Managing Cisco SIP IP Phones (Version 4.4)

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

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 **#.

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

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.

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:

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, 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

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	Optional
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.

Modifying the Phone-Specific SIP Configuration File

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.

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

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 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.

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

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

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.

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:

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

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

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:

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

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.

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:

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.

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.

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.

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

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

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

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.

Before You Begin

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

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.

Accessing Status Information

There are several types of status information that you can access via the settings key. The information that you can obtain via the settings key can aid in system management. To access status information, select settings and then select Status from the Settings menu. From the Status menu, the following three options are available:

In addition to the status messages available via the Setting Status menu, you can also obtain status messages for a current call.

Viewing Status Messages

To view status messages that you can use to diagnose network problems, perform the following steps:

Viewing Network Statistics

To view statistical information about the phone and network performance, perform the following steps:

Viewing the Firmware Version

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

Upgrading from Release 2.2 or Later to Current Release

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).

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.

Upgrading from Release 2.1 or Earlier to Current Release

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 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).

Dual Booting from SCCP or MGCP to a SIP Release

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).

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.

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

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:

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:

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:

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

Table of Contents