Chapter 1. New and Changed Information

New Information for Firmware Release 10.4(1) SR1

New Information for Firmware Release 10.4(1)

Part I. About the Cisco IP Phone

Chapter 2. Technical Details

Overview of the Cisco IP Phone

Physical and Operating Environment Specifications

Cable Specifications

Network and Computer Port Pinouts

Network Port Connector

Computer Port Connector

Phone Power Requirements

Power Outage

Power Reduction

Power Negotiation Over LLDP

Network Protocols

VLAN Interaction

External Devices

USB Port Information

Chapter 3. Cisco IP Phone Hardware

Phone Overview

Cisco IP Phone 8811

Phone Connections

Cisco IP Phone 8841

Phone Connections

Cisco IP Phone 8851

Phone Connections

Cisco IP Phone 8861

Phone Connections

Buttons and Hardware

Navigation

Softkey, Line, and Feature Buttons

Terminology Differences

Part II. Cisco IP Phone Installation

Chapter 4. Cisco IP Phone Installation

Verify the Network Setup

Install the Cisco IP Phone

Set Up the Phone From the Setup Menus

Text and Menu Entry From the Phone

Set up Wireless LAN

Set Up Wireless LAN from Phone

Configure Network Settings

Set Up IPv4

Set Domain Name Field

Set Admin VLAN ID Field

Set SW Port Setup Field

Set PC Port Setup Field

Set Wireless Field

Set Wi-Fi sign in access

Set SSID Field

Set Security Mode Field

Set 802.11 Mode Field

Set DHCP Field

Set IP Address Field

Set Subnet Mask Field

Set Default Router Field

Set DNS Server Fields

Set Up Phone to Not Use DHCP

Verify Phone Startup

Configure the Voice Codecs

Set the Optional Network Servers

VLAN Settings

Cisco Discovery Protocol

LLDP-MED

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

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

LLDP-MED and IEEE 802.X

Configure VLAN Settings

SIP and NAT Configuration

SIP and the Cisco IP Phone

SIP Over TCP

SIP Proxy Redundancy

Dual Registration

Failover and Recovery Registration

RFC3261

SIP NOTIFY XML-Service

SIP Configuration

Configure the Basic SIP Parameters

Configure the SIP Timer Values

Configure the Response Status Code Handling

Configure the RTP Parameters

Configure the SDP Payload Types

Configure the SIP Settings for Extensions

Configure the SIP Proxy Server

Configure the Subscriber Information Parameters

Managing NAT Transversal with Phones

NAT Mapping with Session Border Controller

NAT Mapping with SIP-ALG Router

NAT Mapping with the Static IP Address

Configure NAT mapping with STUN

Dial Plan

Dial Plan Overview

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)

Edit the Dial Plan on the IP Phone

Reset the Control Timers

Regional Parameters and Supplementary Services

Regional Parameters

Set the Control Timer Values

Localize Your Cisco IP Phone

Time and Date Settings

Configure Daylight Saving Time

Select a Display Language on the Phone

Dictionary Server Script

Cisco IP Phone 8800 Series Documentation

Chapter 5. Third-Party Call Control Setup

Determine the Phone MAC Address

Network Configuration

Provisioning

Web-Based Configuration Utility

Access the Web-Based Configuration Utility

Determine the IP Address of the Phone

Allow Web Access to the Cisco IP Phone

Web Administration Tabs

Administrator and User Accounts

Enable User Access to the Phone Interface Menus

Access Administrative Options by Login

Access Administrative Options by IP Address

Part III. Hardware and Accessory Installation

Chapter 6. Cisco IP Phone Accessories

Cisco IP Phone Accessories Overview

Connect the Footstand

Secure the Phone with a Cable Lock

External Speakers and Microphone

Headsets

Audio Quality

Analog Headsets

Enable Wideband on Analog Headsets

Wireless Headsets

Bluetooth Wireless Headsets

Chapter 7. Cisco IP Phone Key Expansion Module

Cisco IP Phone Key Expansion Module Setup Overview

Key Expansion Module Power Information

Connect a Single Key Expansion Module to a Cisco IP Phone

Connect Two or Three Key Expansion Modules to a Cisco IP Phone

Configure the Key Expansion Module in the Configuration Utility

Access Key Expansion Module Setup

Reset the Key Expansion Module to Factory Defaults

Troubleshoot the Key Expansion Module

Chapter 8. Wall Mounts

Non-Lockable Wall Mount Components

Install the Non-Lockable Wall Mount Kit for Phone

Remove the Phone from the Non-Lockable Wall Mount

Non-Lockable Wall Mount Components for Phone with Key Expansion Module

Install Non-Lockable Wall Mount Kit for Phone with Key Expansion Module

Remove the Phone and Key Expansion Module from the Non-Lockable Wall Mount

Adjust the Handset Rest

Part IV. Cisco IP Phone Administration

Chapter 9. Cisco IP Phone Security

Security Features

Domain and Internet Setting

Configure Restricted Access Domains

Configure the Internet Connection Type

DHCP Option Support

Configure the Challenge for the SIP INVITE Messages

Transport Layer Security

Configure SIP Over TLS Signaling Encryption

Documentation, Support, and Security Guidelines

Cisco Product Security Overview

Chapter 10. Cisco IP Phone Customization

Phone Information and Display Settings

Configure the Phone Name

Customize the Startup Screen

Download and Configure the Wallpaper Pictures

Configure the Screen Saver

Configure the LCD Brightness

Configure the Back Light Setting

Configure the Number of Call Appearances Per Line

Call Features Configuration

Enable Call Transfer

Enable Call Forwarding

Enable Conferencing

Enable Do Not Disturb

Configure Voice Mail

Configure Voice Mail for each Extension

Configure the Message Waiting Indicator

Assign a Ring Tone to an Extension

Configure the Audio Settings

User Access Control

Phone Web Server

Configure the Web Server from the Phone Screen Interface

Enable Access to Phone Web Interface

Chapter 11. Phone Features and Setup

Phone Features and Setup Overview

Cisco IP Phone User Support

Telephony Features for Cisco IP Phone

Feature Buttons and Softkeys

Configure a Speed Dial on a Line Key

Configure a Speed Dial on a Key Expansion Module

Busy Lamp Field Configuration on a Monitoring Phone

Configure the Busy Lamp Field for Multiple Users with the Configuration Utility

Configure the Busy Lamp Field in the Phone Configuration File

Configure the Busy Lamp Field for a Single Phone with the Configuration Utility

Configure the Busy Lamp Field on a Key Expansion Module

Configure Alphanumeric Dialing

Configure a Paging Group (Multicast Paging)

Call Park

Add Call Park to a Programmable Line Key

Configure Call Park from the Attendant Console

Configuring Programmable Softkeys

Customize a Programmable Softkey

Configure Speed Dial on a Programmable Softkey

Programmable Softkeys

Configure Provisioning Authority

Configure Provisioning Authority in the Phone Configuration File

Download Problem Reporting Tool Logs

Chapter 12. Corporate and Personal Directory Setup

Personal Directory Setup

LDAP Configuration

Prepare the LDAP Corporate Directory Search

Configure the BroadSoft Settings

Configure the XML Directory Service

Part V. Cisco IP Phone Troubleshooting

Chapter 13. Monitoring Phone Systems

Monitoring Phone Systems Overview

Cisco IP Phone Status

Display the Phone Information Window

Display Status Menu

Display Status Messages Window

Display Network Statistics Window

Display Wireless Statistics Screen

Display Call Statistics Window

Display Current Access Point Window

Cisco IP Phone Web Page

Info

System Information

Reboot History

Product Information

Phone Status

Ext Status

Line 1 Status/Call 2 Status

Wallpaper Picture Download URL

Downloaded Locale Package

Firmware Upgrade Status

Provisioning Status

Custom CA Status

Ethernet Information

Network and Access Port Information

DHCP

Console Logs

Core Dumps

Browser Info

System Configuration

Internet Connection Type

Static IP Settings

Optional Network Configuration

VLAN Settings

Inventory Settings

SIP Parameters

SIP Timer Values

Response Status Code Handling

RTP Parameters

SDP Payload Types

NAT Support Parameters

Configuration Profile

Firmware Upgrade

CA Settings

General Purpose Parameters

Control Timer Values (sec)

Vertical Service Activation Codes

Time

Localization

QoS Settings

General

Line Key

Miscellaneous Line Key Settings

Supplementary Services

BroadSoft Settings

LDAP Corporate Directory Search

XML Service

Extension Mobility

Programmable Softkeys

Hold Reminder Timer

Speed Dial

Supplementary Services

Audio

LCD

Extension

General

Share Line Appearance

Call Forward

NAT Settings

SIP Settings

Call Feature Settings

Supplementary Services

Proxy and Registration

Subscriber Information

Audio Configuration

Dial Plan

Attendant Console

General

Call Parking

Call History

Chapter 14. Troubleshooting

General Troubleshooting Information

Startup Problems

Cisco IP Phone Does Not Go Through the Normal Startup Process

Phone Displays Error Messages

Phone Cannot Connect Using DNS

Configuration File Corruption

Cisco IP Phone Cannot Obtain IP Address

Phone Reset Problems

Phone Resets Due to Intermittent Network Outages

Phone Resets Due to DHCP Setting Errors

Phone Resets Due to Incorrect Static IP Address

Phone Resets During Heavy Network Usage

Phone Does Not Power Up

Phone Cannot Connect to LAN

Audio Problems

No Speech Path

Choppy Speech

General Telephone Call Problems

Phone Call Cannot Be Established

Phone Does Not Recognize DTMF Digits or Digits Are Delayed

Troubleshooting Procedures

Check DHCP Settings

Verify DNS Settings

Additional Troubleshooting Information

Chapter 15. Maintenance

Basic Reset

Perform Factory Reset Phone Keypad

Perform Factory Reset from Phone Menu

Voice Quality Monitoring

Voice Quality Troubleshooting Tips

Voice Quality Reporting

Supported Scenarios for Voice Quality Reporting

Mean Opinion Scores and Codecs

Configure Voice Quality Reporting

Cisco IP Phone Cleaning

View Phone Information

Reboot Reasons

Reboot History on the Phone Web User Interface

Reboot History on the Cisco IP Phone Screen

Reboot History in the Status Dump File

Phone Behavior During Times of Network Congestion

Copyright Page

Chapter 1. New and Changed Information

- New Information for Firmware Release 10.4(1) SR1

- New Information for Firmware Release 10.4(1)

New Information for Firmware Release 10.4(1) SR1

All new features have been added to Telephony Features for Cisco IP Phone.

Revision

Updated Section

Added Configure a Paging Group

Configure a Paging Group (Multicast Paging)

Added Configure Alphanumeric Dialing

Configure Alphanumeric Dialing

New Information for Firmware Release 10.4(1)

All new features have been added to Telephony Features for Cisco IP Phone.

Revision

Updated Section

Added Key Expansion Module

Cisco IP Phone Key Expansion Module Setup Overview

Key Expansion Module Power Information

Configure the Key Expansion Module in the Configuration Utility

Configure a Speed Dial on a Key Expansion Module

Added Speed-Dial on a line key

Configure a Speed Dial on a Line Key

Added Busy Lamp Field configuration on a monitoring phone

Busy Lamp Field Configuration on a Monitoring Phone

Added Provisioning Authority

Configure Provisioning Authority

Added Call Park and Pickup on a line key

Call Park

Added Customization of Programmable Softkey

Configuring Programmable Softkeys

Added Remote Customization

Telephony Features for Cisco IP Phone

Added Voice Quality Reporting using SIP Publish

Voice Quality Reporting

Added Problem Report Tool

Download Problem Reporting Tool Logs

Part I. About the Cisco IP Phone

- Technical Details

- Cisco IP Phone Hardware

Chapter 2. Technical Details

- Overview of the Cisco IP Phone

- Physical and Operating Environment Specifications

- Cable Specifications

- Phone Power Requirements

- Network Protocols

- VLAN Interaction

- External Devices

- USB Port Information

Overview of the Cisco IP Phone

The Cisco IP Phone 8800 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 8800 Series Multiplatform Phones is targeted for solutions that are centered on Third-Party SIP-based IP PBX.

Note

In this document, the terms Cisco IP Phone or phone means Cisco IP Phone 8800 Series Multiplatform Phones.

Physical and Operating Environment Specifications

The following table shows the physical and operating environment specifications for the Cisco IP Phone 8800 Series.

Physical and Operating Specifications

Specification

Value or range

Operating temperature

32° to 104°F (0° to 40°C)

Operating relative humidity

Operating: 10% to 90% (non-condensing)

Non-operating: 10% to 95% (non-condensing)

Storage temperature

14° to 140°F (–10° to 60°C)

Height

9.02 in. (229.1 mm)

Width

10.13 in. (257.34 mm)

Depth

1.57 in. (40 mm)

Weight

2.62 lb (1.19 kg)

Power

100-240 VAC, 50-60 Hz, 0.5 A when using the AC adapter

48 VDC, 0.2 A when using the in-line power over the network cable

Cables

Category 3/5/5e/6 for 10-Mbps cables with 4 pairs

Category 5/5e/6 for 100-Mbps cables with 4 pairs

Category 5e/6 for 1000-Mbps cables with 4 pairs

Note:
Cables have 4 pairs of wires for a total of 8 conductors.

Distance requirements

As supported by the Ethernet Specification, the maximum cable length between each Cisco IP Phone and the switch is assumed to be 330 feet (100 meters).

Cable Specifications

The following information lists the cable specifications:

- RJ-9 jack (4-conductor) for handset and headset connection

- RJ-45 jack for the LAN 10/100/1000BaseT connection (10/100/1000 Network port on the phone)

- RJ-45 jack for a second 10/100/1000BaseT compliant connection (10/100/1000 Computer port on the phone)

- 3.5 mm jack for speaker connection

- 48-volt power connector

- Network and Computer Port Pinouts

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 on the Cisco IP Phone.

- The computer (access) port is the 10/100/1000 PC port on the Cisco IP Phone.

- 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.

Phone Power Requirements

The Cisco IP Phone can be powered with external power or with Power over Ethernet (PoE). A separate power supply provides external power. The switch can provide PoE through the phone Ethernet cable.

Note

When you install a phone that is powered with external power, connect the power supply before you connect the Ethernet cable to the phone. When you remove a phone that is powered with external power, disconnect the Ethernet cable from the phone before you disconnect the power supply.

The following table provides guidelines for Cisco IP Phone power.

Guidelines for Cisco IP Phone Power

Power type

Guidelines

External power: Provided through the CP-PWR-CUBE-4= external power supply

The Cisco IP Phone uses the CP-PWR-CUBE-4 power supply.

External power—Provided through the Cisco IP Phone Power Injector.

The Cisco IP Phone Power Injector may be used with any Cisco IP Phone. Functioning as a midspan device, the injector delivers inline power to the attached phone. The Cisco IP Phone Power Injector connects between a switch port and the IP Phone. It supports a maximum cable length of 100m between the unpowered switch and the IP phone.

PoE power—Provided by a switch through the Ethernet cable attached to the phone.

Cisco IP Phone supports IEEE 802.3af for external add-on devices.

Cisco IP Phones 8851 and 8861 support IEEE 802.3af for smartphone charging. Cisco IP Phone 8861 supports IEEE 802.3at for tablet charging.

To ensure uninterruptible operation of the phone, make sure that the switch has a backup power supply.

Make sure that the CatOS or IOS version that runs on your switch supports your intended phone deployment. See the documentation for your switch for operating system version information.

Support for 802.3at: The Cisco IP Phone can draw more power than IEEE 802.3af, as long as there is 802.3at switch support.

For information about Cisco IP Phone 8800 Key Expansion Module power requirements, see Key Expansion Module Power Information.

The documents in the following table provide more information on the following topics:

- Cisco switches that work with Cisco IP Phones

- Cisco IOS releases that support bidirectional power negotiation

- Other requirements and restrictions about power

Document topics

URL

Cisco IP Phone Power Injector

Click Here

PoE Solutions

Click Here

Cisco Catalyst Switches

Click Here

Integrated Service Routers

Click Here

Cisco IOS Software

Click Here

- Power Outage

- Power Reduction

- Power Negotiation Over LLDP

Power Outage

Your access to emergency service through the phone requires that the phone receive power. If a power interruption occurs, service or emergency calling service dialing does not function until power is restored. If a power failure or disruption occurs, you may need to reset or reconfigure the equipment before you can use service or emergency calling service dialing.

Power Reduction

You can reduce the amount of energy that the Cisco IP Phone consumes by using Power Save mode.

Power Save

In Power Save mode, the backlight on the screen is not lit when the phone is not in use. The phone remains in Power Save mode until the user lifts the handset or presses any button. Set up each phone to enable or disable Power Save settings.

Power Negotiation Over LLDP

The phone and the switch negotiate the power that the phone consumes. Cisco IP Phone operates at multiple power settings, which lowers power consumption when less power is available.

After a phone reboots, the switch locks to one protocol (CDP or LLDP) for power negotiation. The switch locks to the first protocol (containing a power Threshold Limit Value [TLV]) that the phone transmits. If the system administrator disables that protocol on the phone, the phone cannot power up any accessories because the switch does not respond to power requests in the other protocol.

Cisco recommends that Power Negotiation always be enabled (default) when connecting to a switch that supports power negotiation.

If Power Negotiation is disabled, the switch may disconnect power to the phone. If the switch does not support power negotiation, disable the Power Negotiation feature before you power up accessories over PoE. When the Power Negotiation feature is disabled, the phone can power the accessories up to the maximum that the IEEE 802.3af-2003 standard allows.

Note

When CDP and Power Negotiation are disabled, the phone can power the accessories up to 15.4W.

Network Protocols

Cisco IP Phone 8800 Series support several industry-standard and Cisco network protocols required for voice communication. The following table provides an overview of the network protocols that the phones support.

Supported Network Protocols on the Cisco IP Phone 8800 Series

Network protocol

Purpose

Usage notes

Bluetooth

Bluetooth is a wireless personal area network (WPAN) protocol that specifies how devices communicate over short distances.

Cisco IP Phones 8851 and 8861 support Bluetooth 4.0.

Cisco IP Phone 8811, 8841, and 8851NR do not support Bluetooth.

Bootstrap Protocol (BootP)

BootP enables a network device, such as the Cisco IP Phone, to discover certain startup information, such as the IP address.

—

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.

The Cisco IP Phones use 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.

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 the phone to 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.

Note:
If you cannot use option 150, you may try using DHCP option 66, 159, or 160.

Hypertext Transfer Protocol (HTTP)

HTTP is the standard way of transferring information and moving documents across the Internet and the web.

Cisco IP 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.

IEEE 802.11n/802.11ac

The IEEE 802.11 standard specifies how devices communication over a wireless local area network (WLAN).

802.11n operates at the 2.4 GHz and 5 GHz band and 802.11ac operates at the 5 GHz band.

The 802.11 interface is a deployment option for cases when Ethernet cabling is unavailable or undesirable.

Only Cisco IP Phone 8861 and 8865 support WLAN.

Internet Protocol (IP)

IP is a messaging protocol that addresses and sends packets across the network.

To communicate using IP, network devices must have an assigned IP address, subnet, and gateway.

IP addresses, subnets, and gateway identifications are automatically assigned if you are using the Cisco IP Phone 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 for voice products.

The Cisco IP Phone 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:

Click Here

Real-Time Transport Protocol (RTP)

RTP is a standard protocol for transporting real-time data, such as interactive voice, 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 Third-Party Call Control System or 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 addresses the functions of signaling and session management within a packet telephony network. Signaling allows transportation of call information across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.

Cisco IP Phones support the SIP protocol when the phones are operating in IPv4 address mode.

Transmission Control Protocol (TCP)

TCP is a connection-oriented transport protocol.

Cisco IP Phones use TCP to connect to Third-Party Call Control system and to access XML services.

Transport Layer Security (TLS)

TLS is a standard protocol for securing and authenticating communications.

Upon security implementation, Cisco IP Phones use the TLS protocol when securely registering with Third-Party Call Control system.

Trivial File Transfer Protocol (TFTP)

TFTP allows you to transfer files over the network.

On the Cisco IP Phone, TFTP enables you to obtain a configuration file specific to the phone type.

TFTP requires a TFTP server in your network that the DHCP server can automatically identify.

User Datagram Protocol (UDP)

UDP is a connectionless messaging protocol for delivery of data packets.

UDP is used only for RTP streams. SIP signaling on the phones do not support UDP.

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:

Click Here

External Devices

We recommend that you use good-quality external devices that are shielded against unwanted radio frequency (RF) and audio frequency (AF) signals. External devices include headsets, cables, and connectors.

Depending on the quality of these devices and their proximity to other devices, such as mobile phones or two-way radios, some audio noise may still occur. In these cases, we recommend that you take one or more of these actions:

- Move the external device away from the source of the RF or AF signals.

- Route the external device cables away from the source of the RF or AF signals.

- Use shielded cables for the external device, or use cables with a better shield and connector.

- Shorten the length of the external device cable.

- Apply ferrites or other such devices on the cables for the external device.

Cisco cannot guarantee the performance of external devices, cables, and connectors.

Caution

In European Union countries, use only external speakers, microphones, and headsets that are fully compliant with the EMC Directive [89/336/EC].

USB Port Information

Note

- Unpowered hubs are not supported, and powered hubs with more than four ports are not supported.

- USB headsets that connect to the phone through a USB hub are not supported.

Each Cisco IP Phone 8800 Key Expansion Module (KEM) connected to the phone counts as a USB device. If three KEMs are connected to the phone, these count as three USB devices.

Chapter 3. Cisco IP Phone Hardware

- Phone Overview

- Cisco IP Phone 8811

- Cisco IP Phone 8841

- Cisco IP Phone 8851

- Cisco IP Phone 8861

- Buttons and Hardware

- Terminology Differences

Phone Overview

The Cisco IP Phones 8811, 8841, 8851, and 8861 provide voice communication over an Internet Protocol (IP) network. The Cisco IP Phone functions much like a digital business phone, allowing you to place and receive phone calls and to access features such as mute, hold, transfer, speed dial, call forward, and more. In addition, because the phone connects to your data network, it offers enhanced IP telephony features, including access to network information and services, and customizable features and services.

The Cisco IP Phone 8811 has a grayscale LCD screen.

The Cisco IP Phones 8841, 8851, and 8861 has a 24-bit color LCD screen.

The Cisco IP Phones have the following features:

- Programmable feature buttons that support up to 5 lines or that can be programmed for other features

- Gigabit ethernet connectivity

- Bluetooth support for wireless headsets(Cisco IP Phone 8851 and 8861 only)

- Support for an external microphone and speakers (Cisco IP Phone 8861 only)

- Network connectivity by Wi-Fi (Cisco IP Phone 8861 only)

- USB ports:

- one USB port for Cisco IP Phone 8851

- two USB ports for Cisco IP Phone 8861

A Cisco IP Phone, like other network devices, must be configured and managed. These phones encode and decode the following codes:

- G.711 a-law

- G.711 mu-law

- G.722

- G.722.2/AMR-WB

- G.729a/G.729ab

- iLBC

Caution

Using a cell, mobile, or GSM phone, or two-way radio in close proximity to a Cisco IP Phone might cause interference. For more information, see the manufacturer’s documentation of the interfering device.

Cisco IP Phones provide traditional telephony functionality, such as call forwarding and transferring, redialing, speed dialing, conference calling, and voice messaging system access. Cisco IP Phones also provide a variety of other features.

As with other network devices, you must configure Cisco IP Phones to prepare them to access Third-Party Call Control system and the rest of the IP network. By using DHCP, you have fewer settings to configure on a phone. If your network requires it, however, you can manually configure information such as: an IP address, TFTP server, and subnet information.

Cisco IP Phones can interact with other services and devices on your IP network to provide enhanced functionality. For example, you can integrate Third-Party Call Control system with the corporate Lightweight Directory Access Protocol 3 (LDAP3) standard directory to enable users to search for coworker contact information directly from their IP phones. You can also use XML to enable users to access information such as weather, stocks, quote of the day, and other web-based information.

Finally, because the Cisco IP Phone is a network device, you can obtain detailed status information from it directly. This information can assist you with troubleshooting any problems users might encounter when using their IP phones. You can also obtain statistics about a current call or firmware versions on the phone.

To function in the IP telephony network, the Cisco IP Phone must connect to a network device, such as a Cisco Catalyst switch. You must also register the Cisco IP Phone with a Third-Party Call Control system before sending and receiving calls.

Cisco IP Phone 8811

- Phone Connections

Phone Connections

Connect your phone to your organization's IP telephony network as shown in the following diagram.

1

DC adapter port (DC48V).

5

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

2

AC-to-DC power supply (optional).

6

Auxiliary port.

3

AC power wall plug (optional).

7

Handset connection.

4

Network port (10/100/1000 SW) connection. IEEE 802.3at power enabled.

8

Analog headset connection (optional).

Note

The Cisco IP Phone 8811 does not support the Cisco IP Phone 8800 Key Expansion Module.

Cisco IP Phone 8841

- Phone Connections

Phone Connections

Connect your phone to the corporate IP telephony network as shown in the following diagram.

1

DC adaptor port (DC48V).

5

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

2

AC-to-DC power supply (optional).

6

Auxiliary port.

3

AC power wall plug (optional).

7

Handset connection.

4

Network port (10/100/1000 SW) connection. IEEE 802.3at power enabled.

8

Analog headset connection (optional).

Note

The Cisco IP Phone 8841 does not support the Cisco IP Phone 8800 Key Expansion Module.

Cisco IP Phone 8851

- Phone Connections

Phone Connections

Connect your phone to the corporate IP telephony network as shown in the following diagram.

1

DC adaptor port (DC48V).

6

Auxiliary port.

2

AC-to-DC power supply (optional).

7

Handset connection.

3

AC power wall plug (optional).

8

Analog headset connection (optional).

4

Network port (10/100/1000 SW) connection. IEEE 802.3at power enabled.

9

USB port

5

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

Note

Each USB port supports the connection of up to five supported and nonsupported devices. Each device connected to the phone is included in the maximum device count. For example, your phone can support five USB devices (such as two Cisco IP Phone 8800 Key Expansion Modules, one headset, one hub, and one other standard USB device) on the side port. Many third-party USB products count as multiple USB devices, for example, a device containing USB hub and headset can count as two USB devices. For more information, see the USB device documentation.

Cisco IP Phone 8861

- Phone Connections

Phone Connections

Connect your phone to the corporate IP telephony network as shown in the following diagram.

1

DC adaptor port (DC48V).

7

Handset connection.

2

AC-to-DC power supply (optional).

8

Analog headset connection (optional).

3

AC power wall plug (optional).

9

USB port

4

Network port (10/100/1000 SW) connection. IEEE 802.3at power enabled.

10

Audio In/Out ports

5

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

11

USB port

6

Auxiliary port.

Note

Each USB port supports the connection of up to five supported and nonsupported devices. Each device connected to the phone is included in the maximum device count. For example, your phone can support five USB devices (such as three Cisco IP Phone 8800 Key Expansion Modules, one hub, and one other standard USB device) on the side port and five additional standard USB devices on the back port. Many third-party USB products count as multiple USB devices, for example, a device containing USB hub and headset can count as two USB devices. For more information, see the USB device documentation.

Buttons and Hardware

The Cisco IP Phones 8811, 8841, 8851, and 8861 deliver easy-to-use, highly secure voice communications.

Cisco IP Phone 8800 Buttons and Hardware

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 to functions and services.

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

4

Back, Navigation cluster, and Release

Back Return to the previous screen or menu.

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

Release End a connected call or session.

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 is lit.

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).

- Navigation

- Softkey, Line, and Feature Buttons

Navigation

You use the outer ring of the Navigation cluster to scroll through menus. You use the inner Select button of the Navigation cluster to select menu items.

Each item in a menu has an index number. You can enter the index number with the keypad to select the menu item.

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, located on either side of the screen, give you access to phone features and phone lines.

- Feature buttons—Used to view calls on a line or access features such as Speed Dial.

- Line buttons—Used to answer a call or resume a held call. When not used for an active call, used to initiate phone functions, such as the missed calls display.

Feature and line buttons illuminate to indicate status:

- Green—Line is idle.

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

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

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

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

Terminology Differences

The following table highlights some of the differences in terminology found in the Cisco IP Phone 8800 Series Multiplatform Phones User Guide and Cisco IP Phone 8800 Series Multiplatform Phones Administration Guide.

Terminology Differences

User Guide

Administration Guide

Line Status

Busy Lamp Field (BLF)

Message Indicators

Message Waiting Indicator (MWI) or Message Waiting Lamp

Programmable Feature Button

Programmable Button or Programmable Line Key (PLK)

Simplified New Call Window

Simplified New Call Bubble

Voicemail System

Voice Messaging System

Part II. Cisco IP Phone Installation

- Cisco IP Phone Installation

- Third-Party Call Control Setup

Chapter 4. Cisco IP Phone Installation

- Verify the Network Setup

- Install the Cisco IP Phone

- Set Up the Phone From the Setup Menus

- Set up Wireless LAN

- Configure Network Settings

- Verify Phone Startup

- Configure the Voice Codecs

- Set the Optional Network Servers

- VLAN Settings

- SIP and NAT Configuration

- Dial Plan

- Regional Parameters and Supplementary Services

- Cisco IP Phone 8800 Series Documentation

Verify the Network Setup

Upon deployment of a new IP telephony system, system administrators and network administrators must complete several initial configuration tasks to prepare the network for IP telephony service.

For the phone to operate successfully as an endpoint in your network, your network must meet specific requirements.

Note

The phone displays the date and time from Third-Party Call Control. The time displayed on the phone can differ from the Third-Party Call Control time by up to 10 seconds.

Procedure

Step 1

Configure a VoIP Network to meet the following requirements:

- VoIP is configured on your Cisco routers and gateways.

- Third-Party Call Control is installed in your network and is configured to handle call processing.

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 Third-Party Call Control system. To finish installing the phone, configure the network settings on the phone depending on whether you enable or disable DHCP service.

If you used autoregistration, you need to update the specific configuration information for the phone such as associating the phone with a user, changing the button table, or directory number.

Note

Before using external devices, read External Devices.

Procedure

Step 1

Choose the power source for the phone:

- Power over Ethernet (PoE)

- External power supply

For more information, see Phone Power Requirements.

Step 2

Connect the handset to the handset port.
The wideband-capable handset is designed especially for use with a Cisco IP 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 a wireless headset. You can add a wireless headset later if you do not want to connect one now. For more information, see your wireless headset documentation.

Step 5

Connect a straight-through Ethernet cable from the switch to the network port labeled 10/100/1000 SW on the Cisco IP Phone. Each Cisco IP 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 100 Mbps connections; and Category 5e or 6 for 1000 Mbps connections. For more information, see Network and Computer Port Pinouts.

Step 6

Connect a straight-through Ethernet cable from another network device, such as a desktop computer, to the computer port on the Cisco IP 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 100 Mbps connections; and Category 5e or 6 for 1000 Mbps connections. For more information, see Network and Computer Port Pinouts for guidelines.

Step 7

If the phone is on a desk, adjust the footstand. For more information, see Connect the Footstand. With a wall-mounted phone, you might need to adjust the handset rest to ensure that the receiver cannot slip out of the cradle.

Step 8

Monitor the phone startup process. This step verifies that the phone is configured properly.

Step 9

If you are configuring the network settings on the phone, you can set up an IP address for the phone by either using DHCP or manually entering an IP address.
See Set Up the Phone From the Setup Menus.

Step 10

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 11

Make calls with the Cisco IP Phone to verify that the phone and features work correctly.

Step 12

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 Cisco IP Phones.

Set Up the Phone From the Setup Menus

The phone includes many configurable network settings that you may need to modify before the phone is functional for your users. You can access these settings, and change some of them, through menus on the phone.

The phone includes the following setup menus:

- Network Setup: Provides options for viewing and configuring a variety of network settings.

- IPv4 Setup: This submenu provides additional network options.

Note

You can control whether a phone has access to the Settings menu or to options on this menu by modifying the value in the Phone-UI-User-Mode field in the Phone Configuration Utility page. Also, you must modify the attribute of ua in the Resync file of the phone to control the access. For example, when Phone-UI_User_mode is set to Yes and in the resync file the attribute for Speed_Dial_2 are:

- Speed_Dial_2 ua="rw", we can read and write on web of user model and lcd.

- Speed_Dial_2 ua="ro", we can only read on web of user model and lcd.

- Speed_Dial_2 ua="na", there will not be display on lcd and web of user model.

The Phone-UI-User-Mode field accepts these values:

- Yes: Allows access to the Settings menu. It also allows access to the Phone Configuration Utility page for user-mode.

- No: Prevents access to the Settings menu. It also restricts access to the Phone Configuration Utility page for user-mode.

If you cannot access an option on the Admin Settings menu, check the Phone-UI-User-Mode field.

You can configure settings that are display-only on the phone in Third-Party Call Control system.

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings.

Step 3

Enter password if required, then click Sign-In.

Step 4

Select Network Setup.
Note:
For information about the Reset Settings menu, see Maintenance.

Step 5

Perform one of these actions to display the desired menu:

- Use the navigation arrows to select the desired menu and then press Edit.

- Use the keypad on the phone to enter the number that corresponds to the menu.

Step 6

To display a submenu, repeat step 5.

Step 7

To exit a menu, press Back.

- Text and Menu Entry From 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, then 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 Revert before pressing Apply 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.

Related References

Basic Reset

Set up Wireless LAN

Only the Cisco IP Phone 8861 supports Wireless LAN.

Ensure that the Wi-Fi coverage in the location where the wireless LAN is deployed is suitable for transmitting voice packets.

If you have enabled the Wi-Fi connectivity for voice and you're using EAP-FAST or PEAP security mode, authenticate the Wi-Fi network with the WLAN Sign in application. WEP, PSK, and open security modes authenticate on the Wi-Fi network.

A fast-secure roaming method is recommended for Wi-Fi users.

For complete configuration information, see the Cisco IP Phone 8800 Wireless LAN Deployment Guide at this location:

Click Here

The Cisco IP Phone 8800 Wireless LAN Deployment Guide includes the following configuration information:

- Wireless network configuration

- Wireless network configuration in Cisco Unified Communications Manager Administration

- Wireless network configuration on the Cisco IP Phone

Before You Begin

Make sure that Wi-Fi is enabled on the phone, and that the phone is disabled from an Ethernet connection.

Procedure

Step 1

To enable the application, press Applications .

Step 2

Go to Admin settings > Network setup > Wi-Fi Client setup > Wi-Fi Sign in access

Step 3

Enable the Wireless network.

- Set Up Wireless LAN from Phone

Set Up Wireless LAN from Phone

Before the Cisco IP Phone can connect to the WLAN, you must configure the network profile for the phone with the appropriate WLAN settings. You can use the Network setup menu on the phone to access the Wi-Fi client setup submenu and set up the WLAN configuration.

Before You Begin

Configure the wireless LAN from Cisco Unified Communications Manager.

Procedure

Step 1

Press Applications.

Step 2

Select Admin settings > Network setup > Wi-Fi client setup.

Step 3

Set up the wireless configuration as described in the following table.

WiFi Client Setup Menu Options

Option

Description

To change

Wireless

Turns the wireless radio on the Cisco IP Phone on or off.

See Set Wireless Field.

Wi-Fi sign in access

Enables the display of the Wireless Sign in Access window in the main Applications menu.

See Set Wi-Fi sign in access.

IPv4 setup

In the IPv4 Setup configuration submenu, you can do the following:

- Enable or disable the phone to use the IP address that the DHCP server assigns.

- Manually set the IP Address, Subnet Mask, Default Routers, and DNS Server.

For more information about the IPv4 address fields, see Set Up IPv4.

Scroll to IPv4 Setup and select .

MAC address

Unique Media Access Control (MAC) address of the phone.

Display only. Cannot configure.

Domain name

Name of the Domain Name System (DNS) domain in which the phone resides.

See Set Domain Name Field .

SSID

Specifies the Service Set Identifier, a unique identifier for accessing wireless access points.

See Set SSID Field.

You can modify SSID field only if the phone is configured without wireless LAN profile or the profile with allowed profile.

Security mode

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

See Set Security Mode Field.

802.11 mode

Specifies the wireless signal standard that is used in the WLAN.

See Set 802.11 Mode Field .

Step 4

Press Save to make changes or press Revert to discard the connection.

Configure Network Settings

If you are not using DHCP in your network, you must configure these network settings on the Cisco IP Phone after you install the phone on the network:

- IP address

- IP subnet information

If necessary, you may also configure the domain name and the DNS server settings.

The Ethernet Setup menu provides options for viewing and changing a variety of network settings. The following table describes these options and, where applicable, explains how to change them.

Note

Establishing a VPN connection overwrites the Ethernet data fields.

Ethernet Setup Menu Options

Option

Description

To change

IPv4 Setup

In the IPv4 Setup configuration submenu, you can do the following:

- Enable or disable the phone to use the IP address that the DHCP server assigns.

- Manually set the IP Address, Subnet Mask, Default Routers, DNS Server.

This option displays only when the phone is configured in IPv4-only mode.

Scroll to IPv4 Setup and select.

See Set Up IPv4.

MAC Address

Unique Media Access Control (MAC) address of the phone.

Display only. Cannot configure.

Host Name

Unique host name that the DHCP server assigned to the phone.

Select Host Name and enter the host name.

Domain Name

Name of the Domain Name System (DNS) domain in which the phone resides.

See Set Domain Name Field.

Operational VLAN ID

Auxiliary Virtual Local Area Network (VLAN) configured on a Cisco Catalyst switch of which the phone is a member.

If the phone has not received an auxiliary VLAN, this option indicates the Administrative VLAN.

If neither the auxiliary VLAN nor the Administrative VLAN are configured, this option is blank.

Display only. Cannot configure.

The phone obtains the Operational VLAN ID via Cisco Discovery Protocol (CDP) or Link Level Discovery Protocol Media Endpoint Discovery (LLDP-MED). This information comes from the switch to which the phone is attached. To assign a VLAN ID manually, use the Admin VLAN ID option.

Enable VLAN

Allows the phone to interoperate with third-party switches that do not support a voice VLAN. The Admin VLAN ID option must be set before you can change this option.

Scroll to Enable VLAN and press On to enable or Off to disable VLAN.

Admin VLAN ID

Auxiliary VLAN of which the phone is a member.

Used only if the phone does not receive an auxiliary VLAN from the switch; otherwise, this value is ignored.

See Set Admin VLAN ID Field.

CDP

Cisco Discovery Protocol(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.

Scroll to CDP and press On to enable or Off to disable CDP.

LLDP-MED

LLDAP-MED enables the phone to advertise itself to device that use discovery protocol.

Scroll to LLDP-MED and press On to enable or Off to disable LLDP-MED.

SW Port Setup

Speed and duplex of the network port. Valid values specify:

- Auto Negotiate

- 1000 Full:1000-BaseT/full duplex

- 100 Half: 100-BaseT/half duplex

- 100 Full: 100-BaseT/full duplex

- 10 Half: 10-BaseT/half duplex

- 10 Full: 10-BaseT/full duplex

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 Configuration option to the same setting.

See Set SW Port Setup Field.

PC Port Setup

Speed and duplex of the Computer (access) port. Valid values:

- Auto Negotiate

- 1000 Full: 1000-BaseT/full duplex

- 100 Half: 100-BaseT/half duplex

- 100 Full: 100-BaseT/full duplex

- 10 Half: 10-BaseT/half duplex

- 10 Full: 10-BaseT/full duplex

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 SW Port Configuration option to the same setting.

See Set PC Port Setup Field.

NTP Servers

Network Time Protocol server allows the phone to syncronize the date and time displayed.

The phone supports up to 2 NTP servers per IP Protocol version:

- Primary NTP or NTP Server 1

- Secondary NTP or NTP Server 2

Scroll to NTP Server. Select and enter the IP for the NTP server.
Note:
For the NTP servers, a user can enter a FQDN or an IP address.

Resync DHCP Option To Use

Allows the phone to retrieve firmware and profiles.

160, 159, 66,and 150 are the supported DHCP options.

Procedure

Step 1

Press Applications .

Step 2

To access the Network Settings menu, select Admin settings > Network setup.

- Set Up IPv4

- Set Domain Name Field

- Set Admin VLAN ID Field

- Set SW Port Setup Field

- Set PC Port Setup Field

- Set Wireless Field

- Set Wi-Fi sign in access

- Set SSID Field

- Set Security Mode Field

- Set 802.11 Mode Field

- Set DHCP Field

- Set IP Address Field

- Set Subnet Mask Field

- Set Default Router Field

- Set DNS Server Fields

- Set Up Phone to Not Use DHCP

Set Up IPv4

The following table describes the IPv4 Setup menu options.

IPv4 Setup Menu Options

Option

Description

To change

DHCP Enabled

Indicates whether the phone has DHCP enabled or disabled.

When DHCP is enabled, the DHCP server assigns the phone an IP address. When DHCP is disabled, the administrator must manually assign an IP address to the phone.

See Set DHCP Field.

IP Address

Internet Protocol (IP) address of the phone.

If you assign an IP address with this option, you must also assign a subnet mask and default router. See the Subnet Mask and Default Router options in this table.

See Set IP Address Field.

Subnet Mask

Subnet mask used by the phone.

See Set Subnet Mask Field.

Default Router

Default router used by the phone.

See Set Default Router Field.

DNS Server 1

DNS Server 2

Primary Domain Name System (DNS) server (DNS Server 1) and optional backup DNS servers (DNS Server 2) that the phone uses.

See Set DNS Server Fields.

DHCP Address Released

Releases the IP address that DHCP assigned.

This field is editable if DHCP is enabled. If you wish to remove the phone from the VLAN and release the IP address for reassignment, set this option to Yes and press Apply.

Procedure

Step 1

Press Applications .

Step 2

To access the Network Settings menu, select Admin settings > Network setup.

Step 3

To access the IPv4 settings menu, navigate to the IPv4 options from Ethernet Setup > IPv4 setup.

Set Domain Name Field

Procedure

Step 1

Set the DHCP option to Off.

Step 2

Scroll to the Domain Name option, and enter a new domain name.

Step 3

Press Apply.

Set Admin VLAN ID Field

Procedure

Step 1

Scroll and select the Admin. VLAN ID option.

Step 2

Enter a new VLAN ID setting.

Step 3

Press Apply.

Set SW Port Setup Field

Procedure

Step 1

Unlock network configuration options.

Step 2

Scroll and select the SW Port Setup option.

Step 3

Scroll to the setting that you want and press Select.

Set PC Port Setup Field

Procedure

Step 1

Unlock network configuration options.

Step 2

Scroll to the PC Port Setup option and press Select.

Step 3

Scroll to the setting that you want and press Select.

Set Wireless Field

Procedure

Scroll to the Wireless option, and use the toggle switch to change the setting between On and Off.

Set Wi-Fi sign in access

Procedure

Scroll to the Wi-Fi sign in access option, and use the toggle switch to change the setting between On and Off.

Set SSID Field

Procedure

Step 1

Scroll to the SSID option, and enter an SSID.

Step 2

Press Apply.

Set Security Mode Field

Procedure

Step 1

Scroll to the Security Mode option, and highlight the desired value.
Valid values specify:

- None: Access to all access points (APs) without encryption.

- WEP: Open 802.11 authentication that uses Wired Equivalent Privacy (WEP) to encrypt the data. Specifies access to all APs and authentication through WEP keys at the local AP.

- PSK: Shared key authentication uses AES or TKIP encryption.

- EAP-FAST: Extensible Authentication Protocol Flexible Authentication via Secure Tunneling exchanges a username and cryptographically secure password with a RADIUS server in the network where a PAC (Protected Access Credential) establishes a secure tunnel for authentication. EAP-FAST supports WPA and WPA2.

- PEAP(MSCHAPV2): Protected Extensible Authentication Protocol authentication exchanges a username and cryptographically secure password with a RADIUS server in the network. PEAP is a Cisco proprietary version of EAP. PEAP supports WPA and WPA2.

Step 2

Click Apply.

Set 802.11 Mode Field

Procedure

Step 1

Scroll to the 802.11 Mode option, and select the desired value.
Valid values specify:

- Auto: Default value.

- 2.4 GHz

- 5 GHz

You can modify the 802.11 mode field only if the phone does not have a wireless LAN profile configured or the phone is configured with allowed wireless LAN profile.

Step 2

Click Save.

Step 3

Click Apply.

Set DHCP Field

Procedure

Step 1

Scroll to the DHCP option.

Step 2

Press Off to disable DHCP, or press On to enable DHCP.

Set IP Address Field

Procedure

Step 1

Set the DHCP option to Off.

Step 2

Scroll to the IP Address option, and enter a new IP Address.

Step 3

Press Apply.

Set Subnet Mask Field

Procedure

Step 1

Set the DHCP option to Off.

Step 2

Scroll to the Subnet Mask option, and enter a new subnet mask.

Step 3

Press Apply.

Set Default Router Field

Procedure

Step 1

Set the DHCP option to Off.

Step 2

Scroll to the appropriate Default Router option, and enter a new router IP address.

Step 3

Press Apply.

Set DNS Server Fields

Procedure

Step 1

Set the DHCP option to Off.

Step 2

Scroll to the appropriate DNS Server option, and enter a new DNS server IP address.

Step 3

Press Apply.

Step 4

If multiple DNS Servers can be configured, repeat Steps 2 and 3 as needed to assign backup DNS servers.

Set Up Phone to Not Use DHCP

When not using DHCP, you must configure the IP address, subnet mask, DNS server, and default router locally on the phone.

Procedure

Step 1

Press Applications.

Step 2

Choose Admin settings > Network setup > Ethernet Setup > IPv4 Setup.

Step 3

To disable DHCP and manually set an IP address:

1. Set DHCP to Off.

2. Enter the static IP address for phone.

3. Enter the subnet mask.

4. Enter the default router IP addresses.

5. Enter the DNS Server IP address.

Step 4

Press Apply.

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.

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. If the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G729a resource is already allocated (because only one G.729a resource is allowed per IP phone), no other low-bit-rate codec can be allocated for subsequent calls. The only choices are G711a and G711u.

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.

Procedure

Step 1

To configure the voice codecs on each extension, in the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Ext(n), where n is an extension number.

Step 2

In the Audio Configuration section, configure the parameters.

Step 3

Click Submit All Changes.

Set the Optional Network Servers

Optional network servers provide resources such as DNS lookup, network time, logging, and device discovery.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login > advanced > Voice > System.

Step 2

In the Optional Network Configuration section, configure the settings.

Step 3

Click Submit All Changes.

VLAN Settings

If you use a virtual LAN (VLAN), your phone voice packets are tagged with the VLAN ID.

In the VLAN Settings section of the Voice > System window, you can configure these settings:

- Cisco Discovery Protocol (CDP)

- LLDP-MED

- Network Startup Delay

- VLAN ID

- Cisco Discovery Protocol

- LLDP-MED

- Final Network Policy Resolution and QoS

- Configure VLAN Settings

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

- LLDP-MED and IEEE 802.X

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 both CDP and LLDP-MED PDUs at the same time.

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

You can use the same application type for network policy. However, phones receive different Layer 2 or Layer 3 QoS Network policies from multiple network connectivity devices. In such a case, the last valid network policy is accepted.

LLDP-MED and IEEE 802.X

The Cisco IP Phone does not support IEEE 802.X and does not work in a 802.1X wired environment. However, IEEE 802.1X or Spanning Tree Protocols on network devices could result in delay of fast start response from switches.

Configure VLAN Settings

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > System.

Step 2

In the VLAN Settings section, configure the fields.

Step 3

Click Submit All Changes.

SIP and NAT Configuration

- SIP and the Cisco IP Phone

- SIP Configuration

- Managing 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 requesting phone is called the SIP user agent server (UAS), while the receiving 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

- Failover and Recovery Registration

- RFC3261

- 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. Cisco phones support the use of backup SIP Proxy Servers to minimize or eliminate service disruption.

A static list of proxy servers is not always adequate. If your user agent serves different domains, for example, you do not want to configure a static list of proxy servers for each domain into every Cisco IP Phone.

A simple way to support proxy redundancy is to configure a SIP Proxy Server in the Cisco IP Phone configuration profile. The DNS SRV records instruct the phones to contact a SIP Proxy Server in a domain named in SIP messages. The phone consults the DNS server. If configured, the DNS server returns an SRV record that contains a list of SIP Proxy Servers for the domain, with their hostnames, priority, listening ports, and so forth. The Cisco IP Phone tries to contact the hosts in the order of their priority.

If the Cisco IP Phone currently uses a lower-priority proxy server, the phone periodically probes the higher-priority proxy and switches to the higher-priority proxy when available.

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.

Failover and Recovery Registration

- Failover—The phone performs a failover when transport timeout/failure or TCP connection failures; if Try Backup RSC and Retry Reg RSC values are datafilled.

- Recovery—The phone attempts to reregister with the primary proxy while registered or actively connected to the secondary proxy.

- Fallback Behavior

Fallback Behavior

The fallback occurs when the current registration expires or Proxy Fallback Intvl fires.

If the Proxy Fallback Intvl is exceeded, all the new SIP messages go to primary proxy.

For example, when the value for Register Expires is 3600 seconds and Proxy Fallback Intvl is 600 seconds, the fallback triggers 600 seconds later.

When the value for Register Expires is 800 seconds and Proxy Fallback Intvl is 1000 seconds, the fallback triggers at 800 seconds.

After successful registration back to the primary server, all SIP messages go to the primary server.

RFC3261

The Cisco IP Phone supports RFC-3261, 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

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 the RTP Parameters

- Configure the SDP Payload Types

- Configure the SIP Settings for Extensions

- Configure the SIP Proxy Server

- Configure the Subscriber Information Parameters

Configure the Basic SIP Parameters

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the SIP Parameters section, set the SIP parameters.

Step 3

Click Submit All Changes.

Configure the SIP Timer Values

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the SIP Timer Values section, set the SIP timer values in seconds.

Step 3

Click Submit All Changes.

Configure the Response Status Code Handling

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the Response Status Code Handling section, set the values as specified:

- Try Backup RSC—SIP response code that retries a backup server for the current request. Defaults to blank. For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??.

- Retry Reg RSC—SIP response code that the phone retries registration after failing during the last registration. Defaults to blank. For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??.

Step 3

Click Submit All Changes.

Configure the RTP Parameters

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the RTP Parameters section, set the Real-Time Transport Protocol (RTP) parameter values.

Step 3

Click Submit All Changes.

- RTP Packet Contents

RTP Packet Contents

During an active connection, the Cisco IP Phone sends out compound RTCP packets. Each compound RTP packet, except the last one, contains a sender report (SR) and a source description (SDES). The last RTCP packet contains an additional BYE packet. Each SR, except the last one, contains one receiver report (RR); the last SR carries no RR.

The SDES contains CNAME, NAME, and TOOL identifiers:

- CNAME—User ID@Proxy

- NAME—Display Name (or Anonymous if user blocks caller ID)

- TOOL—Vendor/Hardware-platform-software-version.

Configure the SDP Payload Types

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 (comparison is case-sensitive).

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the SDP Payload Types section, set the value as specified:

- AVT Dynamic Payload—Any nonstandard data. Both sender and receiver must agree on a number. Ranges from 96 to 127. Default: 101.

Step 3

Click Submit All Changes.

Configure the SIP Settings for Extensions

Procedure

Step 1

In the phone web user interface, navigate to Admin Login > advanced > Voice > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, set the parameter values.

Step 3

Click Submit All Changes.

Configure the SIP Proxy Server

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Ext(n), where n is an extension number.

Step 2

In the Proxy and Registration section, set the parameter values.

Step 3

Click Submit All Changes.

Configure the Subscriber Information Parameters

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Ext(n), where n is an extension number.

Step 2

In the Subscriber Information section, set the parameter values.

Step 3

Click Submit All Changes.

Managing 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

- NAT Mapping with SIP-ALG Router

- NAT Mapping with a Static IP Address

- NAT Mapping with STUN

- NAT Mapping with Session Border Controller

- NAT Mapping with SIP-ALG Router

- NAT Mapping with the Static IP Address

- Configure NAT mapping with STUN

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 the Static IP Address

You can configure NAT mapping on the phone to ensure interoperability with the service provider.

- You must have an external (public) IP address that is static .

- The NAT mechanism used in the router must be symmetric. See Determining Symmetric or Asymmetric NAT

Use NAT mapping only if the service provider network does not provide a Session Border Controller functionality. To configure NAT mapping on the phone:

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the NAT Support Parameters section, set Handle VIA received, Insert VIA received, Substitute VIA Addr, Handle VIA rport, Insert VIA rport, Send Resp To Src Port fields to Yes.

Step 3

In the NAT Support Parameters section, set a value for the NAT Keep Alive Intvl field.

Step 4

Enter the public IP address for your router in the EXT IP field.

Step 5

Click the Ext(n) tab.

Step 6

In the NAT Settings section, set NAT Mapping Enable to Yes.

Step 7

(Optional) Set NAT Keep Alive Enable to Yes.
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.

Step 8

Click Submit All Changes.

What to Do Next

Configure the firewall settings on your router to allow SIP traffic.

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 Determining 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.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > SIP.

Step 2

In the NAT Support Parameters section, set Handle VIA received, Insert VIA received, Substitute VIA Addr, Handle VIA rport, Insert VIA rport, Send Resp To Src Port fields to Yes.

Step 3

In the NAT Support Parameters section, set STUN Enable field to Yes.

Step 4

Enter the IP address for your STUN server in the STUN Server field.

Step 5

Click the Ext(n) tab.

Step 6

In the NAT Settings section, set NAT Mapping Enable to Yes.

Step 7

(Optional) Set NAT Keep Alive Enable to Yes.
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.

Step 8

Click Submit All Changes.

What to Do Next

Configure the firewall settings on your router to allow SIP traffic.

- Determining Symmetric or Asymmetric NAT

Determining 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:

Procedure

Step 1

Verify that the firewall is not running on your PC. (It can block the syslog port.) By default, the syslog port is 514.

Step 2

Click Voice>System and navigate to Optional Network Configuration.

Step 3

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 4

Set the Debug Level to 3.

Step 5

To capture SIP signaling messages, click the Ext tab and navigate to SIP Settings. Set the SIP Debug Option to Full.

Step 6

To collect information about what type of NAT your router uses click the SIP tab and navigate to NAT Support Parameters.

Step 7

Click Voice>SIP and navigate to NAT Support Parameters.

Step 8

Set STUN Test Enable to Yes.

Step 9

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 10

Click Submit All Changes.

Dial Plan

- Dial Plan Overview

- Edit the Dial Plan on the IP Phone

- Reset the Control Timers

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 > Extension 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 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 )

- 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 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 press 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.

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. The default length of the Dial Plan Timer is 5 seconds.

- 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; if no number is entered after P, the default timer of 5 seconds applies. 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

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

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > 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.

Reset the Control Timers

If you need to edit a timer setting only for a particular digit sequence or type of call, you can edit the dial plan.

Procedure

Step 1

Log in to the phone web user interface.

Step 2

Click Admin Login > advanced > Voice  > Regional.

Step 3

Scroll to the Control Timer Values (sec) section.

Step 4

Enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field.

Step 5

Click Submit All Changes.

Regional Parameters and Supplementary Services

- Regional Parameters

- Set the Control Timer Values

- Localize Your Cisco IP Phone

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:

- Control Timer Values—Displays all values in seconds.

- Vertical Service Activation Codes—Includes Call Back Act Code and Call Back Deact Code.

- Time—Includes local date, local time, time zone, and Daylight Saving Time.

- Localization—Includes Dictionary Server Script, Language Selection, and Locale.

Set the Control Timer Values

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Regional.

Step 2

Configure the values in the fields in the Control Timer Values (sec) section.

Step 3

Click Submit All Changes.

Localize Your Cisco IP Phone

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Regional.

Step 2

Configure the values in the fields in the Time and Localization sections.

Step 3

Click Submit All Changes.

- Time and Date Settings

- Configure Daylight Saving Time

- Select a Display Language on the Phone

- Dictionary Server Script

Time and Date Settings

The Cisco IP Phone obtains the time settings in one of three ways:

- NTP Server—When the phone boots up, it tries to contact the first Network Time Protocol (NTP) server to get the time. The phone periodically synchronizes its time with the NTP server. The synchronization period is fixed at 1 hour. Between updates, the phone tracks time with its internal clock.

- SIP Messages—Each SIP message (request or response) sent to the phone can contain a Date header with the current time information. If the header is present, the phone uses it to set its clock.

- Manual Setup—You can use the phone web user interface to enter the time and date manually. However, the NTP time or SIP Message Date overwrites this value when either is available to the phone. Manual setup requires that you enter the time in 24-hour format only.

The time that the NTP Server and the SIP Date Header serve is expressed in GMT time. The local time is obtained by offsetting the GMT according to the time zone of the region.

You can configure the Time Zone parameter with the phone web user interface or through provisioning. This time can be further offset by the Time Offset (HH/mm) parameter. This parameter must be entered in 24-hour format and can also be configured from the IP phone screen.

The Time Zone and Time Offset (HH/mm) offset values are not applied to manual time and date setup.

Configure Daylight Saving Time

The phone supports automatic adjustment for daylight saving time.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > 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 first Sunday in April and ending at midnight on the last Sunday of October; add 1 hour (USA, North America):

start=4/1/7/0:0:0;end=10/31/7/0:0:0;save=1
start=4/1/7;end=10/-1/7;save=1
start=4/1/7/0;end=10/-1/7/0;save=1

The following example configures daylight saving time for Egypt, starting at midnight on the last Sunday in April and ending at midnight on the last Sunday of September:

start=4/-1/7;end=9/-1/7;save=1 (Egypt)

The following example configures daylight saving time for New Zealand (in version 7.5.1 and higher), starting at midnight on the first Sunday of October and ending at midnight on the third Sunday of March.

start=10/1/7;end=3/22/7;save=1 (New Zealand)

The following example reflects the new change starting in March. DST starts on the second Sunday in March and ends on the first Sunday in November:

start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1

The following example configures the daylight saving time starting on the last Monday (before April 8) and ending on the first Wednesday (after May 8.)

start=4/-8/1;end=5/8/3;save=1

Select a Display Language on the Phone

You can define up to twelve languages, in addition to English, to be available and host the dictionaries for each of the languages on the HTTP or TFTP provisioning server. Language support follows Cisco dictionary principles.

Use the Language Selection parameter to select the phone default display language. The value must match one of the languages that the dictionary server supports. The script (dx value) is as follows:

- <Language_Selection ua =“na”>

- </Language_Selection>

The Language Selection parameter defaults to blank; the maximum number of characters is 512. For example:

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

During startup, the phone checks the selected language and downloads the dictionary from the TFTP/ HTTP provisioning server that the phone configuration indicates. The dictionaries are available at the support website.

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings.

Step 3

Scroll to Set Language.

Step 4

Select the desired language , then press Set.

Dictionary Server Script

The Dictionary Server Script defines the location of the dictionary server, the available languages, and the associated dictionary. The script recognizes up to five language entries. The syntax is:

Dictionary_Server_Script
serv=http://locale_server/locale_path/;d1=French;l1=fr-FR;x1=French/fr-FR.tar;d2=Danish;l2=da-DK;
x2=Danish_Denmark/da-DK.tar;d3=German;l3=de-DE;x3=German_Germany/de-DE.tar;d4=Russian;l4=ru-RU;
x4=Russian/ru-RU.tar;d5=Hebrew;l5=he-IL;x5=Hebrew/he-IL.tar.

Note

TFTP, HTTP, and HTTPS support exists for the dictionary download.

Default to blank; the maximum number of characters is 512. The detailed format is as follows:

serv={server ip port and root path};
d0=language0;l0=locale0;x0=dictionary0 filename;
d1=language1;l1=locale1;x1=dictionary1 filename;
d2=language2;l2=locale2;x2=dictionary2 filename;
d3=language3;l3=locale3;x3=dictionary3 filename;
d4=language4;l4=locale4;x4=dictionary4 filename;
d5=language5;l5=locale5;x5=dictionary5 filename;
d6=language6;l6=locale6;x6=dictionary6 filename;
d7=language3;l7=locale7;x7=dictionary7 filename;
d8=language8;l8=locale8;x8=dictionary8 filename;
d9=language5;l9=locale9;x9=dictionary9 filename;

The following languages locales are supported on the Cisco IP Phone:

- None: English-US

- bg-BG: Bulgarian

- cs-DZ: Czech

- da-DK: Dutch

- fi-FI: Finnish

- fr-FR: French

- de-DE: German

- es-ES: Spanish-ES

- es-MX: Spanish-MX

- gr-GR: Greek

- he-IL: Hebrew

- hr-HR: Croatia

- hu-HU: Hungarian

- it-IT: Italian

- nl-NL: Netherlands

- no-NO: Norwegian

- pl-PL: Polish

- pt-PT: Portuguese

- ru-RU: Russian

- sk-SK: Slovakian

- sv-SE: Swedish

- tr-TR: Turkey

- Localization Configuration Example

Localization Configuration Example

Language Selection: French

(Entry dx must match one of the languages that the dictionary server supports.)

Locale: fr-FR

(Entry lx must be within the Locale option list.)

Cisco IP Phone 8800 Series Documentation

Refer to publications that are specific to your language and phone model, and phone firmware release. Navigate from the following documentation URL:

Click Here

Chapter 5. Third-Party Call Control Setup

- Determine the Phone MAC Address

- Network Configuration

- Provisioning

- Web-Based Configuration Utility

- Administrator and User Accounts

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

Command or Action

Purpose

Step 1

Perform one of the following actions:

- On the phone, press Applications > Phone 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 > System 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.

The Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide describes provisioning in detail.

Web-Based Configuration Utility

Your phone system administrator can allow you to view the phone statistics and modify some or all the parameters. This section describes the features of the Cisco IP Phone that you can modify with the phone web user interface.

- Access the Web-Based Configuration Utility

- Determine the IP Address of the Phone

- Allow Web Access to the Cisco IP Phone

- Web Administration Tabs

Access the Web-Based Configuration Utility

Access the Cisco IP Phone configuration utility from a web browser on a computer that can reach the phone on the subnetwork.

Procedure

Step 1

If the computer is connected to a VPN, exit the VPN.

Step 2

Launch a web browser.

Step 3

Enter the IP address of the phone in your web browser address bar.
For example, http://10.64.84.147

Note:
If your service provider disables access to the configuration utility, contact the service provider to proceed.

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.

Procedure

Step 1

Click Admin Login  > advanced  > Info  > System Status.

Step 2

Scroll to System Information. Current IP displays the IP address.

Allow Web Access to the Cisco IP Phone

To view the phone parameters, enable the configuration profile. To make changes to any of the parameters, you must be able to change the configuration profile. Your system administrator might have disabled the phone option to make the phone web user interface viewable or writable.

For more information, see the Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide.

Procedure

Step 1

Click Admin Login > Voice > System.

Step 2

In the System Configuration section, set Enable Web Server to Yes.

Step 3

To update the configuration profile, click Submit All Changes after you modify the fields in the phone web user interface.
The phone reboots and the changes are applied.

Step 4

To clear all changes that you made during the current session (or after you last clicked Submit All Changes), click Undo All Changes. Values return to their previous settings.

Web Administration Tabs

Each tab contains parameters that are related to a particular feature. Some tasks require that you set multiple parameters in different tabs.

Info briefly describes each parameter that is available on the phone web user interface.

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.

- Enable User Access to the Phone Interface Menus

- Access Administrative Options by Login

- Access Administrative Options by IP Address

Enable User Access to the Phone Interface Menus

Use the admin account to enable or disable access to the phone web user interface by the user account. If the user account has access, users can set parameters, such as speed-dial numbers and caller ID blocking, through the phone web user interface.

Use phone profile provisioning to restrict the ability to configure individual parameters. For more information on provisioning, see the Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide.

Procedure

Step 1

Click Admin Login > advanced > Voice > System.

Step 2

Under System Configuration in the Phone-UI-User-Mode field, choose Yes.

Step 3

Click Submit All Changes.

Access Administrative Options by Login

Procedure

Step 1

Log in to the configuration utility.

Step 2

Click Admin Login.

Step 3

If prompted, enter the Admin Password.

Access Administrative Options by IP Address

Procedure

Enter the IP address of the Cisco IP Phone in a web browser and include the admin/ extension.
For example: http://10.64.84.147/admin/

Part III. Hardware and Accessory Installation

- Cisco IP Phone Accessories

- Cisco IP Phone Key Expansion Module

- Wall Mounts

Chapter 6. Cisco IP Phone Accessories

- Cisco IP Phone Accessories Overview

- Connect the Footstand

- Secure the Phone with a Cable Lock

- External Speakers and Microphone

- Headsets

Cisco IP Phone Accessories Overview

The following table lists the accessories that the Cisco IP Phones 8800 Series support. An "X" indicates support for a particular phone model and a dash (-) indicates no support.

Accessory Support for the Cisco IP Phones 8811, 8841, 8851, and 8861

Accessory

Type

Cisco IP Phone 8811

Cisco IP Phone 8841

Cisco IP Phone 8851

Cisco IP Phone 8861

Third-Party Accessories

Headsets: See Headsets. This section includes information about each headset type.

Analog

X

X

X

X

Analog Wideband

X

X

X

X

Bluetooth

-

-

X

X

USB (wired or wireless)

-

-

X

X

Microphone: See External Speakers and Microphone.

External PC

-

-

-

X

Speakers: See External Speakers and Microphone.

External PC

-

-

-

X

Connect the Footstand

If your phone is placed on a table or desk, connect the footstand to the back of the phone.

Procedure

Step 1

Insert the connectors into the slots.

Step 2

Press the footstand until the connectors snap into place.

Step 3

Adjust the angle of the phone.

Secure the Phone with a Cable Lock

You can secure your phone with a laptop cable lock up to 20 mm wide.

Procedure

Step 1

Take the looped end of the cable lock and wrap it around the object to which you want to secure your phone.

Step 2

Pass the lock through the looped end of the cable.

Step 3

Unlock the cable lock.

Step 4

Press and hold the locking button to align the locking teeth.

Step 5

Insert the cable lock into the lock slot of your phone and release the locking button.

Step 6

Lock the cable lock.

External Speakers and Microphone

External speakers and microphones are plug-and-play accessories. You can connect an external PC-type microphone and powered speakers (with amplifier) on the Cisco IP Phone by using the line in/out jacks. Connecting an external microphone disables the internal microphone and connecting an external speaker disables the internal phone speaker.

Note

Using poor quality external audio devices, playing loudspeakers at very loud volumes, or placing the microphone very close to the loudspeaker may result in undesirable echo for other parties on your speakerphone calls.

Headsets

Although Cisco Systems performs internal testing of third-party headsets for use with Cisco IP Phones, Cisco does not certify nor support products from headset or handset vendors.

The phone reduces some background noise that a headset microphone detects, but if you want to further reduce the background noise and improve the overall audio quality, use a noise cancelling headset.

We recommend the use of good quality external devices; for example, headsets that are screened against unwanted radio frequency (RF) and audio frequency (AF) signals. Depending on the quality of headsets and their proximity to other devices, such as mobile (cell) phones and two-way radios, some audio noise or echo may still occur. Either the remote party or both the remote party and the Cisco IP Phone user may hear an audible hum or buzz. A range of outside sources can cause humming or buzzing sounds; for example, electric lights, electric motors, or large PC monitors.

Note

In some cases, use of a local power cube or power injector may reduce or eliminate hum.

Environmental and hardware inconsistencies in the locations where Cisco IP Phones are deployed mean that no single headset solution is optimal for all environments.

We recommend that customers test headsets in the intended environment to determine performance before making a purchasing decision and deploy on a large scale.

- Audio Quality

- Analog Headsets

- Wireless Headsets

- Bluetooth Wireless Headsets

Related Information

External Device

Audio Quality

Beyond physical, mechanical, and technical performance, the audio portion of a headset must sound good to the user and to the party on the far end. Sound quality is subjective, and Cisco cannot guarantee the performance of any headsets. However, a variety of headsets from leading headset manufacturers are reported to perform well with Cisco IP Phones.

For additional information, see Click Here

Analog Headsets

The phone cannot detect when an analog headset is plugged in. For this reason, the analog headset displays by default in the Accessories window on the phone screen.

Displaying the analog headset as the default allows users to enable wideband for the analog headset.

- Enable Wideband on Analog Headsets

Enable Wideband on Analog Headsets

Although analog headsets are supported on the phone, the phones cannot detect when an analog headset is plugged in. For this reason, by default, the analog headset displays in the Accessories window on the phone screen.

Displaying the analog headset as the default allows users to enable wideband for the analog headset.

The phone is unable to detect whether the headset supports the wideband codec, but the user can enable wideband on analog headsets by following these steps:

Procedure

Step 1

On the Cisco IP Phone, press Applications .

Step 2

Select Accessories.

Step 3

Highlight the analog headset, then press Setup.

Step 4

Press On to turn on the wideband and press Off to turn off the wideband.

Wireless Headsets

You can use a wireless headset with the Cisco IP Phone.

The Cisco website provides information about wireless headsets that work with your IP phone. Go to the following URL:

Click Here

See the wireless headset documentation for information about connecting the headset and using the features.

Bluetooth Wireless Headsets

For a list of supported headsets, see Click Here .

Bluetooth enables low-bandwidth wireless connections within a range of 30 feet (10 meters). The best performance is in the 3- to 6-foot (1- to 2-meter) range. Bluetooth wireless technology operates in the 2.4 GHz band, which is the same as the 802.11b/g band. Interference issues can occur. Cisco recommends that you:

- Use 802.11a, 802.11n or 802.11ac that operates in the 5 GHz band.

- Reduce the proximity of other 802.11b/g devices, Bluetooth devices, microwave ovens, and large metal objects.

The Cisco IP Phone uses a shared key authentication and encryption method to connect with headsets. The Cisco IP Phone can connect with up to fifty headsets, one at a time. The last connected headset is used as the default. Pairing is typically performed once for each headset.

After a device is paired, the Bluetooth connection is maintained as long as both devices (phone and headset) are enabled and within range of each other. The connection typically reestablishes itself automatically if either of the devices powers down then powers up. However, some headsets require user action to reestablish the connection.

The Bluetooth icon  indicates that Bluetooth is on irrespective of whether a device is connected or not.

Potential interference issues can occur. Cisco recommends that you reduce the proximity of other 802.11b/g devices, Bluetooth devices, microwave ovens, and large metal objects. If possible, configure other 802.11 devices to use the 802.11a channels.

For a Bluetooth wireless headset to work, it does not need to be within direct line-of-sight of the phone, but some barriers, such as walls or doors, and interference from other electronic devices, can affect the connection.

When headsets are more than 30 feet (10 meters) away from the Cisco IP Phone, Bluetooth drops the connection after a 15- to 20-second timeout. If the paired headset comes back into range of the Cisco IP Phone and the phone is not connected to another Bluetooth headset, the in-range Bluetooth headset automatically reconnects. For certain phone types that operate in power-save modes, the user can wake up the headset by tapping on the operational button to initiate the reconnect.

You must enable the headset and then add it as a phone accessory.

The phone supports various Handsfree Profile features that enable you to use hands-free devices (such as Bluetooth wireless headsets) to perform certain tasks without having to handle the phone. For example, instead of pressing Redial on the phone, users can redial a number from their Bluetooth wireless headset by following instructions from the headset manufacturer.

These hands-free features apply to Bluetooth wireless headsets that are used with the Cisco IP Phone 8851 and 8861:

- Answer a call

- End a call

- Change the headset volume for a call

- Redial

- Caller ID

- Divert

- Hold and Accept

- Release and Accept

Hands-free devices may differ as to how features are activated. Device manufacturers may also use different terms when referring to the same feature.

Important:
Only one headset type works at any given time, so if you use both a Bluetooth headset and an analog headset that are attached to the phone, enabling the Bluetooth headset disables the analog headset. To enable the analog headset, disable the Bluetooth headset. Plugging a USB headset into a phone that has Bluetooth headset enabled disables both the Bluetooth and analog headset. If you unplug the USB headset, you can either enable the Bluetooth headset or disable the Bluetooth headset to use the analog headset.

Chapter 7. Cisco IP Phone Key Expansion Module

- Cisco IP Phone Key Expansion Module Setup Overview

- Key Expansion Module Power Information

- Connect a Single Key Expansion Module to a Cisco IP Phone

- Connect Two or Three Key Expansion Modules to a Cisco IP Phone

- Configure the Key Expansion Module in the Configuration Utility

- Access Key Expansion Module Setup

- Reset the Key Expansion Module to Factory Defaults

- Troubleshoot the Key Expansion Module

Cisco IP Phone Key Expansion Module Setup Overview

The Cisco IP Phone 8800 Key Expansion Module adds extra line appearances, speed dials, or programmable buttons to the phone. The programmable buttons can be set up as phone line buttons, speed-dial buttons, or phone feature buttons

The following table lists the phones and the number of key expansion modules that each model supports.

Cisco IP Phones and Supported Key Expansion Modules

Cisco IP Phone Model

Supported Key Expansion Modules

Cisco IP Phone 8851

2; providing 72 lines or buttons

Cisco IP Phone 8861

3; providing 108 lines or buttons

Key Expansion Module Power Information

The Cisco IP Phone 8800 Key Expansion Module increases the amount of power that the phone consumes. Although Power over Ethernet (PoE) from the switch may provide enough power to meet the needs of a phone that has multiple key expansion modules attached, use of a power cube may be necessary. The tables that follow help you understand whether your PoE equipment can provide the necessary power.

The Cisco IP Phone 8800 Key Expansion Module power consumption and power scheme is:

Power Consumption

48V DC, 5W per key expansion module

Power Scheme

The phone can power one key expansion module directly. For more information, see the following tables.

If you are charging a smartphone or a tablet, note the following:

- Side USB: Up to 500mA/2.5W charging

- Back USB: Fast charging, Supports up to 2.1A/10.5W charging

Key Expansion Module Power

Cisco IP Phone 8851

Accessories

802.3af PoE

802.3at PoE

Cisco IP Phone Power Cube 4

1 KEM

Supported

Supported

Supported

2 KEM

Not supported

Not supported

Supported

3 KEM

Not supported

Not supported

Not supported

1 KEM + tablet charging

Not supported

Not supported

Not supported

2 KEM + tablet charging

Not supported

Not supported

Not supported

3 KEM + tablet charging

Not supported

Not supported

Not supported

Cisco IP Phone 8861

Accessories

802.3af

803.3at PoE

Cisco IP Phone Power Cube 4

1 KEM

Not supported

Supported

Supported

2 KEM

Not supported

Supported

Supported

3 KEM

Not supported

Supported

Supported

1 KEM + tablet charging

Not supported

Supported

Supported

2 KEM + tablet charging

Not supported

Supported

Supported

3 KEM + tablet charging

Not supported

Supported

Supported

Note

When more than one key expansion module is attached to the Cisco IP Phone 8861, the fast-charge feature using the back USB port is not supported. In this scenario, the back port will slow-charge the device.

Connect a Single Key Expansion Module to a Cisco IP Phone

Procedure

Step 1

Unplug the Ethernet cable from the phone.

Step 2

If installed, remove the footstand from the phone.

Step 3

Locate the accessory connector covers on the side of the phone.
This diagram shows the location.

Step 4

Remove the two accessory connector covers, as shown in the diagram.
Caution:
The slots are designed for the spine connector only. Insertion of other objects will cause permanent damage to the phone.

Step 5

Position the phone so that the front of the phone faces up.

Step 6

Connect one end of the key expansion module spine connector to the accessory connector on the Cisco IP Phone.

1. Align the spine connector with the accessory connector ports.
Note:
Install the connector in the orientation shown in the following diagrams.

2. Firmly press the spine connector into the phone.

This diagram shows the spine connector.

This diagram shows the installation of the spine connector.

Step 7

Connect the other end of the spine connector to the key expansion module as shown as shown in this diagram.

1. Align the spine connector with the key expansion module accessory connector ports.

2. Firmly press the key expansion module into the spine connector.

Step 8

Use a screwdriver to fasten the screws into the phone.
This step ensures that the phone and key expansion module remain connected at all times. This diagram shows the location of the screw holes on the phone and one key expansion module.

Note:
Make sure that the screws are fully inserted into the phone and tightened.

Step 9

(Optional)Install the footstands on the phone and on the key expansion module, and adjust both footstands to rest evenly on the work surface.

Step 10

Plug the Ethernet cable into the phone.

Connect Two or Three Key Expansion Modules to a Cisco IP Phone

Procedure

Step 1

Unplug the Ethernet cable from the phone.

Step 2

If installed, remove the footstand from the phone.

Step 3

Locate the accessory connector covers on the side of the phone.
This diagram shows the location.

Step 4

Remove the two accessory connector covers, as shown in the diagram.
Caution:
The slots are designed for the spine connector only. Insertion of other objects will cause permanent damage to the phone.

Step 5

Position the phone so that the front of the phone faces up.

Step 6

Connect one end of the key expansion module spine connector to the accessory connector on the Cisco IP Phone.

1. Align the spine connector with the accessory connector ports.
Note:
Install the connector in the orientation shown in the following diagrams.

2. Firmly press the spine connector into the phone.

This diagram shows the spine connector.

This diagram shows the installation of the spine connector.

Step 7

Connect the other end of the spine connector to the key expansion module as shown in this diagram.

1. Align the spine connector with the key expansion module accessory connector ports.

2. Firmly press the key expansion module into the spine connector.

The first Key expansion module is now connected to the Cisco IP Phone.

Step 8

Use a second key expansion module spine connector to connect the second key expansion module to the first key expansion module.

Step 9

Use a third key expansion module spine connector to connect the third key expansion module to the second (middle) key expansion module. This figure shows a Cisco IP Phone with three key expansion modules attached.

Step 10

Use a screwdriver to fasten the screws into the phone and into each key expansion module.
This step ensures that the phone and key expansion modules remain connected at all times. This diagram shows the location of the screw holes.

Note:
Make sure that the screws are fully inserted into the phone and tightened.

Step 11

(Optional)Install the footstands on the phone and on the key expansion modules, and adjust all footstands to rest evenly on the work surface.

Step 12

Plug the Ethernet cable into the phone.

Configure the Key Expansion Module in the Configuration Utility

When you set up the Key Expansion Module from the phone web interface, you can configure the key expansion modules to display in one-column or two-column mode.

In one-column mode, each row in the display corresponds to one line. You can access this line by either the button on the left or the button on the right. In this configuration, the key expansion module displays nine lines on page 1, and nine lines on page 2.

In two-column mode, each button on the left and right of the screen is assigned to a different line. The key expansion module displays 18 lines on page 1, and 18 lines on page 2.

Procedure

Step 1

On the Configuration Utility page, click Admin Login > Advanced > Voice > Attendant Console.

Step 2

From the Number of Units list, select the number of supported key expansion modules.

Step 3

Click Submit All Changes.

Access Key Expansion Module Setup

After you install one or more key expansion modules on the phone and configure them in the Configuration Utility page, the phone automatically recognizes the key expansion modules.

When multiple key expansion modules are attached, they are numbered according to the order in which they connect to the phone:

- Key expansion module 1 is the KEM closest to the phone.

- Key expansion module 2 is the KEM in the middle.

- Key expansion module 3 is the KEM farthest to the right.

When the phone automatically recognizes the key expansion modules, you can then choose one of the following softkeys:

- Exit: Returns to the Applications menu.

- Show Details: Provides details about the selected key expansion module.

- Setup: Allows you to configure the brightness of the selected key expansion module. Setting the brightness can also be done using the Preferences menu on the phone. For details, see the Cisco IP Phone 8811, 8841, 8851, and 8861 User Guide for Cisco Unified Communications Manager (SIP).

Procedure

Step 1

On the phone, press Applications .

Step 2

Press Accessories.
All properly installed and configured key expansion modules display in the list of accessories.

Reset the Key Expansion Module to Factory Defaults

This procedure gives the steps for the key expansion module factory reset.

Procedure

Step 1

Power on the key expansion module, press Page 1, and do not release the key. When the LCD turns white, continue pressing Page 1 for at least one second.

Step 2

Release Page 1; LEDs should turn red. Immediately press Page 2 and continue pressing Page 2 for at least one second.

Step 3

Release Page 2; all LEDs should turn amber.

Step 4

Press Lines 5, 14, 1, 18, 10, and 9 in sequence.
The LCD should turn blue, and the spinning loader icon displays in the center.

The key expansion module resets.

Troubleshoot the Key Expansion Module

To obtain Key Expansion Module (KEM) troubleshooting information, follow these steps:

Procedure

Step 1

Open a CLI.

Step 2

Enter the following command to enter debug mode:
debugsh

Step 3

Enter ? to see all available commands and options.

Step 4

Use the applicable commands and options to find the desired KEM information.

Step 5

To exit debug mode, press Ctrl-C.

Chapter 8. Wall Mounts

- Non-Lockable Wall Mount Components

- Non-Lockable Wall Mount Components for Phone with Key Expansion Module

- Adjust the Handset Rest

Non-Lockable Wall Mount Components

This section describes how to install the Cisco IP Phone 8800 Series Wall Mount Kit.

The following figure shows the components of the Cisco IP Phone 8800 Series Wall Mount Kit.

Components

The package contains the following items:

- One phone bracket

- One wall bracket

- Four #8-18 x 1.25-inch Phillips-head screws with four anchors

- Two K30x8mm self-tapping screws

- One 6-inch Ethernet cable

The following figure shows the wall mount kit installed on the phone.

Back view of Wall Mount Kit Installed on Phone

The following figure shows the phone with the wall mount kit from the side.

Side View of Wall Mount Kit Installed on Phone

- Install the Non-Lockable Wall Mount Kit for Phone

- Remove the Phone from the Non-Lockable Wall Mount

Install the Non-Lockable Wall Mount Kit for Phone

The wall mount kit can be mounted on most surfaces, including concrete, brick, and similar hard surfaces. To mount the kit on concrete, brick, or similar hard surfaces, you must provide the appropriate screws and anchors for your wall surface.

Before You Begin

You need these tools to install the bracket:

- #1 and #2 Phillips-head screwdrivers

- Level

- Pencil

You must also install an Ethernet jack for the telephone in the desired location if an Ethernet jack does not currently exist. This jack must be wired appropriately for an Ethernet connection. You cannot use a regular telephone jack.

Procedure

Step 1

Mount the wall bracket in the desired location. You can install the bracket over an Ethernet jack, or you can run the Ethernet network cable to a nearby jack.
Note:
If the jack is to be placed behind the phone, the Ethernet jack must be flush to the wall or recessed.

1. Hold the bracket on the wall, placing it so that the arrow on the back of the bracket is pointing up.

2. Use the level to ensure that the bracket is level and use a pencil to mark the screw holes.

3. Using a #2 Phillips-head screwdriver, carefully center the anchor over the pencil mark and press the anchor into the wall.

4. Screw the anchor clockwise into the wall until it is seated flush.

5. Use the included screws and a #2 Phillips-head screwdriver to attach the bracket to the wall.

Bracket Installation. The following figure shows the bracket installation steps.

Step 2

Attach the phone bracket to the IP Phone.

1. Detach power cord, and any other attached cords from the base of the phone, except the handset cord (and headset cord, if there is a headset).

2. Attach the phone bracket by inserting the tabs into the mounting tabs on the back of the phone. The phone ports should be accessible through the holes in the bracket.

3. Secure the phone bracket to the IP phone with the self-tapping screws, using the #1 Phillips-head screwdriver.

4. Reattach the cords and seat them in the clips that are incorporated into the phone body.
Attach Phone Bracket. The following figure shows how the bracket attaches to the phone.

Step 3

Attach the cables to the phone:

1. Attach the Ethernet cable to the 10/100/1000 SW network port and wall jack.

2. Optional: If you are connecting a network device (such as a computer) to the phone, attach the cable to the 10/100/1000 Computer (PC access) port.

3. Optional: If you are using an external power supply, plug the power cord into the phone and dress the cord by clipping it into the clips that are incorporated into the phone body next to the PC port.

4. Optional: If the cables terminate inside the wall bracket, connect the cables to the jacks.

Attach Cables. The following figure shows the cables.

Step 4

Attach the phone to the wall bracket by inserting the tabs on the top of the wall bracket into the slots on the phone bracket.
For cables that terminate outside of the brackets, use the cable-access openings in the bottom of the bracket to position the power cord and any other cable that does not terminate in the wall behind the bracket. The phone and wall bracket openings together form circular openings with room for one cable per opening.

Attach Phone to Wall Bracket. The following figure shows how you attach the phone to the wall bracket.

Step 5

Press the phone firmly into the wall bracket and slide the phone down. The tabs in the bracket click into position.

Step 6

Proceed to Adjust the Handset Rest.

Remove the Phone from the Non-Lockable Wall Mount

The wall bracket has two tabs that lock the kit together. Use the following illustration to locate the tabs.

Tab Location

Before You Begin

Obtain two Phillips head screwdrivers or other similar devices that have a diameter of 5 millimeters or 3/16ths of an inch.

Procedure

Step 1

Insert a screw driver or other device into the left and right holes in the phone mounting plate. Insert to a depth of about 3/4 of an inch or 2 centimeters.

Step 2

Press firmly inwards to disengage the tabs.
Disengage Tabs

Step 3

Lift the phone to release it from the wall bracket. Pull the phone toward you.

Non-Lockable Wall Mount Components for Phone with Key Expansion Module

This section describes how to install the Cisco IP Phone 8800 Series Wall Mount Kit with Single KEM on a phone when the phone is connected to a Key Expansion Module.

The following figure shows the wall mount kit installed on the phone.

The following figure shows the phone with the wall mount kit from the side.

The following figure shows the components of the Cisco IP Phone 8800 Series Wall Mount Kit with Single KEM.

The package contains the following items:

- One phone bracket

- One wall bracket

- Six #8-18 x 1.25-inch Phillips-head screws with six anchors

- Three K30x8mm self-tapping screws

- One 6-inch Ethernet cable

- Install Non-Lockable Wall Mount Kit for Phone with Key Expansion Module

- Remove the Phone and Key Expansion Module from the Non-Lockable Wall Mount

Install Non-Lockable Wall Mount Kit for Phone with Key Expansion Module

The wall mount kit can be mounted on most surfaces, including concrete, brick, and similar hard surfaces. To mount the kit on concrete, brick, or similar hard surfaces, you must provide the appropriate screws and anchors for your wall surface.

Before You Begin

You need these tools to install the bracket:

- #1 and #2 Phillips-head screwdrivers

- Level

- Pencil

You must also install an Ethernet jack for the telephone in the desired location if an Ethernet jack does not currently exist. This jack must be wired appropriately for an Ethernet connection. You cannot use a regular telephone jack.

Procedure

Step 1

Mount the wall bracket in the desired location. You can install the bracket over an Ethernet jack, or you can run the Ethernet network cable to a nearby jack.
Note:
If the jack is to be placed behind the phone, the Ethernet jack must be flush to the wall or recessed.

1. Hold the bracket on the wall. See the following figure for the orientation of the wall bracket.

2. Use the level to ensure that the bracket is level and use a pencil to mark the screw holes.

3. Using a #2 Phillips-head screwdriver, carefully center the anchor over the pencil mark and press the anchor into the wall.

4. Screw the anchor clockwise into the wall until it is seated flush.

5. Use the included screws and a #2 Phillips-head screwdriver to attach the bracket to the wall.

Step 2

Attach the phone bracket to the IP phone and key expansion assembly.

1. Detach power cord, and any other attached cords from the base of the phone, except the handset cord (and headset cord, if there is a headset).

2. Attach the phone bracket by inserting the tabs into the mounting tabs on the back of the phone. The phone ports should be accessible through the holes in the bracket.

3. Secure the phone bracket to the IP phone with the self-tapping screws using a #1 Philips-head screwdriver.

4. Reattach the cords and seat them in the clips that are incorporated into the phone body.

Step 3

Attach the cords.

1. Attach the Ethernet cable to the 10/100/1000 SW network port and wall jack.

2. Optional: If you are connecting a network device (such as a computer) to the phone, attach the cable to the 10/100/1000 Computer (PC access) port.

3. Optional: If you are using an external power supply, plug the power cord into the phone and dress the cord by clipping it into the clips that are incorporated into the phone body next to the PC port.

4. Optional: If the cables terminate inside the wall bracket, connect the cables to the jacks.

Step 4

Attach the phone to the wall bracket by inserting the tabs on the top of the phone bracket into the slots on the wall bracket.
For cables that terminate outside of the bracket, use the cable-access openings in the bottom of the bracket to position the power cord and any other cable that does not terminate in the wall behind the bracket. The phone and wall bracket openings together form circular openings with room for one cable per opening.

Step 5

Proceed to Adjust the Handset Rest.

Remove the Phone and Key Expansion Module from the Non-Lockable Wall Mount

The wall bracket has two tabs that lock the kit together. Use the following illustration to locate the tabs.

Before You Begin

Obtain two Phillips head screwdrivers or other similar devices that have a diameter of 5 millimeters or 3/16ths of an inch.

Procedure

Step 1

Insert a screw driver or other device into the left and right holes in the phone mounting plate. Insert to a depth of about 3/4 of an inch or 2 centimeters.

Step 2

Press firmly inwards to disengage the tabs.

Step 3

Lift the phone to release it from the wall bracket. Pull the phone toward you.

Adjust the Handset Rest

If your phone is wall-mounted or if the handset slips out of the cradle too easily, you may need to adjust the handset rest to ensure that the receiver does not slip out of the cradle.

Procedure

Step 1

Remove the handset from the cradle and pull the plastic tab from the handset rest.

Step 2

Rotate the tab 180 degrees.

Step 3

Hold the tab between two fingers, with the corner notches facing you.

Step 4

Line up the tab with the slot in the cradle and press the tab evenly into the slot. An extension protrudes from the top of the rotated tab.

Step 5

Return the handset to the handset rest.

Part IV. Cisco IP Phone Administration

- Cisco IP Phone Security

- Cisco IP Phone Customization

- Phone Features and Setup

- Corporate and Personal Directory Setup

Chapter 9. Cisco IP Phone Security

- Security Features

- Documentation, Support, and Security Guidelines

Security Features

Security features ensure that calls are secure and authenticated.

- Domain and Internet Setting

- Configure the Challenge for the SIP INVITE Messages

- Transport Layer Security

- Configure SIP Over TLS Signaling Encryption

Domain and Internet Setting

- Configure Restricted Access Domains

- Configure the Internet Connection Type

- DHCP Option Support

Configure Restricted Access Domains

If you enter domains, the Cisco IP Phone responds only to SIP messages only from the identified servers.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > System.

Step 2

In the System Configuration section, in the Restricted Access Domains field, enter fully qualified domain names (FQDNs) for each SIP server that you want the phone to respond to. Separate FQDNs with semicolons.

Example:voiceip.com; voiceip1.com

Step 3

Click Submit All Changes.

Configure the Internet Connection Type

You can set the connection type to one of the following:

- 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 device lease on the IP address. If a phone loses the IP address for any reason, or if some other device on the network is assigned the same IP address, the communication between the SIP proxy and the phone is either severed or degraded. Whenever an expected SIP response is not received within a programmable amount of time after the corresponding SIP command is sent, the DHCP Timeout on Renewal parameter causes the device to request a renewal of its IP address. 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.

- Static IP - A static IP address for the phone.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > System.

Step 2

In the Internet Connection Type section, use the Connection Type drop-down list box to choose the connection type:

- Dynamic Host Configuration Protocol (DHCP)

- Static IP

Step 3

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

- Gateway—Gateway IP address

Step 4

Click Submit All Changes.

DHCP Option Support

The following table lists the DHCP options that are supported on the Cisco IP Phone.

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

DHCP option 60

Vendor class identifier

DHCP option 66

TFTP server name

DHCP option 125

Vendor-identifying vendor-specific information

DHCP option 150

TFTP server

DHCP option 159

Provisioning server IP

DHCP option 160

Provisioning URL

Configure the Challenge for the SIP INVITE Messages

The phone can challenge the SIP INVITE (initial) message in a session. The challenge restricts the SIP servers that are permitted to interact with the devices on a service provider network. This practice significantly increases the security of the VoIP network through prevention of malicious attacks against the device.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, choose Yes from the Auth INVITE drop-down list box.

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 messages between the service provider SIP proxy and the end user. SIP over TLS encrypts only the signaling messages, not the media.

TLS has two layers:

- 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.

The Cisco IP Phone uses UDP as the standard for SIP transport, but the phone also supports SIP over TLS for added security.

Configure SIP Over TLS Signaling Encryption

Procedure

Step 1

To enable TLS for the phone, in the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, select TLS from the SIP Transport drop-down list box.

Step 3

Click Submit All Changes.

Documentation, Support, and Security Guidelines

For information on obtaining documentation, obtaining support, providing documentation feedback, reviewing security guidelines, and also recommended aliases and general Cisco documents, see the monthly What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at:

Click Here

Subscribe to the What’s New in Cisco Product Documentation as a Really Simple Syndication (RSS) feed and set content to be delivered directly to your desktop using a reader application. The RSS feeds are a free service and Cisco currently supports RSS Version 2.0.

- Cisco Product Security Overview

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 Click Here .

Chapter 10. Cisco IP Phone Customization

- Phone Information and Display Settings

- Call Features Configuration

- Configure Voice Mail

- Assign a Ring Tone to an Extension

- Configure the Audio Settings

- Phone Web Server

Phone Information and Display Settings

The phone web user interface allows you to customize settings such as the phone name, background picture, logo, and screen saver.

- Configure the Phone Name

- Customize the Startup Screen

- Download and Configure the Wallpaper Pictures

- Configure the Screen Saver

- Configure the LCD Brightness

- Configure the Back Light Setting

- Configure the Number of Call Appearances Per Line

Configure the Phone Name

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Phone.

Step 2

Under General, enter the phone name in the Station Display Name field.
This name displays on the phone LCD in the top left corner.

Step 3

Click Submit All Changes.

Customize the Startup Screen

You can create a text or 128-by-48 pixel by 1-bit deep image logo to display when the Cisco IP Phone boots up. A logo displays during the boot sequence for a short period after the Cisco logo displays.

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > Phone.

Step 2

To display a text logo, enter text in the Text Logo field with following requirements:

- Enter up to two lines of text.

- Each line must be less than 32 characters.

- 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.

Step 3

To display a picture logo:

- Enter the path in the Logo Picture Download URL field.

For example:

http://10.64.84.147/pictures/image04_128x48.png

The supported phone image file attributes are: Bitmap format, one bit-per-pixel color, size 128-by-48 pixels. You can also use a TFTP server.

- Change Select Logo to Logo Picture.

Step 4

Click Submit All Changes.
The phone reboots, retrieves the .png file, and displays the picture when it next boots.

Download and Configure the Wallpaper Pictures

You can download a picture to customize the background on the phone screen.

Procedure

Step 1

Enter the path in the Wallpaper Picture Download URL field.
The URL must include the TFTP or HTTP server name (or IP address), directory, and filename.

Example:

- http://10.64.84.147/pictures/image04_800x480x24.png

The supported phone image file attributes are: PNG format, 24 bit-per-pixel color, size 800-by-480 pixels. You can also use a TFTP server.

Step 2

After download, you can navigate through phone LCD menu, press Applications > Preference > Wallpaper to set the new download picture as background picture.
Note:
Wallpaper can be set or deleted only through phone LCD menu.

Configure the Screen Saver

You can configure a screen saver for the Cisco IP Phone. When the phone is idle for a specified time, it enters screen saver mode.

Any button press returns the phone to normal mode. If a user password is set, the user must enter it to exit screen saver mode.

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > Phone.

Step 2

In the General section, in the Screen Saver Enable field, choose Yes to enable.

Step 3

In the Screen Saver Wait field, enter the number of seconds of idle time to elapse before the screen saver starts.

Step 4

In the Screen Saver Icon field, choose the display type:

- Station Time: The time displays in the middle of the IP phone screen.

- Lock: The phone displays a moving padlock icon. When the phone is locked, the status line displays a scrolling message Press any key to unlock your phone.

- Cisco: The phone displays the Cisco logo.

- DateTime: The phone displays the date and time on the phone screen.

- Wallpaper Picture: The phone displays the background picture.

Step 5

Click Submit All Changes.

Configure the LCD Brightness

Procedure

Step 1

Click Admin Login > advanced > Voice  > User.

Step 2

Under LCD, in the LCD Brightness field, enter a number value from 1 to 15. The higher the number, the greater the brightness on the IP phone screen.

Step 3

Click Submit All Changes.

Configure the Back Light Setting

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > User.

Step 2

Under LCD, use the Back Light Timer drop-down list box to choose the number of seconds of idle time that can elapse before the back light turns off.
Note:
The Cisco IP Phone 7811 does not have Back Light Timer option.

Step 3

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.

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > Phone.

Step 2

In the Miscellaneous Line Key Settings section, use the Call Appearances Per Line drop-down list box to specify the number of calls per line to allow.

Step 3

Click Submit All Changes.

Call Features Configuration

- Enable Call Transfer

- Enable Call Forwarding

- Enable Conferencing

- Enable Do Not Disturb

Enable Call Transfer

Procedure

Step 1

Click Admin Login > advanced > Voice > Phone.

Step 2

Under Supplementary Services, choose Yes for each of the transfer services that you want to enable:

- Attn Transfer Serv—Attended call transfer service. The user answers the call before transferring it.

- Blind Transfer Serv—Blind call transfer service. The user transfers the call without speaking to the caller.

Step 3

To disable a transfer service, set the field to No.

Step 4

Click Submit All Changes.

Enable Call Forwarding

Procedure

Step 1

Click Admin Login > advanced > Voice > Phone.

Step 2

Under Supplementary Services, choose Yes for each of the call forwarding services that you want to enable:

- Cfwd All Serv—Forwards all calls.

- Cfwd Busy Serv—Forwards calls only if the line is busy.

- Cfwd No Ans Serv—Forwards calls only if the line is not answered.

Step 3

Click Submit All Changes.

Enable Conferencing

Procedure

Step 1

In the phone web user interface, navigate to Admin Login  > advanced  > Voice  > Phone.

Step 2

Under Supplementary Services, choose Yes in the Conference Serv drop-down list box.

Step 3

Click Submit All Changes.

Enable Do Not Disturb

You can allow users to turn the Do Not Disturb feature on or off. The feature plays a message to the caller to say that the user is unavailable. On the Cisco IP Phone, users can press the Ignore softkey to divert a ringing call to another destination.

If the feature is enabled for the phone, users turn the feature on or off with an LCD softkey.

Procedure

Step 1

In the phone web user interface, navigate to Admin Login   > advanced  > Voice  > Phone.

Step 2

Under Supplementary Services, choose Yes in the DND Serv drop-down list box.

Step 3

Click Submit All Changes.

Configure Voice Mail

You can configure the internal or external phone number or URL for the voice mail system. If you are using an external voice mail service, the number must include any digits required to dial out and any required area code

Procedure

Step 1

Click Admin Login > advanced > Voice > Phone.

Step 2

Under General, enter the Voice Mail Number.

Step 3

Click Submit All Changes. The phone reboots.

- Configure Voice Mail for each Extension

- Configure the Message Waiting Indicator

Configure Voice Mail for each Extension

Procedure

Step 1

Click Admin Login > advanced > Voice > Extn.

Step 2

Under Call Feature Settings, enter the Voice Mail Server.

Step 3

(Optional) Enter the Voice Mail Subscribe Interval; the expiration time in seconds, of a subscription to a voice mail server.

Step 4

Click Submit All Changes.
The phone reboots.

Configure the Message Waiting Indicator

You can configure the Message Waiting Indicator for separate extensions on the phone. The Message Waiting Indicator lights based on the presence of new voicemail messages in the mailbox.

You can enable the indicator at the top of your IP phone to light when voice mail is left, or display a seeing message waiting notification.

Procedure

Step 1

Click Admin Login > advanced > Voice > Extn.

Step 2

Under Call Feature Settings in the Message Waiting , choose Yes to enable.

Assign a Ring Tone to an Extension

Procedure

Step 1

Click Admin Login > advanced > Voice > Ext(n), where (n) is the number of an extension.

Step 2

Under Call Feature Settings, use the Default Ring (n) drop-down list box to specify one of the following:

- No Ring

- 1 through 9—Choose one of the available ring tones.

Step 3

Click Submit All Changes.

Configure the Audio Settings

The user can modify volume settings by pressing the volume control button on the phone, then pressing the Save softkey.

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > User.

Step 2

In the Audio section, configure a volume level between 1 and 10, with 1 being the lowest level:

- Ringer Volume—Sets the ringer volume.

- Speaker Volume—Sets the volume for the full-duplex speakerphone.

- Headset Volume—Sets the headset volume.

- Handset Volume—Sets the handset volume.

Step 3

Click Submit All Changes.

- User Access Control

User Access Control

The Cisco IP Phone respects only the “ua” user access attribute. For a specific parameter, the “ua” attribute defines access by the user account to the administration web server. If the “ua” attribute is not specified, the phone applies the factory default user access for the corresponding parameter. This attribute does not affect access by the admin account.

Note

The value of the element attribute encloses within double quotes.

The “ua” attribute must have one of the following values:

- na – no access

- ro – read-only

- rw – read/write

Phone Web Server

The web server allows administrators and users to log in to the phone by using a phone web user interface. Administrators and users have different privileges and see different options for the phone based on their role.

- Configure the Web Server from the Phone Screen Interface

- Enable Access to Phone Web Interface

Configure the Web Server from the Phone Screen Interface

Use this procedure to enable the phone web user interface from the phone screen.

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings > Enable Web Server.

Step 3

Select Edit.

Step 4

Press y/n to toggle the selection to Yes and enable.

Step 5

Click OK  > Save.

Enable Access to Phone Web Interface

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > System.

Step 2

Under the System Configuration section, choose Yes from the Enable Web Server drop-down list box.

Step 3

In the Web Server Port field, enter the port to access the web server. The default is port 80.

Step 4

In the Enable Web Admin Access drop-down list box, you can enable or disable local access to the Admin Login of the phone web user interface. Defaults to Yes (enabled).

Step 5

In the Admin Password field, enter a password if you want the system administrator to log in to the phone web user interface with a password. The password prompt appears when an administrator clicks Admin Login. The minimum password length can be 4 characters or the maximum password length is 127 characters.
Note:
Password can contain any character except:

- Space key

- 13 stars (*************)

Step 6

In the User Password field, enter a password if you want users to log in to the phone web user interface with a password. The password prompt appears when users click User Login. The minimum password length can be 4 characters or the maximum password length is 127 characters.
Note:
Password can contain any character except:

- Space key

- 13 stars (*************)

Step 7

Click Submit All Changes.

Chapter 11. Phone Features and Setup

- Phone Features and Setup Overview

- Cisco IP Phone User Support

- Telephony Features for Cisco IP Phone

- Feature Buttons and Softkeys

- Configure a Speed Dial on a Line Key

- Configure a Speed Dial on a Key Expansion Module

- Busy Lamp Field Configuration on a Monitoring Phone

- Configure Alphanumeric Dialing

- Configure a Paging Group (Multicast Paging)

- Call Park

- Configuring Programmable Softkeys

- Configure Provisioning Authority

- Download Problem Reporting Tool Logs

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 Speed Dial, 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

- list of features supported

- user guide or quick reference for your voicemail system

Telephony Features for Cisco IP Phone

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.

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 a user to monitor the call state of a directory number associated with a speed-dial button on the phone.

Busy Lamp Field (BLF) Pickup

Provides enhancements to BLF speed dial. Allows you to configure a Directory Number (DN) that a user can monitor for incoming calls. When the DN receives an incoming call, the system alerts the monitoring user, who can then pick up the call.

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 options include Call Forward All, Call Forward Busy, Call Forward No Answer.

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 will:

- Log missed calls for a shared line

- Log 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.

Conference

Allows a user to talk simultaneously with multiple parties by calling each participant individually.

Allows a noninitiator in a standard (ad hoc) 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 (2048 to 65535) for Real-Time Transport Protocol (RTP) and secure Real-Time Transport Protocol (sRTP).

The default RTP and sRTP port range is 16384 to 16538.

You configure the RTP and sRTP port range in the SIP Profile.

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.

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 configuration required unless you want to use Music On Hold. See "Music On Hold" in this table for information.

- 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.

HTTPS for Phone Services

Increases security by requiring communication using HTTPS.

Note:
When the web is in HTTPS mode, the phone is a HTTPS servers.

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.

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 the call.

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.

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.

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.

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 password) 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.

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 Forward All to line buttons.

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.

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.

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 Calling ID and Calling Number

The phones can display both the calling ID and calling number for incoming calls. The IP phone LCD display size limits the length of the calling ID and the calling number that display.

The Show Calling ID and Calling Number feature applies to the incoming call alert only and does not change the function of the Call Forward and Hunt Group features.

See "Caller ID" in this table.

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.

Speed Dial

Dials a specified number that has been previously stored.

SSH Access

Allows you to enable or disable the SSH Access setting.

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.

Web Access Enable by Default

Web services are enabled by default.

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. An "X" 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 Cisco IP Phone administration.

Features with Corresponding Buttons and Softkeys

Feature Name

Dedicated Feature Button

Programmable Feature Button

Softkey

Answer

X

X

Call Forward All

X

X

Call Park

X

X

Call Park Line Status

X

Call Pickup (Pick Up)

X

X

Call Pickup Line Status

X

Conference

X

X (only displayed during connected call conference scenario)

Divert

X

Do Not Disturb

X

X

Hold

X

X

Mute

X

PLK Support for Queue Status

X

X

Redial

X

X

Speed Dial

X

X

Speed Dial Line Status

X

Transfer

X

X (only displayed during connected call transfer scenario)

Configure a 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, name, and assigned extension number on the phone screen. The user presses the line key to dial the assigned extension.

When you add the name in the string, the user only sees the extension number with the speed-dial icon.

Procedure

Step 1

On the Configuration Utility page, click Admin Login > advanced > Voice > Phone.

Step 2

Select a Line Key on which to configure speed-dial.

Step 3

Select Disabled to disable the extension.

Step 4

In the Extended Function field, 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;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 +.

- nme= XXXX is the name displayed on the phone for the speed-dial line key. Replace XXXX with a name.

Step 5

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.

Procedure

Step 1

On the Configuration Utility page, click Admin Login > Advanced > Voice > Unit.

Step 2

Select a Key Expansion Module line key on which to enable the speed dial and

Step 3

Enter a string in this format:
fnc=sd;ext=9999@$PROXY;vid=1;nme=xxxx

where:

- fnc= sd means function=speed dial

- ext= 9999 is the phone that the line key calls. Replace 9999 with a 4-digit number.

- nme= XXXX is the name displayed on the phone for the speed-dial line key. Replace XXXX with a name.

Step 4

Click Submit All Changes.

Busy Lamp Field Configuration on a Monitoring Phone

If a user needs to monitor a coworker's availability to receive a call, you can configure a busy lamp field on the user's phone (monitoring phone). With this feature, colored LEDs show whether a coworker's line (monitored line) is busy or free to take a call.

If this feature is configured on your phone, the following LED colors display on a line key:

- A green LED—Monitored line is available.

- A red LED—Monitored line is busy.

- A red fast blinking LED—Call is ringing to the monitored line.

- An amber LED—Configuration error occurred when this feature was being set up.

You can configure busy lamp field so a user can answer an incoming call on the monitored line. With busy lamp field pickup, the user can answer an incoming call by selecting the blinking line key.

You can also configure the busy lamp field to work with speed dial or call pickup to give the user more flexibility when handling calls.

There are two ways to configure the busy lamp field on a phone that you want to monitor:

- Configuration of a busy lamp field for a specific line key or user

- Configuration of a busy lamp field for multiple users (BroadSoft only)

- Configure the Busy Lamp Field for Multiple Users with the Configuration Utility

- Configure the Busy Lamp Field in the Phone Configuration File

- Configure the Busy Lamp Field for a Single Phone with the Configuration Utility

- Configure the Busy Lamp Field on a Key Expansion Module

Configure the Busy Lamp Field for Multiple Users with the Configuration Utility

If a phone registers to a BroadSoft server, you can configure the busy lamp field for several users at once.

The busy lamp field, speed dial, and call pickup features are configured for each user specified by BLF List URI unless overridden by an extended function.

Before You Begin

The BLF List URI must be configured on the server.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > advanced > Voice > Attendant Console.

Step 2

Set Use Line Keys For BLF List to Yes.

Step 3

Enter a string in the BLF List URI field in this format:
parameter@domain name.

Step 4

(Optional)To configure the busy lamp field with call pickup, enter *97 in the Call Pickup Code field.

Step 5

Click Submit All Changes.

Configure the Busy Lamp Field in the Phone Configuration File

If the phone is registered to a BroadSoft server, you can use the phone configuration file to configure the busy lamp field.

Procedure

Step 1

Edit the BLF_List_URI parameter of the phone configuration file that is available in the BroadSoft server.

Step 2

Add the List URI: sip: parameter @ domain name.
The List URI must match the one defined in the BroadSoft server.

Step 3

Save the changes.

Configure the Busy Lamp Field for a Single Phone with the Configuration Utility

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.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > advanced > Voice > Phone.

Step 2

Select a line key on which to configure a busy lamp field.

Step 3

Select Disabled to disable the extension.

Step 4

In the Extended Function field, enter a string in this format:
fnc=blf;sub=xxxx@$PROXY;usr=yyyy@$PROXY

Where:

- fnc= blf means function=busy lamp field

- sub= the subscribing phone, the phone that monitors the status of other phones with busy lamp field. For a BroadSoft server, 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 user that the busy lamp field monitors. yyyy is user id of the phone that the busy lamp field monitors. Replace yyyy with the exact user id of the monitored phone. $PROXY is the server. Replace $PROXY with the server address or name.

Step 5

(Optional)You can configure the busy lamp field to work with any combination of speed dial or call pickup. To enable the busy lamp field to work with speed dial or call pickup, enter a string in the following format in the Extended Function field:
fnc=blf+sd+cp;sub=xxxx@$PROXY;usr=yyyy@$PROXY.

Where:

sd= speed dial

cp= call pickup

Step 6

Click Submit All Changes.

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.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > Advanced > Voice > Unit.

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 subscribing phone, the phone that monitors the status of other phones with busy lamp field. 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. 8888 is 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 speed dial or call pickup, enter a string in the following format in the Extended Function field: Extended Function field, enter a string in this format:
fnc=blf+sd+cp;sub=xxxx@$PROXY;usr=yyyy@$PROXY.

Where:

sd= speed dial

cp= call pickup

Step 5

Click Submit All Changes.

Configure Alphanumeric Dialing

You can configure a phone with alphanumeric dialing so that a user can place a call using alphanumeric characters. When you assign a speed-dial line with alphanumeric dialing, the user sees the alphanumeric characters on the line key label. On the Configuration Utility page, you configure alphanumeric dialing with speed-dial and blf.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > Advanced > Voice > Ext.

Step 2

In the Enable URI Dialing 1 field, select Yes to enable alphanumeric dialing.

Step 3

Click Submit All Changes.

Configure a Paging Group (Multicast Paging)

You can configure multicast paging so that users can page all the phones at once or page a group of phones without involving a server. On the Configuration Utility page, you configure a phone as a part of a paging group and can subscribe them to the same multicast address. This enables users to direct pages to specific groups of phones. When you assign each paging group with a unique number, the user dials the paging group number to start paging. All phones that are subscribed to the same multicast address (also configured on the Configuration Utility page) receive the page. The user hears a paging tone of three short beeps when there is an incoming paging call.

Keep these things in mind:

- Your network must support multicasting so that all devices in the same paging group are able to join the corresponding multicast group.

- If the phone is on an active call when a group page starts, the incoming page is ignored.

- Group paging is one way and uses the G711 codec. The paged phone can only listen to the call from the originator.

- Incoming pages are ignored when DND is enabled.

- When paging occurs, the speaker on the paged phones automatically powers on unless the handset or the headset is in use.

- If the phone is on an active call when a group page starts, the incoming page is ignored. When the call ends, the page is answered, if the page is active.

- When multiple pages occur, the pages are answered in chronological order. Until the active page ends, the next page is not answered.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > Advanced > Voice > Phone.

Step 2

In the Multiple Paging Group Parameters section, enter a string in the Group Paging Script field in this format:
pggrp=multicast-address:port;[name=xxxx;]num=yyy;[listen={yes|no}]];

where:

- multicast-address = Multicast IP address of the phone that listens for and receives pages.

- port = Port on which to page; you must use different ports for each paging group.

- name (optional) = xxxx is the name of the paging group. Replace xxxx with a name. The name can consist maximum of 64 characters.

- num= yyy is a unique number that the user dials to access the paging group. Replace yyy with a number. The number can consist maximum of 64 characters and the allowed range is 1024 to 32767.

- listen = Indicates whether the phone listens on the page group. Only the first two groups with listen set to yes listen to group pages. If the field is not defined, the default value is no, so you must set this field to listen to the group pages.

You can add more paging groups by appending to the configuration string. Here is an example.

pggrp=224.168.168.168:34560;name=All;num=500;listen=yes;
pggrp=224.168.168.168:34561;name=GroupA;num=501;listen=yes;
pggrp=224.168.168.168:34562;name=GroupB;num=502;
pggrp=224.168.168.168:34563;name=GroupC;num=503;

This example creates four paging groups: All, GroupA, GroupB, and GroupC. Users dial 500 to send pages to all phones, 501 to send pages to phones configured as part of the GroupA group, 502 to send pages to phones configured as part of the GroupB group, and 503 to send pages to phones configured as part of the GroupC group. The configured phone receives pages directed to the groups All and GroupA.

Step 3

Click Submit All Changes.

Call Park

If your network uses a BroadSoft server, you can configure call park for your users. 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 fast blinking LED—A call is parked.

- Add Call Park to a Programmable Line Key

- Configure Call Park from the Attendant Console

Add Call Park to a Programmable 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.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > Advanced > Voice > Phone.

Step 2

Select a line key on which to configure a busy lamp field.

Step 3

Select Disabled to disable the extension.

Step 4

In the Extended Function field, enter a string in this format:
For a private line, enter fnc=park;sub=$USER@$PROXY;nme=CallPark-Slot1.

For a shared line, enter fnc=prk;sub=$USER@$PROXY;nme=Call-Park1;orbit=<DN of primary line>.

Step 5

Click Submit All Changes.

Configure Call Park from the Attendant Console

If a phone is registered to a BroadSoft server, 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.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > Advanced > Voice > Attendant Console.

Step 2

Enter *68 in the Call Park Code field.

Step 3

Enter *88 in the Call Unpark Code field.

Step 4

Click Submit All Changes.

Configuring Programmable Softkeys

You can customize the softkeys displayed on the phone. The default softkeys (when the phone is in an idle state) are Redial, Directory, Call Forward, and Do Not Disturb. Other softkeys are available during specific call states (for example, if a call is on hold, the Resume softkey displays).

Procedure

Step 1

Click Admin Login > advanced > Voice > Phone

Step 2

Under Programmable Softkeys, edit the softkeys depending on the call state that you want the softkey to display. For more information, see Programmable Softkeys
In the Programmable Softkeys section, each phone state is displayed and the softkeys that are available to display during that state are listed. Each softkey is separated by a semicolon. Softkeys are shown in the format:

softkeyname  |[  position  ]

where softkeyname is the name of the key and position is where the key is displayed on the IP phone screen. Positions are numbered, with position one displayed on the lower left of the IP phone screen, followed by positions two through four. Additional positions (over four) are accessed by pressing the right arrow key on the phone. If no position is given for a softkey, the key will float and appears in the first available empty position on the IP phone screen.

Step 3

Click Submit All Changes.

- Customize a Programmable Softkey

- Configure Speed Dial on a Programmable Softkey

- Programmable Softkeys

Customize a Programmable Softkey

The phone provides sixteen programmable softkeys (fields PSK1 through PSK16). You can define the fields by a speed-dial script.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > advanced > 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.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > advanced > 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;nme=name

Where:

- fnc= function of the key (speed dial)

- extensionname=extension being dialed or the star code action to perform

- 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).

Step 5

Click Submit All Changes.

Programmable Softkeys

The following table lists each softkey and the phone state under which the softkey displays. You can have a maximum of 16 softkeys for each call state field.

Keyword

Key Label

Definition

Available Phone States

answer

Answer

Answers an incoming call.

Ringing

barge

Barge

Allows another user to interrupt a shared call.

Shared-Active

bargesilent

Barge Silent

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

Call

Calls the selected item in a list.

Call History, New Call Recents

callhist

Received calls/Missed calls/Placed Calls/All Calls

Filter button for Call History. Toggles between Received calls, Missed calls, Placed Calls, All Calls.

Call Hist Search, Call History

callhistdetail

Details

Displays details for the selected item in the call history list.

Call History

calllist

Calls

Displays a list of current calls when there is one or more call.

Connected

cancel

Cancel

Cancels a call (for example, when conferencing a call and the second party is not answering.

Dialing Input, Off-Hook, Off-Hook Consult, On-Hook, New Call Recents

cfwd

Forward all

Forwards all calls to a specified number.

Idle, Off-Hook, Shared-Active, Hold, Shared-Held

clearlist

Clear list

Clears the displayed call history list.

Call History Search, Call History

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, Start-Conf

delchar

delChar - backspace Icon

Deletes a character when entering text.

Dialing Input, Dialing Consult, On-Hook, Call History Search

delete

Delete

Deletes the selected item in the call history list.

Call History

dial

Dial

Dials a number.

Dialing Input, Dialing Consult

dnd

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

editdial

Edit dial

Allows the user to edit the number for the selected item in the call history list before dialing.

Call History List

endcall

End Call

Ends a call.

Connected, Progressing, Start-Xfer, Start-Conf, Conferencing, Dialing Consult, Hold

exit

Exit

Exits from the current screen.

Call Hist Search, Call History

gpickup

Group Pickup

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

Ignore

Ignores 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

newcall

New Call

Begins a new call.

Idle, Hold, Shared-Active, Shared-Held

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

CallPickup

Allows a user to answer a call ringing on another extension by entering the extension number.

Idle, Off-Hook

redial

Redial

Displays the redial list.

Idle, Connected, Start-Conf, Start-Xfer, Off-Hook, Off-Hook Consult, Hold

resume

Resume

Resumes a call that is on hold.

Hold, Shared-Held

save

Save

Saves the selected item in the call history list to local contacts.

Call History

showdetail

Show detail

Shows the details of the current call.

Connected, Ringing, Hold, Conference, Shared-Held

signin

Sign in/Sign out

Logs user in to or out of Extension Mobility.

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

Configure Provisioning Authority

You can set up provisioning authority so that users can access their personalized phone settings from other phones. For example, people who work different shifts or who work at different desks during the week can share an extension, yet have their own personalized settings.

The Sign in softkey appears on the phone when you enable provisioning authority on the phone. Users enter their usernames and passwords to access their personal phone settings. Users can also ignore the sign-in and use the phone as a guest. After signing on, users have access to their personal directory numbers on the phone. When the user signs out, the phone reverts to a basic profile with limited features.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > advanced > Voice > Phone.

Step 2

In the Configuration Profile section, set the Profile Rule field to the phone configuration file's URL.

Example:http://http://192.0.2.1:80/dms/CP-8851-3PCC/8851System.xml The EM Enable and EM User Domain fields are filled in, based on the information provided in the phone configuration file.

Step 3

(Optional)If the Programmable Softkey Enable field in the Programmable Softkeys section is set to Yes, add signin to Idle Key List.

Example:newcall|1;signin|2

Step 4

Click Submit All Changes.

- Configure Provisioning Authority in the Phone Configuration File

Configure Provisioning Authority in the Phone Configuration File

You can enable provisioning authority in the default configuration file for your phones, so that you don't need to set up the feature manually for each phone.

Procedure

Step 1

In the phone configuration file, set the following parameters:

1. Set the Provisioning Authority profile rules in the Profile_Rule parameters.

Example:

<Profile_Rule ua="na">("$EMS" eq "mobile" and "$MUID" ne "" and "$MPWD" ne "")?[--uid $MUID$PDOM --pwd $MPWD] http://10.74.121.51:80/dms/CP-8851-3PCC/8851System.xml|http://10.74.121.51:80/dms/CP-8851-3PCC/8851System.xml</Profile_Rule>

2. Set the EM_Enable parameter to Yes.

Example:

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

3. Enter the enter the domain for the phone, or the authentication server in the EM_User_Domain parameter.

Example:

<EM_User_Domain ua="na">@10.74.121.51</EM_User_Domain>

Step 2

Save the configuration file and upload it to your provisioning server.

Step 3

On the Configuration Utility page, select Admin Login > advanced > Voice > Provisioning.

Step 4

Enter the filepath to the configuration file in one of the Profile Rule fields.

Example:http://<SERVER IP ADDRESS>:80/dms/td_8861/8861System.xml

Step 5

Click Submit All Changes.

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.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > 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.

Chapter 12. Corporate and Personal Directory Setup

- Personal Directory Setup

- LDAP Configuration

- Configure the BroadSoft Settings

- Configure the XML Directory Service

Personal Directory Setup

The Personal Directory allows a user to store a set of personal numbers.

Personal Directory consists of the following features:

- Personal Address Book (PAB)

- Speed Dials

Users can use these methods to access Personal Directory features:

- From a web browser: Users can access the PAB and Speed Dials features from the Configuration Utility web page.

- From the Cisco IP Phone: Choose Contacts to search the corporate directory or the user personal directory.

To configure Personal Directory from a web browser, users must access their Configuration Utility. You must provide users with a URL and sign-in information.

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 the following equipment and services:

- An LDAP server, such as OpenLDAP or Microsoft Active Directory Server 2003.

- Prepare the LDAP Corporate Directory Search

Prepare the LDAP Corporate Directory Search

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > System.

Step 2

In the Optional Network Configuration section, in the Primary DNS field, enter the IP address of the DNS server.
This step is required only if you are using Active Directory with authentication set to MD5.

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.

Step 4

Click the Phone tab.

Step 5

In the LDAP Corporate Directory Search section, use the LDAP Dir Enable drop-down list box to choose Yes.
This action enables LDAP and causes the name that is defined in the LDAP Corp Dir Name field to appear in the phone directory.

Step 6

Configure the LDAP fields. See the LDAP Corporate Directory Search

Step 7

Click Submit All Changes.

Configure the 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 following table describes the access restrictions that apply to the BroadSoft settings:

Field

Access Restriction

Directory Name

Admin password is required (if set)

XSI Host Server

Admin password is required (if set)

Directory Type

Enterprise/Group/Personal

Directory User ID

None

Directory Password

None

Procedure

Step 1

Click Admin Login  > advanced  > Voice  > Phone.

Step 2

In the Broadsoft Settings section, choose Yes from the Directory Enable drop down list box.

Step 3

In the XSI Host Server field, enter the name of server.

Step 4

In the Directory Name field, enter the name of directory.

Step 5

In the Directory Type field, choose the type of BroadSoft directory from the drop down list.

Step 6

In the Directory User ID field, enter the BroadSoft user id of the phone user.

Step 7

In the Directory Password field, enter the password.

Step 8

Click Submit All Changes.

Configure the XML Directory Service

Procedure

Step 1

In the Configuration Utility page, click Admin Login  > advanced  > Voice  > Phone.

Step 2

In the XML Directory Service Name field, enter the name of XML directory.

Step 3

In the XML Directory Service URL field, enter the url where XML directory is located.

Step 4

In the XML User Name field, enter the username of XML service.

Step 5

In the XML Password field, enter the password of XML service.

Step 6

Click Submit All Changes.

Part V. Cisco IP Phone Troubleshooting

- Monitoring Phone Systems

- Troubleshooting

- Maintenance

Chapter 13. Monitoring Phone Systems

- Monitoring Phone Systems Overview

- Cisco IP Phone Status

- Cisco IP Phone Web Page

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.

For more information about troubleshooting the Cisco IP Phone, see Troubleshooting.

Cisco IP Phone Status

The following sections describes how to view model information, status messages, and network statistics on the Cisco IP Phone 7800 Series.

- 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.

For more information about troubleshooting the Cisco IP Phone 8800 Series, see Troubleshooting.

- Display the Phone Information Window

- Display Status Menu

Display the Phone Information Window

Procedure

Step 1

Press Applications .

Step 2

Select Phone Information.
When a user password is set, a corresponding icon (lock or certificate) displays at the top-right corner of the phone screen.

Step 3

To exit the Model Information screen, press Exit or .

Display Status Menu

Procedure

Step 1

To display the Status menu, press Applications .

Step 2

Select Admin Settings > Status.

Step 3

To exit the Status menu, press Back.

- Display Status Messages Window

- Display Network Statistics Window

- Display Wireless Statistics Screen

- Display Call Statistics Window

- Display Current Access Point Window

Display Status Messages Window

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings > Status > Status Messages.

Step 3

To remove current status messages, press Clear.

Step 4

To exit the Status menu, press Back.

- Status Messages Fields

Status Messages Fields

The following table describes the status messages that display on the Status Messages screen of the phone.

Status Messages on the Cisco IP Phone

Message

Description

Possible Explanation and Action

TFTP Size Error

The configuration file is too large for file system on the phone.

Power cycle the phone.

ROM Checksum Error

Downloaded software file is corrupted.

Obtain a new copy of the phone firmware and place it in the TFTPPath directory. You should only copy files into this directory when the TFTP server software is shut down; otherwise, the files may be corrupted.

Duplicate IP

Another device is using the IP address that is assigned to the phone.

If the phone has a static IP address, verify that you did not assigned a duplicate IP address.

If you are using DHCP, check the DHCP server configuration.

Error Updating Locale

One or more localization files could not be found in the TFTP Path directory or were not valid. The locale was not changed.

From the Third-Party Call Control system, check that the following files are located within subdirectories in the TFTP File Management:

- Located in subdirectory with same name as network locale:

- tones.xml

- Located in subdirectory with same name as user locale:

- glyphs.xml

- dictionary.xml

- kate.xml

IP Address Released

The phone is configured to release the IP address.

The phone remains idle until it is power cycled or until you reset the DHCP address.

IPv4 DHCP Timeout

IPv4 DHCP server did not respond.

Network is busy: The errors should resolve themselves when the network load reduces.

No network connectivity between the IPv4 DHCP server and the phone: Verify the network connections.

IPv4 DHCP server is down: Check configuration of IPv4 DHCP server.

Errors persist: Consider assigning a static IPv4 address.

IPv4 DNS Timeout

IPv4 DNS server did not respond.

Network is busy: The errors should resolve themselves when the network load reduces.

No network connectivity between the IPv4 DNS server and the phone: Verify the network connections.

IPv4 DNS server is down: Check configuration of the IPv4 DNS server.

DNS unknown IPv4 Host

IPv4 DNS could not resolve the name of the TFTP server.

Consider using IPv4 addresses rather than host names

Load Rejected HC

The application that was downloaded is not compatible with the phone hardware.

Occurs if you attempted to install a version of software on this phone that did not support hardware changes on this phone.

No Default Router

DHCP or static configuration did not specify a default router.

If the phone has a static IP address, verify that the default router is configured.

If you are using DHCP, the DHCP server has not provided a default router. Check the DHCP server configuration.

No IPv4 DNS Server

A name was specified but DHCP or static IP configuration did not specify a IPv4 DNS server address.

If the phone has a static IP address, verify that the IPv4 DNS server is configured.

If you are using DHCP, the DHCP server has not provided a IPv4 DNS server. Check the DHCP server configuration.

Timed Out

Supplicant attempted 802.1X transaction but timed out to due the absence of an authenticator.

Authentication typically times out if 802.1X is not configured on the switch.

Version Error

The name of the phone load file is incorrect.

Make sure that the phone load file has the correct name.

XmlDefault.cnf.xml, or .cnf.xml corresponding to the phone device name

Name of the configuration file.

None. This message indicates the name of the configuration file for the phone.

Display Network Statistics Window

To display the Network Statistics screen, perform these steps:

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings > Status > Network Statistics.

Step 3

To reset the Rx Frames, Tx Frames, and Rx Broadcasts statistics to 0, press Clear.

Step 4

To exit the Status menu, press Back.

- Network Statistics Fields

Network Statistics Fields

The following table describes the information in the Network Statistics screen.

Network Statistics Fields

Item

Description

Tx Frames

Number of packets sent by the phone

Tx Broadcasts

Number of broadcast packets sent by the phone

Tx Unicast

Total number of unicast packets transmitted by the phone

Rx Frames

Number of packets received by the phone

Rx Broadcasts

Number of broadcast packets received by the phone

Rx Unicast

Total number of unicast packets received by the phone

CDP Neighbor Device ID:

- CDP Neighbor IP Address

- CDP Neighbor Port

Identifier of a device connected to this port discovered by CDP protocol.

Restart Cause: One of these values:

- Hardware Reset (Power-on reset)

- Software Reset (memory controller also reset)

- Software Reset (memory controller not reset)

- Watchdog Reset

- Unknown

- None

Cause of the last reset of the phone

Port 1

Link state and connection of the PC port (for example, Auto 100 Mb Full-Duplex means that the PC port is in a link-up state and has auto-negotiated a full-duplex, 100-Mbps connection)

Port 2

Link state and connection of the Network port

IPv4

Information on the DHCP status. This includes the following states:

- CDP BOUND

- CDP INIT

- DHCP BOUND

- DHCP DISABLED

- DHCP INIT

- DHCP INVALID

- DHCP REBINDING

- DHCP REBOOT

- DHCP RENEWING

- DHCP REQUESTING

- DHCP RESYNC

- DHCP UNRECOGNIZED

- DHCP WAITING COLDBOOT TIMEOUT

- SET DHCP COLDBOOT

- SET DHCP DISABLED

- DISABLED DUPLICATE IP

- SET DHCP FAST

Display Wireless Statistics Screen

This procedure only applies to the wireless Cisco IP Phone 8861.

To display the Wireless Statistics screen, follow these steps:

Procedure

Step 1

Press Applications .

Step 2

Select Admin settings>Status > Wireless Statistics.

Step 3

To reset the Wireless statistics to 0, press Clear.

Step 4

To exit the Wireless Statistics screen, press Exit.

- WLAN Statistics

WLAN Statistics

The following table describes the WLAN statistics on the phone.

WLAN Statistics on the Cisco Unified IP Phone

Item

Description

tx bytes

Number of bytes that the phone transmitted.

rx bytes

Number of bytes that the phone received.

tx packets

Number of packets that the phone transmitted.

rx packets

Number of packets that the phone received.

tx packets dropped

The number of packets dropped during transmission.

rx packets dropped

The number of packets dropped during reception.

tx packets errors

The number of erroneous packets that phone transmitted.

rx packets errors

The number of erroneous packets that phone received.

Tx frames

The number of successfully transmitted MSDU.

tx multicast frames

The nunber of successfully transmitted multicast MSDU.

tx retry

The number of MSDU that is successfully transmitted after one or more retransmissions.

tx multi retry

The number of multicast MSDU that is successfully transmitted after one or more retransmissions.

tx failure

This number of MSDU that is not transmitted successfully due to the number of transmit attempts exceeding the retry limit.

rts succeess

This counter shall increment when a CTS is received in response to an RTS.

rts failure

This counter shall increment when a CTS is not received in response to an RTS.

ack failure

This counter shall increment when an ACK is not received when expected.

rx duplicate frames

The number of received frame that the Sequence Control field indicates is a duplicate.

rx fraagmented packets

The number of successfully received MPDU of type Data or Management.

Roaming count

The number of succesful roaming.

Display Call Statistics Window

You can access the Call Statistics screen on the phone to display counters, statistics, and voice-quality metrics of the most recent call.

Note

You can also remotely view the call statistics information by using a web browser to access the Streaming Statistics web page. This web page contains additional RTCP statistics that are not available on the phone.

A single call can use multiple voice streams, but data is captured for only the last voice stream. A voice stream is a packet stream between two endpoints. If one endpoint is put on hold, the voice stream stops even though the call is still connected. When the call resumes, a new voice packet stream begins, and the new call data overwrites the former call data.

To display the Call Statistics screen for information about the latest voice stream, follow these steps:

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings > 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

Receiver Codec

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

Sender Codec

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

Receiver Size

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

Sender Size

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

Rcvr Packets

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

Note:
This number is not necessarily identical to the number of RTP voice packets that were received since the call began because the call might have been placed on hold.

Sender Packets

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

Note:
This number is not necessarily identical to the number of RTP voice packets that were transmitted since the call began because the call might have been placed on hold.

Avg Jitter

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

Max Jitter

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

Receiver Discarded

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

Note:
The phone discards payload type 19 comfort noise packets that Cisco Gateways generate, because they increment this counter.

Rcvr Lost Packets

Missing RTP packets (lost in transit).

Voice-Quality Metrics

Cumulative Conceal Ratio

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

Interval Conceal Ratio

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

Max Conceal Ratio

Highest interval concealment ratio from start of the voice stream.

Conceal Seconds

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

Severely Conceal Seconds

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

Latency

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

Display Current Access Point Window

The Current Access Point screen displays statistics about the access point that the Cisco IP Phone 8861 uses for wireless communications.

Procedure

Step 1

Press Applications .

Step 2

Select Admin settings > Status > Current Access Point.

Step 3

To exit the Current Access Point screen, press Exit.

- Current Access Points Fields

Current Access Points Fields

The following table describes the fields in the Current Access Point screen.

Current Access Point Items

Item

Description

AP name

Name of the AP, if it is CCX-compliant; otherwise, the MAC address displays here.

MAC address

MAC address of the AP.

Frequency

The latest frequency where this AP was observed.

Current channel

The latest channel where this AP was observed.

Last RSSI

The latest RSSI in which this AP was observed.

Beacon interval

Number of time units between beacons. A time unit is 1.024 ms.

Capability

This field contains a number of subfields that are used to indicate requested or advertised optional capabilities.

Basic rates

Data rates that the AP requires and the AP at which the station must be capable of operating.

Optional rates

Data rates that the AP supports and the AP that are optional for the station to operate at.

Supported VHT(rx) rates

VHT Supported RX MCS Set received from AP.

Supported VHT(tx) rates

VHT Supported TX MCS Set received from AP.

Supported HT MCS

HT Supported MCS Set received from AP.

DTIM period

Every nth beacon is a dtime period. After each DTIM beacon, the AP sends any broadcast or multicast packets that are queued for power-save devices.

Country code

A two-digit country code. Country information might not be display if the country information element (IE) is not present in the beacon.

Channels

A list of supported channels (from the country IE).

Power constraint

The amount of power by which the maximum transmit power should be reduced from the regulatory domain limit.

Power limit

Maximum transmit power in dBm that is permitted for that channel.

Channel utilization

The percentage of time, normalized to 255, in which the AP sensed the medium was busy, as indicated by the physical or virtual carrier sense (CS) mechanism.

Station count

The total number of STAs currently associated with this AP.

Admission capacity

An unsigned integer that specifies the remaining amount of medium time that is available through explicit admission control, in units of 32 microseconds per second.

If the value is 0, the AP does not support this information element and the capacity is unknown.

WMM supported

Support for Wi-Fi multimedia extensions.

UAPSD Supported

The AP supports Unscheduled Automatic Power Save Delivery. May only be available if WMM is supported. This feature is critical for talk time and for achieving maximum call density on the wireless IP Phone.

Proxy ARP

CCX-compliant AP supports responding to IP ARP requests on behalf of the associated station. This feature is critical to standby time on the wireless IP Phone.

CCX version

If the AP is CCX compliant, this field shows the CCX version.

Best Effort

Contains information related to the Best Effort queue.

Background

Contains information related to the Background queue.

Video

Contains information related to the Video queue.

Voice

Contains information related to the Voice queue.

Cisco IP Phone Web Page

This section describes the information that you can obtain from the phone web page. You can use this information to remotely monitor the operation of a phone and to assist with troubleshooting.

- Info

- Call History

Related Tasks

Access the Web-Based Configuration Utility

Determine the IP Address of the Phone

Allow Web Access to the Cisco IP Phone

Info

The fields on this tab are read-only and cannot be edited.

System Information

Parameter

Description

Connection Type

Indicates the type of internet connection for the phone:

- DHCP

- Static IP

Current IP

Displays the current IP address assigned to the IP phone.

Host Name

Displays the current host name assigned to the phone.

Domain

Displays the network domain name of the phone.

Default: cisco.com

Current Netmask

Displays the network mask assigned to the phone.

DNS from DHCP

Displays the IP address assigned by DHCP server.

Primary DNS

Displays the primary DNS server assigned to the phone.

Current Gateway

Displays the default router assigned to the phone.

Secondary DNS

Displays the secondary DNS server assigned to the phone.

Reboot History

For information about Reboot History, see the Reboot Reasons.

Product Information

Parameter

Description

Product Name

Model number of the Cisco IP Phone.

Software Version

Version number of the Cisco IP Phone firmware.

MAC Address

Hardware address of the Cisco IP Phone.

Customization

For an RC unit, this field indicates whether the unit has been customized or not. Pending indicates a new RC unit that is ready for provisioning. If the unit has already retrieved its customized profile, this field displays the name of the company that provisioned the unit.

Serial Number

Serial number of the Cisco IP Phone.

Hardware Version

Version number of the Cisco IP Phone hardware.

VID

Device version ID of the phone, associated with PID.

Client Certificate

Status of the client certificate, which authenticates the Cisco IP Phone for use in the ITSP network. This field indicates if the client certificate is properly installed in the phone.

Phone Status

Parameter

Description

Current Time

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

Elapsed Time

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

Operational VLAN ID

ID of the VLAN currently in use if applicable.

SW Port

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

Ext Status

Parameter

Description

Registration State

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

Last Registration At

Last date and time the line was registered.

Next Registration In

Number of seconds before the next registration renewal.

Mapped SIP Port

Port number of the SIP port mapped by NAT.

Line 1 Status/Call 2 Status

Each line and call on the phone display with the following parameters.

Parameter

Description

Call State

Status of the call.

Duration

Duration of the call.

Remote Address

Address of the remote device.

Local Address

Address of the local device.

Start Time

Starting time of the call

Type

Direction of the call.

Peer Name

Name of the internal phone.

Peer Phone

Phone number of the internal phone.

Sender Packets

Number of RTP voice packets transmitted because voice stream was opened.

Note:
This number is not necessarily identical to the number of RTP voice packets transmitted since the call began because the call might have been placed on hold.

Sender Octets

Total number of octets sent by the phone.

Sender Codec

Type of voice stream transmitted (RTP streaming audio from codec): G.729, iLBC, G.711 u-law, or G.711 A-law.

Rcvr Lost Packets

Missing RTP packets (lost in transit).

Avg Jitter

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

Rcvr Codec

Type of voice stream received (RTP streaming audio from codec): G.729, iLBC, G.711 u-law, or G.711 A-law.

Rcvr Packets

Number of RTP voice packets received because voice stream was opened.

Note:
This number is not necessarily identical to the number of RTP voice packets received since the call began because the call might have been placed on hold

Rcvr Octets

Total number of octets received by the phone.

MOS-LQK

Score that is an objective estimate of the mean opinion score (MOS) for listening quality (LQK) that rates from 5 (excellent) to 1 (bad). This score is based on audible concealment events due to frame loss in the preceding 8-second interval of the voice stream.

Note:
The MOS LQK score can vary based on the type of codec that the Cisco IP Phone uses.

AVG MOS LQK

Average MOS LQK score observed for the entire voice stream.

Min MOS LQK

Lowest MOS LQK score observed from start of the voice stream.

Max MOS LQK

Baseline or highest MOS LQK score observed from start of the voice stream.

These codecs provide the following maximum MOS LQK score under normal conditions with no frame loss:

- G.711 gives 4.5

- G.722 gives 4.5

- G.729 A /AB gives 3.7

- iLBC gives 3.9

MOS LQK Version

Version of the Cisco proprietary algorithm used to calculate MOS LQK scores

Cumulative Conceal Ratio

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

Interval Conceal Ratio

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

Max Conceal Ratio

Highest interval concealment ratio from start of the voice stream.

Conceal Secs:

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

Severely Conceal Secs

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

Latency

Number of milliseconds for latency.

Max Jitter

Number of milliseconds for receiver jitter.

Rcvr Discarded

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

Note:
The phone discards payload type 19 comfort noise packets that are generated by Cisco gateways, which increments this counter.

Wallpaper Picture Download URL

Parameter

Description

Last Wallpaper Download Status

Displays the last downloaded wallpaper picture status.

Downloaded Locale Package

Parameter

Description

Locale Download Status

Displays the downloaded locale package status.

Downloaded Dictionary Info

Dictionary downloaded from the TFTP/HTTP provisioning server indicated in the phone.

Downloaded Font Info

Displays the downloaded font name.

Firmware Upgrade Status

Parameter

Description

Firmware Upgrade Status 1

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

Firmware Upgrade Status 2

Firmware Upgrade Status 3

Provisioning Status

Parameter

Description

Provisioning Status 1

Displays the provisioning status (resync) of the phone.

Provisioning Status 2

Provisioning Status 3

Note

The Upgrade and Provisioning Status are displayed in reverse chronological order (like reboot history) displaying status with time and reason.

Custom CA Status

These fields display the status of provisioning using a custom Certificate Authority (CA).

Parameter

Description

Custom CA Provisioning Status

Indicates whether provisioning using a custom CA succeeded or failed:

- 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.

Custom CA certificates are configured in the Provisioning tab. For more information about custom CA certificates, see the Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide.

Ethernet Information

Parameter

Description

TxFrames

Total number of packets that the phone transmitted.

TxBroadcasts

Total number of broadcast packets that the phone transmitted.

TxMulticasts

Total number of multicast packets that the phone transmitted.

TxUnicasts

Total number of unicast packets that the phone transmitted.

RxFrames

Total number of packets received by the phone.

RxBroadcasts

Total number of broadcast packets that the phone received.

RxMulticasts

Total number of multicast packets that the phone received.

RxUnicasts

Total number of unicast packets that the phone received.

RxPacketNoDes

Total number of shed packets that the no Direct Memory Access (DMA) descriptor causes.

Network and Access Port Information

Parameter

Description

RxtotalPkt

Total number of packets that the phone received.

RxcrcErr

Total number of packets that were received with CRC failed.

RxalignErr

Total number of packets between 64 and 1522 bytes in length that were received and that had a bad Frame Check Sequence (FCS).

Rxmulticast

Total number of multicast packets that the phone received.

Rxbroadcast

Total number of broadcast packets that the phone received.

Rxunicast

Total number of unicast packets that the phone received.

RxshortErr

Total number of received FCS error packets or Align error packets that are less than 64 bytes in size.

RxshortGood

Total number of received good packets that were less than 64-bytes size.

RxlongGood

Total number of received good packets that were greater than 1522 bytes in size.

RxlongErr

Total number of received FCS error packets or Align error packets that were greater than 1522 bytes in size.

Rxsize64

Total number of received packets, including bad packets, that were between 0 and 64 bytes in size.

Rxsize65to127

Total number of received packets, including bad packets, that were between 65 and 127 bytes in size.

Rxsize128to255

Total number of received packets, including bad packets, that were between 128 and 255 bytes in size.

Rxsize256to511

Total number of received packets, including bad packets, that were between 256 and 511 bytes in size.

Rxsize512to1023

Total number of received packets, including bad packets, that were between 512 and 1023 bytes in size.

Rxsize1024to1518

Total number of received packets, including bad packets, that were between 1024 and 1518 bytes in size.

RxtokenDrop

Total number of packets that were dropped due to lack of resources (for example, FIFO overflow).

TxexcessDefer

Total number of packets that were delayed from transmitting due to busy medium.

TxlateCollision

Number of times that collisions occurred later than 512-bit times after the start of packet transmission.

TxtotalGoodPkt

Total number of good packets (multicast, broadcast, and unicast) that the phone received.

Txcollisions

Total number of collisions that occurred while a packet was transmitted.

TxexcessLength

Total number of packets that were not transmitted because the packet experienced 16 transmission attempts.

Txbroadcast

Total number of broadcast packets that the phone transmitted.

Txmulticast

Total number of multicast packets that the phone transmitted.

lldpFramesOutTotal

Total number of LLDP frames that the phone sent out.

lldpAgeoutsTotal

Total number of LLDP frames that timed out in the cache.

lldpFramesDiscardedTotal

Total number of LLDP frames that were discarded when any of the mandatory TLVs is missing, out of order, or contains out of range string length.

lldpFramesInErrorsTotal

Total number of LLDP frames that were received with one or more detectable errors.

lldpFramesInTotal

Total number of LLDP frames that the phone received.

lldpTLVDiscardedTotal

Total number of LLDP TLVs that were discarded.

lldpTLVUnrecognizedTotal

Total number of LLDP TLVs that were not recognized on the phone.

CDPNeighborDeviceId

Identifier of a device connected to this port that CDP discovered.

CDPNeighborIP

IP address of the neighbor device discovered that CDP discovered.

CDPNeighborPort

Neighbor device port to which the phone is connected discovered by CDP.

LLDPNeighborDeviceId

Identifier of a device connected to this port discovered by LLDP discovered.

LLDPNeighborIP

IP address of the neighbor device that LLDP discovered.

LLDPNeighborPort

Neighbor device port to which the phone connects that LLDP discovered.

PortSpeed

Speed and duplex information.

DHCP

Parameter

Description

IPv4

Information on the DHCP status.

Console Logs

Displays the syslog output of the phone in the reverse order, where messages is the latest one. The display includes hyperlinks to individual log files. The console log files include debug and error messages received on the phone.

Parameter

Description

Debug Message 0

messages

Debug Message 1

messages.0

Debug Message 2

messages.1

Debug Message 3

messages.2

Debug Message 4

messages.3

Debug Message 5

messages.4

Debug Message 6

messages.5

Debug Message 7

messages.6

Debug Message 8

messages.7

Core Dumps

Parameter

Description

Core Dump

Core Dump is a recorded state of the working memory of a program at a specific time, generally when the program has terminated abnormally.

Browser Info

Parameter

Description

Loading Time

The amount of elapsed time when the page is loaded on the browser.

Note:
Safari and IE 9.0 and previous do not support this parameter.

Browser Version

Version of the browser. For example, Firefox 31

OS Version

Version of the Windows operating system.

Platform

The platform to which the browser is compiled.

Width

Current width of the browser.

Height

Current height of the browser.

System Configuration

Parameter

Description

Restricted Access Domains

This feature is used when implementing software customization.

Enable Web Server

Enable/disable web server of the IP phone.

Default: Yes

Enable Protocol

Choose the type of protocol:

- Http

- Https

Enable Direct Action Url

Enables the direct action of the URL.

Default: Yes

Session Max Timeout

Maximum timeout of the session.

Default: 3600

Session Idle Timeout

Idle timeout of the session.

Default: 3600

Web Server Port

Port number of the phone web user interface.

Default: 80

Enable Web Admin Access

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

Default: Yes

Admin Password

Password for the administrator.

Default: No password

User Password

Password for the user.

Default: Blank

Phone-UI-User-Mode

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

Default: No

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

Internet Connection Type

Parameter

Description

Connection Type

Choose the type of Internet connection:

- DHCP

- Static IP

Static IP Settings

Parameter

Description

Static IP

If static IP was chosen as the type of internet connection, displays the static IP address assigned to the phone.

Netmask

If static IP was chosen as the type, display IP mask.

Gateway

Default router IP address. Blank if DHCP assigned.

Optional Network Configuration

Parameter

Description

Host Name

The hostname of the Cisco IP Phone.

Domain

The network domain of the Cisco IP Phone.

If you are using LDAP, see the LDAP Configuration.

Primary DNS

DNS server used by the Cisco IP Phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the primary DNS server.

Default: Blank

If using LDAP, see the LDAP Configuration.

Secondary DNS

DNS server used by the Cisco IP Phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the secondary DNS server.

Default: Blank

Syslog Server

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

Debug Level

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

Default: 0

Layer 2 Logging

Used for IP phone network layer debugging purposes. Do not use except when advised to do so by Cisco technical support, as this may impact system performance. Set to No by default.

Primary NTP Server

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

Default: Blank

Secondary NTP Server

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

Default: Blank

SSH Access

The administrator can be configure this parameter to control the SSH console.

Default: No

DNS Server Order

Specifies the method for selecting the DNS server:

- Manual-Dhcp

- Manual

- Dhcp-Manual

DNS Query Sequential

Perform parallel or sequential DNS Query:

- No–With parallel DNS query mode, the IP phone sends the same request to all DNS servers at the same time. When doing a DNS lookup, the IP phone accepts the first incoming reply.

- Yes–The IP phone sends DNS queries sequentially.

DNS Cache TTL Ignore

When set to Yes, the DNS query results are not cached. When set to No, the phone caches the A/AAAA/SRV/CNAME record according to the TTL responses.

Default: Yes

SSH User ID

The administrator can set the User ID for SSH login.

Default: Blank

SSH Password

The administrator can set the Password for SSH login.

Default: Blank

Enable SSLv3

Choose Yes to enable SSLv3. Choose No to disable.

Default: No

VLAN Settings

Parameter

Description

Enable VLAN

Choose Yes to enable VLAN. Choose No to disable.

Enable CDP

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

Enable LLDP-MED

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

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

Network Startup Delay

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

VLAN ID

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

Inventory Settings

Parameter

Description

Asset ID

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

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

Changing the Asset ID field causes the phone to reboot.

SIP Parameters

Parameter

Description

Max Forward

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

Default: 70

Max Redirection

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

Default: 5

SIP User Agent Name

Used in outbound REGISTER requests.

Default: $VERSION

If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed

SIP Server Name

Server header used in responses to inbound responses.

Default: $VERSION

SIP Reg User Agent Name

User-Agent name to be used in a REGISTER request. If this is not specified, the <SIP User Agent Name> is also used for the REGISTER request.

Default: Blank

SIP Accept Language

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

There is no default. If empty, the header is not included.

Use Compact Header

If set to yes, the phone uses compact SIP headers in outbound SIP messages. If inbound SIP requests contain normal headers, the phone substitutes incoming headers with compact headers. If set to no, the phones use normal SIP headers. If inbound SIP requests contain compact headers, the phones reuse the same compact headers when generating the response, regardless of this setting.

Default: No

Talk Package

Enables support for the BroadSoft Talk Package that lets users answer or resume a call by clicking a button in an external application.

Default: No

Hold Package

Enables support for the BroadSoft Hold Package, which lets users place a call on hold by clicking a button in an external application.

Default: No

Conference Package

Enables support for the BroadSoft Conference Package that enables users to start a conference call by clicking a button in an external application.

Default: No

RFC 2543 Call Hold

If set to yes, unit includes c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either case.

Default: Yes

SIP TCP Port Min

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

Default: 5060

SIP TCP Port Max

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

Default: 5080

Caller ID Header

Provides the option to take the caller ID from PAID-RPID-FROM, PAID-FROM, RPID-PAID-FROM, RPID-FROM, or FROM header.

Default: PAID-RPID-FROM

Display Diversion Info

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

Max INVITE Retry Attempts

Maximum number of INVITE retry attempts by the phone.

Default: 6

Max NON-INVITE Retry Attempts

Maximum number of NON-INVITE retry attempts by the phone.

Default: 6

SIP Timer Values

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

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 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 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.

Response Status Code Handling

Parameter

Description

Try Backup RSC

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

Default: Blank

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??

Retry Reg RSC

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

Default: Blank

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??

RTP Parameters

Parameter

Description

RTP Port Min

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

Default: 16384

RTP Port Max

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

Default: 16538

RTP Packet Size

Packet size in seconds, which can range from 0.01 to 0.13. Valid values must be a multiple of 0.01 seconds.

Default: 0.02

RTCP Tx Enable

Set to Yes to enable Real-Time Transport Control Protocol (RTCP) sender report on an active connection.

Default: No

SDP Payload Types

Parameter

Description

AVT Dynamic Payload

AVT dynamic payload type. Ranges from 96-127.

Default: 101

NAT Support Parameters

Parameter

Description

Handle VIA received

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

Default: No

Handle VIA rport

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

Default: No

Insert VIA received

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

Default: No

Insert VIA rport

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

Default: No

Substitute VIA Addr

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

Default: No

Send Resp To Src Port

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

Default: No

STUN Enable

Enables the use of STUN to discover NAT mapping.

Default: No

STUN Test Enable

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

Default: No

STUN Server

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

Default: Blank

EXT IP

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

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

Default: Blank

EXT RTP Port Min

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

Default: 0

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Default: 15

Configuration Profile

Parameter

Description

Provision Enable

Allows or denies resync actions.

Default: 160,159,66,150

Resync On Reset

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

Default: Yes

Resync Random Delay

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

Default: 2

Resync At (HHmm)

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

Default: Blank

Resync At Random Delay

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

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

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

Default: 600

Resync Periodic

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

Default: 3600

Resync Error Retry Delay

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

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

Default: 3600

Forced Resync Delay

The resync typically takes place when the voice lines are idle. When a voice line is active and a resync is due, the IP Telephony device delays the resync procedure until the line becomes idle. However, it waits no longer than the Forced Resync Delay (seconds). A resync might cause configuration parameter values to change. This causes a firmware reboot and terminates any voice connection active at the time of the resync.

Default: 14400

Resync From SIP

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

Default: Yes

Resync Trigger 1

Resync Trigger 2

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

Default: Blank

Resync Fails On FNF

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

Default: Yes

Profile Rule
Profile Rule B
Profile Rule C
Profile Rule D

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

Default: /$PSN.xml

Resync Option To Use

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

Default: 160,159,66,150

Log Request Msg

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

Default:

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

Log Success Msg

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

Default:

$PN $MAC –Successful % $SCHEME://$SERVIP:$PORT$PATH -- $ERR

Log Failure Msg

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

Default:

$PN $MAC -- %s failed: $ERR

User Configurable Resync

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

Default: Yes

Firmware Upgrade

Parameter

Description

Upgrade Enable

Allows firmware update operations independent of resync actions.

Default: Yes

Upgrade Error Retry Delay

The interval applied in the event of an upgrade failure. The firmware upgrade error timer activates after a failed firmware upgrade attempt and is initialized with this value. The next firmware upgrade attempt occurs when this timer counts down to zero.

Default: 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:

protocol://server[:port]/profile_pathname

For example:

tftp://192.168.1.5/image/sip88xx.10-3-1-9-3PCC.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

For information about the Provisioning page, see the Cisco IP Phone 7800 Series and Cisco IP Phone 8800 Series Multiplatform Phones Provisioning Guide.

CA Settings

Parameter

Description

Custom CA Rule

The URL to download Custom CA.

Default: Blank

General Purpose Parameters

Parameter

Description

GPP A - GPP P

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

- Encryption keys.

- URLs.

- Multistage provisioning status information.

- Post request templates.

- Parameter name alias maps.

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

Default: Blank

Control Timer Values (sec)

Parameter

Description

Call Back Expires

Expiration time in seconds of a call-back activation. Range: 0–65535 seconds.

Call Back Retry Intvl

Call back retry interval in seconds. Range: 0–255 seconds.

Call Back Delay

Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the phone still considers the call as failed and keeps on retrying.

Default: 0.5

Interdigit Long Timer

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

Default: 10

Interdigit Short Timer

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

Default: 3

Vertical Service Activation Codes

Parameter

Description

Call Back Act Code

Activation code for Call Back

Default: *66

Call Back Deact Code

Deactivation code for Call Back

Default: *86

Block CID Act Code

Blocks caller ID on all outbound calls

Default: *67

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

Default: *68

Block ANC Act Code

Blocks all anonymous calls.

Default: *77

Block ANC Deact Code

Removes blocking of all anonymous calls.

Default: *87

Time

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.

Default: GMT-08:00

Time Offset (HH/mm)

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

Default: 00/00

Ignore DHCP Time Offset

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

Default: Yes.

Daylight Saving Time Rule

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

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

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

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

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

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

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

Daylight Saving Time Rule (continued)

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

- HH stands for hours (0-23).

- mm stands for minutes (0-59).

- ss stands for seconds (0-59).

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

Daylight Saving Time Enable

Enables Daylight Saving Time.

Default: Yes

Localization

Parameter

Description

Dictionary Server Script

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

Default: Blank

Language Selection

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

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

Default: Blank

The maximum number of characters is 512. For example:

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

Locale

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

Default: en-US

QoS Settings

Parameter

Description

SIP TOS Value

TOS field value in UDP IP packets carrying a SIP message.

Default: 0x60

RTP TOS Value

ToS/DiffServ field value in UDP IP packets carrying RTP data.

Default: 0xb8

General

Parameter

Description

Station Display Name

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

Voice Mail Number

A phone number or URL to check voice mail.

Default: None

Text Logo

Text logo to display when the phone boots up. A service provider, for example, can enter logo text as follows:

- Up to 2 lines of text

- Each line must be fewer than 32 characters

- Insert a new line character (\n) between lines

- Insert escape code %0a

For example,

Super\n%0aTelecom

displays:

Super

Telecom

Use the + character to add spaces for formatting. For example, you can add multiple + characters before and after the text to center it.

Logo Picture Download URL

URL locating the (.png) file to display during the boot up time.

For more information, see the Phone Information and Display Settings.

Wallpaper Picture Download URL

URL locating the (.png) file to display on the phone screen background.

For more information, see the Phone Information and Display Settings.

Select Logo

Select from None, PNG Picture, or Text Logo.

Default: None

Screen Saver Enable

Enables a screen saver on the phone. When the phone is idle for a specified time, it enters screen saver mode.

Default: No

Screen Saver Wait

Amount of idle time before screen saver displays.

Default: 300

Screen Saver Refresh Period

Number of seconds before the screen saver should refresh (if, for example, you chose a rotation of pictures).

Screen Saver Icon

In screen saver mode, the display unit can display:

- A Wallpaper picture.

- Station time in the middle of the screen.

- A moving Cisco icon. When the phone is locked, the status line displays a scrolling message Press any key to unlock your phone

- Cisco Logo

- The station date and time in the middle of the screen.

Co-branding Banner Picture Download URL

URL to download a .gif, .png, or .jpeg) image on the web GUI for co-branding.

Line Key

Parameter

Description

Extension

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

Default: Line Key n

Short Name

Specifies the user name for Line Key.

Default: $USER

Miscellaneous Line Key Settings

Parameter

Description

Line ID Mapping

Specifies the shared call appearance line ID mapping. If Vertical First is set, the first call makes the LED flash. If Horizontal first is set, the second call makes the same LED flash.

Default: Vertical First

SCA Barge-In-Enable

Enables the SCA Barge-In.

Default: No

Call Appearances Per Line

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

Default: 2

Supplementary Services

Parameter

Description

Conference Serv

Enable/disable three-way conference service.

Default: Yes

Attn Transfer Serv

Enable/disable attended-call-transfer service.

Default: Yes

Blind Transfer Serv

Enable/disable blind-call-transfer service.

Default: Yes

DND Serv

Enable/disable do not disturb service.

Default: Yes

Block ANC Serv

Enable/disable block-anonymous-call service.

Default: Yes

Call Back Serv

Enable/disable call-back service.

Default: Yes

Block CID Serv

Enable/disable blocking outbound Caller-ID service.

Default: Yes

Cfwd All Serv

Enable/disable call-forward-all service.

Default: Yes

Cfwd Busy Serv

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

Default: Yes

Cfwd No Ans Serv

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

Default: Yes

BroadSoft Settings

Parameter

Description

Directory Enable

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

Default: No

XSI Host Server

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

Default: Blank

Directory Name

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

Default: Blank

Directory Type

Select the type of BroadSoft directory:

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

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

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

Default: Enterprise

Directory User ID

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

Default: Blank

Directory Password

Alphanumeric password associated with the User ID.

Default: Blank

LDAP Corporate Directory Search

Parameter

Description

LDAP Dir Enable

Choose Yes to enable LDAP.

Default: No

LDAP Corp Dir Name

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

Default: Blank

LDAP Server

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

nnn.nnn.nnn.nnn

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

Default: Blank

LDAP Auth Method

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

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

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

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

Default: None

LDAP Client DN

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

dc=cv2bu,dc=com

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

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

Default: Blank

LDAP Username

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

Default: Blank

LDAP Password

Enter the password for the LDAP username.

Default: Blank

LDAP Search Base

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

dc=cv2bu,dc=com

Default: Blank

LDAP Last Name Filter

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

Default: Blank

LDAP First Name Filter

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

Default: Blank

LDAP Search Item 3

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

Default: Blank

LDAP Item 3 Filter

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

Default: Blank

LDAP Search Item 4

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

Default: Blank

LDAP Item 4 Filter

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

Default: Blank

LDAP Display Attrs

Format of LDAP results displayed on phone, where:

- a—Attribute name

- cn—Common name

- sn—Surname (last name)

- telephoneNumber—Phone number

- n—Display name

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

- t—type

When t=p, that is, t is of type phone number, then 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 will be ignored.

- p—phone number

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

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

Default: Blank

LDAP Number Mapping

Can be blank if not needed.

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

Default: Blank

XML Service

Parameter

Description

XML Directory Service Name:

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

Default: Blank

XML Directory Service URL

URL where the XML Directory is located.

Default: Blank

XML User Name

XML service username for authentication purposes

Default: Blank

XML Password

XML service password for authentication purposes

Default: Blank

Extension Mobility

Parameter

Description

EM Enable

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

Default: No

EM User Domain

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

Default: Blank

Programmable Softkeys

Parameter

Description

Programmable Softkey Enable

Enables programmable softkeys.

Idle Key List

Softkeys that display when the phone is idle.

Dialing Input Key List

Softkeys that display when the user must enter dialing data.

Progressing Key List

Softkeys that display when a call is attempting to connect.

Connected Key List

Softkeys that display when a call is connected.

Start-Xfer Key List

Softkeys that display when a call transfer has been initiated.

Start-Conf Key List

Softkeys that display when a conference call has been initiated.

Conferencing Key List

Softkeys that display when a conference call is in progress.

Hold Key List

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

Ringing Key List

Softkeys that display when a call is incoming.

Shared Active Key List

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

Shared Held Key List

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

Off Hook Consult Key List

Softkeys that display when the call is active and the transfer, conference, or the bxfter softkeys are pressed.

Off Hook Key List

Softkeys that display when the receiver is lifted, or the headphone or speaker phone buttons are pressed.

Call History Key List

Softkeys that display when Call History menu is navigated.

On Hook Dial Key List

Softkeys that display when the receiver is not lifted or the speaker is not pressed but you input the number that you want to call.

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.

Hold Reminder Timer

This parameter specifies the time delay (in seconds), that a ring splash is heard on an active call when another call was placed on hold.

Default: 0

Speed Dial

You can configure speed dials on the Cisco IP Phone from the LCD GUI or the web GUI.

Speed Dial 2 to 9: Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. Press the digit key (2-9) to dial out the assigned number.

Default: Blank

Supplementary Services

Parameter

Description

CW Setting

Enables or disables the Call Waiting service.

Default: Yes

Block CID Setting

Enables or disables the Block CID service.

Default: No

Block ANC Setting

Enables or disables the Block ANC service.

Default: No

Cfwd Global Mode

Choose the Cfwd mode for phone (All Lines or Per Line).

Set All lines, to apply to all lines when Cfwd All Dest changes in any single line. Set Per line, to only affect the specifed line when Cfwd All Dest change changed.

Default: All Lines

DND Global Mode

Choose the DND mode for phone (All Lines or Per Line).

Default: All Lines

Secure Call Setting

Enables or disables Secure Call.

Default: No

Auto Answer Page

Enables or disables automatic answering of paged calls.

Default: Yes

Time Format

Choose the time format for the phone (12 or 24 hour).

Default: 12hr

Date Format

Choose the date format for the phone (month/day or day/month).

Default: month/day

Shared Line DND Cfwd Enable

Enable/disable the Shared Line DND Call Forward.

Audio

Parameter

Description

Ringer Volume

Sets the default volume for the ringer.

Default: 9

Speaker Volume

Sets the default volume for the speakerphone.

Default: 8

Handset Volume

Sets the default volume for the handset.

Default: 10

Headset Volume

Sets the default volume for the headset.

Default: 10

LCD

Parameter

Description

Back Light Timer (minutes)

Select the number of minutes before the back light should turn off (1m, 5m, or 30m) or Always On.

Default: 5m

Brightness

Enter a number value from 1 to 15. The higher the number, the greater the brightness on the IP phone screen.

Default: 10

Extension

In a configuration profile, the Line parameters must be appended with the appropriate numeral to indicate the line to which the setting applies. For example:

[1] to specify line one
[2] to specify line two

- General

- Share Line Appearance

- Call Forward

- NAT Settings

- SIP Settings

- Call Feature Settings

- Supplementary Services

- Proxy and Registration

- Subscriber Information

- Audio Configuration

- Dial Plan

General

Parameter

Description

Line Enable

To enable this line for service, select yes. Otherwise, select No.

Default: Yes

Server Type

Choose the server type:

- Broadsoft

- Metaswitch

- Asterisk

- Other

Share Line Appearance

Parameter

Description

Share Ext

Indicates whether this extension is to be shared with other Cisco IP phones or private.

Default: Yes

Shared User ID

The user identified assigned to the shared line appearance.

Default: Blank

Subscription Expires

Number of seconds before the SIP subscription expires. Before the subscription expiration, the phone gets NOTIFY messages from the SIP server on the status of the shared phone extension.

Default: 3600

Restrict MWI

When enabled, the message waiting indicator lights only for messages on private lines.

Default: No

Call Forward

Parameter

Description

Cfwd All Dest

Enter the extensions to which the call is forwarded.

Cfwd Busy Dest

Enter the extensions to forward calls to when the line is busy.

Default: voicemail

Cfwd No Ans Dest

Enter the extension to forward calls to when the call is not answered.

Default: voicemail

Cfwd No Ans Delay

Enter the delay in time (in seconds) to wait before forwarding a call that is unanswered.

Default: 20 seconds

NAT Settings

Parameter

Description

NAT Mapping Enable

To use externally mapped IP addresses and SIP/ RTP ports in SIP messages, select yes. Otherwise, select no.

Default: No

NAT Keep Alive Enable

To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.

Default: No

NAT Keep Alive Msg

Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent.

Default: $NOTIFY

SIP Settings

Parameter

Description

SIP Transport

Select from UDP, TCP, or TLS.

Default: UDP

SIP UDP Port

Port number of the SIP message listening and transmission port.

Default: 5060

SIP 100REL Enable

Support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests. Select Yes to enable.

Default: No

EXT SIP Port

The external SIP port number.

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

Select Yes to enable.

Default: Yes

SIP Remote-Party-ID

The Remote-Party-ID header to use instead of the From header. Select Yes to enable.

Default: Yes

SIP P-Asserted-Identity

Enables the use of the P-Asserted-Identity header field in SIP header which is inserted by a User Agent Client (UAC) in the same Trusted Domain as the first proxy.

Refer-To Target Contact

Indicates the refer-to target. Select Yes to send the SIP Refer to the contact.

Default: No

SIP Debug Option

How SIP messages are received at or sent from the proxy listen port to the log. Select:

- Default—No messages.

- Current—Logs all the current SIP messages in full text.

- Full—Logs all SIP messages in full text.

Sticky 183

When enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select Yes. Otherwise, select No.

Default: No

Auth INVITE

When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. To enable this feature, select Yes.

Default: No

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

To enable this optional parameter, select Yes.

Default: No

Voice Quality Report Address

Enter a domain name or an IP address in this field to configure the generation of voice quality report for each extension on the phone.

Call Feature Settings

Parameter

Description

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.

Default Ring

Type of ring heard. Choose from No Ring or 1 through 10.

Conference Bridge URL

URL used to join into a conference call, generally in the form of the word conference or user@IPaddress:port.

Voice Mail Server

Identifies the SpecVM server for the phone, generally the IP address, and port number of the VM server.

Voice Mail Subscribe Interval

The expiration time, in seconds, of a subscription to a voice mail server.

Auto Ans Page On Active Call

Determines the behavior of the phone when a page call arrives.

Feature Key Sync

Enable/disable the Feature Key synchronization

Supplementary Services

Parameter

Description

DND Setting Line

Enables or disables Do Not Disturb.

Proxy and Registration

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.

The port number is optional.

Default: 5060

Outbound Proxy

All outbound requests are sent as the first hop. Enter an IP address or domain name.

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.

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.

Register

Enables periodic registration with the proxy. This parameter is ignored if a proxy is not specified. To enable this feature, select Yes.

Default: Yes

Make Call Without Reg

Enables making outbound calls without successful (dynamic) registration by the phone. If set to no, the dial tone plays only when registration is successful. To enable this feature, select Yes.

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.

The range is from 32 to 2000000.

Default: 3600 seconds

Use DNS SRV

Enables DNS SRV lookup for the proxy and outbound proxy. To enable this feature, select Yes. Otherwise, select 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.

The range is from 0 to 65535.

Default: 3600 seconds

Dual Registration

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.

Subscriber Information

Parameter

Description

Display Name

Name displayed as the caller ID.

User ID

Extension number for this line.

Password

Password for this line.

Default: Blank (no password required)

Auth ID

Authentication ID for SIP authentication.

Default: Blank

Reversed Auth Realm

The IP address for an authentication realm other than the proxy IP address. The default value is blank; the proxy IP address is used as the authentication realm.

The parameter for extension 1 appears as follows in the phone configuration file:

<Reversed_Auth_Realm_1_ ua=”na”>
</Reversed_Auth_Realm_1_>

Audio Configuration

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.) Select one of the following:

- G711u

- G711a

- G729a

- G729ab

- G722

- G722.2

- iLBC

Default: G711u

Use Pref Codec Only

To use only the preferred codecs for all calls, select Yes. (The call fails if the far end does not support these codecs.) Otherwise, select No.

Default: No

Second Preferred Codec

If the first codec fails, this codec is tried.

Default: Unspecified

Third Preferred Codec

If the second codec fails, this codec is tried.

Default: Unspecified

G711u Enable

Enables use of the G.711u codec.

Default: Yes

G711a Enable

Enables use of the G.711a codec.

Default: Yes

G729a Enable

To enable the use of the G.729a codec at 8 kbps, select Yes. Otherwise, select No.

Default: Yes

G729ab Enable

To enable the use of the G.729ab codec, select Yes. Otherwise, select No.

Default: Yes

G722 Enable

Enables use of the G.722 codec.

Default: Yes

G722.2 Enable

Enables use of the G.722.2 codec.

Default: No

iLBC Enable

Enables use of the iLBC codec.

Default: Yes

Silence Supp Enable

To enable silence suppression so that silent audio frames are not transmitted, select Yes. Otherwise, select 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.

Use Remote Pref Codec

To use, Use Remote Preferred Codec, select Yes. Otherwise, select No.

Default: No

Dial Plan

For more information about dial plan, see the Dial Plan Overview.

Attendant Console

- General

- Call Parking

General

Parameter

Description

BLF List URL

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

Default: Blank

Use Line Keys For BLF List

Options to enable or disable the line keys for BLF.

Default: No

BLF Label Display Mode

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

Default: Blank

Call Parking

Parameter

Description

Call Park

The Call Park code field contains the start code used for parking a call.

Defaults to *68.

Call Unpark

The Unpark code field contains the star code used for retrieving a parked call.

Defaults to *88.

Call Pickup

The star code used for picking up a ringing call.

Defaults to *98.

Call History

Displays the call history for the phone. To change the information displayed, select the type of call history from the drop-down list:

- All Calls

- Received Calls

- Placed Calls

- Missed Calls

Chapter 14. Troubleshooting

- General Troubleshooting Information

- Startup Problems

- Phone Reset Problems

- Phone Cannot Connect to LAN

- Audio Problems

- General Telephone Call Problems

- Troubleshooting Procedures

- Additional Troubleshooting Information

General Troubleshooting Information

The following table provides general troubleshooting information for the Cisco IP Phone.

Cisco IP Phone troubleshooting

Summary

Explanation

Connecting a Cisco IP Phone to another Cisco IP Phone

Cisco does not support connecting an IP phone to another IP Phone through the PC port. Each IP Phone should connect directly to a switch port. If phones are connected together in a line by using the PC port, the phones do not work.

Prolonged broadcast storms cause IP phones to reset, or be unable to make or answer a call

A prolonged Layer 2 broadcast storm (lasting several minutes) on the voice VLAN may cause IP phones to reset, lose an active call, or be unable to initiate or answer a call. Phones may not come up until a broadcast storm ends.

Moving a network connection from the phone to a workstation

If you power your phone through the network connection, you must be careful if you decide to unplug the network connection of the phone and plug the cable into a desktop computer.

Caution:
The network card in the computer cannot receive power through the network connection; if power comes through the connection, the network card can be destroyed. To protect a network card, wait 10 seconds or longer after unplugging the cable from the phone before plugging it into a computer. This delay gives the switch enough time to recognize that there is no longer a phone on the line and to stop providing power to the cable.

Changing the telephone configuration

By default, the network configuration options are locked to prevent users from making changes that could impact their network connectivity. You must unlock the network configuration options before you can configure them.

Note:   If the administrator password is not set in common phone profile, then user can modify the network settings.

Codec mismatch between the phone and another device

The RxType and the TxType statistics show the codec that is used for a conversation between this Cisco IP Phone and the other device. The values of these statistics should match. If they do not, verify that the other device can handle the codec conversation, or that a transcoder is in place to handle the service.

Sound sample mismatch between the phone and another device

The RxSize and the TxSize statistics show the size of the voice packets that are used in a conversation between this Cisco IP Phone and the other device. The values of these statistics should match.

Loopback condition

A loopback condition can occur when the following conditions are met:

- The SW Port Configuration option in the Network Configuration menu on the phone is set to 10 Half (10-BaseT/half duplex).

- The phone receives power from an external power supply.

- The phone is powered down (the power supply is disconnected).

In this case, the switch port on the phone can become disabled and the following message appears in the switch console log:

HALF_DUX_COLLISION_EXCEED_THRESHOLD

To resolve this problem, reenable the port from the switch.

Startup Problems

After you install a Cisco IP Phone into your network and add it to Cisco Unified Communications Manager, the phone should start up as described in the related topic below.

If the phone does not start up properly, see the following sections for troubleshooting information.

- Cisco IP Phone Does Not Go Through the Normal Startup Process

- Phone Displays Error Messages

- Configuration File Corruption

- Cisco IP Phone Cannot Obtain IP Address

Cisco IP Phone Does Not Go Through the Normal Startup Process

Problem

When you connect a Cisco IP Phone to the network port, the phone does not go through the normal startup process as described in the related topic and the phone screen does not display information.

Cause

If the phone does not go through the startup process, the cause may be faulty cables, bad connections, network outages, lack of power, or the phone may not be functional.

Solution

To determine whether the phone is functional, use the following suggestions to eliminate other potential problems.

- Verify that the network port is functional:

- Exchange the Ethernet cables with cables that you know are functional.

- Disconnect a functioning Cisco IP Phone from another port and connect it to this network port to verify that the port is active.

- Connect the Cisco IP Phone that does not start up to a different network port that is known to be good.

- Connect the Cisco IP Phone that does not start up directly to the port on the switch, eliminating the patch panel connection in the office.

- Verify that the phone is receiving power:

- If you are using external power, verify that the electrical outlet is functional.

- If you are using in-line power, use the external power supply instead.

- If you are using the external power supply, switch with a unit that you know to be functional.

- If the phone still does not start up properly, power up the phone with the handset off-hook. When the phone is powered up in this way, it attempts to launch a backup software image.

- If the phone still does not start up properly, perform a factory reset of the phone.

- After you attempt these solutions, if the phone screen on the Cisco IP Phone does not display any characters after at least five minutes, contact a Cisco technical support representative for additional assistance.

Phone Displays Error Messages

Problem

Status messages display errors during startup.

Solution

While the phone cycles through the startup process, you can access status messages that might provide you with information about the cause of a problem. See Display Status Messages Window section for instructions about accessing status messages and for a list of potential errors, their explanations, and their solutions.

- Phone Cannot Connect Using DNS

Phone Cannot Connect Using DNS

Problem

The DNS settings may be incorrect.

Solution

If you use DNS to access the TFTP server or Third-Party Call Control Manager, you must ensure that you specify a DNS server.

Configuration File Corruption

Problem

If you continue to have problems with a particular phone that other suggestions in this chapter do not resolve, the configuration file may be corrupted.

Solution

Get a new configuration file remotely from the provisioning server using resync.

Cisco IP Phone Cannot Obtain IP Address

Problem

If a phone cannot obtain an IP address when it starts up, the phone may not be on the same network or VLAN as the DHCP server, or the switch port to which the phone connects may be disabled.

Solution

Ensure that the network or VLAN to which the phone connects has access to the DHCP server, and ensure that the switch port is enabled.

Phone Reset Problems

If users report that their phones are resetting during calls or while the phones are idle on their desk, you should investigate the cause. If the network connection and Third Party Call Control connection are stable, a Cisco IP Phone should not reset.

Typically, a phone resets if it has problems in connecting to the Ethernet network or to Third Party Call Control.

- Phone Resets Due to Intermittent Network Outages

- Phone Resets Due to DHCP Setting Errors

- Phone Resets Due to Incorrect Static IP Address

- Phone Resets During Heavy Network Usage

- Phone Does Not Power Up

Phone Resets Due to Intermittent Network Outages

Problem

Your network may be experiencing intermittent outages.

Solution

Intermittent network outages affect data and voice traffic differently. Your network might be experiencing intermittent outages without detection. If so, data traffic can resend lost packets and verify that packets are received and transmitted. However, voice traffic cannot recapture lost packets. Rather than retransmitting a lost network connection, the phone resets and attempts to reconnect to the network. Contact the system administrator for information on known problems in the voice network.

Phone Resets Due to DHCP Setting Errors

Problem

The DHCP settings may be incorrect.

Solution

Verify that you have properly configured the phone to use DHCP. Verify that the DHCP server is set up properly. Verify the DHCP lease duration. We recommend that you set the lease duration to 8 days.

Phone Resets Due to Incorrect Static IP Address

Problem

The static IP address assigned to the phone may be incorrect.

Solution

If the phone is assigned a static IP address, verify that you have entered the correct settings.

Phone Resets During Heavy Network Usage

Problem

If the phone appears to reset during heavy network usage, it is likely that you do not have a voice VLAN configured.

Solution

Isolating the phones on a separate auxiliary VLAN increases the quality of the voice traffic.

Phone Does Not Power Up

Problem

The phone does not appear to be powered up.

Solution

In most cases, a phone restarts if it powers up by using external power but loses that connection and switches to PoE. Similarly, a phone may restart if it powers up by using PoE and then connects to an external power supply.

Phone Cannot Connect to LAN

Problem

The physical connection to the LAN may be broken.

Solution

Verify that the Ethernet connection to which the Cisco IP Phone connects is up. For example, check whether the particular port or switch to which the phone connects is down and that the switch is not rebooting. Also ensure that no cable breaks exist.

Audio Problems

The following sections describe how to resolve audio problems.

- No Speech Path

- Choppy Speech

No Speech Path

Problem

One or more people on a call do not hear any audio.

Solution

When at least one person in a call does not receive audio, IP connectivity between phones is not established. Check the configuration of routers and switches to ensure that IP connectivity is properly configured.

Choppy Speech

Problem

A user complains of choppy speech on a call.

Cause

There may be a mismatch in the jitter configuration.

Solution

Check the AvgJtr and the MaxJtr statistics. A large variance between these statistics might indicate a problem with jitter on the network or periodic high rates of network activity.

General Telephone Call Problems

The following sections help troubleshoot general telephone call problems.

- Phone Call Cannot Be Established

- Phone Does Not Recognize DTMF Digits or Digits Are Delayed

Phone Call Cannot Be Established

Problem

A user complains about not being able to make a call.

Cause

The phone does not have a DHCP IP address. The phones display the message Configuring IP or Registering.

Solution

1. Verify the following:

1. The Ethernet cable is attached.

2. The Third-Party Call Control system is active.

2. Audio server debug and capture logs are enabled for both phones. If needed, enable Java debug.

Phone Does Not Recognize DTMF Digits or Digits Are Delayed

Problem

The user complains that numbers are missed or delayed when the keypad is used.

Cause

Pressing the keys too quickly can result in missed or delayed digits.

Solution

Keys should not be pressed rapidly.

Troubleshooting Procedures

These procedures can be used to identify and correct problems.

- Check DHCP Settings

- Verify DNS Settings

Check DHCP Settings

Procedure

Step 1

On the Cisco IP Phone, press Applications .

Step 2

Select Admin Settings > Network Setup > Ethernet Setup > IPv4 Setup, and look at the following options:

- DHCP Option: Check the DHCP option for enabled or disabled.

- IP Address, Subnet Mask, Default Router: If you have assigned a static IP address to the phone, you must manually enter settings for these options.

Step 3

If you are using DHCP, check the IP addresses that your DHCP server distributes.
See the Understanding and Troubleshooting DHCP in Catalyst Switch or Enterprise Networks document, available at this URL:

Click Here

Verify DNS Settings

To verify DNS settings, follow these steps:

Procedure

Step 1

Press Applications .

Step 2

Select Admin Settings > Network Setup > IPv4 Setup > DNS Server 1.

Step 3

You should also verify that a CNAME entry was made in the DNS server for the TFTP server and for the Third-Party Call Control.
You must also ensure that DNS is configured to do reverse lookups.

Additional Troubleshooting Information

If you have additional questions about troubleshooting your phone, go to the following Cisco website and navigate to the desired phone model:

Click Here

Chapter 15. Maintenance

- Basic Reset

- Voice Quality Monitoring

- Voice Quality Reporting

- Cisco IP Phone Cleaning

- View Phone Information

- Reboot Reasons

- Phone Behavior During Times of Network Congestion

Basic Reset

Performing a basic reset of a Cisco IP Phone provides a way to recover if the phone experiences an error and provides a way to reset or restore various configuration and security settings.

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 appropriate for your situation.

Basic Reset Methods

Operation

Action

Explanation

Restart phone

Press Applicationsand choose Admin Settings>Reset settings>Cold Reboot

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

To reset settings, press Applicationsand choose Admin Settings>Reset settings>Factory Reset

Restores phone configuration or settings to factory default.

- Perform Factory Reset Phone Keypad

- Perform Factory Reset from Phone Menu

Perform Factory Reset Phone Keypad

Use these steps to reset the phone to factory default settings using the phone keypad.

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

Press and hold # and plug the phone back in.

Step 4

When the phone boots up, the headset button, the speaker button, and the mute button light up. When the light on the Mute button turns off, 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.

Perform Factory Reset from Phone Menu

Procedure

Step 1

Press Applications .

Step 2

Scroll to Admin Settings>Reset settings and select Factory Reset.

Step 3

To restore phone configuration or settings to factory default, press Ok.

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.

- Voice Quality Troubleshooting Tips

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.

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 sends the SIP Publish message when a call ends.

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 the codecs and MOS scores. For other codecs, the phone does not send 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 enable voice quality reporting on the phone with the web interface. Each extension on a phone has a separate voice quality report. For each extension on the phone, use the corresponding Voice Quality Report Address field to configure the generation of voice quality report.

Procedure

Step 1

On the Configuration Utility page, select Admin Login > advanced > Voice > Ext x.
Where:

- Ext x = the extension number on the phone

Step 2

In SIP Settings, enter a value in the Voice Quality Report Address x field. You can enter either a domain name or an IP address in this field.
You can also add a port number along with the domain name or an IP address in this field. 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

Click Submit All Changes.

Cisco IP Phone Cleaning

To clean your Cisco IP Phone, use only a dry soft cloth to gently wipe the phone and the phone screen. Do not apply liquids or powders directly to the phone. As with all non-weatherproof electronics, liquids and powders can damage the components and cause failures.

When the phone is in sleep mode, the screen is blank and the Select button is not lit. When the phone is in this condition, you can clean the screen, as long as you know that the phone will remain asleep until after you finish cleaning.

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.

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/>

Phone Behavior During Times of Network Congestion

Anything that degrades network performance can affect Cisco IP Phone voice and video 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.

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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:

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