried to generate the Problem Report Tool (PRT) logs on the phone, the generation of the PRT logs succeeded. However, the phone failed to upload the PRT logs to the remote server. The phone screen showed the Error: 109 or Report Problem together with an unavailable URL of a compressed file (for example, tar.gz).

Solution

Ensure that the web server is enabled on the phone, see Configure the Network from the Phone.

The Error: 109 indicates that the PRT upload rule is incorrect.

The Report problem indicates that the PRT upload rule is empty.

To resolve the issue, you must enter a correct PRT upload rule on the phone administration web page.

Saved Passwords Become Invalid after Downgrade

Problem

You update certain passwords on a phone that uses Firmware Release 11.3(6) or later, and then downgrade the phone to Firmware Release 11.3(5) or older. In this scenario, the updated or saved passwords become invalid after the downgrade.

On the phone with Firmware Release 11.3(6) or later, even though you change the password back to the original one, this issue still occurs after the downgrade.

Solution

For the Firmware Release 11.3(6) or later, if you update the passwords, you must reconfigure the passwords to avoid the downgrade issue. If not, this issue doesn’t occur after the downgrade.

The following table shows the passwords that are affected by the downgrade issue:

Password List

Category

Password Type

System Configuration

User Password

Admin Password

Wi-Fi Profile (1-4)

Wi-Fi Password

WEP Key

PSK Passphrase

XSI Phone Service

Login Password

SIP Password

Broadsoft XMPP

Password

XML Service

XML Password

LDAP

Password

Call Feature Settings

Auth Page Password

Subscriber Information

Password

XSI Line Service

Login Password

TR-069

ACS Password

Connection Request Password

BACKUP ACS Password

Failed to Onboard the Phone to Webex

Problem

A phone onboards with the EDOS device activation that uses phone MAC address, and it onboards to the Webex cloud. An administrator deletes the phone user from an organization in Webex Control Hub and then assigns the phone to another user. In this scenario, the phone fails to onboard to the Webex cloud even though it can connect to Webex Calling service. Specifically, the status of the phone in Control Hub is shown as "Offline".

Solution

Manually perform a factory reset on the phone after a user is deleted in Control Hub. For more information about how to perform a factory reset, see one of the following topics for details:

- Factory Reset the Phone with the Keypad

- Perform Factory Reset from Phone Menu

- Factory Reset the Phone from Phone Web Page

Phone Display Problems

Your users may see unusual screen displays. Use the following sections to troubleshoot the problem.

- Phone Displays Irregular Fonts

- Phone Screen Displays Boxes Instead of Asian Characters

Phone Displays Irregular Fonts

Problem

The phone screen has smaller fonts than expected or there are unusual characters displayed. Examples of unusual characters are letters from a different alphabet from the characters that the locale uses.

Cause

Possible causes are:

- TFTP server does not have the correct set of locale and font files

- XML files or other files are specified as a font file

- The font and locale files did not download successfully.

Solution

- Font files and locale files must be in the same directory.

- Do not add or change files in the locale and font folder structure.

- On the phone web page, select Admin Login > Advanced > Info > Status and scroll to the Locale Download Package section to verify that the locale and font files downloaded successfully. If they did not, try the download again.

Phone Screen Displays Boxes Instead of Asian Characters

Problem

The phone is set for an Asian language, but the phone shows square boxes instead of Asian characters.

Cause

Possible causes are:

- TFTP server does not have the correct set of locale and font files.

- The font and locale files did not download successfully.

Solution

- Font files and locale files must be in the same directory.

- On the phone web page, select Admin Login > Advanced > Info > Status and scroll to the Locale Download Package section to verify that the locale and font files downloaded successfully. If they did not, try the download again.

Report All Phone Issues from Phone Web Page

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. You can generate PRT logs using the phone web page and upload them to a remote log server.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info > Debug Info.

Step 2

In the Problem Reports section, click Generate PRT.

Step 3

Enter the following information in the Report Problem screen:

1. Enter the date that you experienced the problem in the Date field. The current date appears in this field by default.

2. Enter the time that you experienced the problem in the Time field. The current time appears in this field by default.

3. In the Select Problem drop-down list box, choose the description of the problem from the available options.

Step 4

Click Submit in the Report Problem screen.

The Submit button is enabled only if you select a value in the Select Problem drop-down list box.

You get a notification alert on the Phone Web page that indicates if the PRT upload was successful or not.

Report Phone Issues from Webex Control Hub

You can issue a phone problem report remotely from the Webex Control Hub, after the phone successfully onboards to Webex cloud.

Before you begin

- Access the customer view in https://admin.webex.com/.

- Access the phone administration web page. See Access the Phone Web Interface.

- Problem Report Tool is configured successfully. URL specified in the PRT Upload Rule field is valid. See, Configure Problem Report Tool.

Procedure

Step 1

From the Webex Control Hub, generate the problem report of a phone.

For more information, see Webex for Cisco BroadWorks Solution Guide.

Step 2

(Optional) Check the PRT generation status in any of the following ways:

- Access the phone administration web page, select Info > Status > PRT Status. The PRT Generation Status shows that the Control Hub triggered PRT generation is successful and the PRT Upload Status shows that the uplaod is successful.

- On the phone, select Applications > Status > Last problem report info. The screen displays the report status is uploaded. The report generation time, the report upload time, and the PRT file name have the same value as shown in the phone administration web page.

When you do not generate a PRT or you factory reset the phone, then Last problem report info doesn't appear.

- Access the Webex Control Hub Help Desk and check the values of PRT generation. The values are identical with the values shown on the phone and on the phone administration web page.

Factory Reset the Phone from Phone Web Page

You can factory reset the phone from the phone web page. The reset only happens if the phone is idle. If the phone is not idle, the phone web page shows a message that the phone is busy and that you need to try again.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info > Debug Info.

Step 2

In the Factory Reset section, click Factory Reset.

Step 3

Click Confirm factory reset.

Reboot the Phone from the Phone Web Page

You can reboot the phone from the phone web page for changes to take effect.

Procedure

Enter the URL in a supported web browser.

You can enter URL in the format:

http://<Phone IP>/admin/reboot

where:

Phone IP = actual or VPN IP address of your phone.

/admin = path to access admin page of your phone.

reboot = command that you need to enter in the phone web page to reboot your phone.

After you enter the URL in the web browser, the phone reboots immediately.

Reboot the Phone from the Webex Control Hub

You can reboot the phone from the Webex Control Hub remotely, after the phone successfully onboards to Webex cloud. You can only reboot a phone that is in idle state. If it’s in use, such as in a call, the phone doesn’t reboot.

Before you begin

- Access the customer view in https://admin.webex.com/.

- Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

From the Webex Control Hub, reboot a phone.

For more information, see Webex for Cisco BroadWorks Solution Guide.

Step 2

(Optional) You can check the reboot reason from any of the following ways after the phone reboots successfully:

- Access the phone administration web page, select Info > Status > Reboot History. The reboot reason shows as cloud triggered.

- On the phone, select Applications > Status > Reboot history. The Reboot history screen shows that the reboot is cloud triggered.

Report a Phone Problem Remotely

You can initiate a phone problem report remotely. The phone generates a problem report using the Cisco Problem Report Tool (PRT), with the problem description "Remote PRT Trigger". If you have configured an upload rule for problem reports, the phone uploads the problem report according to the upload rule.

You can see the status of the problem report generation and upload on the phone administration web page. When a problem report is successfully generated, you can download the problem report from the phone administration web page.

Procedure

To initiate a phone problem report remotely, initiate a SIP-NOTIFY message from the server to the phone, with the Event specified as prt-gen.

Capture Packets

For troubleshooting purposes you may need to gather a packet capture from an IP Phone.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info > Debug Info.

Step 2

In the Problem Report Tool section, click the Start Packet Capture button in the Packet Capture field.

Step 3

Choose All to capture all packets that the phone receives and select Host IP Address to capture packets only when source or destination is the IP address of the phone.

Step 4

Make phone calls to and from the selected phone.

Step 5

When you want to stop the packet capture, click Stop Packet Capture.

Step 6

Click Submit.

You see a file in the Capture File field. This file contains the filtered packets.

Voice Quality Troubleshooting Tips

When you observe significant and persistent changes to metrics, use the following table for general troubleshooting information.

Changes to Voice Quality Metrics

Metric Change

Condition

Conceal Ratio and Conceal Seconds increase significantly

Network impairment from packet loss or high jitter.

Conceal Ratio is near or at zero, but the voice quality is poor.

- Noise or distortion in the audio channel such as echo or audio levels.

- Tandem calls that undergo multiple encode/decode such as calls to a cellular network or calling card network.

- Acoustic problems coming from a speakerphone, handsfree cellular phone or wireless headset.

Check packet transmit (TxCnt) and packet receive (RxCnt) counters to verify that voice packets are flowing.

MOS LQK scores decrease significantly

Network impairment from packet loss or high jitter levels:

- Average MOS LQK decreases may indicate widespread and uniform impairment.

- Individual MOS LQK decreases may indicate bursty impairment.

Cross-check the conceal ratio and conceal seconds for evidence of packet loss and jitter.

MOS LQK scores increase significantly

- Check to see if the phone is using a different codec than expected (RxType and TxType).

- Check to see if the MOS LQK version changed after a firmware upgrade.

Note

Voice quality metrics do not account for noise or distortion, only frame loss.

- Phone Behavior During Times of Network Congestion

Phone Behavior During Times of Network Congestion

Anything that degrades network performance can affect Cisco IP Phone audio quality, and in some cases, can cause a call to drop. Sources of network degradation can include, but are not limited to, the following activities:

- Administrative tasks, such as an internal port scan or security scan

- Attacks that occur on your network, such as a Denial of Service attack

To reduce or eliminate any adverse effects to the phones, schedule administrative network tasks during a time when the phones are not being used or exclude the phones from testing.

Where to Find Additional Information

If you have additional questions about troubleshooting your phone, see the Cisco IP Phone 6800, 7800, and 8800 Series Multiplatform Phones Troubleshooting FAQ in the the following Cisco website:

https://www.cisco.com/c/en/us/support/collaboration-endpoints/ip-phone-6800-series-multiplatform-firmware/products-tech-notes-list.html

Chapter 19. Monitoring Phone Systems

- Monitoring Phone Systems Overview

- Cisco IP Phone Status

- Reboot Reasons

Monitoring Phone Systems Overview

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:

- Device information

- Network setup information

- Network statistics

- Device logs

- Streaming statistics

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.

Cisco IP Phone Status

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.

- Display the Phone Information Window

- View Phone Information

- View the Phone Status

- View the Status Messages on the Phone

- View Download Status

- Determine the IP Address of the Phone

- View the Network Status

- Voice Quality Monitoring

- Display Call Statistics Screen

- View the Customization State in the Configuration Utility

Display the Phone Information Window

Procedure

Step 1

Press Applications .

Step 2

Select Status > Product information.

If the user is connected to a secure or authenticated server, a corresponding icon (lock or certificate) displays in the Phone Information Screen to the right of the server option. If the user is not connected to a secure or authenticated server, no icon appears.

The Product information screen might show the following information:

- Product name

- Serial number

- MAC address

- Software version

- Configuration version

The information displays only when it has been configured in the configuration file (cfg.xml).

- Hardware version

- VID (version ID)

- Certificate

- Customization

Step 3

To exit the Model Information screen, press .

View Phone Information

Procedure

To check the current status of the Cisco IP Phone, click the Info tab.

The Info tab shows information about all phone extensions, including phone statistics and the registration status.

View the Phone Status

Procedure

Step 1

Press Applications .

Step 2

Select Status > Phone status > Phone status.

You can view the following information:

- Elapsed time—Total time elapsed since the last reboot of the system

- Tx (Packets)—Transmitted packets from the phone.

- Rx (Packets)—Received packets from the phone.

View the Status Messages on the Phone

Procedure

Step 1

Press Applications .

Step 2

Select Status > Status messages.

You can view a log of the various phone statuses since provisioning was last done.

Note

Status messages reflect UTC time and are not affected by the timezone settings on the phone.

Step 3

Press Back.

View Download Status

You can view the download status from the phone web page when your user has difficulties with phone registration.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info > Download Status.

Step 2

View the firmware upgrade, provisioning, and custom CA status details as described in the Firmware Upgrade Status, Provisioning Status, Custom CA Status, and Screen Status.

Step 3

View the Manufacture Installed Certificate (MIC) renewal status details in the MIC Cert Refresh Status section.

Determine the IP Address of the Phone

A DHCP server assigns the IP address, so the phone must be booted up and connected to the subnetwork.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Info > Status.

Step 2

Scroll to IPv4 Information. Current IP displays the IP address.

Step 3

Scroll to IPv6 Information. Current IP displays the IP address.

View the Network Status

Procedure

Step 1

Press Applications .

Step 2

Select Status > Network status.

You can view the following information:

- Network type—Indicates the type of Local Area Network (LAN) connection that the phone uses.

- Network status—Indicates if the phone is connected to a network.

- IPv4 status—IP address of the phone. You can see information on IP address, Addressing type, IP status, Subnet mask, Default router, Domain Name Server (DNS) 1, DNS 2 of the phone.

- IPv6 status —IP address of the phone. You can see information on IP address, Addressing type, IP status, Subnet mask, Default router, Domain Name Server (DNS) 1, DNS 2 of the phone.

- VLAN ID—VLAN ID of the phone.

- MAC address—Unique Media Access Control (MAC) address of the phone.

- Host name—Displays the current host name assigned to the phone.

- Domain—Displays the network domain name of the phone. Default: cisco.com

- Switch port link—Status of the switch port.

- Switch port config—Indicates speed and duplex of the network port.

- PC port config—Indicates speed and duplex of the PC port.

- PC port link—Indicates speed and duplex of the PC port.

Voice Quality Monitoring

To measure the voice quality of calls that are sent and received within the network, Cisco IP Phones use these statistical metrics that are based on concealment events. The DSP plays concealment frames to mask frame loss in the voice packet stream.

- Concealment Ratio metrics—Show the ratio of concealment frames over total speech frames. An interval conceal ratio is calculated every 3 seconds.

- Concealed Second metrics—Show the number of seconds in which the DSP plays concealment frames due to lost frames. A severely "concealed second" is a second in which the DSP plays more than five percent concealment frames.

Note

Concealment ratio and concealment seconds are primary measurements based on frame loss. A Conceal Ratio of zero indicates that the IP network is delivering frames and packets on time with no loss.

You can access voice quality metrics from the Cisco IP Phone using the Call Statistics screen or remotely by using Streaming Statistics.

Display Call Statistics Screen

You can access the Call statistics menu on the phone to display detailed information of the recent calls. For example, call type, caller name, caller number.

Procedure

Step 1

Press Applications .

Step 2

Select Status > Phone status > Call statistics.

Step 3

To exit the Status menu, press Back.

- Call Statistics Fields

Call Statistics Fields

The following table describes the items on the Call Statistics screen.

Call Statistics Items for the Cisco IP Phone

Item

Description

Call type

An outbound or inbound call.

Peer name

The name of the person who made or answered the call.

Peer phone

The phone number of the person who made or answered the call.

Encode codec

The method used to compress the outgoing audio.

Decode codec

The method used to decompress the incoming audio.

Call time

The time a call was made or answered.

Call ID

An identifier of the caller.

View the Customization State in the Configuration Utility

After the RC download from the EDOS server completes, you can view the customization state of a phone using the web interface.

Here are the descriptions of the remote customization states:

- Open—The phone has booted for the first time and is not configured.

- Aborted—Remote customization is aborted due to other Provisioning like DHCP options.

- Pending—The profile has been downloaded from the EDOS server.

- Custom-Pending—The phone has downloaded a redirect URL from the EDOS server.

- Acquired—In the profile downloaded from the EDOS server, there is a redirect URL for provision configuration. If the redirect URL download from the provisioning server is successful, this state is displayed.

- Unavailable—Remote customization has stopped because the EDOS server responded with an empty provisioning file and the HTTP response was 200 OK.

Procedure

Step 1

On the Phone Web page, select Admin Login > Info > Status.

Step 2

In the Product Information section, you can view the customization state of the phone in the Customization field.

If any provisioning is failing, you can view the details in the Provisioning Status section on the same page.

Reboot Reasons

The phone stores the most recent five reasons that the phone was refreshed or rebooted. When the phone is reset to factory defaults, this information is deleted.

The following table describes the reboot and refresh reasons for the Cisco IP Phone.

Reason

Description

Upgrade

The reboot was a result of an upgrade operation (regardless whether the upgrade completed or failed).

Provisioning

The reboot was the result of changes made to parameter values by using the IP phone screen or phone web user interface, or as a result of synchronization.

SIP Triggered

The reboot was triggered by a SIP request.

RC

The reboot was triggered as a result of remote customization.

User Triggered

The user manually triggered a cold reboot.

IP Changed

The reboot was triggered after the phone IP address changed.

You can view the reboot history as follows:

- From the phone web user interface

- From the IP phone screen

- From the phone Status Dump file (http://phoneIP/status.xml or http://phoneIP/admin/status.xml)

- Reboot History on the Phone Web User Interface

- Reboot History on the Cisco IP Phone Screen

- Reboot History in the Status Dump File

Reboot History on the Phone Web User Interface

On the Info > System Status page, the Reboot History section displays the device reboot history, the five most recent reboot dates and times, and a reason for the reboot. Each field displays the reason for the reboot and a time stamp that indicates when the reboot took place.

For example:

Reboot Reason 1: [08/13/14 06:12:38] User Triggered
Reboot Reason 2: [08/10/14 10:30:10] Provisioning
Reboot Reason 3: [08/10/14 10:28:20] Upgrade

The reboot history displays in reverse chronological order; the reason for the most recent reboot displays in Reboot Reason 1.

Reboot History on the Cisco IP Phone Screen

Reboot History is located under Apps > Admin Settings > Status menu. In the Reboot History window, the reboot entries displays in reverse chronological order, similar to the sequence that displays on the phone web user interface.

Reboot History in the Status Dump File

The reboot history is stored in the Status Dump file
(http://<phone_IP_address>/admin/status.xml).

In this file, tags Reboot_Reason_1 to Reboot_Reason_3 store the reboot history, as shown in this example:

<Reboot_History>
<Reboot_Reason_1>[08/10/14 14:03:43]Provisioning</Reboot_Reason_1>
<Reboot_Reason_2>[08/10/14 13:58:15]Provisioning</Reboot_Reason_2>
<Reboot_Reason_3>[08/10/14 12:08:58]Provisioning</Reboot_Reason_3>
<Reboot_Reason_4>
<Reboot_Reason_5>
<Reboot_History/>

Chapter 20. Maintenance

- Basic Reset

Basic Reset

Performing a basic reset of a Cisco IP Phone provides a way to recover when the phone experiences an error. The reset provides a way to reset or restore various configuration and security settings.

Note

When you set up emergency calls, the phone requests an updated location whenever a person restarts the phone.

The following table describes the ways to perform a basic reset. You can reset a phone with any of these operations after the phone has started up. Choose the operation that is applicable for your situation.

Basic Reset Methods

Operation

Action

Explanation

Restart phone

Press Services, Applications , or Directories and then press **#**.

Resets any user and network setup changes that you have made, but that the phone has not written to its Flash memory, to previously saved settings, then restarts the phone.

Reset settings

Restores phone configuration or settings to factory default.

To reset settings, press Applications > Admin Settings > Custom Reset.

Restores phone configuration or settings to noncustomized default.

Note

When you set up emergency calls, the phone requests an updated location whenever the you do the following actions:

- Registers the phone with the call server.

- Restarts the phone (phone is registered).

- Changes the network interface that is used for the SIP registration.

- Changes the IP address of the phone.

- Factory Reset the Phone with the Keypad

- Perform Factory Reset from Phone Menu

- Factory Reset the Phone from Phone Web Page

- Identify Phone Issues with a URL in the Phone Web Page

Factory Reset the Phone with the Keypad

Use these steps to reset the phone to factory default settings using the phone keypad.

You have two methods to perform the factory reset by using the keypad:

- Method 1 (recommended): Press # > 123456789*0#

- Method 2: Press 0 > 369#

Before you begin

You must know if your phone is an original hardware release or if the hardware has been updated and re-released.

Procedure

Step 1

Unplug the phone:

- If using PoE, unplug the LAN cable.

- If using the power cube, unplug the power cube.

Step 2

Wait 5 seconds.

Step 3

Do one of the following actions:

- Method 1: Press and hold # and plug the phone back in.

- Method 2: Press and hold 0 and plug the phone back in.

Only the Cisco IP Phone 6821 support the method.

The phone begins the reboot process. The headset button and the speaker button light up.

Step 4

On earlier hardware versions, the Mute button lights up. Wait for the Mute button to turn off.

Step 5

Do one of the following actions:

- Method 1: Press 123456789*0# in sequence.

When you press 1, the lights on the headset button turns off. The light on the Select button flashes when a button is pressed.

After you press these buttons, the phone goes through the factory reset process.

If you press the buttons out of sequence, the phone powers on normally.

Caution

Do not power down the phone until it completes the factory reset process, and the main screen appears.

- Method 2: Press 369# in sequence.

For Cisco IP Phone 6821, after you press these buttons, the phone still remains on the same screen, and all LEDs change to solid green.

Step 6

If you use the Method 2, unplug and plug in the phone again to reboot it.

After the phone reboots, the main screen appears.

Perform Factory Reset from Phone Menu

Procedure

Step 1

Press Applications .

Step 2

Select Device administration > Factory reset.

Step 3

To restore phone configuration or settings to factory default, press OK.

Factory Reset the Phone from Phone Web Page

You can restore your phone to its original manufacturer settings from the phone web page. After you reset the phone, you can reconfigure it.

Procedure

Reset your phone from the phone web page from one of the methods:

- Enter the URL in a supported web browser and click Confirm Factory Reset.
You can enter URL in the format:

http://<Phone IP>/admin/factory-reset
where:

Phone IP = actual IP address of your phone.

/admin = path to access admin page of your phone.

factory-reset = command that you need to enter in the phone web page to factory-reset your phone.

- On the phone web page, select Admin Login > Advanced > Info > Debug Info.Click Factory Reset in the Factory Reset section and confirm the factory reset message in the next screen. Click Submit All Changes.

Identify Phone Issues with a URL in the Phone Web Page

When the phone doesn't work or doesn't register, a network error or any misconfiguration might be the cause. To identify the cause, add a specific IP address or a domain name to the phone admin page. Then, try to access so that the phone can ping the destination and display the cause.

Procedure

In a supported web browser, enter a URL that consists of your phone IP address and the destination IP that you want to ping. Enter the URL using the format:

http:/<Phone IP>/admin/ping?<ping destination>, where:

<Phone IP> = actual IP address of your phone.

/admin = path to the access admin page of your phone.

<ping destination> = any IP address or domain name that you want to ping.

The ping destination allows only alphanumeric characters, ‘-’, and “_” (underscores). Otherwise the phone shows an error on the web page. If the <ping destination> includes spaces, the phone uses only the first part of the address as the pinging destination.

For example, to ping the 192.168.1.1 address:

http://<Phone IP>/admin/ping?192.168.1.1

Technical Details

- Network and Computer Port Pinouts

- Network Protocols

- VLAN Interaction

- Disable the USB Port

- SIP and NAT Configuration

- Cisco Discovery Protocol

- LLDP-MED

- Final Network Policy Resolution and QoS

Network and Computer Port Pinouts

Although both the network and computer (access) ports are used for network connectivity, they serve different purposes and have different port pinouts:

- The network port is the 10/100/1000 SW port.

Note

The Cisco IP Phone 6821 Multiplatform Phones and Cisco IP Phone 6861 Multiplatform Phones have a 10/100 SW port.

- The computer (access) port is the 10/100/1000 PC port.

Note

Cisco IP Phone 6861 Multiplatform Phones doesn't have a PC port.

- Network Port Connector

- Computer Port Connector

Network Port Connector

The following table describes the network port connector pinouts.

Network Port Connector Pinouts

Pin Number

Function

1

BI_DA+

2

BI_DA-

3

BI_DB+

4

BI_DC+

5

BI_DC-

6

BI_DB-

7

BI_DD+

8

BI_DD-

Note

BI stands for bidirectional, while DA, DB, DC, and DD stand for Data A, Data B, Data C, and Data D respectively.

Computer Port Connector

The following table describes the computer port connector pinouts.

Computer (Access) Port Connector Pinouts

Pin Number

Function

1

BI_DB+

2

BI_DB-

3

BI_DA+

4

BI_DD+

5

BI_DD-

6

BI_DA-

7

BI_DC+

8

BI_DC-

Note

BI stands for bidirectional, while DA, DB, DC, and DD stand for Data A, Data B, Data C, and Data D respectively.

Network Protocols

Cisco IP Phones support several industry-standard and Cisco network protocols that are required for voice communication. The following table provides an overview of the network protocols that the phones support.

Supported Network Protocols on the Cisco IPPhone

Network Protocol

Purpose

Usage Notes

Bootstrap Protocol (BootP)

BootP enables a network device, such as the Cisco IP Phone, to discover certain startup information, such as its IP address.

—

Cisco Discovery Protocol (CDP)

CDP is a device-discovery protocol that runs on all Cisco-manufactured equipment.

A device can use CDP to advertise its existence to other devices and receive information about other devices in the network.

The Cisco IPPhone uses CDP to communicate information such as auxiliary VLAN ID, per port power management details, and Quality of Service (QoS) configuration information with the Cisco Catalyst switch.

Domain Name Server (DNS)

DNS translates domain names to IP addresses.

Cisco IP Phones have a DNS client to translate domain names into IP addresses.

Dynamic Host Configuration Protocol (DHCP)

DHCP dynamically allocates and assigns an IP address to network devices.

DHCP enables you to connect an IP phone into the network and have the phone become operational without the need to manually assign an IP address or to configure additional network parameters.

DHCP is enabled by default. If disabled, you must manually configure the IP address, subnet mask, and gateway on each phone locally.

We recommend that you use the DHCP custom option 160, 159.

Hypertext Transfer Protocol (HTTP)

HTTP is the standard protocol for transfer of information and movement of documents across the Internet and the web.

CiscoIP Phones use HTTP for XML services, provisioning, upgrade and for troubleshooting purposes.

Hypertext Transfer Protocol Secure (HTTPS)

Hypertext Transfer Protocol Secure (HTTPS) is a combination of the Hypertext Transfer Protocol with the SSL/TLS protocol to provide encryption and secure identification of servers.

Web applications with both HTTP and HTTPS support have two URLs configured. Cisco IP Phones that support HTTPS choose the HTTPS URL.

A lock icon is displayed to the user if the connection to the service is via HTTPS.

Internet Protocol (IP)

IP is a messaging protocol that addresses and sends packets across the network.

To communicate with IP, network devices must have an assigned IP address, subnet, and gateway.

IP addresses, subnets, and gateways identifications are automatically assigned if you are using the Cisco IPPhone with Dynamic Host Configuration Protocol (DHCP). If you are not using DHCP, you must manually assign these properties to each phone locally.

Link Layer Discovery Protocol (LLDP)

LLDP is a standardized network discovery protocol (similar to CDP) that is supported on some Cisco and third-party devices.

The Cisco IP Phone supports LLDP on the PC port.

Link Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED)

LLDP-MED is an extension of the LLDP standard developed for voice products.

The Cisco IPPhone supports LLDP-MED on the SW port to communicate information such as:

- Voice VLAN configuration

- Device discovery

- Power management

- Inventory management

For more information about LLDP-MED support, see the LLDP-MED and Cisco Discovery Protocol white paper at this URL: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper0900aecd804cd46d.shtml

Network Transport Protocol (NTP)

NTP is a networking protocol for clock synchronization between computer systems over packet-switched, variable-latency data networks.

Cisco IP Phones have an NTP client integrated into the software.

Real-Time Transport Protocol (RTP)

RTP is a standard protocol for transporting real-time data, such as interactive voice and video, over data networks.

Cisco IP Phones use the RTP protocol to send and receive real-time voice traffic from other phones and gateways.

Real-Time Control Protocol (RTCP)

RTCP works in conjunction with RTP to provide QoS data (such as jitter, latency, and round trip delay) on RTP streams.

RTCP is disabled by default.

Session Description Protocol (SDP)

SDP is the portion of the SIP protocol that determines which parameters are available during a connection between two endpoints. Conferences are established by using only the SDP capabilities that all endpoints in the conference support.

SDP capabilities, such as codec types, DTMF detection, and comfort noise, are normally configured on a global basis by a Third-Party Call Control System or a Media Gateway in operation. Some SIP endpoints may allow configuration of these parameters on the endpoint itself.

Session Initiation Protocol (SIP)

SIP is the Internet Engineering Task Force (IETF) standard for multimedia conferencing over IP. SIP is an ASCII-based application-layer control protocol (defined in RFC 3261) that can be used to establish, maintain, and terminate calls between two or more endpoints.

Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

Secure Real-Time Transfer protocol (SRTP)

SRTP is an extension of the Real-Time Protocol (RTP) Audio/Video Profile and ensures the integrity of RTP and Real-Time Control Protocol (RTCP) packets providing authentication, integrity, and encryption of media packets between two endpoints.

Cisco IP Phones use SRTP for media encryption.

Transmission Control Protocol (TCP)

TCP is a connection-oriented transport protocol.

—

Transport Layer Security (TLS)

TLS is a standard protocol for securing and authenticating communications.

When security is implemented, Cisco IP Phones use the TLS protocol when securely registering with the third-party call control system.

Trivial File Transfer Protocol (TFTP)

TFTP allows you to transfer files over the network.

On the Cisco IPPhone, TFTP enables you to obtain a configuration file specific to the phone type.

TFTP requires a TFTP server in your network, which can be automatically identified from the DHCP server.

User Datagram Protocol (UDP)

UDP is a connectionless messaging protocol for delivery of data packets.

UDP is used only for RTP streams. SIP uses UDP, TCP, and TLS.

Related Tasks

Verify the Network Setup

Verify Phone Startup

VLAN Interaction

The Cisco IP Phone contains an internal Ethernet switch, enabling forwarding of packets to the phone, and to the computer (access) port and the network port on the back of the phone.

If a computer is connected to the computer (access) port, the computer and the phone share the same physical link to the switch and share the same port on the switch. This shared physical link has the following implications for the VLAN configuration on the network:

- The current VLANs might be configured on an IP subnet basis. However, additional IP addresses might not be available to assign the phone to the same subnet as other devices that connect to the same port.

- Data traffic present on the VLAN supporting phones might reduce the quality of VoIP traffic.

- Network security may indicate a need to isolate the VLAN voice traffic from the VLAN data traffic.

You can resolve these issues by isolating the voice traffic onto a separate VLAN. The switch port to which the phone connects would be configured for separate VLANs for carrying:

- Voice traffic to and from the IP phone (auxiliary VLAN on the Cisco Catalyst 6000 series, for example)

- Data traffic to and from the PC that connects to the switch through the computer (access) port of the IP phone (native VLAN)

Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows a large number of phones to be added to an existing network that does not have enough IP addresses for each phone.

For more information, see the documentation that is included with a Cisco switch. You can also access switch information at this URL:

http://cisco.com/en/US/products/hw/switches/index.html

Disable the USB Port

If you don't allow the users to use the USB port for certain purposes, you can disable it on the phone web page. The only USB port locates at the back of the phone. The disabled USB port doesn't provide any function. For example, it doesn't recognize the USB headset. Also, it doesn't charge any connected device.

The Cisco IP Phone 6871 contains only one USB port, the back USB port.

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

Under the Power Settings section, set the parameter Disable Back USB Port to Yes to turn off the back USB port.

You can configure this parameter in the phone configuration XML file (cfg.xml) by entering a string in this format:

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

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

SIP and NAT Configuration

- SIP and the Cisco IP Phone

- NAT Transversal with Phones

SIP and the Cisco IP Phone

The Cisco IP Phone uses Session Initiation Protocol (SIP), which allows interoperation with all IT service providers that support SIP. SIP is an IETF-defined signaling protocol that controls voice communication sessions in an IP network.

SIP handles signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management controls the attributes of an end-to-end call.

In typical commercial IP telephony deployments, all calls go through a SIP Proxy Server. The receiving phone is called the SIP user agent server (UAS), while the requesting phone is called the user agent client (UAC).

SIP message routing is dynamic. If a SIP proxy receives a request from a UAS for a connection but cannot locate the UAC, the proxy forwards the message to another SIP proxy in the network. When the UAC is located, the response routes back to the UAS, and the two UAs connect using a direct peer-to-peer session. Voice traffic transmits between UAs over dynamically assigned ports using Real-time Protocol (RTP).

RTP transmits real-time data such as audio and video; RTP does not guarantee real-time delivery of data. RTP provides mechanisms for the sending and receiving applications to support streaming data. Typically, RTP runs on top of UDP.

- SIP Over TCP

- SIP Proxy Redundancy

- Dual Registration

- RFC3311

- SIP NOTIFY XML-Service

SIP Over TCP

To guarantee state-oriented communications, the Cisco IP Phone can use TCP as the transport protocol for SIP. This protocol provides guaranteed delivery that assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent.

TCP overcomes the problem of UDP port-blocking by corporate firewalls. With TCP, new ports do not need to be open or packets dropped, because TCP is already in use for basic activities, such as internet browsing or e-commerce.

SIP Proxy Redundancy

An average SIP Proxy Server can handle tens of thousands of subscribers. A backup server allows an active server to be temporarily switched out for maintenance. The phone supports the use of backup servers to minimize or eliminate service disruption.

A simple way to support proxy redundancy is to specify a SIP Proxy Server in the phone configuration profile. The phone sends a DNS NAPTR or SRV query to the DNS server. If configured, the DNS server returns SRV records that contain a list of servers for the domain, with their hostnames, priority, listening ports, and so forth. The phone tries to contact the servers in the order of the priority. The server with a lower number has a higher priority. Up to six NAPTR records and twelve SRV records are supported in a query.

When the phone fails to communicate with the primary server, the phone can failover to a lower-priority server. If configured, the phone can restore the connection back to the primary. Failover and failback support switches between servers with different SIP transport protocols. The phone doesn't perform failback to the primary server during an active call until the call ends and the failback conditions are met.

Example of Resource Records from the DNS Server

as1bsoft     3600    IN NAPTR 50   50  "s"  "SIPS+D2T"     ""  _sips._tcp.tlstest
3600    IN NAPTR 90   50  "s"  "SIP+D2T"      ""  _sip._tcp.tcptest
3600    IN NAPTR 100  50  "s"  "SIP+D2U"      ""  _sip._udp.udptest

_sips._tcp.tlstest  SRV 1 10 5061 srv1.sipurash.com.
SRV 2 10 5060 srv2.sipurash.com.
_sip._tcp.tcptest   SRV 1 10 5061 srv3.sipurash.com.
SRV 2 10 5060 srv4.sipurash.com.
_sip._udp.udptest   SRV 1 10 5061 srv5.sipurash.com.
SRV 2 10 5060 srv6.sipurash.com.

srv1     3600    IN    A   1.1.1.1
srv2     3600    IN    A   2.2.2.2
srv3     3600    IN    A   3.3.3.3
srv4     3600    IN    A   4.4.4.4
srv5     3600    IN    A   5.5.5.5
srv6     3600    IN    A   6.6.6.6

The following example shows the priority of the servers from the perspective of the phone.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS            UP
2nd            2.2.2.2            TLS            UP
3rd            3.3.3.3            TCP            UP
4th            4.4.4.4            TCP            UP
5th            5.5.5.5            UDP            UP
6th            6.6.6.6            UDP            UP

The phone always sends SIP messages to the available address with the top priority and with the status UP in the list. In the example, the phone sends all the SIP messages to the address 1.1.1.1. If the address 1.1.1.1 in the list is marked with the status DOWN, the phone communicates with 2.2.2.2 instead. The phone can restore the connection back to 1.1.1.1 when the specified failback conditions are met. For more details about failover and failback, see SIP Proxy Failover and SIP Proxy Fallback.

- SIP Proxy Failover

- SIP Proxy Fallback

SIP Proxy Failover

The phone performs a failover in any of these cases:

- The phone sends SIP messages and doesn't get responses from the server.

- The server responds with a code that matches the specified code in Try Backup RSC.

- The phone gets a TCP disconnection request.

We strongly recommend that you set the Auto Register When Failover to Yes when SIP Transport is set to Auto.

You can also configure this extension-specific parameters in the configuration file:

<SIP_Transport_n_ ua="na">Auto</SIP_Transport_n_>

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

where n is the extension number.

Phone Failover Behavior

When the phone fails to communicate with the currently connected server, it refreshes the server list status. The unavailable server is marked with the status DOWN in the server list. The phone tries to connect to the top-priority server with the status UP in the list.

In the following example, the addresses 1.1.1.1 and 2.2.2.2 aren’t available. The phone sends SIP messages to 3.3.3.3, which has the top priority among the servers with the status UP.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS          DOWN
2nd            2.2.2.2            TLS          DOWN
3rd            3.3.3.3            TCP          UP
4th            4.4.4.4            TCP          UP
5th            5.5.5.5            UDP          UP
6th            6.6.6.6            UDP          UP

In the following example, there are two SRV records from the DNS NAPTR response. For each SRV record, there are three A records (IP addresses).

Priority       IP Address     SIP Protocol    Server            Status
1st            1.1.1.1            UDP         SRV1              DOWN
2nd            1.1.1.2            UDP         SRV1              UP
3rd            1.1.1.3            UDP         SRV1              UP
4th            2.2.2.1            TLS         SRV2              UP
5th            2.2.2.2            TLS         SRV2              UP
6th            2.2.2.3            TLS         SRV2              UP

Let's assume that the phone failed to connect to 1.1.1.1 and then registered to 1.1.1.2. When 1.1.1.2 goes down, phone behavior depends on the setting of Proxy Fallback Intvl.

- When Proxy Fallback Intvl is set to 0, the phone tries with the addresses in this order: 1.1.1.1, 1.1.1.3, 2.2.2.1, 2.2.2.2, 2.2.2.3.

- When Proxy Fallback Intvl is set to a value other than zero, the phone tries with the addresses in this order: 1.1.1.3, 2.2.2.1, 2.2.2.2, 2.2.2.3.

SIP Proxy Fallback

The proxy fallback requires a value other than zero specified in the Proxy Fallback Intvl field on the Ext (n) tab in the phone web interface. If you set this field to 0, the SIP proxy failback feature is disabled. You can also configure this extension-specific parameter in the configuration file in this format:

<Proxy_Fallback_Intvl_n_ ua="na">60</Proxy_Fallback_Intvl_n_>

where n is the extension number.

The time when the phone triggers a failback depends on the phone configuration and the SIP transport protocols in use.

To enable the phone to perform failback between different SIP transport protocols, set SIP Transport to Auto on the Ext (n) tab in the phone web interface. You can also configure this extension-specific parameter in the configuration file with the following XML string:

<SIP_Transport_n_ ua="na">Auto</SIP_Transport_n_>

where n is the extension number.

Failback from a UDP Connection

The failback from a UDP connection is triggered by SIP messages. In the following example, the phone first failed to register to 1.1.1.1 (TLS) at the time T1 since there's no response from the server. When SIP Timer F expires, the phone registers to 2.2.2.2 (UDP) at the time T2 (T2=T1+SIP Timer F). The current connection is on 2.2.2.2 via UDP.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS          DOWN        T1 (Down time)
2nd            2.2.2.2            UDP          UP
3rd            3.3.3.3            TCP          UP

The phone has the following configuration:

<Proxy_Fallback_Intvl_n_ ua="na">60</Proxy_Fallback_Intvl_n_>
<Register_Expires_n_ ua="na">3600</Register_Expires_n_>
<SIP_Timer_F ua="na">16</SIP_Timer_F>

where n is the extension number.

The phone refreshes the registration at time T2 (T2=(3600-16)*78%). The phone checks the address list for the availability of the IP addresses and the down time. If T2-T1 > = 60, the failed server 1.1.1.1 resumes back to UP and the list is updated to the following. The phone sends SIP messages to 1.1.1.1.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS           UP
2nd            2.2.2.2            UDP           UP
3rd            3.3.3.3            TCP           UP

Failback from a TCP or TLS Connection

The failback from a TCP or TLS connection is triggered by the parameter Proxy Fallback Intvl. In the following example, the phone failed to register to 1.1.1.1 (UDP) at the time T1 and thus registered to 2.2.2.2 (TCP). The current connection is on 2.2.2.2 via TCP.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            UDP          DOWN        T1 (Down time)
2nd            2.2.2.2            TCP          UP
3rd            3.3.3.3            TLS          UP

The phone has the following configuration:

<Proxy_Fallback_Intvl_n_ ua="na">60</Proxy_Fallback_Intvl_n_>
<Register_Expires_n_ ua="na">3600</Register_Expires_n_>
<SIP_Timer_F ua="na">16</SIP_Timer_F>

where n is the extension number.

The proxy fallback interval (60 seconds) counts down from T1. The phone triggers proxy failback at the time of T1+60. If you set the proxy fallback interval to 0 in this example, the phone keeps the connection on 2.2.2.2.

Dual Registration

The phone always registers to both primary (or primary outbound) and alternate (or alternate outbound) proxies. After registration, the phone sends out Invite and Non-Invite SIP messages through primary proxy first. If there is no response for the new INVITE from the primary proxy, after timeout, the phone attempts to connect with the alternate proxy. If the phone fails to register to the primary proxy, it sends an INVITE to the alternate proxy without trying the primary proxy.

Dual registration is supported on a per-line basis. Three added parameters can be configured through web user interface and remote provisioning:

- Alternate Proxy—Default is empty.

- Alternate Outbound Proxy—Default is empty.

- Dual Registration—Default is NO (turned off).

After you configure the parameters, reboot the phone for the feature to take effect.

Note

Specify a value for primary proxy (or primary outbound proxy) and alternate proxy (or alternate outbound proxy) for the feature to function properly.

- Dual Registration and DNS SRV Limitations

- Dual Registration and Alternate Proxy

Dual Registration and DNS SRV Limitations

- When Dual Registration is enabled, DNS SRV Proxy Fallback or Recovery must be disabled.

- Do not use Dual Registration along with other Fallback or Recovery mechanisms. For example: Broadsoft mechanism.

- There is no recovery mechanism for feature request. However, the administrator can adjust the reregistration time for a prompt update of the registration state for primary and alternate proxy.

Dual Registration and Alternate Proxy

When the Dual Register parameter is set to No, Alternate Proxy is ignored.

RFC3311

The Cisco IP Phone supports RFC-3311, the SIP UPDATE Method.

SIP NOTIFY XML-Service

The Cisco IP Phone supports the SIP NOTIFY XML-Service event. On receipt of a SIP NOTIFY message with an XML-Service event, the phone challenges the NOTIFY with a 401 response if the message does not contain correct credentials. The client must furnish the correct credentials using MD5 digest with the SIP account password for the corresponding line of the IP phone.

The body of the message can contain the XML event Message. For example:

<CiscoIPPhoneExecute>
<ExecuteItem Priority="0" URL="http://xmlserver.com/event.xml"/>
</CiscoIPPhoneExecute>

Authentication:

challenge  = MD5( MD5(A1) ":" nonce ":" nc-value ":" cnonce ":" qop-value
":" MD5(A2) )
where A1 = username ":" realm ":" passwd
and A2 = Method ":" digest-uri

NAT Transversal with Phones

Network Address Translation (NAT) allows multiple devices to share a single, public, routable, IP address to establish connections over the Internet. NAT is present in many broadband access devices to translate public and private IP addresses. For VoIP to coexist with NAT, NAT traversal is required.

Not all service providers provide NAT traversal. If your service provider does not provide NAT traversal, you have several options:

- NAT Mapping with Session Border Controller: We recommend that you choose an service provider that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the service provider, you have more choices in selecting a router.

- NAT Mapping with SIP-ALG Router: NAT mapping can be achieved by using a router that has a SIP Application Layer Gateway (ALG). By using a SIP-ALG router, you have more choices in selecting an service provider.

- NAT Mapping with a Static IP Address: NAT mapping with an external (public) static IP address can be acheived to ensure interoperability with the service provider. The NAT mechanism used in the router must be symmetric. For more information, see Determine Symmetric or Asymmetric NAT.

Use NAT mapping only if the service provider network does not provide a Session Border Controller functionality. For more information on how to configure NAT mapping with a static IP, see Configure NAT Mapping with the Static IP Address.

- NAT Mapping with STUN: If the service provider network does not provide a Session Border Controller functionality and if the other requirements are met, it is possible to use Session Traversal Utilities for NAT (STUN) to discover the NAT mapping. For information on how to configure NAT mapping with STUN, see Configure NAT mapping with STUN.

- NAT Mapping with Session Border Controller

- NAT Mapping with SIP-ALG Router

NAT Mapping with Session Border Controller

We recommend that you choose an service provider that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the service provider, you have more choices in selecting a router.

NAT Mapping with SIP-ALG Router

NAT mapping can be achieved by using a router that has a SIP Application Layer Gateway (ALG). By using a SIP-ALG router, you have more choices in selecting an service provider.

Cisco Discovery Protocol

Cisco Discovery Protocol (CDP) is negotiation-based and determines which virtual LAN (VLAN) the Cisco IP Phone resides in. If you are using a Cisco switch, Cisco Discovery Protocol (CDP) is available and is enabled by default. CDP has these attributes:

- Obtains the protocol addresses of neighboring devices and discovers the platform of those devices.

- Shows information about the interfaces your router uses.

- Is media and protocol-independent.

If you are using a VLAN without CDP, you must enter a VLAN ID for the Cisco IP Phone.

LLDP-MED

The Cisco IP Phone supports Link Layer Discovery Protocol for Media Endpoint Devices (LLDP-MED) for deployment with Cisco or other Third-Party network connectivity devices that use a Layer 2 auto discovery mechanism. Implementation of LLDP-MED is done in accordance with IEEE 802.1AB (LLDP) Specification of May 2005, and ANSI TIA-1057 of April 2006.

The Cisco IP Phone operates as a LLDP-MED Media End Point Class III device with direct LLDP-MED links to Network Connectivity Devices, according to the Media Endpoint Discovery Reference Model and Definition (ANSI TIA-1057 Section 6).

The Cisco IP Phone supports only the following limited set of Type-Length-Values (TLV) as an LLDP-MED Media Endpoint device class III:

- Chassis ID TLV

- Port ID TLV

- Time to live TLV

- Port Description TLV

- System Name TLV

- System Capabilities TLV

- IEEE 802.3 MAC/PHY Configuration/Status TLV (for wired network only)

- LLDP-MED Capabilities TLV

- LLDP-MED Network Policy TLV (for application type=Voice only)

- LLDP-MED Extended Power-Via-MDI TLV (for wired network only)

- LLDP-MED Firmware Revision TLV

- End of LLDPDU TLV

The outgoing LLDPDU contains all the preceding TLVs if applicable. For the incoming LLDPDU, the LLDPDU is discarded if any of the following TLVs are missing. All other TLVs are not validated and ignored.

- Chassis ID TLV

- Port ID TLV

- Time to live TLV

- LLDP-MED Capabilities TLV

- LLDP-MED Network Policy TLV (for application type=Voice only)

- End of LLDPDU TLV

The Cisco IP Phone sends out the shutdown LLDPDU if applicable. The LLDPDU frame contains the following TLVs:

- Chassis ID TLV

- Port ID TLV

- Time to live TLV

- End of LLDPDU TLV

There are some restrictions in the implementation of LLDP-MED on the Cisco IP Phones:

- Storage and retrieval of neighbor information are not supported.

- SNMP and corresponding MIBs are not supported.

- Recording and retrieval of statistical counters are not supported.

- Full validation of all TLVs does not take place; TLVs that do not apply to the phones are ignored.

- Protocol state machines as stated in the standards are used only for reference.

- Chassis ID TLV

- Port ID TLV

- Time to Live TLV

- End of LLDPDU TLV

- Port Description TLV

- System Name TLV

- System Capabilities TLV

- Management Address TLV

- System Description TLV

- IEEE 802.3 MAC/PHY Configuration/Status TLV

- LLDP-MED Capabilities TLV

- Network Policy TLV

- LLDP-MED Extended Power-Via-MDI TLV

- LLDP-MED Inventory Management TLV

Chassis ID TLV

For the outgoing LLDPDU, the TLV supports subtype=5 (Network Address). When the IP address is known, the value of the Chassis ID is an octet of the INAN address family number followed by the octet string for the IPv4 address used for voice communication. If the IP address is unknown, the value for the Chassis ID is 0.0.0.0. The only INAN address family supported is IPv4. Currently, the IPv6 address for the Chassis ID is not supported.

For the incoming LLDPDU, the Chassis ID is treated as an opaque value to form the MSAP identifier. The value is not validated against its subtype.

The Chassis ID TLV is mandatory as the first TLV. Only one Chassis ID TLV is allowed for the outgoing and incoming LLDPDUs.

Port ID TLV

For the outgoing LLDPDU, the TLV supports subtype=3 (MAC address). The 6 octet MAC address for the Ethernet port is used for the value of Port ID.

For the incoming LLDPDU, the Port ID TLV is treated as an opaque value to form the MSAP identifier. The value is not validated against its subtype.

The Port ID TLV is mandatory as the second TLV. Only one Port ID TLV is allowed for the outgoing and incoming LLDPDUs.

Time to Live TLV

For the outgoing LLDPDU, the Time to Live TTL value is 180 seconds. This differs from the 120-second value that the standard recommends. For the shutdown LLDPDU, the TTL value is always 0.

The Time to Live TLV is mandatory as the third TLV. Only one Time to Live TLV is allowed for the outgoing and incoming LLDPDUs.

End of LLDPDU TLV

The value is 2-octet, all zero. This TLV is mandatory and only one is allowed for the outgoing and incoming LLDPDUs.

Port Description TLV

For the outgoing LLDPDU, in the Port Description TLV, the value for the port description is the same as “Port ID TLV” for CDP. The incoming LLDPDU, the Port Description TLV, is ignored and not validated. Only one Port Description TLV is allowed for outgoing and incoming LLDPDUs.

System Name TLV

For the Cisco IP Phone, the value is SEP+MAC address.

Example: SEPAC44F211B1D0

The incoming LLDPDU, the System Name TLV, is ignored and not validated. Only one System Name TLV is allowed for the outgoing and incoming LLDPDUs.

System Capabilities TLV

For the outgoing LLDPDU, in the System Capabilities TLV, the bit values for the 2 octet system capabilities fields should be set for Bit 2 (Bridge) and Bit 5 (Phone) for a phone with a PC port. If the phone does not have a PC port, only Bit 5 should be set. The same system capability value should be set for the enabled capability field.

For the incoming LLDPDU, the System Capabilities TLV is ignored. The TLV is not validated semantically against the MED device type.

The System Capabilities TLV is mandatory for outgoing LLDPDUs. Only one System Capabilities TLV is allowed.

Management Address TLV

The TLV identifies an address associated with the local LLDP agent (that may be used to reach higher layer entities) to assist discovery by network management. The TLV allows the inclusion of both the system interface number and an object identifier (OID) that are associated with this management address, if either or both are known.

- TLV information string length—This field contains the length (in octets) of all the fields in the TLV information string.

- Management address string length—This field contains the length (in octets) of the management address subtype + management address fields.

System Description TLV

The TLV allows the network management to advertise the system description.

- TLV information string length—This field indicates the exact length (in octets) of the system description.

- System description—This field contains an alphanumeric string that is the textual description of the network entity. The system description includes the full name and version identification of the system hardware type, software operating system, and networking software. If implementations support IETF RFC 3418, the sysDescr object should be used for this field.

IEEE 802.3 MAC/PHY Configuration/Status TLV

The TLV is not for autonegotiation, but for troubleshooting purposes. For the incoming LLDPDU, the TLV is ignored and not validated. For the outgoing LLDPDU, for the TLV, the octet value autonegotiation support/status should be:

- Bit 0—Set to 1 to indicate that the autonegotiation support feature is supported.

- Bit 1—Set to 1 to indicate that autonegotiation status is enabled.

- Bit 2-7—Set to 0.

The bit values for the 2 octets PMD autonegotiation advertised capability field should be set to:

- Bit 13—10BASE-T half duplex mode

- Bit 14—10BASE-T full duplex mode

- Bit 11—100BASE-TX half duplex mode

- Bit 10—100BASE-TX full duplex mode

- Bit 15—Unknown

Bit 10, 11, 13 and 14 should be set.

The value for 2 octets operational MAU type should be set to reflect the real operational MAU type:

- 16—100BASE-TX full duplex

- 15—100BASE-TX half duplex

- 11—10BASE-T full duplex

- 10—10BASE-T half duplex

For example, usually, the phone is set to 100BASE-TX full duplex. The value 16 should then be set. The TLV is optional for a wired network and not applicable for a wireless network. The phone sends out this TLV only when in wired mode. When the phone is not set for autonegotiation but specific speed/duplexity, for the outgoing LLDPDU TLV, bit 1 for the octet value autonegotiation support/status should be clear (0) to indicate that autonegotiation is disabled. The 2 octets PMD autonegotiation advertised capability field should be set to 0x8000 to indicate unknown.

LLDP-MED Capabilities TLV

For the outgoing LLDPDU, the TLV should have the device type 3 (End Point Class III) with the following bits set for 2-octet Capability field:

Bit Position

Capability

0

LLDP-MED Capabilities

1

Network Policy

4

Extended Power via MDI-PD

5

Inventory

For the incoming TLV, if the LLDP-MED TLV is not present, the LLDPDU is discarded. The LLDP-MED Capabilities TLV is mandatory and only one is allowed for the outgoing and incoming LLDPDUs. Any other LLDP-MED TLVs will be ignored if they present before the LLDP-MED Capabilities TLV.

Network Policy TLV

In the TLV for the outgoing LLDPDU, before the VLAN or DSCP is determined, the Unknown Policy Flag (U) is set to 1. If the VLAN setting or DSCP is known, the value is set to 0. When the policy is unknown, all other values are set to 0. Before the VLAN is determined or used, the Tagged Flag (T) is set to 0. If the tagged VLAN (VLAN ID > 1) is used for the phone, the Tagged Flag (T) is set to 1. Reserved (X) is always set to 0. If the VLAN is used, the corresponding VLAN ID and L2 Priority will be set accordingly. VLAN ID valid value is range from 1-4094. However, VLAN ID=1 will never be used (limitation). If DSCP is used, the value range from 0-63 is set accordingly.

In the TLV for the incoming LLDPDU, Multiple Network Policy TLVs for different application types are allowed.

LLDP-MED Extended Power-Via-MDI TLV

In the TLV for the outgoing LLDPDU, the binary value for Power Type is set to “0 1” to indicate the power type for phone is PD Device. The Power source for the phone is set to “PSE and local” with binary value “1 1”. The Power Priority is set to binary “0 0 0 0” to indicate unknown priority while the Power Value is set to maximum power value. The Power Value for the Cisco IP Phone is 12900mW.

For the incoming LLDPDU, the TLV is ignored and not validated. Only one TLV is allowed in the outgoing and incoming LLDPDUs. The phone will send out the TLV for the wired network only.

The LLDP-MED standard was originally drafted in the context of Ethernet. Discussion is ongoing for LLDP-MED for Wireless Networks. Refer to ANSI-TIA 1057, Annex C, C.3 Applicable TLV for VoWLAN, table 24. It is recommended that the TLV is not applicable in the context of the wireless network. This TLV is targeted for use in the context of PoE and Ethernet. The TLV, if added, will not provide any value for network management or power policy adjustment at the switch.

LLDP-MED Inventory Management TLV

This TLV is optional for Device Class III. For the outgoing LLDPDU, we support only Firmware Revision TLV. The value for the Firmware Revision is the version of firmware on the phone. For the incoming LLDPDU, the TLVs are ignored and not validated. Only one Firmware Revision TLV is allowed for the outgoing and incoming LLDPDUs.

Final Network Policy Resolution and QoS

- Special VLANs

- Default QoS for SIP Mode

- QoS Resolution for CDP

- QoS Resolution for LLDP-MED

- Coexistence with CDP

- LLDP-MED and Multiple Network Devices

Special VLANs

VLAN=0, VLAN=1, and VLAN=4095 are treated the same way as an untagged VLAN. Because the VLAN is untagged, Class of Service (CoS) is not applicable.

Default QoS for SIP Mode

If there is no network policy from CDP or LLDP-MED, the default network policy is used. CoS is based on configuration for the specific extension. It is applicable only if the manual VLAN is enabled and manual VLAN ID is not equal to 0, 1, or 4095. Type of Service (ToS) is based on configuration for the specific extension.

QoS Resolution for CDP

If there is a valid network policy from CDP:

- If the VLAN=0, 1, or 4095, the VLAN will not be set, or the VLAN is untagged. CoS is not applicable, but DSCP is applicable. ToS is based on the default as previously described.

- If the VLAN > 1 and VLAN < 4095, the VLAN is set accordingly. CoS and ToS are based on the default as previously described. DSCP is applicable.

- The phone reboots and restarts the fast start sequence.

QoS Resolution for LLDP-MED

If CoS is applicable and if CoS = 0, the default is used for the specific extension as previously described. But the value shown on L2 Priority for TLV for outgoing LLDPDU is based on the value used for extension 1. If CoS is applicable and if CoS != 0, CoS is used for all extensions.

If DSCP (mapped to ToS) is applicable and if DSCP = 0, the default is used for the specific extension as previously described. But the value show on DSCP for TLV for outgoing LLDPDU is based on value used for the extension 1. If DSCP is applicable and if DSCP != 0, DSCP is used for all extensions.

If the VLAN > 1 and VLAN < 4095, the VLAN is set accordingly. CoS and ToS are based on the default as previously described. DSCP is applicable.

If there is a valid network policy for the voice application from LLDP-MED PDU and if the tagged flag is set, the VLAN, L2 Priority (CoS), and DSCP (mapped to ToS) are all applicable.

If there is a valid network policy for the voice application from LLDP-MED PDU and if the tagged flag is not set, only the DSCP (mapped to ToS) is applicable.

The Cisco IP Phone reboots and restarts the fast start sequence.

Coexistence with CDP

If both CDP and LLDP-MED are enabled, the network policy for the VLAN determines the last policy set or changed with either one of the discovery modes. If both LLDP-MED and CDP are enabled, during startup the phone sends CDP and LLDP-MED PDUs.

Inconsistent configuration and behavior for network connectivity devices for CDP and LLDP-MED modes could result in an oscillating rebooting behavior for the phone due to switching to different VLANs.

If the VLAN is not set by CDP and LLDP-MED, the VLAN ID that is configured manually is used. If the VLAN ID is not configured manually, no VLAN is supported. DSCP is used and the network policy determines LLDP-MED if applicable.

LLDP-MED and Multiple Network Devices

If the same application type is used for the network policy but different Layer 2 or Layer 3 QoS Network policies are received by the phones from multiple network connectivity devices, the last valid network policy is honored. To ensure deterministic and consistent of Network Policy, multiple network connectivity devices should not send out conflicting network policies for the same application type.

TR-069 Parameter Comparison

- XML and TR-069 Parameter Comparison

XML and TR-069 Parameter Comparison

This table shows the XML parameters that the phones use, with their TR-069 counterpart.

TR-069 Parameter

XML Parameter

Device.Services.VoiceService.

N/A

Device.Services.VoiceService.{i}.

N/A

Device.Services.VoiceService.{i}.Capabilities.

N/A

Device.Services.VoiceService.{i}.Capabilities.ButtonMap

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.{i}.

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.{i}.BitRate

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.{i}.Codec

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.{i}.EntryID

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.{i}.PacketizationPeriod

N/A

Device.Services.VoiceService.{i}.Capabilities.Codecs.{i}.SilenceSuppression

N/A

Device.Services.VoiceService.{i}.Capabilities.DigitMap

N/A

Device.Services.VoiceService.{i}.Capabilities.DSCPCoupled

N/A

Device.Services.VoiceService.{i}.Capabilities.EthernetTaggingCoupled

N/A

Device.Services.VoiceService.{i}.Capabilities.FaxPassThrough

N/A

Device.Services.VoiceService.{i}.Capabilities.FaxT38

N/A

Device.Services.VoiceService.{i}.Capabilities.FileBasedRingGeneration

N/A

Device.Services.VoiceService.{i}.Capabilities.FileBasedToneGeneration

N/A

Device.Services.VoiceService.{i}.Capabilities.MaxLineCount

N/A

Device.Services.VoiceService.{i}.Capabilities.MaxProfileCount

N/A

Device.Services.VoiceService.{i}.Capabilities.MaxSessionCount

N/A

Device.Services.VoiceService.{i}.Capabilities.MaxSessionsPerLine

N/A

Device.Services.VoiceService.{i}.Capabilities.ModemPassThrough

N/A

Device.Services.VoiceService.{i}.Capabilities.NumberingPlan

N/A

Device.Services.VoiceService.{i}.Capabilities.PatternBasedRingGeneration

N/A

Device.Services.VoiceService.{i}.Capabilities.PatternBasedToneGeneration

N/A

Device.Services.VoiceService.{i}.Capabilities.PSTNSoftSwitchOver

N/A

Device.Services.VoiceService.{i}.Capabilities.Regions

N/A

Device.Services.VoiceService.{i}.Capabilities.RingDescriptionsEditable

N/A

Device.Services.VoiceService.{i}.Capabilities.RingFileFormats

N/A

Device.Services.VoiceService.{i}.Capabilities.RingGeneration

N/A

Device.Services.VoiceService.{i}.Capabilities.RingPatternEditable

N/A

Device.Services.VoiceService.{i}.Capabilities.RTCP

N/A

Device.Services.VoiceService.{i}.Capabilities.RTPRedundancy

N/A

Device.Services.VoiceService.{i}.Capabilities.SignalingProtocols

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.EventSubscription

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.Extensions

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.ResponseMap

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.Role

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.TLSAuthenticationKeySizes

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.TLSAuthenticationProtocols

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.TLSEncryptionKeySizes

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.TLSEncryptionProtocols

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.TLSKeyExchangeProtocols

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.Transports

N/A

Device.Services.VoiceService.{i}.Capabilities.SIP.URISchemes

N/A

Device.Services.VoiceService.{i}.Capabilities.SRTP

N/A

Device.Services.VoiceService.{i}.Capabilities.SRTPEncryptionKeySizes

N/A

Device.Services.VoiceService.{i}.Capabilities.SRTPKeyingMethods

N/A

Device.Services.VoiceService.{i}.Capabilities.ToneDescriptionsEditable

N/A

Device.Services.VoiceService.{i}.Capabilities.ToneFileFormats

N/A

Device.Services.VoiceService.{i}.Capabilities.ToneGeneration

N/A

Device.Services.VoiceService.{i}.Capabilities.VoicePortTests

N/A

Device.Services.VoiceService.{i}.VoiceProfile.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.DTMFMethod

DTMF_Tx_Method_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Enable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.AnonymousCalEnable

Block_CID_Setting

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.AnonymousCallBlockEnable

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallerIDEnable

Block_CID_Setting

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallerIDName

Display_Name_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallForwardOnBusyNumber

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallForwardOnNoAnswerNumber

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallForwardOnNoAnswerRingCount

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallForwardUnconditionalEnable

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallForwardUnconditionalNumber

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallReturnEnable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallTransferEnable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.CallWaitingEnable

CW_Setting

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.ConferenceCallingSessionCount

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.ConferenceCallingStatus

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.DoNotDisturbEnable

DND_Setting

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.MaxSessions

Call_Appearances_Per_Line

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.MessageWaiting

Message_Waiting_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.MWIEnable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.RepeatDialEnable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallingFeatures.X_CISCO_SharedLineDNDCfwdEnable

Shared_Line_DND_Cfwd_Enable

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.CallState

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.BitRate

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.Codec

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.Enable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.EntryID

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.PacketizationPeriod

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.Priority

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.List.{i}.SilenceSuppression

Silence_Supp_Enable_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.ReceiveBitRate

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.ReceiveCodec

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.ReceiveSilenceSuppression

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.TransmitBitRate

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.TransmitCodec

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.TransmitPacketizationPeriod

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.TransmitSilenceSuppression

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.X_CISCO_PreferredCodec

Preferred_Codec_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.X_CISCO_PreferredCodec2

Second_Preferred_Codec_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.X_CISCO_PreferredCodec3

Third_Preferred_Codec_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.X_CISCO_UsePrefCodecOnly

Use_Pref_Codec_Only_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Codec.X_CISCO_CodecNegotiation

Codec_Negotiation_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.DirectoryNumber

User_ID_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Enable

Line_Enable_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.PhyReferenceList

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.RingMuteStatus

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.RingVolumeStatus

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.{i}.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.{i}.FarEndIPAddress

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.{i}.FarEndUDPPort

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.{i}.LocalUDPPort

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.{i}.SessionDuration

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Session.{i}.SessionStartTime

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.AuthPassword

Password_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.AuthUserName

User_ID_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.SIPEventSubscribeNumberOfElements

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.URI

SIP_URI_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_AuthID

Auth_ID_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_DisplayName

Display_Name_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_UseDNSSRV

Use_DNS_SRV_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_UserEqualPhone

User_Equal_Phone_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_SetG729annexb

Set_G729_annexb_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_BlindAttnXferEnable

Blind_Attn-Xfer_Enable_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_FeatureKeySync

Feature_Key_Sync_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.X_CISCO_DNSSRVAutoPrefix

DNS_SRV_Auto_Prefix_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Status

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.VoiceProcessing.

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.VoiceProcessing.EchoCancellationEnable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.VoiceProcessing.EchoCancellationInUse

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.VoiceProcessing.EchoCancellationTail

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.X_CISCO_DialPlan

Dial_Plan_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.X_CISCO_DefaultRing

Default_Ring_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.MaxSessions

Call_Appearances_Per_Line

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Name

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.NumberOfLines

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Region

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.Reset

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.DSCPMark

RTP_TOS_DiffServ_Value_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.LocalPortMax

RTP_Port_Max

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.LocalPortMin

RTP_Port_Min

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.RTCP.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.RTCP.Enable

RTCP_Tx_Interval

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.RTCP.TxRepeatInterval

RTCP_Tx_Interval

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.SRTP.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.SRTP.Enable

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.SRTP.EncryptionKeySizes

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.SRTP.KeyingMethods

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.TelephoneEventPayloadType

AVT_Dynamic_Payload

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.X_CISCO_RTPPacketSize

RTP_Packet_Size

Device.Services.VoiceService.{i}.VoiceProfile.{i}.RTP.X_CISCO_RTPBeforeACK

RTP_Before_ACK

Device.Services.VoiceService.{i}.VoiceProfile.{i}.ServiceProviderInfo.

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.ServiceProviderInfo.ContactPhoneNumber

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.ServiceProviderInfo.EmailAddress

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.ServiceProviderInfo.Name

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.ServiceProviderInfo.URL

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SignalingProtocol

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.DSCPMark

SIP_TOS_DiffServ_Value_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.InviteExpires

INVITE_Expires

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.Organization

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.OutboundProxy

Outbound_Proxy_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.OutboundProxyPort

Outbound_Proxy_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.ProxyServer

Proxy_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.ProxyServerPort

Proxy_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.ProxyServerTransport

SIP_Transport_<1>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.RegisterExpires

Register_Expires_<i>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.RegisterRetryInterval

Reg_Retry_Intvl

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.RegistersMinExpires

Reg_Min_Expires

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.ReInviteExpires

ReINVITE_Expires

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.SIPEventSubscribeNumberOfElements

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.SIPResponseMapNumberOfElements

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerB

SIP_Timer_B

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerD

SIP_Timer_D

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerF

SIP_Timer_F

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerH

SIP_Timer_H

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerJ

SIP_Timer_J

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerT1

SIP_T1

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerT2

SIP_T2

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.TimerT4

SIP_T4

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.UserAgentDomain

N/A

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.UserAgentPort

SIP_Port_<1>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.UserAgentTransport

SIP_Transport_<1>_

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.X_CISCO_SubMinExpires

Sub_Min_Expires

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.X_CISCO_SubMaxExpires

Sub_Max_Expires

Device.Services.VoiceService.{i}.VoiceProfile.{i}.SIP.X_CISCO_SubRetryIntvl

Sub_Retry_Intvl

Device.Services.VoiceService.{i}.VoiceProfile.{i}.STUNEnable

STUN_Enable

Device.Services.VoiceService.{i}.VoiceProfileNumberOfEntries

N/A

Device.Services.VoiceService.{i}.X_CISCO_SIP.

Device.Services.VoiceService.{i}.X_CISCO_SIP.G711uCodecName

G711u_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.G711aCodecName

G711a_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.G729aCodecName

G729a_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.G729bCodecName

G729b_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.G722CodecName

G722_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.G7222CodecName

G722.2_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.iLBCCodecName

iLBC_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.OPUSCodecName

OPUS_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.AVTCodecName

AVT_Codec_Name

Device.Services.VoiceService.{i}.X_CISCO_SIP.G7222BEDynamicPayload

G722.2_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.G7222OADynamicPayload

G722.2_OA_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.iLBC20msDynamicPayload

iLBC_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.iLBC30msDynamicPayload

iLBC_30ms_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.OPUSDynamicPayload

OPUS_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.AVTDynamicPayload

AVT_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.AVT16kHzDynamicPayload

AVT_16kHz_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.AVT48kHzDynamicPayload

AVT_48kHz_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.INFOREQDynamicPayload

INFOREQ_Dynamic_Payload

Device.Services.VoiceService.{i}.X_CISCO_SIP.DisplayAnonymousFromHeader

Display_Anonymous_From_Header

Device.Services.VoiceService.{i}.X_CISCO_SIP.RedirectKeepAlive

Redirect_Keep_Alive

Device.Services.VoiceService.{i}.X_CISCO_Regional.

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.DialTone

Dial_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.OutsideDialTone

Outside_Dial_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.PromptTone

Prompt_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.BusyTone

Busy_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.ReorderTone

Reorder_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.OffHookWarningTone

Off_Hook_Warning_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.RingBackTone

Ring_Back_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.CallWaitingTone

Call_Waiting_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.ConfirmTone

Confirm_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.MWIDialTone

MWI_Dial_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.CfwdDialTone

Cfwd_Dial_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.HoldingTone

Holding_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.ConferenceTone

Conference_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.SecureCallIndicationTone

Secure_Call_Indication_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.PageTone

Page_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.AlertTone

Alert_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.MuteTone

Mute_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.UnmuteTone

Unmute_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.SystemBeep

System_Beep

Device.Services.VoiceService.{i}.X_CISCO_Regional.Tones.CallPickupTone

Call Pickup_Tone

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence1

Cadence_1

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence2

Cadence_2

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence3

Cadence_3

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence4

Cadence_4

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence5

Cadence_5

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence6

Cadence_6

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence7

Cadence_7

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence8

Cadence_8

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.Cadence9

Cadence_9

Device.Services.VoiceService.{i}.X_CISCO_Regional.Cadences.

Device.Services.VoiceService.{i}.X_CISCO_Regional.ControlTimer.ReorderDelay

Reorder_Delay

Device.Services.VoiceService.{i}.X_CISCO_Regional.ControlTimer.InterdigitLongTimer

Interdigit_Long_Timer

Device.Services.VoiceService.{i}.X_CISCO_Regional.ControlTimer.InterdigitShortTimer

Interdigit_Short_Timer

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.NumberOfUnits

Number_of_Units

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.ServerType

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.SubscribeRetryInterval

Subscribe_Retry_Interval

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.BXferOnSpeedDialEnable

Bxfer_On_Speed_Dial_Enable

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.AttendantConsoleLCDContrast

Attendant_Console_LCD_Brightness

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.BXferToStarcodeEnable

Bxfer_To_Starcode_Enable

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.Unit.

N/A

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.Unit.{i}.

N/A

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.Unit.{i}.Key.

N/A

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.Unit.{i}.Key.{i}.

N/A

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.Unit.{i}.Key.{i}.Config

Unit_<i>_Key_<i>_

Device.Services.VoiceService.{i}.X_CISCO_AttConsole.Unit.{i}.NumberOfKey

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LineKey.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LineKey.{i}.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LineKey.{i}.ExtendedFunction

Extended_Function_<i>_

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LineKey.{i}.Extension

Extension_<i>_

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LineKey.{i}.ShareCallApparence

Share_Call_Appearance_<i>_

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LineKey.{i}.ShortName

Short_Name_<i>_

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.NumberOfLineKey

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.StationName

Station_Name

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.GroupPagingScript

Group_Paging_Script

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.VoiceMailNumber

Voice_Mail_Number

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.BluetoothMode

Bluetooth_Mode

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Line

Line

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring1

Ring1

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring2

Ring2

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring3

Ring3

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring4

Ring4

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring5

Ring5

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring6

Ring6

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring7

Ring7

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring8

Ring8

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring9

Ring9

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring10

Ring10

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring11

Ring11

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.Ringtone.Ring12

Ring12

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.ConferenceServ

Coference_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.AttnTransferServ

Attn_Transfer_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.BlindTransferServ

Blind_Transfer_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.DNDServ

DND_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.BlockANCServ

Block_ANC_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.BlockCIDServ

Block_CID_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.SecureCallServ

Secure_Call_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.CfwdAllServ

Cfwd_All_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.CfwdBusyServ

Cfwd_Busy_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.CfwdNoAnsServ

Cfwd_No_Ans_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.PagingServ

Paging_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.CallParkServ

Call_Park_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.CallPickUpServ

Call_Pick_Up_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.ACDLoginServ

ACD_Login_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.GroupCallPickUpServ

Group_Call_Pick_Up_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.ServiceAnncServ

Service_Annc_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.CallRecordingServ

Call_Recording_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.SuppServices.ReversePhoneLookupServ

Reverse_Phone_Lookup_Serv

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.ProgrammableSoftkeyEnable

Programmable_Softkey_Enable

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.IdleKeyList

Idle_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.MissedCallKeyList

Missed_Call_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.OffHookKeyList

Off_Hook_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.DialingInputKeyList

Dialing_Input_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.ProgressingKeyList

Progressing_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.ConnectedKeyList

Connected_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.StartXferKeyList

Start-Xfer_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.StartConfKeyList

Start-Conf_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.ConferencingKeyList

Conferencing_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.ReleasingKeyList

Releasing_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.HoldKeyList

Hold_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.RingingKeyList

Ringing_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.SharedActiveKeyList

Shared_Active_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.SharedHeldKeyList

Shared_Held_Key_List

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK1

PSK_1

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK2

PSK_2

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK3

PSK_3

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK4

PSK_4

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK5

PSK_5

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK6

PSK_6

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK7

PSK_7

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK8

PSK_8

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK9

PSK_9

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK10

PSK_10

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK11

PSK_11

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK12

PSK_12

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK13

PSK_13

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK14

PSK_14

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK15

PSK_15

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.ProgramSoftkeys.PSK16

PSK_16

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.

N/A

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.LDAPDirEnable

LDAP_Dir_Enable

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.CorpDirName

LDAP_Corp_Dir_Name

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.Server

LDAP_Server

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.SearchBase

LDAP_Search_Base

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.ClientDN

LDAP_Client_DN

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.UserName

LDAP_User_Name

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.Password

LDAP_Password

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.AuthMethod

LDAP_Auth_Method

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.LastNameFilter

LDAP_Last_Name_Filter

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.FirstNameFilter

LDAP_First_Name_Filter

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.SearchItem3

LDAP_Search_Item_3

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.SearchItem3Filter

LDAP_Item_3_Filter

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.SearchItem4

LDAP_Search_Item_4

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.SearchItem4Filter

LDAP_Item_4_Filter

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.DisplayAttrs

LDAP_Display_Attrs

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.NumberMapping

LDAP_Number_Mapping

Device.Services.VoiceService.{i}.X_CISCO_PhoneSetting.LDAP.StartTLSEnable

LDAP_StartTLS_Enable

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.

N/A

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.RingerVolume

Ringer_Volume

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.SpeakerVolume

Speaker_Volume

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.HandsetVolume

Handset_Volume

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.HeadsetVolume

Headset_Volume

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.PhoneBackground

Phone_Background

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.PictureDownloadURL

Picture_Download URL

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.ElectronicHookSwitchControl

Ehook_Enable

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.ScreenSaverEnable

Screen_Saver_Enable

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.ScreenSaverType

Screen_Saver_Type

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.MissCallShortcut

Miss_Call_Shortcut

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.AlertToneOff

Alert_Tone_Off

Device.Services.VoiceService.{i}.X_CISCO_UserSetting.LogoURL

Logo_URL

Device.Services.VoiceService.{i}.X_CISCO_StarCode.

N/A

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateBlockAnonymousCall

Block_ANC_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateBlockCallerId

Block_CID_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateBlockCallerIdNextCall

Block_CID_Per_Call_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateCallForwardAll

Cfwd_All_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateCallForwardBusy

Cfwd_Busy_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateCallForwardNoAnswer

Cfwd_No_Ans_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateCallWaiting

CW_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateCallWaitingNextCall

CW_Per_Call_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateDoNotDisturb

DND_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateSecureCall

Secure_All_Call_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.ActivateSecureCallNextCall

Secure_One_Call_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.BlindTransfer

Blind_Transfer_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.CallPark

Call_Park_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.CallPickup

Call_Pickup_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.CallReturn

Call_Return_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.CallUnpark

Call_Unpark_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateBlockAnonymousCall

Block_ANC_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateBlockCallerId

Block_CID_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateBlockCallerIdNextCall

Block_CID_Per_Call_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateCallForwardAll

Cfwd_All_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateCallForwardBusy

Cfwd_Busy_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateCallForwardNoAnswer

Cfwd_No_Ans_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateCallWaiting

CW_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateCallWaitingNextCall

CW_Per_Call_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateDoNotDisturb

DND_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateSecureCal

Secure_No_Call_Act_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.DeactivateSecureCallNextCall

Secure_One_Call_Deact_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.GroupCallPickup

Group_Call_Pickup_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PagingCode

Paging_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodecG711a

Prefer_G711a_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodecG711u

Prefer_G711u_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodecG722

Prefer_G722_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodecG7222

Prefer_G722.2_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodecG729a

Prefer_G729a_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodeciLBC

Prefer_iLBC_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.PreferCodecOPUS

Prefer_OPUS_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodecG711a

Force_G711a_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodecG711u

Force_G711u_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodecG722

Force_G722_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodecG7222

Force_G722.2_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodecG729a

Force_G729a_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodeciLBC

Force_iLBC_Code

Device.Services.VoiceService.{i}.X_CISCO_StarCode.UseOnlyCodecOPUS

Force_OPUS_Code

N/A

N/A

*(1) We support such TR-069 configuration, but no corresponding parameter on Web/GUI

N/A

*(2) We support such TR-069 configuration, but can only be set to 'Yes"

N/A

*(3) i=0 G.711MuLaw i=1 G.711ALaw i=2 G.729a i=3 G.722 i=4 G.722.2 i=5 iLBC i=6 (88xx iSAC) (78xx OPUS) i=7 OPUS (88xx)

N/A

*(4) Only available on 8851/8861/8865

N/A

*(5) This parameter is for global setting, not per extension

N/A

*(6) This will leads to codec <i> on line <i> enable/disable, for codec <i>, please refer to *(4)

N/A

*(7) Only with sidecar. On mountlake it is named Attendant Console LCD Contrast

N/A

Device.

N/A

Device.DeviceSummary

N/A

Device.Services.

N/A

Device.Services.VoiceServiceNumberOfEntries

Device.DeviceInfo.

N/A

Device.DeviceInfo.Manufacturer

N/A

Device.DeviceInfo.ManufacturerOUI

N/A

Device.DeviceInfo.ModelName

N/A

Device.DeviceInfo.Description

N/A

Device.DeviceInfo.ProductClass

N/A

Device.DeviceInfo.SerialNumber

N/A

Device.DeviceInfo.HardwareVersion

N/A

Device.DeviceInfo.SoftwareVersion

N/A

Device.DeviceInfo.EnabledOptions

N/A

Device.DeviceInfo.AdditionalHardwareVersion

N/A

Device.DeviceInfo.AdditionalSoftwareVersion

N/A

Device.DeviceInfo.ProvisioningCode

N/A

Device.DeviceInfo.DeviceStatus

N/A

Device.DeviceInfo.UpTime

N/A

Device.ManagementServer.

N/A

Device.ManagementServer.URL

N/A

Device.ManagementServer.Username

N/A

Device.ManagementServer.Password

N/A

Device.ManagementServer.PeriodicInformEnable

N/A

Device.ManagementServer.PeriodicInformInterval

N/A

Device.ManagementServer.PeriodicInformTime

N/A

Device.ManagementServer.ParameterKey

N/A

Device.ManagementServer.ConnectionRequestURL

N/A

Device.ManagementServer.ConnectionRequestUsername

N/A

Device.ManagementServer.ConnectionRequestPassword

N/A

Device.GatewayInfo.

N/A

Device.GatewayInfo.ManufacturerOUI

N/A

Device.GatewayInfo.ProductClass

N/A

Device.GatewayInfo.SerialNumber

N/A

Device.Time.

N/A

Device.Time.NTPServer1

Primary_NTP_Server

Device.Time.NTPServer2

Secondary_NTP_Server

Device.Time.CurrentLocalTime

N/A

Device.Time.LocalTimeZone

Time_Zone

Device.Time.X_CISCO_TimeFormat

Time_Format

Device.Time.X_CISCO_DateFormat

Date_Format

Device.LAN.

N/A

Device.LAN.X_CISCO_IPMode

IP_Mode

Device.LAN.AddressingType

Connection_Type

Device.LAN.IPAddress

Static_IP

Device.LAN.SubnetMask

NetMask

Device.LAN.DefaultGateway

Gateway

Device.LAN.DNSServers

Primary_DNS

Device.LAN.MACAddress

N/A

Device.LAN.DHCPOptionNumberOfEntries

N/A

Device.LAN.DHCPOption.

N/A

Device.LAN.DHCPOption.{i}.

N/A

Device.LAN.DHCPOption.{i}.Request

DHCP_Option_To_Use

Device.LAN.DHCPOption.{i}.Tag

DHCP_Option_To_Use

Device.LAN.DHCPOption.{i}.Value

DHCP_Option_To_Use

Device.Ethernet.

N/A

Device.Ethernet.X_CISCO_CDP

Enable_CDP

Device.Ethernet.X_CISCO_LLDP

Enable_LLDP-MED

Device.Ethernet.X_CISCO_EnableVLAN

Enable_VLAN

Device.Ethernet.X_CISCO_VLANID

VLAN_ID

Device.X_CISCO_Language.

N/A

Device.X_CISCO_Language.DictionaryServerScript

Dictionary_Server_Script

Device.X_CISCO_Language.LanguageSelection

Language_Selection

Device.X_CISCO_Language.Locale

Locale

Device.X_CISCO_XmlService.

N/A

Device.X_CISCO_SecuritySettings.TLSCipherList

TLS_Cipher_List

Device.X_CISCO_XmlService.Password

XML_Password

Device.X_CISCO_XmlService.UserName

XML_User_Name

Device.X_CISCO_XmlService.XMLAppServiceName

XML_Application_Service_Name

Device.X_CISCO_XmlService.XMLAppServiceURL

XML_Application_Service_URL

Device.X_CISCO_XmlService.XMLDirServiceName

XML_Directory_Service_Name

Device.X_CISCO_XmlService.XMLDirServiceURL

XML_Directory_Service_URL

Device.X_CISCO_XmlService.CISCOXMLEXEEnable

CISCO_XML_EXE_Enable

Device.X_CISCO_XmlService.CISCOXMLEXEAuthMode

CISCO_XML_EXE_AUTH_MODE

Device.X_CISCO_RestrictedAccessDomains

Restricted_Access_Domains

Device.X_CISCO_EnableWebServer

Enable_Web_Server

Device.X_CISCO_WebProtocol

Enable_Protocol

Device.X_CISCO_EnableDirectActionUrl

Enable_Direct_Action_Url

Device.X_CISCO_SessionMaxTimeout

Session_Max_Timeout

Device.X_CISCO_SessionIdleTimeout

Session_Idle_Timeout

Device.X_CISCO_WebServerPort

Web_Server_Port

Device.X_CISCO_EnableWebAdminAccess

Enable_Web_Admin_Access

Device.X_CISCO_HostName

Host_Name

Device.X_CISCO_Domain

Domain

Device.X_CISCO_UpgradeErrorRetryDelay

Upgrade_Error_Retry_Delay

Device.X_CISCO_UpgradeRule

Upgrade_Rule

Device.X_CISCO_ProfileRule

Profile_Rule

Device.X_CISCO_UserConfigurableResync

User_Configurable_Resync

Device.X_CISCO_HTTPReportMethod

HTTP_Report_Method

Device.X_CISCO_CWMPV1dot2Support

CWMP_V1.2_Support

Introduction

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.

The following information is for FCC compliance of Class A devices: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio-frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in which case users will be required to correct the interference at their own expense.

The following information is for FCC compliance of Class B devices: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If the equipment causes interference to radio or television reception, which can be determined by turning the equipment off and on, users are encouraged to try to correct the interference by using one or more of the following measures:

- Reorient or relocate the receiving antenna.

- Increase the separation between the equipment and receiver.

- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

- Consult the dealer or an experienced radio/TV technician for help.

Modifications to this product not authorized by Cisco could void the FCC approval and negate your authority to operate the product.

The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCB’s public domain version of the UNIX operating system. All rights reserved. Copyright © 1981, Regents of the University of California.

NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED "AS IS" WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE.

IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.

Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.

All printed copies and duplicate soft copies of this document are considered uncontrolled. See the current online version for the latest version.

Cisco has more than 200 offices worldwide. Addresses and phone numbers are listed on the Cisco website at www.cisco.com/go/offices.

Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL: https://www.cisco.com/c/en/us/about/legal/trademarks.html. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (1721R)