Cisco SIP IP Phone Administrator Guide, Release 5.1
Managing Cisco SIP IP Phones (Version 5.1)
Downloads: This chapterpdf (PDF - 537.0KB) The complete bookPDF (PDF - 2.31MB) | Feedback

Managing Cisco SIP IP Phones

Table Of Contents

Managing Cisco SIP IP Phones

Changing Your Configuration

Modifying the Network Settings

Entering Configuration Mode

Unlocking Configuration Mode

In Release 4.2 or Later

In Release 4.1 or Earlier

Locking Configuration Mode

In Release 4.2 or Later

In Release 4.1 or Earlier

Changing the Network Settings

Modifying the SIP Settings

Modifying SIP Parameters via a TFTP Server

Modifying the Default SIP Configuration File

Modifying the Phone-Specific SIP Configuration File

Modifying the SIP Parameters Directly on Your Phone

Modifying Call Preferences

Setting the Date, Time, and Daylight Saving Time

Erasing the Locally Defined Settings

Erasing the Locally Defined Network Settings

Erasing the Locally Defined SIP Settings

Viewing the Firmware Version

Upgrading the Cisco SIP IP Phone Firmware

Upgrading to Release 5.0 and Release 5.1

Upgrading from Release 2.2 or Later to Current Release

Upgrading from Release 2.1 or Earlier to Current Release

Dual Booting from SCCP or MGCP to a SIP Release

Performing an Image Upgrade and Remote Reboot


Managing Cisco SIP IP Phones


This chapter provides information on the following:

Changing Your Configuration

Modifying the Network Settings

Modifying the SIP Settings

Modifying Call Preferences

Setting the Date, Time, and Daylight Saving Time

Erasing the Locally Defined Settings

Viewing the Firmware Version

Upgrading the Cisco SIP IP Phone Firmware

Performing an Image Upgrade and Remote Reboot

Changing Your Configuration

You can change your Cisco SIP IP phone configuration by any of the following methods:

Using your phone buttons and soft keys. You must first follow the instructions in the "Entering Configuration Mode" section.

Editing the default and phone-specific configuration files on the TFTP server. See the "Modifying SIP Parameters via a TFTP Server" section.

Using Telnet or a console to connect to your Cisco SIP IP phone and using the command-line interface (CLI). You will need to the phone IP address. Press Settings, select Network Configuration, and scroll down to IP Address to find this address. The default Telnet password is "cisco."


Note Use the CLI only to debug and troubleshoot your Cisco SIP IP phone.


You can change the following parameters:

Network settings. See the "Modifying the Network Settings" section.

SIP settings. See the "Modifying the SIP Settings" section.

Call preferences settings. See the "Modifying the SIP Settings" section.

XML URL settings. See the "Modifying the SIP Settings" section.

Date, time, and daylight saving time settings. See the "Setting the Date, Time, and Daylight Saving Time" section.

Modifying the Network Settings

You can display and configure the network settings of a Cisco SIP IP phone. The network settings include information such as the phone's Dynamic Host Configuration Protocol (DHCP) server, MAC address, IP address, and domain name.

Entering Configuration Mode

When you access the network configuration information on your Cisco SIP IP phone, you will notice that there is a padlock symbol located in the upper-right corner of your LCD. By default, the network configuration information is locked. Before you can modify any of the network configuration parameters, you must unlock the phone.

Unlocking Configuration Mode

There are two methods to unlock the configuration mode in Cisco SIP IP phones: one method for phones that have Release 4.2 or later and one method for phones that have Release 4.1 or earlier.

In Release 4.2 or Later

In Release 4.2 and later, an "Unlock Config" item displays in the phone settings menu. When the user selects Unlock Config, the user is prompted to enter a phone password using the alphanumeric entry function of the keypad. The phone password is set using the phone_password configuration parameter. When the correct password is entered, the configuration is unlocked and the settings can be changed.

When the Network Configuration or SIP Configuration menus display, the lock icon in the upper-right corner of your LCD will indicate an unlocked state. The unlocked symbol indicates that you can modify the network and SIP configuration settings.

When the Settings menu is exited, the phone will automatically relock the configuration.

In Release 4.1 or Earlier

To unlock the Cisco SIP IP phone for releases before Cisco Release 4.2, press **#.


Note Pressing **# activates the configuration mode for your phone; however there is no indication that an action has taken place.


If the Network Configuration or SIP Configuration panel is displayed, the lock icon in the upper-right corner of your LCD changes to an unlocked state. If you are located elsewhere in the Cisco SIP IP phone menus, the next time you access the Network Configuration or the SIP Configuration menus, the unlocked icon displays, and you can modify the network and SIP configuration settings.

Locking Configuration Mode

There are two methods to lock the configuration mode in Cisco SIP IP phones: one method for phones that have Release 4.2 or later and one method for phones that have Release 4.1 or earlier.

In Release 4.2 or Later

When the configuration has been successfully locked, the menu item displayed is "Lock Config." If you select this item, the configuration will relock. Also, if you exit the Settings menu the configuration will relock. Refer to the "Unlocking Configuration Mode" section for more information.

When the Network Configuration or SIP Configuration menus display, the lock icon in the upper-right corner of your LCD will indicate a locked state. The lock symbol indicates that you cannot modify the network and SIP configuration settings.

In Release 4.1 or Earlier

To lock the Cisco SIP IP phone when you are done modifying the settings, press **#.

If the Network Configuration or SIP Configuration panel is displayed, the lock icon in the upper-right corner of your LCD changes to a locked state. If you are located elsewhere in the Cisco SIP IP phone menus, the next time you access the Network Configuration or the SIP Configuration panels, the lock icon will be displayed in a locked state.

The lock symbol indicates that you cannot modify the network and SIP configuration settings.

Changing the Network Settings

Before You Begin

When configuring network settings, remember the following:

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

Review the guidelines on using the Cisco SIP IP phone menus documented in the "Using the Cisco SIP IP Phone Menu Interface" section on page 2-15.

After making your changes, relock configuration mode as described in the "Locking Configuration Mode" section.

To change the network settings, perform the following steps:


Step 1 Press the settings key. The Settings menu displays.

Step 2 Highlight Network Configuration.

Step 3 Press the Select soft key. The Network Configuration menu displays. Table 3-1 lists the network parameters available from the Network Configuration menu.

Step 4 When done, press the Save soft key. The phone programs the new information into Flash memory and resets.



Caution When you have completed your changes, ensure that you lock the phone as described in the "Locking Configuration Mode" section.

Table 3-1 Network Configuration Parameters 

Parameter
Can Edit?
Description

Admin. VLAN Id

Yes, but if you have an administrative VLAN assigned on the Catalyst switch, that setting overrides any changes made on the phone.

Unique identifier of the VLAN to which the phone is attached. The value in this field is used only in switched networks that are not Cisco networks.

Alternate TFTP

Yes.

Whether to use an alternate TFTP server. This field enables an administrator to specify the remote TFTP server rather than the local one. Possible values for this parameter are Yes and No. The default is No. When Yes is specified, the IP address in the TFTP Address parameter must be changed to the address of the alternate TFTP server.

Default Routers 1 through 5

Yes, but DHCP must be disabled.

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 unavailable.

DHCP Address Released

Yes.

Whether the IP address of the phone can be released for reuse in the network. When you set this field 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 moving the phone to a new network segment, you should first release the DHCP address.

DHCP Enabled

Yes.

Whether the phone will use DHCP to configure network settings (IP address, subnet mask, domain name, default router list, DNS server list, and TFTP address). Valid values for this field are Yes and No. By default, DHCP is enabled on the phone. To manually configure your IP settings, you must first disable DHCP.

DHCP Server

No.

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

DNS Servers 1 through 5

Yes, but DHCP must be disabled.

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.

Domain Name

Yes.

Name of the DNS domain in which the phone resides.

Dynamic DNS Server 1 and 2

No.

You can specify the IP address of a new dynamic DNS server. If a new DNS server is specified, it is used for any further DNS requests after the phone uses the initial DNS address upon bootup. The DNS addresses are used in the following order:

1. dyn_dns_addr_1 (if present)

2. dyn_dns_add_2 (if present)

3. DNS Server 1

4. DNS Server 2

5. DNS Server 3

6. DNS Server 4

7. DNS Server 5

The dynamic DNS address is not stored in Flash memory.

Dynamic TFTP Server

No.

You can specify the IP address of a new dynamic TFTP server. After initially querying the default TFTP server, the phone will rerequest the default and MAC-specific configuration files from the new TFTP server. The dynamic TFTP server is not stored in Flash memory.

Erase Configuration

Yes.

Whether to erase all of the locally defined network settings on the phone and reset the values to the defaults. Selecting Yes reenables DHCP. For more information on erasing the local configuration, see the "Erasing the Locally Defined Settings" section.

Host Name

No.

Unique host name assigned to the phone. The value in this field is always SIPmac where mac is the MAC address of the phone.

HTTP Proxy Address

Yes.

The 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

Yes.

The port number of the outbound proxy port. The default is 80.

IP Address

Yes, but DHCP must be disabled.

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

MAC Address

No.

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

Network Media Type

Yes.

Ethernet port negotiation mode. Valid values are as follows:

Auto—Port is autonegotiated. (This is the default value.)

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.

Network Port 2 Device Type

Yes.

The device type that is connected to port 2 of the phone. Valid values are as follows:

Hub/Switch (default)

PC

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.

Operational VLAN Id

No.

Unique identifier of the VLAN of which the phone is a member. This identifier is obtained through Cisco Discovery Protocol (CDP).

Subnet Mask

Yes, but DHCP must be disabled.

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

TFTP Server

Yes, but DHCP must be disabled.

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


Modifying the SIP Settings

You can modify the SIP parameters of a Cisco SIP IP phone. When modifying SIP parameters, remember the following:

Parameters defined in the default configuration file override the values stored in Flash memory.

Parameters defined in the phone-specific configuration file override the values specified in the default configuration file.

Parameters entered locally are used by the phone until the next reboot if a phone-specific configuration file exists.

If you choose not to configure the phone via a TFTP server, you must manage the phone locally.

Table 3-2 lists each of the SIP parameters that you can configure. In the Configuration File column, the name of a parameter as you would specify it in a configuration file is listed. In the menu columns (SIP Configuration, Network Configuration, Call Preferences, and Time and Date), the name of the same parameter as it would appear on the user interface is listed. If NA appears in a menu column, the parameter cannot be defined using that menu.

Table 3-2 Summary of SIP Parameters 

Configuration File
SIP Configuration Menu
Network Configuration Menu
Call
Preferences
Time and
Date

anonymous_call_block

Anonymous Call Block

autocomplete

Auto-Complete Numbers

callerid_blocking

Caller ID Blocking

call_hold_ringback

Call Hold Ringback

call_waiting

Call Waiting

cnf_join_enable

date_format

Date Format

dial_template

dnd_control

Do Not Disturb

dst_auto_adjust

dst_offset

dst_start_day

dst_start_day_of_week

dst_start_month

dst_start_time

dst_start_week_of_month

dst_stop_day

dst_stop_day_of_week

dst_stop_month

dst_stop_time

dst_stop_week_of_month

dtmf_avt_payload

dtmf_db_level

dtmf_inband

dtmf_outofband

Out of Band DTMF

enable_vad

Enable VAD

end_media_port

End Media Port

image_version

language

linex_authname (line1 to line6)

Authentication Name

linex_displayname (line1 to line6)

Display Name

linex_name (line1 to line6)

Name

linex_password (line1 to line6)

Authentication Password

linex_shortname (line1 to line6)

Shortname

messages_uri

Messages URI

nat_address

NAT Address

nat_enable

NAT Enabled

nat_received_processing

network_media_type

Network Media Type

network_port2_type

Network Port 2 Device Type

outbound_proxy

Outbound Proxy

outbound_proxy_port

Outbound Proxy Port

phone_label

Phone Label

phone_password

phone_prompt

preferred_codec

Preferred Codec

proxy_backup

Backup Proxy

proxy_backup_port

Backup Proxy Port

proxy_emergency

Emergency Proxy

proxy_emergency_port

Emergency Proxy Port

proxy_register

Register with Proxy

proxyN_address (N=1 to 6)

Proxy Address

proxyN_port (N=1 to 6)

Proxy Port

remote_party_id

sip_invite_retx

sip_retx

sntp_mode

sntp_server

start_media_port

Start Media Port

sync

tftp_cfg_dir

TFTP Directory

time_format_24hr

Time format 24-hr

time_zone

Time Zone

timer_invite_expires

timer_register_expires

Register Expires

timer_t1

timer_t2

tos_media

user_info

voip_control_port

VoIP Control Port


Modifying SIP Parameters via a TFTP Server

If you have set up your phones to retrieve their SIP parameters via a TFTP server as described in the "Modifying SIP Parameters via a TFTP Server" section, you can also modify your SIP parameters using the configuration files.

As explained in the "Configuring SIP Parameters" section on page 2-4, there are two configuration files that you can use to define the SIP parameters: the default configuration file and the phone-specific configuration file. If used, the default configuration file must be stored in the root directory of your TFTP server. The phone-specific configuration file can be stored in the root directory of the TFTP server or in a subdirectory in which phone-specific configuration files are stored.

While it is not required, Cisco recommends that you use the default configuration file to define values for SIP parameters that are common to all phones. Doing so will make controlling and maintaining your network easier. You can then define only those parameters that are specific to a phone in the phone-specific configuration file. Phone-specific parameters should be defined only in a phone-specific configuration file, or they should be manually configured. Phone-specific parameters should not be defined in the default configuration file.

Modifying the Default SIP Configuration File

In the default configuration file (SIPDefault.cnf), Cisco recommends that you maintain the SIP parameters that are common to all your phones. By maintaining these parameters in the default configuration file, you can perform global changes, such as upgrading the image version, without having to modify the phone-specific configuration file for each phone.

Before You Begin

Ensure that you have downloaded the SIPDefault.cnf file from Cisco.com to the root directory of your TFTP server.

Review the guidelines documented in the "Configuring SIP Parameters" section on page 2-4.


Note Refer to the "Setting the Date, Time, and Daylight Saving Time" section for more information.



Step 1 Using an ASCII editor, open the SIPDefault.cnf file and define or modify values for the SIP parameters shown in Table 3-3, as necessary.

Step 2 Save the file with the same filename, SIPDefault.cnf, to the root directory of your TFTP server.


Table 3-3 Default SIP Configuration File Parameters 

Parameter
Required or Optional
Description

anonymous_call_block

Optional

Configures anonymous call block. Valid values are as follows:

0—Disabled by default, but can be turned on and off using the user interface. When disabled, anonymous calls are received.

1—Enabled by default, but can be turned on and off using the user interface. When enabled, anonymous calls are rejected

2—Disabled permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

3—Enabled permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

The default value is 0.

autocomplete

Optional

Configures automatic completion of numbers. Valid values are 0 (disable autocompletion) or 1 (enable autocompletion). The default is 1.

call_hold_ringback

Optional

Operates the same way that DND operates. It is selectable by the user in the Services->Call Preferences menu. When this value is enabled, the phone will ring if the handset is placed on-hook and there is a call currently on hold. If the value is disabled, the phone will not ring in this situation. Acceptable values:

0—Off by default, but can be turned on and off locally using the user interface.

1—On by default, but can be turned on and off locally using the user interface.

2—Off permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

3—On permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

The default value is 0.

call_waiting

Optional

Configures call waiting. Valid values are as follows:

0—Disabled by default, but can be turned on and off using the user interface. When disabled, call waiting calls are not received.

1—Enabled by default, but can be turned on and off using the user interface. When enabled, call waiting calls are accepted.

2—Disabled permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

3—Enabled permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

The default value is 1.

callerid_blocking

Optional

Configures caller ID blocking. When enabled, the phone blocks its own number or e-mail address from phones that have caller identification enabled. Valid values are as follows:

0—Disabled by default, but can be turned on and off using the user interface. When disabled, the caller identification is included in the Request-URI header field.

1—Enabled by default, but can be turned on and off using the user interface. When enabled, "Anonymous" is included in place of the user identification in the Request-URI header field.

2—Disabled permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

3—Enabled permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

The default value is 0.

cnf_join_enable

Optional

Specifies whether or not the conference bridge, when the it hangs up, should attempt to join the two leaf nodes. Valid values are as follows:

0—Do not join two leaf nodes.

1—Join two leaf nodes.

The default value is 1.

date_format

Optional

Specifies the format for dates. Valid values are as follows:

M/D/Y—Month/day/year

D/M/Y—Day/month/year

Y/M/D—Year/month/day

Y/D/M—Year/day/month

Y-M-D—Year-month-day

YY-M-D—4-digit year-month-day

The default is M/D/Y.

directory_url

Optional

Specifies the URL of the external directory server. This URL is accessed when the Directory key is pressed and the External Directory option is selected. For example, use directory_url: "http://10.10.10.10/CiscoServices/Directory.asp".

dnd_control

Optional

Specifies Do Not Disturb (DND). Valid values are as follows:

0—Off by default, but can be turned on and off locally using the user interface.

1—On by default, but can be turned on and off locally using the user interface. The phone blocks all calls placed to the phone and logs those calls in the Missed Calls directory.

2—Off permanently and cannot be turned on and off locally using the user interface. Specify this parameter in the phone-specific configuration file.

3—On permanently and cannot be turned on and off locally using the user interface. This setting sets the phone to be a "call out" phone only. Specify this parameter in the phone-specific configuration file.

The default value is 0.

dst_auto_adjust

Optional

Configures the date, time, and DST. See the "Setting the Date, Time, and Daylight Saving Time" section for more information.

dst_offset

dst_start_day

dst_start_day_of_week

dst_start_month

dst_start_time

dst_start_week_of_month

dst_stop_day

dst_stop_day_of_week

dst_stop_month

dst_stop_time

dst_stop_week_of_month

dtmf_avt_payload

Optional

Configures the payload type for Audio/Video Transport (AVT) packets. Range is from 96 to 127. If the value specified exceeds 127, the phone defaults to 101.

dtmf_db_level

Optional

Specifies the in-band DTMF digit tone level. Valid values are as follows:

1—6 dB below nominal

2—3 dB below nominal

3—nominal

4—3 dB above nominal

5—6 dB above nominal

The default is 3.

dtmf_inband

Optional

Configures the in-band signaling format. Valid values are 1 (generate DTMF digits in-band) and 0 (do not generate DTMF digits in-band). The default is 1.

dtmf_outofband

Optional

Configures the out-of-band signaling (for tone detection on the IP side of a gateway).

Note The Cisco SIP IP phone supports out-of-bound signaling using the AVT tone method.

Valid values are as follows:

none—Do not generate DTMF digits out-of-band.

avt—If requested by the remote side, generate DTMF digits out-of-band (and disable in-band DTMF signaling); otherwise, do not generate DTMF digits out-of-band.

avt_always—Always generate DTMF digits out-of-band. This option disables in-band DTMF signaling.

The default is avt.

dyn_dns_addr_1

Optional

Specifies the IP address of a new dynamic DNS server. If a new DNS server is specified, it is used for any further DNS requests after the phone uses the initial DNS address upon bootup. The DNS addresses are used in the following order:

1. dyn_dns_addr_1 (if present)

2. dyn_dns_add_2 (if present)

3. DNS Server 1

4. DNS Server 2

5. DNS Server 3

6. DNS Server 4

7. DNS Server 5

The dynamic DNS address is not stored in Flash memory. Only dotted IP addresses are accepted. This value can be cleared by removing it from the configuration file or changing its value to a null value " " or "UNPROVISIONED."

dyn_dns_addr_2

Optional

Specifies a second dynamic DNS to be used for DNS requests.

dyn_tftp_addr

Optional

Specifies the IP address of a new dynamic TFTP server. After initially querying the default TFTP server, the phone will rerequest the default and MAC-specific configuration files from the new TFTP server. The dynamic TFTP server is not stored in Flash memory. The number of dyn_tftp_addr values supported by the phone is limited to prevent the phone from bouncing between two TFTP servers. Only dotted IP addresses are accepted. This value can be cleared by removing it from the configuration file or changing its value to a null value " " or "UNPROVISIONED."

enable_vad

Optional

Enables or disables VAD. Valid values are:

0—Disable

1—Enable

The default is 0.

end_media_port

Optional

Configures the Real-Time Transport Protocol (RTP) end range for media. Valid values are from 16,384 to 32,766. Default is 32,766.

http_proxy_addr

Optional

Specifies the 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

Optional

Specifies the number of the HTTP proxy port. The default is 80.

image_version

Required

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. 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.

language

Optional

This parameter is for future use. English is the only value that is currently supported.

logo_url

Optional

Specifies the location of the company logo file. This logo appears on the phone display. The background space allocated for the image is 90 x 56 pixels. Images that are larger than this will automatically be scaled down to 90 x 56 pixels. The recommended file size for the image is from 5 to 15 Kb. For example, use logo_url: "http://10.10.10.10/companylogo.bmp".

Note This parameter supports Windows 256 color bitmap format only. CMXML PhoneImage objects are not supported for this parameter. Using anything other than a Windows bit-map (.bmp) file can cause unpredictable results.

messages_uri

Optional

Configures the voice-mail number when the messages button is pressed.

nat_address

Optional

Specifies the WAN IP address of the NAT or firewall server. You can use either a dotted IP address or a DNS name (a record only).

nat_enable

Optional

Enables or disables NAT. Valid values are:

0—Disable

1—Enable

The default is 0.

When NAT is enabled, the Contact header appears as follows:

Contact: 
sip:lineN_name@nat_address:voip_control_port

If nat_address is invalid or UNPROVISIONED, the Contact header appears as follows:

Contact: 
sip:lineN_name@phone_ip_address:voip_control_port

and the Via header appears as follows:

Via: SIP/2.0/UDP 
phone_ip_address:voip_control_port

If NAT is enabled, the Session Description Protocol (SDP) message uses the nat_address and an RTP port between the start_media_port and the end_media_port range in the C and M fields. All RTP traffic is sourced from the port advertised in the SDP.

nat_received_processing

Optional

Enables or disables NAT received processing. Valid values are:

0—Disable

1—Enable

The default is 0.

If nat_received_processing is enabled, and received= tag is in the Via header of the 200 OK response from a REGISTER, the IP address in the received= tag is used instead of the nat_address in the Contact header. If this switch occurs, the phone unregisters the old IP address and reregisters with the new IP address.

network_media_type

Optional

Specifies the Ethernet port negotiation mode. Valid values are as follows:

Auto—Port is autonegotiated.

Full100—Port is configured to be a full-duplex, 100-MB connection.

Half100—Port is configured to be a half-duplex, 100-MB connection.

Full10—Port is configured to be a full-duplex, 10-MB connection.

Half10—Port is configured to be a half-duplex, 10-MB connection.

The default is Auto.

network_port2_type

Optional

Configures the device type that is connected to port 2 of the phone. Valid values are as follows:

Hub/Switch (default)

PC

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 results in spanning-tree loops and network confusion.

outbound_proxy

Optional

Specifies the IP address of the outbound proxy server. You can use either a dotted IP address or a DNS name.

outbound_proxy_port

Optional

Specifies the port number of the outbound proxy server. The default is 5060. When outbound proxy is enabled, all SIP requests are sent to the outbound proxy server instead of the proxyN_address. All responses continue to reconcile the normal Via processing rules. The media stream is not routed through the outbound proxy.

NAT and outbound proxy modes can be independently enabled or disabled. The received= tag is added to the Via header of all responses if there is no received= tag in the uppermost Via header and if the source IP address is different from the IP address in the uppermost Via header. Responses are sent back to the source under the following conditions:

If a received= tag is in the uppermost Via header, the response is sent back to the IP address contained in the received= tag.

If there is no received= tag and the IP address in the uppermost Via header is different from the source IP address, the response is sent back to the source IP. Otherwise, the response is sent back to the IP address in the uppermost Via header.

phone_password

Optional

Specifies a password to be used for console or Telnet access. The default password is "cisco."

phone_prompt

Optional

Specifies the prompt to be displayed when using Telnet or console access. The default phone prompt is "SIP Phone."

preferred_codec

Optional

SPecifies the codec to use when a call is initiated. Valid values are g711alaw, g711ulaw, g729a, and none. The default is g711ulaw.

proxy_backup

Optional

Specifies the IP address of the backup proxy server or gateway. Enter this address in IP dotted-decimal notation.

proxy_backup_port

Optional

Specifies the port number of the backup proxy server. Default is 5060.

proxy_emergency

Optional

Specifies the IP address of the emergency proxy server or gateway. Enter this address in IP dotted-decimal notation.

proxy_emergency_port

Optional

Specifies the port number of the emergency proxy server. Default is 5060.

proxy_register

Optional

Specifies that the phone must register with a proxy server during initialization. Valid values are 0 and 1. Specify 0 to disable registration during initialization. Specify 1 to enable registration during initialization. The default is 0.

After a phone has initialized and registered with a proxy server, changing the value of this parameter, using manual configuration, to 0 unregisters the phone from the proxy server. To reinitiate a registration, change the value of this parameter back to 1.

Note If you enable registration, and authentication is required, you must specify values for the linex_authname and linex_password parameters (where x is a number from 1 to 6) in the phone-specific configuration file. For information on configuring the phone-specific configuration file, refer to the "Modifying the Phone-Specific SIP Configuration File" section.

proxy1_address

Required

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

proxy1_port

Optional

Specifies the port number of the primary SIP proxy server. This is the port on which the SIP client listens for messages. The default is 5060.

Note For additional phone lines, the proxyN_address and proxyN_port parameters can be used to assign different proxy addresses to different phone lines. "N" in the parameters represents a phone line. The value of "N" can be from 2 to 6. If the value of "N" is not specified in the proxyN_address parameter, the phone uses the proxy1_address parameter as the default.

proxyN_address

Optional

Specifies the IP address or DNS name of the SIP proxy server that will be used by phone lines other than line 1. For IP address, use the IP dotted-decimal notation. If the proxyN_address 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. If you want to use a dotted IP address, the proxyN_address parameters should be configured as dotted IP addresses.

proxyN_port

Optional

Specifies the port number of the SIP proxy server that will be used by phone lines other than line 1.

remote_party_id

Optional

Specifies the Remote-Party-ID header supports network verification and screening of a call participant's identity (for example, name and number) and provides privacy for call participants. Valid values are as follows:

0—Remote party ID is disabled. The phone does not send or accept the Remote Party ID.

1—Remote party ID is enabled. The phone sends the Remote Party ID, and can accept the Remote Party ID.

The default value is 0.

semi_attended_transfer

Optional

Defines whether or not the caller can transfer the second leg of an attended transfer while the call is ringing. Valid values are as follows:

0—Semi-attended transfer is disabled.

1—Semi-attended transfer is enabled.

The default value is 1.

services_url

Optional

Specifies the URL of the services BTXML files. This URL is accessed when the Services button is pressed. For example, use services_url: "http://10.10.10.10/CiscoServices/Services.asp"

sip_invite_retx

Optional

Specifies the maximum number of times that an INVITE request will be retransmitted. The valid value is any positive integer. The default is 6.

sip_retx

Optional

Specifies the maximum number of times that a SIP message other than an INVITE request will be retransmitted. The valid value is any positive integer. The default is 10.

sntp_mode

Optional

See the "Setting the Date, Time, and Daylight Saving Time" section for more information.

sntp_server

 

start_media_port

Optional

Specifies the start RTP range for media. Valid values are from 16,384 to 32,766. Default is 16,384.

sync

Optional

Specifies the value against which to compare the value in the syncinfo.xml file before a remote reboot is performed. Valid value is a character string up to 32 characters long.

telnet_level

Optional

Enables Telnet for the phone. Valid values are as follows:

0—Telnet is disabled.

1—Telnet is enabled, no privileged commands.

2—Telnet is enabled and privileged commands can be executed.

The default value is 0.

tftp_cfg_dir

Required1

Specifies the path to the TFTP subdirectory in which phone-specific configuration files are stored.

time_format_24hr

Optional

Specifies the whether a 12- or 24-hour time format is displayed by default on the user interface. Valid values are as follows:

0—12-hour format is displayed by default but can be changed to a 24-hour format using the user interface.

1—24-hour format is displayed by default but can be changed to a 12-hour format using the user interface.

2—12-hour format is displayed and cannot be changed to a 24-hour format using the user interface.

3—24-hour format is displayed and cannot be changed to a 12-hour format using the user interface.

The default value is 1.

time_zone

Optional

See the "Setting the Date, Time, and Daylight Saving Time" section for more information.

timer_invite_expires

Optional

Specifies the amount of time, in seconds, after which a SIP INVITE expires. This value is used in the Expire header field. The valid value is any positive number; however, Cisco recommends 180 seconds. The default is 180.

timer_register_expires

Optional

Specifies the amount of time, in seconds, after which a REGISTRATION request expires. This value is inserted into the Expire header field. The valid value is any positive number; however, Cisco recommends 3600 seconds. The default is 3600.

timer_t1

Optional

Specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages. The valid value is any positive integer. The default is 500.

timer_t2

Optional

Specifies the highest value, in milliseconds, of the retransmission timer for SIP messages. The valid value is any positive integer greater than timer_t1. The default is 4000.

tos_media

Optional

Specifies the Type of service (ToS) level for the media stream being used. Valid values are as follows:

0—IP_ROUTINE

1—IP_PRIORITY

2—IP_IMMEDIATE

3—IP_FLASH

4—IP_OVERRIDE

5—IP_CRITIC

The default is 5.

user_info

Optional

Configures the "user=" parameter in the REGISTER message. Valid values are as follows:

none—No value is inserted.

phone—The value user=phone is inserted in the To, From, and Contact Headers for REGISTER.

ip—The value user=ip is inserted in the To, From, and Contact Headers for REGISTER.

The default value is none.

voip_control_port

Optional

Specifies the UDP port used for SIP messages. All SIP REQUESTS use voip_control_port as the UDP source port when nat_enable = 1. Valid values are from 1025 to 65,535. Default is 5060.

1 Required if phone-specific configuration files are located in a subdirectory.


The following is a sample SIP default configuration file:

# Image Version
image_version: "P0S3-xx-y-zz"
 
# Proxy Server
proxy1_address: "proxy.company.com"
proxy2_address: ""
proxy3_address: "" 
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""
 
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:""
proxy3_port:""
proxy4_port:""
proxy5_port:""
proxy6_port:""

# Emergency Proxy info
proxy_emergency: "1.2.3.4"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: "1.2.3.4
proxy_backup_port: "5060"
 
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"
 
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "180"
 
# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "g711ulaw"
 
# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"
 
# In-band DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"
 
# 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 ms
timer_t2: "4000"                  ; Default 4 sec
sip_retx: "10"                     ; Default 11
sip_invite_retx: "6"               ; Default 7
timer_invite_expires: "180"        ; Default 180 sec
 
# Setting for Message speed dial to Voice mail
messages_uri: "9195551000"

#********* Release 2 new configuration parameters **********
 
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
 
# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "172.16.10.150"
#sntp_server: "sntp.company.com"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "April"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "1"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
dst_stop_time: "2"
dst_auto_adjust: "1"
 
# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user 
control)
dnd_control: "0"                  ; Default 0 (Do Not Disturb feature is off)
 
# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user 
control)
callerid_blocking: "0"            ; Default 0 (Disable 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 0 (Disable blocking of anonymous calls)
 
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101"           ; Default 101
 
# XML file that specifies the dial plan desired
dial_template: "dialplan"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"

#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"

#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"

#Enable or Disable VAD (0-disabled (default), 1-enabled)
enable_vad: 0

telnet_level: 0
phone_password: "cisco"

#URL for External XML Services and Phone Logo
services_url: "http://www.company.com/phone/services.asp"
directory_url: "http://www.company.com/phone/companydirectory.asp"
logo_url: "http://www.company.com/phone/logo.bmp"

Modifying the Phone-Specific SIP Configuration File

Before you begin modifying the configuration file, :

Review the guidelines documented in the "Modifying the Default SIP Configuration File" section.

Line parameters (those identified as linex) define a line 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. Each line can have a different proxy configured.

To modify the phone-specific SIP configuration file, open the file in an ASCII text editor. In the file, define values for the SIP parameters shown in Table 3-4. For all variables, x is a number 1 through 6.

Table 3-4 Phone-Specific Configuration Parameters  

Parameter
Required or Optional
Description

linex_authname

Required1

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 the linex_authname parameter for a line when 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_displayname

Optional

Identification as it should appear for caller identification purposes. 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, nothing is used.

linex_name

Required

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.

linex_password

Required1

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 linex_password parameter for a line when registration is enabled, the value defined for line 1 is used. If a value is not defined for line 1, the default line1_password is UNPROVISIONED.

linex_shortname

Optional

Name or number associated with the linex_name as you want it to display on the phone's LCD if the linex_name length exceeds the allowable space in 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 linex_name variable is displayed.

phone_label

Optional

Label to display on the top status line of the LCD. This field is for end-user display only. For example, a phone label can display "John Doe's phone." Up to 11 characters can be used to specify the phone label.

   

Save the file to your TFTP server (in the root directory or in a subdirectory that contains all the phone-specific configuration files). Name the file SIPXXXXYYYYZZZZ.cnf where XXXXYYYYZZZZ is the MAC address of the phone. The MAC address must be in uppercase, and the extension, cnf, must be in lower case (for example, SIP00503EFFD842.cnf).

1 Required for line 1 when registration is enabled and the proxy server requires authentication.


The following is a sample phone-specific configuration file:

line1_displayname: "jdoe43"
line1_name:         "43"
line2_displayname: "jdoe44"
line2_name:         "44"
line3_displayname: "pgatour"
line3_name:         "duval"
line4_displayname: "jdoe46"
line4_name:         "46"
line5_displayname: "jdoe47"
line5_name:         "47"
line6_displayname: "jdoe48"
line6_name:         "48"
phone_label:        "jdoe4X"
phone_prompt: "John-43"

proxy1_address: 1.2.3.4
proxy2_address: 1.2.3.4
proxy3_address: 1.2.3.4
proxy4_address: 1.2.3.4
proxy5_address: 1.2.3.4
proxy6_address: 1.2.3.4
proxy1_port: 5060 
proxy2_port: 5060 
proxy3_port: 5060
proxy4_port: 5060
proxy5_port: 5060
proxy6_port: 5060

callerid_blocking: 0
dtmf_outofband: avt
network_media_type: auto
tos_media: 5
dtmf_avt_payload: 101
time_zone: EST
call_waiting: 1
cnf_join_enable : 1
semi_attended_transfer : 1

Modifying the SIP Parameters Directly on Your Phone

If you did not configure the SIP parameters using a TFTP server, you can configure them directly on your phone after you have connected the phone.

Before You Begin

Unlock configuration mode as described in the "Unlocking Configuration Mode" section. By default, the SIP parameters are locked to ensure that end users cannot modify settings that might affect their call capabilities.

Review the guidelines on using the Cisco SIP IP phone menus documented in the "Using the Cisco SIP IP Phone Menu Interface" section on page 2-15.

Line parameters (those identified as linex) define a line 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.

When configuring the Preferred Codec and Out of Band DTMF parameters, press the Change soft key until the option that you desire is displayed and then press the Save soft key.

After making your changes, relock configuration mode as described in the "Locking Configuration Mode" section.


Step 1 Press the settings key. The Settings menu appears.

Step 2 Highlight SIP Configuration. The SIP Configuration menu appears.

Step 3 Highlight Line 1 Settings.

Step 4 Press the Select soft key. The Line 1 Configuration menu appears.

Step 5 Highlight and press the Select soft key to configure the parameters shown in Table 3-5, as necessary.

Step 6 Press the Back soft key to exit the Line 1 Configuration menu.

Step 7 To configure additional lines on the phone, highlight the next Line x Settings, press the Select soft key and repeat Step 5 and Step 6, and then continue with Step 8.

Step 8 In addition to the line settings, you can highlight and press Select to configure the parameters on the SIP Configuration menu shown in Table 3-6.

Step 9 When done, press the Save soft key to save your changes and exit the SIP Configuration menu.



Caution When you have completed your changes, ensure that you lock the phone as described in the "Locking Configuration Mode" section.

Table 3-5 SIP Configuration Parameters 

Parameter
Required or Optional
 

Authentication Password

Required1

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.

Authentication Name

Required1

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

Display Name

Optional

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

Required

Description 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.

Proxy Address

Required

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

Optional

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

Short Name

Optional

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 3444 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.


Table 3-6 Additional SIP Configuration Parameters 

Parameter
Required or Optional
 

Backup Proxy

Optional

IP address of the backup proxy server or gateway. Enter this address in IP dotted-decimal notation.

Backup Proxy Port

Optional

Port number of the backup proxy server. Default is 5060.

Emergency Proxy

Optional

IP address of the emergency proxy server or gateway. Enter this address in IP dotted-decimal notation.

Emergency Proxy Port

Optional

Port number of the emergency proxy. Default is 5060.

Enable VAD

Optional

Specifies whether VAD is enabled or disabled.

End Media Port

Optional

The end RTP range for media. Valid values are 16,384 to 32,766. Default is 32,766.

Messages URI

Optional

Number to call to check voice mail. This number is called when the Messages key is pressed.

NAT Address

Optional

The WAN IP address of the NAT or firewall server. You can use either a dotted IP address or a DNS name (a record only).

NAT Enabled

Optional

Choose No to disable NAT and Yes to enable NAT.

Out of Band DTMF

Optional

Whether to detect and generate the out-of-band signaling (for tone detection on the IP side of a gateway) and if so, when. The Cisco SIP IP phone supports out-of-bound signaling via the AVT tone method. Valid values are as follows:

none—Do not generate DTMF digits out-of-band.

avt—If requested by the remote side, generate DTMF digits out-of-band (and disable in-band DTMF signaling); otherwise, do not generate DTMF digits out-of-band.

avt_always—Always generate DTMF digits out-of-band. This option disables in-band DTMF signaling.

The default is avt.

Outbound Proxy

Optional

The IP address of the outbound proxy server. You can use either a dotted IP address or a DNS name.

Outbound Proxy Port

Optional

The port number of the outbound proxy server. The default is 5060.

Phone Label

Optional

Label to display on the top status line of the LCD. This field is for end-user display only. For example, a phone's label can display "John Doe's phone." Up to 11 characters can be used when specifying the phone label.

Preferred Codec

Optional

Codec to use when initiating a call. Valid values are g711alaw, g711ulaw, and g729a. The default is g711ulaw.

Register Expires

Optional

The amount of time, in seconds, after which a REGISTRATION request expires. This value is used the Expire header field. The valid value is any positive number; however, Cisco recommends 3600 seconds. The default is 3600.

Register with Proxy

Optional

Whether the phone must register with a proxy server during initialization. Valid values are Yes and No. Select the No soft key to disable registration during initialization. Select the Yes soft key to enable registration during initialization. The default is No. After a phone has initialized and registered with a proxy server, changing the value of this parameter to No unregisters the phone from the proxy server. To reinitiate a registration, change the value of this parameter back to Yes.

Note If you enable registration, and authentication is required, you must specify values for the Authentication Name and Authentication Password parameters.

Start Media Port

Optional

The start RTP range for media. Valid values are from 16,384 to 32,766. Default is 16,384.

TFTP Directory

Required1

Path to the TFTP subdirectory in which phone-specific configuration files are stored.

VoIP Control Port

Optional

The UDP port used for SIP messages. All SIP REQUESTS use voip_control_port as the UDP source port when nat_enable = 1. Valid values are from 1,025 to 65,535. Default is 5060.

1 Required if phone-specific configuration files are located in a subdirectory.


Modifying Call Preferences

The call preferences can be modified only if the configuration variable for each setting has been set to 0 or 1 by the system administrator. If the variables are configured as 2 or 3, the call preferences cannot be modified with the Call Preferences menu. See Table 3-2 for descriptions of each parameter.

You can modify the call preferences on each phone using the Call Preferences menu as follows:


Step 1 Press the settings key. The Settings menu displays.

Step 2 Highlight Call Preferences. The Call Preferences configuration menu displays.

Step 3 Press the Select soft key.

Step 4 Highlight and press the Select soft key to configure the parameters as follows:

Anonymous Call Block

Auto-Complete Numbers

Caller ID Blocking

Call Hold Ringback

Call Waiting

Do Not Disturb

Step 5 Press the Back soft key to exit.

Step 6 When done, press the Save soft key to save your changes and exit.


Setting the Date, Time, and Daylight Saving Time

The current date and time is supported on the Cisco SIP IP phone via SNTP and is displayed on the phone's LCD. In addition to supporting the current date and time, daylight saving time (DST) and time zone settings are also supported. DST can be configured to be obtained via an absolute (for example, starts on April 1 and ends on October 1) or relative (for example, starts on the first Sunday in April and ends on the last day of October) configuration.

The format for the date can be set using the date_format parameter.

International time zone abbreviations are supported and are case sensitive (must be in all capital letters).

Cisco recommends that date- and time-related parameters be defined in the SIPDefault.cnf file. The time zone parameter can be set manually on the phone or in the configuration file.

Before You Begin

When configuring the date, time, time zone, and DST settings, remember the following:

Review the guidelines and restrictions documented in the "Modifying the Default SIP Configuration File" section.

Determine whether you want to configure absolute DST or relative DST.

The SNTP parameters specify how the phone will obtain the current time from an SNTP server. Review the guidelines in Table 3-7 and Table 3-8 before configuring the SNTP parameters.

Table 3-7 lists the actions that take place when a null value (0.0.0.0) is specified in the sntp_server parameter.

Table 3-7 Actions Based on sntp_mode When the sntp_server Parameter Is Set to a Null Value

sntp_server=0.0.0.0
sntp_mode=
unicast
sntp_mode=
multicast
sntp_mode=
anycast
sntp_mode=
directedbroadcast
Sends

Nothing.

No known server with which to communicate.

Nothing.

When in multicast mode, 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

Nothing.

No known server with which to communicate.

SNTP data via 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.


Table 3-8 lists the actions that take place when a valid IP address is specified in the sntp_server parameter.

Table 3-8 Actions Based on sntp_mode When the sntp_server Parameter Is Set to an IP Address

sntp_server
= 192.168.1.9
sntp_mode=
unicast
sntp_mode=
multicast
sntp_mode=
anycast
sntp_mode=
directedbroadcast
Sends

SNTP request to the SNTP server.

Nothing.

When in multicast mode, SNTP requests are not sent.

SNTP request to the SNTP server.

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.



Step 1 Using an ASCII editor, open the SIPDefault.cnf file and define or modify values for the following SNTP-specific SIP parameters as necessary:

sntp_mode—(Required) Mode in which the phone listens for the SNTP server. Valid values are unicast, multicast, anycast, or directedbroadcast.

See Table 3-7 and Table 3-8 for an explanation on how these values work, depending on the sntp_server parameter value.

sntp_server—(Required) IP address of the SNTP server from which the phone will obtain time data.

See Table 3-7 and Table 3-8 for an explanation on how these values work, depending on the sntp_server parameter value.

time_zone—(Required) Time zone in which the phone is located. Valid values are the time zone abbreviations shown in Table 3-9. These abbreviations are case sensitive and must be in all capital letters.

Table 3-9 Time Zone Abbreviations

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), USSR-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)


Step 2 To configure common DST settings, specify values for the following parameters:

dst_offset—Offset from the phone's time when DST is in effect. When DST is over, the specified offset is no longer applied to the phone's time. Valid values are hour/minute, -hour/minute, +hour/minute, hour, -hour, and +hour.

dst_auto_adjust—Whether or not DST is automatically adjusted on the phones. Valid values are 0 (disable automatic DST adjustment) or 1 (enable automatic DST adjustment). The default is 1.

dst_start_month—Month in which DST starts. Valid values are January, February, March, April, May, June, July, August, September, October, November, and December or 1 through 12 with January being 1 and December being 12. When specifying the name of a month, the value is not case sensitive. In the United States, the default value is April.

dst_stop_month—Month in which DST ends. Valid values are January, February, March, April, May, June, July, August, September, October, November, and December or 1 through 12 with January being 1 and December being 12. When specifying the name of a month, the value is not case sensitive. In the United States, the default value is October.

dst_start_time—Time of day on which DST begins. Valid values are hour/minute (02/00) or hour (02:00). In the United States, the default value is 02:00.

dst_stop_time—Time of day on which DST ends. Valid values are hour/minute (02/00) or hour (02:00). In the United States, the default value is 02:00.

Step 3 To configure absolute DST, specify values for the following parameters, or to configure relative DST, proceed to Step 4:

dst_start_day—Day of the month on which DST begins.

Valid values are 1 through 31 for the days of the month or 0 when specifying relative DST to indicate that this field be ignored and that the value in the dst_start_day_of_week parameter be used instead.

dst_stop_day—Day of the month on which DST ends.

Valid values are 1 through 31 for the days of the month or 0 when specifying relative DST to indicate that this field be ignored and that the value in the dst_stop_day_of_week parameter be used instead.

Step 4 To configure relative DST, specify values for the following parameters:

dst_start_day_of_week—Day of the week on which DST begins.

Valid values are Sunday or Sun, Monday or Mon, Tuesday or Tue, Wednesday or Wed, Thursday or Thu, Friday or Fri, Saturday or Sat, or Sunday or Sun or 1 through 7 with 1 being Sunday and 7 being Saturday. When specifying the name of the day, the value is not case sensitive. In the United States, the default value is Sunday.

dst_start_week_of_month—Week of month on which DST begins.

Valid values are 1 through 6 and 8, with 1 being the first week and each number thereafter being subsequent weeks and 8 specifying the last week in the month regardless of which week the last week is. In the United States, the default value is 1.

dst_stop_day_of_week—Day of the week on which DST ends.

Valid values are Sunday or Sun, Monday or Mon, Tuesday or Tue, Wednesday or Wed, Thursday or Thu, Friday or Fri, Saturday or Sat, or Sunday or Sun or 1 through 7, with 1 being Sunday and 7 being Saturday. When specifying the name of the day, the value is not case sensitive. In the United States, the default value is Sunday.

dst_stop_week_of_month—Week of month on which DST ends.

Valid values are 1 through 6 and 8, with 1 being the first week and each number thereafter being subsequent weeks and 8 specifying the last week in the month regardless of which week the last week is. In the United States, the default value is 8.

Step 5 Save the file with the same filename, SIPDefault.cnf, to the root directory of your TFTP server.


The following is a sample configuration for 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

The following is a sample configuration for 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 : 

Erasing the Locally Defined Settings

You can erase the locally defined network and SIP settings that have been configured in the phone.

Erasing the Locally Defined Network Settings

When you erase the locally defined network settings, the values are reset to the defaults.

Before You Begin

Unlock configuration mode as described in the "Unlocking Configuration Mode" section.

If DHCP has been disabled on a phone, clearing the phone's settings reenables it.

Select the Erase Config parameter by pressing the down arrow to scroll to and highlight the parameter or by pressing the number that represents the parameter (located to the left of the parameter name on the LCD).


Step 1 Press the settings key. The Settings menu appears.

Step 2 Highlight Network Configuration.

Step 3 Press the Select soft key. The Network Configuration settings are displayed.

Step 4 Highlight Erase Configuration.

Step 5 Press the Yes soft key.

Step 6 Press the Save soft key. The phone programs the new information into Flash memory and resets.


Erasing the Locally Defined SIP Settings

When you erase the locally defined SIP settings, the values are reset to the defaults.


Note If your system has been set up to have the phones retrieve their SIP parameters using a TFTP server, you must edit the configuration file in which a parameter is defined to delete the parameter. When deleting a parameter, remove the variable in the file or change its value to a null value " " or "UNPROVISIONED". If both the variable and its value are removed, the phone uses the setting for that variable that it has stored in Flash memory.



Note 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.


Before You Begin

Unlock configuration mode as described in the "Unlocking Configuration Mode" section.


Step 1 Press the settings key. The Settings menu appears.

Step 2 Highlight SIP Configuration.

Step 3 Press the Select soft key. The SIP Configuration settings are displayed.

Step 4 Highlight the parameter for which you want to erase the setting.

Step 5 Press the Edit soft key.

Step 6 Press the << soft key to delete the current value.

Step 7 Press the Validate soft key to save your change and exit the Edit panel.

Step 8 If modifying a line parameter, press the Back soft key to exit the Line Configuration panel.

Step 9 Press the Save soft key. The phone programs the new information into Flash memory and resets.


Viewing the Firmware Version

To view the firmware version, perform the following steps:


Step 1 Press the Settings key. The Settings menu appears.

Step 2 Highlight Status.

Step 3 Press the Select soft key. The Setting Status menu appears.

Step 4 Highlight Firmware Versions.

Step 5 Press the Select soft key. The Firmware Versions panel appears.

The following information is displayed on this panel:

Application Load ID—Current software image on the phone.

Boot Load ID—Bootstrap loader image version that is manufactured on the phone. This image name does not change.

Step 6 To exit the Firmware Versions panel, press the Exit soft key.


Upgrading the Cisco SIP IP Phone Firmware

You can use one of two methods to upgrade the firmware on your Cisco SIP IP phones. You can upgrade the firmware on one phone at a time using the phone-specific configuration, or you can upgrade the firmware on a system of phones using the default configuration file.

To upgrade the firmware, you specify the image_version in the phone-specific configuration file. To upgrade the firmware on a system of phones, specify the image_version in the default configuration file and do not define the image_version in the phone-specific configuration files.

Before You Begin

Ensure that the latest version of the Cisco SIP IP phone firmware has been copied from Cisco.com to the root directory of your TFTP server.

See the upgrade scenarios in Table 3-10 to determine how to upgrade.

Table 3-10 Upgrade Scenarios 

Image Name
Use Section

P0S3-xx-y-zz

Upgrading to Release 5.0 and Release 5.1

P0S30202, P0S30203 and P0S3-03-y-xx

Upgrading from Release 2.2 or Later to Current Release

P0S30100, P0S30200, P0S30201, and P0S3Zxxx

Upgrading from Release 2.1 or Earlier to Current Release

P003xxxx or P003xxxxxxxx (these images are loaded on the Cisco SIP IP phone when it is shipped)

Dual Booting from SCCP or MGCP to a SIP Release

P0S3-xx-y-zz

Dual Booting from SCCP or MGCP to a SIP Release


Upgrading to Release 5.0 and Release 5.1

When you upgrade to Release 5.0 or Release 5.1, you will download a ZIP archive instead of a file as in earlier releases. Contained in the archive are the unsigned (.bin) and signed (.sbn) binary images. Specific information for each release is as follows:

Release 5.0

Cisco has added image authentication to IP phone protocols, which means that tampering with the binary image before the image is downloaded to the phone is not allowed. Any tampering with the image will cause the phone to fail the authentication process and reject the image. Once you download the Release 5.0 image, you cannot downgrade to any earlier releases.

Release 5.1

Release 5.1 is the second release of the signed Cisco IP phone image. Release 5.1 is compatible with Release 5.0 and later releases. Release 5.1 addresses the user interface responsiveness and voice clipping issues.

Procedure


Step 1 Unzip the ZIP archive to extract the binary images and any notes or readme text files. Read these text files for any special directions regarding the images.

Step 2 Using a text editor, open the configuration file and update the image version specified in the image_version variable. The version name in the image_version variable should match the version name (without the .sbn or .bin extension) of the latest firmware that you downloaded (for example, P0S3-xx-y-zz).

Step 3 Reset each phone.

The phone contacts the TFTP server and requests its configuration files. The phone compares the image defined in the file to the image that it has stored in Flash memory. If the phone determines that the image defined in the file differs from the image in Flash memory, it downloads the image defined in the configuration file (which is stored in the root directory on the TFTP server). Once the new image has been downloaded, the phone programs that image into Flash memory and then reboots.



Note If you do not define the image_version parameter in the default configuration file, only phones that have an updated phone-specific configuration file with the new image version and that have been restarted use the latest firmware image. All other phones use the older version until their configuration files have been updated with the new image version.


Upgrading from Release 2.2 or Later to Current Release


Step 1 Copy the new image P0S3-xx-y-zz.bin, where xx is the release major version, y is the release minor version, and zz is the maintenance number, from Cisco.com to the root directory of the TFTP server.

Step 2 Using a text editor, open the configuration file and update the image version specified in the image_version variable. The version name in the image_version variable should match the version name (without the .bin extension) of the latest firmware that you downloaded (for example, P0S3-xx-y-zz).

Step 3 Reset each phone.

The phone contacts the TFTP server and requests its configuration files. The phone compares the image defined in the file to the image that it has stored in Flash memory. If the phone determines that the image defined in the file differs from the image in Flash memory, it downloads the image defined in the configuration file (which is stored in the root directory on the TFTP server). Once the new image has been downloaded, the phone programs that image into Flash memory and then reboots.



Note If you do not define the image_version parameter in the default configuration file, only phones that have an updated phone-specific configuration file with the new image version and that have been restarted use the latest firmware image. All other phones use the older version until their configuration files have been updated with the new image version.


Upgrading from Release 2.1 or Earlier to Current Release


Step 1 Copy the P0S30202.bin binary image from Cisco.com to the root directory of the TFTP server.

Step 2 If you are dual booting from a Cisco IP phone running the Skinny Client Control Protocol (SCCP) or MGCP protocol, open the OS79XX.TXT file with a text editor and change the file to include P0S30202.

Step 3 Open the phone configuration file with a text editor and edit the image_version variable to read P0S30202.

Step 4 Reset each phone.

The phone contacts the TFTP server and requests its configuration files. The phone compares the image defined in the file to the image that it has stored in Flash memory. If the phone determines that the image defined in the file differs from the image in Flash memory, it downloads the image defined in the configuration file (which is stored in the root directory on the TFTP server). Once the new image has been downloaded, the phone programs that image into Flash memory and then reboots.

Step 5 Copy the new image P0S3-xx-y-zz.bin, where xx is the release major version, y is the release minor version, and zz is the maintenance number, from Cisco.com to the root directory of the TFTP server.

Step 6 Using a text editor, open the configuration file and update the image version specified in the image_version variable. The version name in image_version variable should match the version name (without the .bin extension) of the latest firmware that you downloaded (for example, P0S3-xx-y-zz).

Step 7 Reset each phone.


Dual Booting from SCCP or MGCP to a SIP Release


Step 1 Copy the P0S30202.bin binary image from Cisco.com to the root directory of the TFTP server.

Step 2 If you are dual booting from a Cisco IP phone running the SCCP or MGCP protocol, open the OS79XX.TXT file with a text editor and change the file to include P0S30202.

Step 3 Copy the new image P0S3-xx-y-zz.bin, where xx is the release major version, y is the release minor version, and zz is the maintenance number, from Cisco.com to the root directory of the TFTP server.

Step 4 Using a text editor, open the configuration file and update the image version specified in the image_version variable. The version name in image_version variable should match the version name (without the .bin extension) of the latest firmware that you downloaded (for example, P0S3-xx-y-zz).

Step 5 Reset each phone.

The phone contacts the TFTP server and requests its configuration files. The phone compares the image defined in the file to the image that it has stored in Flash memory. If the phone determines that the image defined in the file differs from the image in Flash memory, it downloads the image defined in the configuration file (which is stored in the root directory on the TFTP server). Once the new image has been downloaded, the phone programs that image into Flash memory and then reboots.


Performing an Image Upgrade and Remote Reboot

With Version 2.0 and newer of the Cisco SIP IP phone, you can perform an image upgrade and remote reboot using NOTIFY messages and the syncinfo.xml file. The dialplan.xml file can also be pushed down to the phones using a NOTIFY with a check-sync Event header.


Note To perform an image upgrade and remote reboot, a SIP proxy server and a TFTP server must exist in the phone network.


To upgrade the firmware image and perform a remote reboot, perform the following steps:


Step 1 Using an ASCII editor, open the SIPDefault.cnf file located in the root directory of your TFTP server and change the image_version parameter to the name of the latest image.

Step 2 Using an ASCII editor, open the syncinfo.xml file located in the root directory of your TFTP server and specify values for the image version and sync parameter as follows:

<IMAGE VERSION="image_version" SYNC="sync_number"/>

Where:

image_version is the image version of the phone. The asterisk (*) can be used as a wildcard character.

sync_number is the synchronization level of the phone. The default synchronization level for the phone is 1. A valid value is a character string of up to 32 characters.

Step 3 Send a NOTIFY message to the phone. In the NOTIFY message, ensure that an Event header that is equal to "check-sync" is included. The following is a sample NOTIFY message:

NOTIFY sip:lineX_name@ipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP ipaddress:5060;branch=1
Via: SIP/2.0/UDP ipaddress
From: <sip:webadim@ipaddress>
To: <sip:lineX_name@ipaddress>
Event: check-sync
Date: Mon, 10 Jul 2000 16:28:53 -0700
Call-ID: 1349882@ipaddress
CSeq: 1300 NOTIFY
Contact: <sip:webadmin@ipaddress>
Content-Length: 0

After the remote reboot process is initiated on the phone via the NOTIFY message, the following actions take place:

1. If the phone is currently in an idle state, the phone waits 20 seconds and then contacts the TFTP server for the syncinfo.xml file. If the phone is not in an idle state, the phone waits until it is in an idle state for 20 seconds and then contacts the TFTP server for the syncinfo.xml file.

2. The phone reads the syncinfo.xml file and performs the following as appropriate:

a. Determines whether the current image is specified. If so, the phone proceeds to Step c. If not, the phone proceeds to Step b.

b. Determines whether there is a wildcard entry (*) in the image version parameter. If so, the phone proceeds to Step c. If not, the phone proceeds to Step d.

c. Determines if the synchronization value is different than what is stored on the phone. If so, the phone proceeds to Step e. If not, the phone proceeds to Step d.

d. The phone does nothing.

e. The phone reboots.

The phone the performs a normal reboot process as described in the "Overview of the Initialization Process" section on page 2-1, sees the new image, and upgrades to the new image with a synchronization value of what is specified in the syncinfo.xml file.