Cisco SIP IP Administrator Guide, Version 8.0
Initializing Cisco SIP IP Phones
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Initializing Cisco Unified IP Phones

Table Of Contents

Initializing Cisco Unified IP Phones

Prerequisites

Overview of the Initialization Process

About Configuration Files

How to Customize the Default Configuration File

How to Customize a Phone-Specific Configuration File

Configuring the SIP Parameters Manually

Unlocking the Phone

Setting and Restoring Network Parameters

Configuring Network Parameters Manually

Setting and Restoring Phone-Specific Parameters

Setting End-User Call Preferences

How to Set the Date and Time

Time-Zone Configuration Examples

How to Create Dial Plans

Dial-Plan Configuration Examples

How to Verify Initialization

Where to Go Next


Initializing Cisco Unified IP Phones


This chapter describes the initial firmware installation tasks and configuration process for the Cisco Unified IP Phone 7960G and 7940G in a Session Initiation Protocol (SIP) network. It provides information on the following:

Prerequisites

Overview of the Initialization Process

About Configuration Files

How to Customize the Default Configuration File

How to Customize a Phone-Specific Configuration File

Configuring the SIP Parameters Manually

How to Set the Date and Time

How to Create Dial Plans

How to Verify Initialization

Where to Go Next

Prerequisites

Ensure that your network meets the following requirements:

A working IP network is established and configured for SIP.

For information on configuring IP, refer to the Cisco IOS IP Configuration Guide.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/ip_vcg.htm

VoIP is configured on your Cisco routers.

For information on configuring VoIP, refer to the Cisco IOS Voice Configuration Library.
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vcl.htm

VoIP gateways are configured for SIP.

A TFTP server is configured on your network and has the files that the phone requests.


Note Files are available from cisco.com.


When the phone initializes, it requests the following from the TFTP server:

Latest firmware image (P0S3-xx-y-zz.bin)

The Cisco SIP IP phone firmware image. The xx variable represents the major version number, the y variable represents the minor version number, and the zz variable represents the subversion number.


Note Applies to Cisco SIP IP Phone Release 3.0 and later.


For more information, see the "How to Upgrade Your Cisco SIP IP Phone Firmware Image and Reboot Remotely" section.

Dual-boot file (OS79XX.TXT)

After downloading this file, you must use an ASCII editor to open it and specify the filename (without the file extension) of the image version that you plan to run on your phones.

Default configuration file (SIPDefault.cnf)

For more information, see the "About Configuration Files" section and the "How to Customize the Default Configuration File" section.

Phone-specific configuration file ( SIPXXXXYYYYZZZZ.cnf)

XXXXYYYYZZZZ is the MAC address of the phone.

Ring-list file (RINGLIST.DAT)

A file listing audio files that are the custom ring type options for the phones. These audio files must also be in the root directory of the TFTP server.

For more information, see the "How to Customize Cisco Unified IP Phone 7960G and 7940G Rings" section.

Synchronization file (syncinfo.xml)

Controls the image version and associated synchronization value to be used for remote reboots.

Dial-plan file (dialplan.xml)

For more information, see the "How to Upgrade Your Cisco SIP IP Phone Firmware Image and Reboot Remotely" section.

For information about configuring your TFTP server, refer to your operating-system documentation.

A DHCP server is configured on your network.

The phone can use DHCP to obtain IP addresses. Configuration options are as follows:

dhcp option #1 (IP subnet mask)

dhcp option #3 (default IP gateway)

dhcp option #6 (DNS server IP address)

dhcp option #15 (domain name)

dhcp option #50 (IP address)

dhcp option #66 (TFTP server IP address)

If you do not configure DHCP options on the DHCP server, you must manually configure them on the phone. For information on configuring a DHCP server, refer to your operating-system documentation.

A proxy server is active and configured to receive and forward SIP messages.


Note Refer to the Cisco 7940 and 7960 IP Phones Firmware Upgrade Matrix for additional prerequisites.


Overview of the Initialization Process

For an overview of the initialization process, refer to Cisco IP Phone 7960 and 7940 Firmware Upgrade Matrix, which is available at this URL:

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/
frmwrup.htm

Basic phone configuration follows these high-level steps:


Step 1 Download the required files from Cisco.com to the TFTP server as described in "Prerequisites".

Step 2 If you are configuring SIP parameters via a TFTP server, create and store the configuration files as described in "How to Customize the Default Configuration File" and "How to Customize a Phone-Specific Configuration File". If you are not configuring the SIP parameters via a TFTP server, manually configure the required parameters as described in "Configuring the SIP Parameters Manually".

Step 3 If you are using DCHP to configure the phone network settings, configure the required network parameters on your DHCP server as described in "Prerequisites". If you are not using DHCP to configure network parameters, manually configure the required network parameters as described in "Configuring Network Parameters Manually".

Step 4 Connect the phone to the network as described in "Technical Specifications of the Cisco Unified IP Phone 7960G and 7940G", and to a power supply as described in "Connecting the Phone to Power".


About Configuration Files

Configuration files reside in a TFTP server subdirectory (you can specify the location of this subdirectory with the tftp_cfg_dir parameter). For more information, refer to the Cisco 7940 and 7960 IP Phones Firmware Upgrade Matrix.


Note Be sure to customize configuration files before you power up the phone. When powered up, the phone automatically loads parameters stored in flash memory and then requests configuration files from the TFTP server.


There are two configuration files that can be downloaded from the TFTP server:

A default configuration file, named SIPDefault.cnf, which is downloaded by all phones.

A phone-specific configuration file, which is downloaded by a specific phone after the default configuration file.

The name of each phone-specific configuration file is unique and is based on the MAC address of the phone. The format of the filename must be SIPXXXXYYYYZZZZ.cnf, where XXXXYYYYZZZZ is the MAC address of the phone. The MAC address must be in uppercase; the .cnf extension must be in lowercase (for example, SIP00503EFFD842.cnf).


Note You can find the MAC address of a phone on the middle sticker adhered to the base of the phone. You can also view it on the Network Configuration menu.


Each configuration file contains phone parameters you can set. When setting parameters, note the following:

Parameters in the default configuration file override those stored in the phone's flash memory.

Parameters in the phone-specific configuration file override those stored in the default configuration file.

If configuration files are not used to set parameters on the phone, you must set up the phone manually.

When a phone is rebooted, the manually set values of parameters are overridden by the values found in the configuration files (if the same parameters exist in at least one of the configuration files).

Since the configuration files are reloaded each time the phone is rebooted, you can avoid overriding a phone's local values by removing from the configuration files the parameters that set those values. You can also avoid overriding local values by preventing the reload of the configuration files.

Each parameter, or variable, in a configuration file is a one-line entry that must use the following format:

variable-name : value ; optional comments

Configuration-file variable entries must adhere to the following rules:

Associate only one value with one variable.

Separate variable names and values with colons.

Set only one variable per line.

Indicate the end of a line with <lf> or <cr><lf>.

Put the variable and value on the same line, and do not break the line.

You can include white space before or after a variable or value. You can include any character within them. However, if white spaces are needed within the value, you must enclose the value in single or double quotes. If the value is enclosed in quotes, the end quote must be the same as the start quote.

You can include comments after the value. Use the semicolon (;) or pound (#) delimiters to distinguish the comments.

You can include comment lines.

You can include blank lines.

You can use any case for variables; they are not case sensitive.

How to Customize the Default Configuration File

When you reboot a phone on your network, it automatically downloads the default configuration file SIPDefault.cnf from your TFTP server.

This section describes how to customize the SIPDefault.cnf file to allow you to set common parameters on all of the phones on your network. Maintaining parameters—such as whether phones must register with a proxy server and the codec that phones must use when initiating a call—in the default configuration file allows you to perform global changes. For example, you can upgrade the image version without having to customize the phone-specific configuration file for each phone.


Note For a complete alphabetical list of configurable parameters, see "SIP IP Phone Parameters."


Prerequisites

If you have an existing system from a release earlier than Release 7.x, upgrade your system firmware as described in the "How to Upgrade Your Cisco SIP IP Phone Firmware Image and Reboot Remotely" section before proceeding.

Procedure


Step 1 Obtain the default configuration file as follows:

a. Go to the Cisco.com SIP IP 7940/7960 phone software-download site at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.

b. Download the SIPDefault.cnf file to the root directory of your TFTP server or to a subdirectory in which all phone-specific configuration files are stored.

Step 2 Using an ASCII text editor such as vi, open the file.

Step 3 Modify the following required parameters:

image_version—Specifies the firmware version that the Cisco SIP IP phone should run. Enter the name of the image version (as it is released by Cisco). Do not enter the extension.


Note You cannot change the image version by changing the filename because the version is also built into the file header. Trying to change the image version by changing the filename causes the firmware to fail when it compares the version in the header against the filename.


proxy1_address—IP address of the SIP proxy server that is used by the phones. Enter the address in IP dotted-decimal notation or use the FQDN.

tftp_cfg_dir—Specifies the path to the TFTP subdirectory in which phone-specific configuration files are stored. Required only if the phone-specific configuration files are located in a subdirectory.

Step 4 Modify additional parameters as needed.

Step 5 Save the file to the root directory of your TFTP server or to a subdirectory in which all phone-specific configuration files are stored.


Configuration Example

The following is an example of the SIPDefault configuration file that you downloaded from Cisco.com:

# SIP Default Configuration File

# Image Version
image_version: P0S3-06-0-00

# Proxy Server
proxy1_address: 172.16.255.255
proxy2_address:	""; Can be dotted IP or FQDN
proxy3_address:	""; Can be dotted IP or FQDN
proxy4_address:	""; Can be dotted IP or FQDN
proxy5_address:	""; Can be dotted IP or FQDN
proxy6_address:	""; Can be dotted IP or FQDN

# Proxy Server Port (default - 5060)
proxy1_port: 5060
proxy2_port: 5060
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 0

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711ulaw

# TOS bits in media stream [0-5] (Default - 5)
tos_media: 5

# Out of band DTMF Settings
#(none-disable, avt-avt enable (default), avt_always-always avt)
dtmf_outofband: avt

# DTMF dB Level Settings
#(1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3

# SIP Timers
timer_t1: 500; Default 500 msec
timer_t2: 4000; Default 4 sec
sip_retx: 10; Default 10
sip_invite_retx: 6; Default 6
timer_invite_expires: 180 	; Default 180 sec

####### New Parameters added in Release 2.0 #######

# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: ""; Example:  ./sip_phone/

# Time Server
#(There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: ""; SNTP Server IP Address
sntp_mode: anycast (default); unicast, multicast, or directedbroadcast
time_zone: EST; Time Zone Phone is in
dst_offset: 1; Offset from Phone's time when DST is in effect
dst_start_month: April; Month in which DST starts
dst_start_day: ""; Day of month in which DST starts
dst_start_day_of_week: Sun; Day of week in which DST starts
dst_start_week_of_month: 1; Week of month in which DST starts
dst_start_time: 02; Time of day in which DST starts
dst_stop_month: Oct; Month in which DST stops
dst_stop_day: ""; Day of month in which DST stops
dst_stop_day_of_week: Sunday; Day of week in which DST stops
dst_stop_week_of_month: 8; Week of month in which DST stops 8=last week of month
dst_stop_time: 2; Time of day in which DST stops
dst_auto_adjust: 1; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)

# Do Not Disturb Control
#(0-off (default), 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: 0;

# Caller ID Blocking
#(0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: 0; (Default is 0 - disabled and sending all calls as anonymous)

# Anonymous Call Blocking
#(0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0; (Default is 0 - disabled and blocking of anonymous calls)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101; Default 101

# Sync value of the phone used for remote reset
sync: 1; Default 1

####### New Parameters added in Release 2.1 #######

# Backup Proxy Support
proxy_backup: ""; Dotted IP of Backup Proxy
proxy_backup_port: 5060; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: ""; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060; Emergency Proxy port (default is 5060)

# Configurable VAD option
enable_vad: 0; VAD setting 0-disable (Default), 1-enable

####### New Parameters added in Release 2.2 ######

# NAT/Firewall Traversal
nat_enable: 0; 0-Disabled (default), 1-Enabled
nat_address: ""; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060; UDP port used for SIP messages (default - 5060)
start_media_port: 16384; Start RTP range for media (default - 16384)
end_media_port: 32766; End RTP range for media (default - 32766)
nat_received_processing: 0; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support
outbound_proxy: ""; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060; default is 5060

####### New Parameter added in Release 3.0 #######

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: 1; 0-Disabled, 1-Enabled (default)

####### New Parameters added in Release 3.1 #######

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: 1; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to Telnet into the phone)
telnet_level: 1; 0-Disabled (default), 1-Enabled, 2-Privileged

####### New Parameters added in Release 4.0 #######

# XML URLs
services_url: ""; URL for external Phone Services
directory_url: ""; URL for external Directory location
logo_url: ""; URL for branding logo to be used on phone display

# HTTP Proxy Support
http_proxy_addr: ""; Address of HTTP Proxy server
http_proxy_port: 80; Port of HTTP Proxy Server (80-default)

# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""; restricted to dotted IP
dyn_dns_addr_2: ""; restricted to dotted IP
dyn_tftp_addr: ""; restricted to dotted IP

# Remote Party ID
remote_party_id: 0; 0-Disabled (default), 1-Enabled

How to Customize a Phone-Specific Configuration File

You can define parameters that are specific to a particular phone, such as the lines configured on a phone and the defined users for those lines, in a phone-specific configuration file. Parameters defined in the phone-specific configuration file override those specified in the default configuration file.


NoteIf you configure a line to use an e-mail address, that line can be called only by using the e-mail address. Similarly, if you configure a line to use a number, that line can be called only by using the number.

Each line can have a different proxy configured.

Define the dial_template parameter in a phone-specific configuration file only if that phone needs to use a different dial plan than the default.

For a complete alphabetical list of configurable parameters, see "SIP IP Phone Parameters."


Procedure


Step 1 Obtain the phone-specific configuration file as follows:

a. Go to the Cisco.com SIP IP 7940/7960 phone software-download site at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.

b. Download the SIP<mac-addr>.cnf file to the root directory of your TFTP server or to a subdirectory in which all phone-specific configuration files are stored.

Step 2 Do the following for each phone that you plan to install:

a. Using an ASCII text editor such as vi, create and open a SIP<mac-addr>.cnf file for the phone.

b. Modify the following end-user call-preference parameters as needed to permit or deny end-user use or customization:

anonymous_call_block

autocomplete

callerid_blocking

call_hold_ringback

call_waiting

dnd_control

c. Modify the following required parameter:

linex_name—Number or e-mail address used when registering. When entering a number, enter the number without any dashes. For example, enter 555-1212 as 5551212. When entering an e-mail address, enter the e-mail ID without the host name.

d. The following parameters are required for line 1 if registration is enabled and the proxy server requires authentication:

linex_authname—Name used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for this parameter for a line, and registration is enabled, the value defined for line 1 is used. If a value is not defined for line 1, the default line1_authname is UNPROVISIONED.

linex_password—Number or e-mail address used when registering. When entering a number, enter the number without any dashes. For example, enter 555-1212 as 5551212. When entering an e-mail address, enter the e-mail ID without the host name.

e. Save the file to the root directory of your TFTP server or to a subdirectory that contains all the phone-specific configuration files.

Name the file SIP<mac-addr>.cnf. Type the MAC address in uppercase and the extension, cnf, in lowercase (for example, SIP00503EFFD842.cnf).


Configuration Example

The following is an example of the phone-specific configuration file that you downloaded from Cisco.com.

# SIP Configuration Generic File

# Line 1 appearance
line1_name: 1234567

# Line 1 Registration Authentication
line1_authname: "UNPROVISIONED"

# Line 1 Registration Password
line1_password: "UNPROVISIONED"

# Line 2 appearance
line2_name: football

# Line 2 Registration Authentication
line2_authname: "UNPROVISIONED"

# Line 2 Registration Password
line2_password: "UNPROVISIONED"

####### New Parameters added in Release 2.0 #######

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: ""; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "User ID"

# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: ""

####### New Parameters added in Release 3.0 ######

# Phone Prompt (The prompt that will be displayed on console and Telnet)
phone_prompt: "SIP Phone"; Limited to 15 characters (Default - SIP Phone)

# Phone Password (Password to be used for console or Telnet login)
phone_password: "cisco"; Limited to 31 characters (Default - cisco)

# User classification used when Registering [ none (default), phone, ip ]
user_info: none

Configuring the SIP Parameters Manually

After the phone has been connected to power and initialized and the configuration files have been downloaded, you can modify your configuration using the phone's Settings menus.

This section contains the following procedures, which can be executed in the order listed:

Unlocking the Phone

Setting and Restoring Network Parameters

Setting and Restoring Phone-Specific Parameters

Setting End-User Call Preferences


TipTo select a parameter, press the down arrow to scroll to and highlight the parameter, or press the number that represents the parameter (located to the left of the parameter on the LCD).

During configuration, use * for dots (periods) or press the "." softkey when available on the LCD.

During configuration:

To enter a number, press the Number softkey. To enter a name, press the Alpha softkey.

To enter a new value, use the buttons on the dial pad.

If entering letters, use the numbers on the dial pad that are associated with a particular letter. For example, the 2 key has the letters A, B, and C. For a lowercase a, press the 2 key once. To scroll through the available letters and numbers, press the key repeatedly.

To delete any mistakes, press the << softkey.

To cancel all changes and exit a menu during configuration, press Cancel.

After editing a parameter, press the Validate softkey to save the value that you have entered and exit the Edit panel.


Unlocking the Phone

By default, the Localization, Network, and SIP menus on the Cisco IP Phone 7960G/7940G are locked. You must unlock these menus before you can edit the values of parameters in them.

When the phone displays a menu that is locked, a padlock icon appears at the end of the menu's name. If the menu is unlocked, the padlock icon appears open.

When you exit the Settings menu, lockable menus are automatically relocked.

Prerequisites

Set the phone password with the phone_password parameter in the phone-specific configuration file.

To unlock the phone, follow these steps:

Procedure


Step 1 Press Settings > Unlock Config.

The password prompt appears.

Step 2 Enter the phone's password.

Step 3 Press Accept.

Locked menus are unlocked. The Unlock Config menu choice changes to Lock Config and lockable menus remain unlocked while you work within the Settings menu.


Note When you exit the Settings menu, lockable menus automatically relock.


Step 4 To manually relock the phone, select Settings > Lock Config.


Setting and Restoring Network Parameters

To modify network parameters using the phone menus, follow these steps:

Procedure


Step 1 Unlock the Network Configuration menu.

See the "Unlocking the Phone" section.

Step 2 Select Settings > Network Configuration.

The Network Configuration menu appears.

Step 3 Select the parameters you want to change and set them to the desired values.

Network parameters that can be modified are listed in Table 3-1. Network parameters that cannot be modified are listed in Table 3-2. For a complete alphabetical list of configurable parameters, see "SIP IP Phone Parameters."

Step 4 Select Save.

The phone programs the new information into flash memory and resets.

Step 5 Relock the phone.


Table 3-1 Network Parameters That Can Be Modified 

Parameter
Description

Admin. VLAN Id1

Unique identifier of the VLAN to which the phone is attached (for use in switched networks that are not Cisco networks).

Alternate TFTP

Whether to use an alternate remote TFTP server rather than the local one.

Valid values are Yes and No. If you set this parameter to Yes, you must change the IP address in the TFTP server parameter to the address of the alternate TFTP server. Default is No.

Default Router 1 to 52

IP address (1) of the default gateway used by the phone and (2 to 5) of the gateways that the phone attempts to use as an alternate gateway if the default gateway is unavailable.

DHCP Address Released

Whether the IP address of the phone can be released for reuse in the network.

Valid values are Yes and No. When set to Yes, the phone sends a DHCP release message to the DHCP server and goes into a release state. The release state provides enough time to remove the phone from the network before the phone attempts to acquire another IP address from the DHCP server. When you move the phone to a new network segment, first release the DHCP address.

DHCP Enabled

Whether the phone uses DHCP to configure network settings (IP address, subnet mask, domain name, default router list, DNS server list, and TFTP address). Valid values are Yes and No. Default is Yes.

To manually configure your IP settings, including the TFTP server's IP address, set this parameter to No. If not providing the TFTP server's IP address manually, set DHCP to Yes.

DNS Servers 1 to 52

IP address of the DNS server used by the phone to resolve names to IP addresses. The phone attempts to use DNS servers 2 to 5 if DNS server 1 is unavailable.

Domain Name

Name of the DNS domain in which the phone resides.

Erase Configuration

Whether to erase all of the locally defined network settings on the phone and reset the values to the defaults.

Valid values are Yes and No. Yes reenables DHCP. For information on erasing the local configuration, see the "Setting and Restoring Network Parameters" section.

GARP Enabled

Enables or disables generation of the Gratuitous ARP packets from the phone.

HTTP Proxy Address

IP address of the HTTP proxy server. You can use either a dotted IP address or a DNS name (a record only).

HTTP Proxy Port

Port number of the outbound proxy port. Default is 80.

IP Address2

IP address of the phone that is assigned by DHCP or that is locally configured.

Network Media Type

Ethernet port negotiation mode. Valid values are as follows:

Auto—Port is autonegotiated.

Full-100—Port is configured to be a full-duplex, 100-MB connection.

Half-100—Port is configured to be a half-duplex, 100-MB connection.

Full-10—Port is configured to be a full-duplex, 10-MB connection.

Half-10—Port is configured to be a half-duplex, 10-MB connection.

Default is Auto.

Network Port 2 Device Type

Device type that is connected to port 2 of the phone. Valid values are Hub/Switch and PC. Default is Hub/Switch.

Note If the value is PC, port 2 can be connected only to a PC. If you are not sure about the connection, use the default value. Using a value of PC and connecting port 2 to a switch could result in spanning-tree loops and network confusion.

Subnet Mask2

IP subnet mask used by the phone. A subnet mask partitions the IP address into a network and a host identifier.

TFTP Server2

IP address of the TFTP server.

1 If you have an administrative VLAN setting assigned on the Cisco Catalyst switch, that setting overrides any changes made on the phone.

2 DHCP must be disabled.


Table 3-2 Network Parameters That Cannot Be Modified 

Parameter
Description

DHCP Server

IP address of the DHCP server from which the phone received its IP address and additional network settings.

Dynamic DNS Server 1 and 2

IP address of a dynamic DNS server.

Dynamic TFTP Server

IP address of a dynamic TFTP server.

Host Name

Unique host name assigned to the phone.

MAC Address

Factory-assigned unique 48-bit hexadecimal MAC address of the phone.

Operational VLAN Id

Unique identifier of the VLAN of which the phone is a member.


To restore network parameters, follow these steps:

Procedure


Step 1 Unlock the Network Configuration menu.

See the "Unlocking the Phone" section.

Step 2 Select Settings > Network Configuration.

The Network Configuration menu appears.

Step 3 To restore all parameters to their defaults, select Erase Config > Yes.


Note If DHCP is disabled on a phone, restoring default phone settings reenables DHCP.


Step 4 Select Save.

The phone programs the new information into flash memory and resets.


Configuring Network Parameters Manually

If you are not using DHCP to configure your network parameters, you must manually configure them.

Prerequisites

1. Connect your phone as described inthe "Installing the Phone on the Wall" section.

2. Unlock configuration mode as described in the "Unlocking the Phone" section. By default, the network parameters are locked to ensure that end users cannot modify settings that might affect their network connectivity.

3. Review the guidelines on using the Cisco SIP IP phone menus.

4. If configuring a domain name:

a. Press the Number softkey to enter a numerical ID or press the Alpha softkey to enter a name.

b. If entering letters, use the numbers on the dial pad associated with a particular letter. For example, the 2 key has the letters A, B, and C. For a lowercase a, press the 2 key once. To scroll through the available letters and numbers, press the key repeatedly.

c. Press the << softkey to delete any mistakes.

For a complete list of the SIP parameters that you can configure, see "SIP IP Phone Parameters."

To manually configure your phone's network parameters, follow these steps:

Procedure


Step 1 Press the Settings key.

The Settings menu is displayed.

Step 2 Highlight Network Configuration.

Step 3 Press the Select softkey.

The Network Configuration menu is displayed.

Step 4 Highlight DHCP Enabled.

Step 5 Press the No softkey.

DHCP is now disabled.

Step 6 Highlight and configure each of the following parameters:

IP Address—IP address of the phone.

Subnet Mask—IP subnet mask used by the phone.

TFTP Server—IP address of the TFTP server from which the phone downloads its configuration files and firmware images.

Default routers 1 through 5—IP address of the default gateway used by the phone. Default routers 2 through 5 are the IP addresses of the gateways that the phone attempts to use as an alternate gateway if the primary gateway is not available.

Domain Name—Name of the DNS domain in which the phone resides.

DNS servers 1 through 5—IP address of the DNS server used by the phone to resolve names to IP addresses. The phone attempts to use DNS servers 2 through 5 if DNS server 1 is unavailable.

Step 7 Press the Save softkey.

The phone programs the new information into flash memory and resets.


Setting and Restoring Phone-Specific Parameters

Phone users can modify the phone-specific configuration settings using the phone menus. If a phone-specific configuration file exists, the phone uses locally configured parameters until the next reboot. If a phone-specific configuration file does not exist, you must configure the phone locally with parameters specific to that phone.

To configure the preferred codec and out-of-band DTMF parameters, press Change until the option appears, then press Save.

If your system has been set up to have the phones retrieve the configuration file from a TFTP server, you must use the server's configuration file to change the parameter value to a null value (" ") or to "UNPROVISIONED." The phone uses the setting for that variable that it has stored in flash memory.

If the telnet_level parameter is set to allow privileged commands to be executed, the entire SIP configuration can be erased. Use the erase_protflash command so that the phone can retrieve its configuration files.

Prerequisites

Define the line parameters (those identified as linex) on the phone. If you configure a line to use an e-mail address, that line can be called only by using an e-mail address. Similarly, if you configure a line to use a number, that line can be called only by using the number.

To set phone-specific parameters, follow these steps:

Procedure


Step 1 Unlock the phone.

See "Unlocking the Phone" section.

Step 2 Select Settings > SIP Configuration.

The SIP Configuration menu appears.

Step 3 Set the required parameters.

To set a required parameter, select it and set it as desired. The following are required parameters that you must set now if you did not set them in the default configuration file as described in the "How to Customize the Default Configuration File" section:

line1_name—Number or e-mail address for use when registering. Enter a number without dashes. For example, enter 555-0100 as 5550100. Enter an e-mail ID without the host name.

proxy1_address—IP address of the SIP proxy server that is used by the phones. Enter the address in IP dotted-decimal notation or use the FQDN. The "x" argument is representative of server addresses. If the parameter is provisioned with an FQDN, the phone sends REGISTER and INVITE messages by using the FQDN in the Req-URI, To, and From fields.

proxy1_port—Port number of the SIP proxy server that is used by line 1.

If the proxy server with which the phone communicates has authentication enabled, set the following parameters as well:

line1_authname—Name used by the phone for authentication if a registration is challenged by the proxy server during initialization. Default is UNPROVISIONED.

line1_password—Password used by the phone for authentication if a registration is challenged by the proxy server during initialization. Default is UNPROVISIONED.

Step 4 Set additional parameters as needed.

To set a parameter, select it and set it as desired. Phone-specific parameters are listed in Table 3-3.

Step 5 Select Save.

The phone programs the new information into flash memory and resets.


To restore parameters to default values, follow these steps:

Procedure


Step 1 Unlock the phone.

See the "Unlocking the Phone" section.

Step 2 Select Settings > SIP Configuration.

The SIP Configuration menu appears.

Step 3 Highlight the parameter whose setting you want to restore

Step 4 Select Edit followed by <--.

Step 5 Select Accept.

Step 6 If necessary, select Back to exit the menu.


Table 3-3 Phone-Specific Parameters 

Parameter
Description

Authentication Name1

Name used by the phone for authentication if a registration is challenged by the proxy server during initialization.

Authentication Password1

Password used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for the Authentication Password parameter when registration is enabled, the default logical password is used. The default logical password is SIPmac-address, where mac-address is the MAC address of the phone.

Display Name

Identification as it should appear for caller identification. For example, instead of jdoe@company.com appearing on phones that have caller ID, you can specify John Doe in this parameter to have John Doe appear on the callee end instead. If a value is not specified for this parameter, the Name value is used.

Name

Description phone number or e-mail address used when registering. When entering a number, enter the number without any dashes. For example, enter 555-0100 as 5550100.

Proxy Address

IP address of the primary SIP proxy server that will be used by the phone. Enter this address in IP dotted-decimal notation.

Proxy Port

Port number of the primary SIP proxy server. This is the port that the SIP client will use. The default is 5060.

Short Name

Name or number associated with the linex_name as you want it to display on the phone LCD if the linex_name value exceeds the display area. For example, if the linex_name value is the phone number 111-222-333-4444, you can specify 34444 for this parameter to have 34444 display on the LCD instead. Alternatively, if the value for the linex_name parameter is the e-mail address "username@company.com," you can specify the "username" to have just the username appear on the LCD instead. This parameter is used for display only. If a value is not specified for this parameter, the value in the Name variable is displayed.

1 Required when registration is enabled and the registrar challenges registration.


Setting End-User Call Preferences

End users can modify call preferences from their own phones by setting certain parameters. Only call preferences whose configuration variable has been set to 0 or 1 can be modified. If the configuration variables are set to 2 or 3, the corresponding call preferences cannot be modified with the Call Preferences menu. See the following procedure for more information.

To set end-user call preferences, follow these steps:

Procedure


Step 1 On the IP phone, select Settings > Call Preferences.

Step 2 Highlight and set any of the following optional preferences by pressing the Yes or No softkeys:

anonymous_call_block—Allow (default) or block incoming anonymous calls.

auto_answer—Define the intercom line number (cannot be set in configuration file).

autocomplete—Enable or disable (default) automatic completion of stored numbers. Specific to phone.

callerid_blocking—Enable or disable (default) caller ID blocking. Specific to phone.

call_hold_ringback—Enable or disable (default) a ringback from calls on hold after you hang up another call. Specific to phone.

call_waiting—Enable (default) or disable acceptance of call waiting calls. Specific to phone.

dnd_control—Enable or disable (default) the Do Not Disturb feature. Specific to phone.

stutter_msg_waiting—Enable or disable (default) a stutter tone when a message is waiting.

For more information on the values each of these parameters can take, see "SIP IP Phone Parameters." For more information on setting parameters, see the "How to Customize a Phone-Specific Configuration File" section.

Step 3 Press the Save softkey.

The phone programs the new information into flash memory and resets.


How to Set the Date and Time

You can set date, time, and daylight savings time (DST) parameters. The current date and time is supported on the Cisco Unified IP Phone 7960G and 7940G using Simple Network Time Protocol (SNTP) and is displayed on the LCD. DST and time-zone settings are also supported.

International time-zone abbreviations are supported and must be in all capital letters.


Note We recommend that you set date- and time-related parameters in the default file for all phones. Alternatively, you can set the time-zone parameter manually on the phone or in the phone-specific configuration files.


Prerequisites

Determine the type of DST that you want to configure:

Absolute DST (for example, starts on April 1 and ends on October 1)

Relative DST (for example, starts on the first Sunday in April and ends on the last Sunday of October)

Review the list of common and absolute DST parameters from "SIP IP Phone Parameters."

Review the information on SNTP in Table 3-4. SNTP parameters specify how the phone obtains the current time from an SNTP server.

Determine your time zone from Table 3-5.

Procedure


Step 1 Using an ASCII text editor such as vi, open the SIPDefault.cnf file.

Step 2 Modify the following SNTP parameters as needed:

sntp_mode

sntp_server

time_zone

Step 3 Modify the following common DST parameters as needed:

dst_offset

dst_auto_adjust

dst_start_month

dst_stop_month

dst_start_time

dst_stop_time

Step 4 Do one of the following:

Modify the following absolute DST parameters as needed:

dst_start_day

dst_stop_day

Modify the following relative DST parameters as needed:

dst_start_day_of_week

dst_start_week_of_month

dst_stop_day_of_week

dst_stop_week_of_month

Step 5 Save the file to the root tftp directory of your TFTP server.


Note To adjust the phone display to European Day-Month-Year format, add the following entry to the SIPDefault.cnf file: date_format:D/M/Y.



Table 3-4 describes the effects on SNTP mode when the SNTP server is null (not assigned an IP address) or when it is assigned a valid IP address.

Table 3-4 Effects on SNTP Mode 

SNTP Server
SNTP Mode
Unicast
Multicast
Anycast 1
Directed Broadcast
SNTP server parameter is null

Sends

No known server with which to communicate.

SNTP requests are not sent.

SNTP packet to the local network broadcast address.

After the first SNTP response is received, the phone switches to unicast mode with the server being set as the one who first responded.

SNTP packet to the local network broadcast address.

After the first SNTP response is received, the phone switches to multicast mode.

Receives

No known server with which to communicate.

Multicast data using the SNTP/NTP multicast address from the local network broadcast address from any server on the network.

Unicast SNTP data from the SNTP server that first responded to the network broadcast request.

SNTP data from the SNTP/NTP multicast address and the local network broadcast address from any server on the network.

SNTP server parameter is a valid IP address.

Sends

SNTP request to the SNTP server.

SNTP requests are not sent.

If the mode is anycast and the SNTP server parameter is a valid IP address, the phone sends the request to the broadcast address in version 7.4.

SNTP packet to the SNTP server.

After the first SNTP response is received, the phone switches to multicast mode.

Receives

SNTP response from the SNTP server and ignores responses from other SNTP servers.

SNTP data via the SNTP/NTP multicast address from the local network broadcast address.

SNTP response from the SNTP server and ignores responses from other SNTP servers.

SNTP data from the SNTP/NTP multicast address and the local network broadcast address and ignores responses from other SNTP servers.

1 If sntp_mode is set to anycast, the sntp_server address will be ignored and subsequent sntp requests will be sent to the first

1 sntp server that responded (the first sntp request must be unconditionally sent to the broadcast address).


Table 3-5 includes the time-zone information that you need to configure the SNTP mode and server parameters.

Table 3-5 Time-Zone Information 

Abbreviation
GMT Offset
Cities
Time-Zone Names

IDL

GMT-12:00

Eniwetok

IDL (International Date Line),
IDLW (International Date Line West)

NT

GMT-11:00

Midway

BT (Bering Time), NT (Nome Time)

AHST

GMT-10:00

Hawaii

AHST (Alaska-Hawaii Standard Time),
HST (Hawaiian Standard Time),
CAT (Central Alaska Time)

IMT

GMT-09:30

Isle Marquises

Isle Marquises

YST

GMT-09:00

Yukon

YST (Yukon Standard Time)

PST

GMT-08:00

Los Angeles

PST (Pacific Standard Time)

MST

GMT-07:00

Phoenix

MST (Mountain Standard Time),
PDT (Pacific Daylight Time)

CST

GMT-06:00

Dallas, Mexico City

CST (Central Standard Time),
MDT (Mountain Daylight Time),
Chicago

EST

GMT-05:00

New York

EST (Eastern Standard Time),
CDT (Central Daylight Time),
NYC

AST

GMT-04:00

La Paz

AST (Atlantic Standard Time),
EDT (Eastern Daylight Time)

NST

GMT-03:30

Newfoundland

NST (Newfoundland Standard Time)

BST

GMT-03:00

Buenos Aires

BST (Brazil Standard Time),
ADT (Atlantic Daylight Time),
GST (Greenland Standard Time)

AT

GMT-02:00

Mid-Atlantic

AT (Azores Time)

WAT

GMT-01:00

Azores

WAT (West Africa Time)

GMT

GMT 00:00

London

GMT (Greenwich Mean Time),
WET (Western European Time),
UT (Universal Time)

CET

GMT+01:00

Paris

CET (Central European Time),
MET (Middle European Time),
BST (British Summer Time),
MEWT (Middle European Winter Time),
SWT (Swedish Winter Time),
FWT (French Winter Time)

EET

GMT+02:00

Athens, Rome

EET (Eastern European Time),
USSR-zone1,
MEST (Middle European Summer Time),
FST (French Summer Time)

BT

GMT+03:00

Baghdad, Moscow

BT (Baghdad Time), USSR-zone2

IT

GMT+03:30

Tehran

IT (Iran Time)

ZP4

GMT+04:00

Abu Dhabi

USSR-zone3,
ZP4 (GMT Plus 4 Hours)

AFG

GMT+04:30

Kabul

Afghanistan

ZP5

GMT+05:00

Islamabad

USSR-zone4,
ZP5 (GMT Plus 5 Hours)

IST

GMT+05:30

Bombay, Delhi

IST (Indian Standard Time)

ZP6

GMT+06:00

Colombo

USSR-zone5,
ZP6 (GMT Plus 6 Hours)

SUM

GMT+06:30

North Sumatra

NST (North Sumatra Time)

WAST

GMT+07:00

Bangkok, Hanoi

SST (South Sumatra Time),
USSR-zone6,
WAST (West Australian Standard Time)

HST

GMT+08:00

Beijing, Hong Kong

CCT (China Coast Time),
HST (Hong Kong Standard Time), USSR-zone7,
WADT (West Australian Daylight Time)

JST

GMT+09:00

Tokyo, Seoul

JST (Japan Standard Time/Tokyo),
KST (Korean Standard Time),
SSR-zone8

CAST

GMT+09:30

Darwin

SAST (South Australian Standard Time), CAST (Central Australian Standard Time)

EAST

GMT+10:00

Brisbane, Guam

GST (Guam Standard Time),
USSR-zone9,
EAST (East Australian Standard Time)

EADT

GMT+11:00

Solomon Islands

USSR-zone10,
EADT (East Australian Daylight Time)

NZST

GMT+12:00

Auckland

NZT (New Zealand Time/Auckland),
NZST (New Zealand Standard Time),
IDLE (International Date Line East)


Time-Zone Configuration Examples

Absolute DST Configuration

The following is an example of an absolute DST configuration:

time_zone : PST
dst_offset : 01/00
dst_start_month : April
dst_start_day : 1
dst_start_time : 02/00
dst_stop_month : October
dst_stop_day : 1
dst_stop_time : 02/00
dst_stop_autoadjust : 1

Relative DST Configuration

The following is an example of a relative DST configuration:

time_zone : PST
dst_offset : 01/00
dst_start_month : April
dst_start_day : 0
dst_start_day_of_week : Sunday
dst_start_week_of_month : 1
dst_start_time : 02/00
dst_stop_month : October
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 8
dst_stop_time : 02/00
dst_stop_autoadjust : 1

How to Create Dial Plans

Dial plans enable the Cisco SIP IP phone to support automatic dialing and generation of a secondary dial tone. If a single dial plan is used for a system of phones, the dial plan is best specified in the default configuration file.

However, you can also create multiple dial plans and specify which phones are to use which dial plan by defining the dial_template parameter in the phone-specific configuration file. If one phone in a system of phones needs to use a different dial plan than the rest, you need to define a dial plan for that phone in its phone-specific configuration file.

Special Characters

You can specify the pound sign (#) and asterisk (*) as dialed digits, if needed.

The # is processed as a "dial now" event by default. You can override this by specifying # in the dial-plan template, in which case the phone does not dial immediately when the # is pressed but does continue to match the dial-plan template that specifies the #. The # is not matched by the wildcard character * or the period (.).

The * is processed as a wildcard character. You can override this by preceding the * with the backward slash (\) escape sequence, resulting in the sequence \*. The phone automatically strips the \ so that it does not appear in the outgoing dial string. When * is received as a dialed digit, it is matched by the wildcard characters * and period (.).

You can also specify the comma (,) as a secondary dial tone, if needed, and you can set which tones are played. Choose from the following tone names:

Bellcore-Alerting

Bellcore-None

CallWaiting-3

Bellcore-Busy

Bellcore-Outside (default)

CallWaiting-4

Bellcore-BusyVerify

Bellcore-Permanent

Cisco-BeepBonk

Bellcore-CallWaiting

Bellcore-Reminder

Cisco-Zip

Bellcore-Confirmation

Bellcore-Reorder

Cisco-ZipZip

Bellcore-Hold

Bellcore-Stutter

 

Bellcore-Inside

CallWaiting-2

 

Tone names are case insensitive. All tone names should begin with a common prefix.


Note For more information on Bellcore tones, refer to Bellcore GR-506-CORE. For more information on tones in BTS 10200 Softswitch features, refer to the Cisco BTS 10200 Softswitch website at http://www.cisco.com/en/US/partner/products/hw/vcallcon/ps531/index.html.


Prerequisites

Ensure that your dial plans adhere to the following:

They are written with the understanding that rules are matched from start to finish with the longest matching rule taken as the one to use. Matches against a period are not counted as part of the longest length.

They are in XML format.

They are stored on your TFTP server.

Specify which dial plan a phone is to use by specifying the path to the dial plan in the dial_template parameter. Define the dial_template parameter in either the default configuration file or a phone-specific configuration file.


Note To simplify maintenance and control, define this parameter in the default configuration file. Define it in a phone-specific configuration file only if that phone needs to use a different dial plan than the one being used by the other phones in the same system.


Procedure


Step 1 Using an ASCII text editor such as vi, open a new file.

Step 2 Type the following to indicate the start of the dial-plan template:

<DIALTEMPLATE>

Step 3 For each of the numbering schemes that you require, add the following string to the template, each starting on a separate line:

TEMPLATE MATCH="pattern" Timeout="sec" User="type" Rewrite="xxx" Route="route" Tone="tone"

Arguments are as follows:

Argument
Description
TEMPLATE MATCH="pattern"

Dial-pattern string to match. You can use the following wildcards:

A period (.) matches any character.

An asterisk (*) matches one or more characters.

A comma (,) causes the phone to generate a secondary dial tone.

Timeout="sec"

Time, in seconds, before the system times out and dials the number as entered by the user. To have the number dial immediately, specify 0.

User="type"

Tag to be added automatically to the dialed number. Valid values are IP and Phone. This tag is not case sensitive.

Rewrite="xxx"

Alternate string to be dialed instead of the number that the user dials. The following apply:

Rewrite rules are matched from start to finish with the longest matching rule taken as the one to use.

Matches against a period are not counted as part of the length.

A complete rule is matched only if it has more nonwildcard matches than an incomplete rule.

Comments are allowed with the following syntax:

<!-- comment -->

Rules allow for substitution of up to five replacement strings as well as picking off of replaced digits one at a time.

For example, a match string of ab..cd..ef* and an input string of ab12cd34ef5678 result in the following.

Rewrite
Output
Notes
 
%s
ab12cd34ef5678

 
%0
ab12cd34ef5678

 
%1
12


%2
34

 
%3
5678

 
%4

Null output.

 
%5

Null output.

 
XYZ....
XYZ1234

 
X.Y.Z...
X1Y2Z345

 
919%1%2%3
91912345678

 
AB...X%1X..
AB123X12X45

12 appears twice.

 
X%1X%1X%1
X12X12X12

You can reuse the string.

 
X%s%%
Xab12cd34ef5678%

%% produces a %.

 
919
919

No need to use the input.

 
............
12345678

Nothing goes in for the extra dots.

Route="route"

Proxy to which to route the call. Valid values are default, emergency, and FQDN. FQDN is treated the same as default proxy. This entry is not case sensitive.

Tone="tone"

User-specific dial tone. If no tone is specified, the default secondary dial tone plays. If a comma (,) is specified followed by a tone, the phone plays the indicated tone instead of the secondary dial tone. If a tone is specified but there is no comma, the tone is ignored.

You can specify up to three different secondary dial tones in a single dial-plan template. The tones play in the order in which they appear in the template. Multiple tone entries are condensed into a single comma. For example, the phone interprets the match string 9,,,234 as 9,234 and treats the three commas as a single comma.


Step 4 If desired, specify a comment at the end of each string to denote the type of plan.

For example:

<!-- Long Distance --> or <!-- Corporate Dial Plan -->

Step 5 Type the following to indicate the end of the dial-plan template:

</DIALTEMPLATE>

Step 6 Give the file a unique name specific to the dial plan that it defines and save it with a .xml extension to your TFTP server.

Step 7 If the dial plan applies to a specific phone, add the path to the dial plan (without specifying the file type of .xml) via the dial_template parameter in the phone-specific configuration file. If the dial plan applies to a system of phones, add the path to the dial plan via the dial_template parameter in the default configuration file.


Dial-Plan Configuration Examples

Using the Pound-Sign (#) Character

The following example uses the pound sign (#) as a dialed digit:

<DIALTEMPLATE>
	<TEMPLATE MATCH="123#45#6" TIMEOUT="0" User="Phone"/> <!-- Match `#' -->
	<TEMPLATE MATCH="34#..."    TIMEOUT="0" User="Phone"/> <!-- Match `#' -->
	<TEMPLATE MATCH="*"      TIMEOUT="15" User="Phone"/>
</DIALTEMPLATE>

In the example above, the 123#45#6 string is matched if the user dials 123#45#6. Pressing the pound sign (#) does not cause the phone to dial immediately because # is explicitly specified. However, dialing 1# or 123#4# causes the phone to dial immediately.

Using the Backward-Slash (\) and Asterisk (*) Characters

The following example uses the backward slash (\) and asterisk (*) as a dialed digit:

<DIALTEMPLATE>
	<TEMPLATE MATCH="12\*345" TIMEOUT="0" User="Phone"/> <!-- Match * Char -->
	<TEMPLATE MATCH="*"        TIMEOUT="10" User="Phone"/> <!-- Wildcard -->
</DIALTEMPLATE>

If you use the backslash (\) on a character other than the asterisk (*), the \ is ignored and the \\ character is matched. If you need to explicitly specify the \ character in a dial plan, use \\. The \ is not sent out as part of the dialed digit string because the phone removes it before sending the dial string.


Note The \* character is matched by the "." character.


Specifying a Secondary Dial Tone

The following example specifies two different tones:

<DIALTEMPLATE>
	<TEMPLATE MATCH="7,..." TIMEOUT="0" /> <!-- Default Secondary Dial Tone -->
	<TEMPLATE MATCH="9,..." TIMEOUT="0" Tone="Zip" /> <!-- Play Zip Tone -->
	<TEMPLATE MATCH="8,...." TIMEOUT="0" Tone="Hold" /> <!-- Play Hold Tone -->
	<TEMPLATE MATCH="8,123,...." TIMEOUT="0" Tone="Hold" Tone="Zip" /> <!--Play Hold Tone 
                                 after 8, Play Zip Tone after 123-->
</DIALTEMPLATE>

How to Verify Initialization

The initialization process establishes network connectivity and makes the phone operational in your IP network.

Procedure


Step 1 After the phone has power connected to it, ensure that the phone cycles through the following steps:

a. The following flash on and off in sequence: Headset button, Mute button, and Speaker button.

b. The Cisco Systems, Inc. copyright appears on the LCD.

c. The following messages appear:

Configuring VLAN—The phone configures the Ethernet connection.

Configuring IP—The phone contacts the DHCP server to obtain network parameters and the IP address of the TFTP server.

Requesting Configuration—The phone contacts the TFTP server to request its configuration files and compares firmware images.

Upgrading Software—The phone displays this message only if it determines that an image upgrade is required. After upgrading the image, the phone automatically reboots to run the new image.

d. The main LCD displays the following:

Primary directory number

Softkeys

If the phone successfully cycles through these steps, it has started up properly.


Where to Go Next

See "Managing Cisco SIP IP Phones," for information on upgrading firmware and performing other management tasks.

See "Monitoring Cisco SIP IP Phones," for information on debugging and on viewing network statistics.