Cisco IP Phone Installation

Verify the Network Setup

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

Procedure

Step 1

Configure a VoIP Network to meet the following requirements:

  • VoIP is configured on your routers and gateways.

Step 2

Set up the network to support one of the following:

  • DHCP support

  • Manual assignment of IP address, gateway, and subnet mask

Install the Conference Phone (7832)

After the phone connects to the network, the phone startup process begins, and the phone registers with Third-party Call Control System. You need to configure the network settings on the phone if you disable the DHCP service.

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

After the phone connects, it determines if a new firmware load should be installed on the phone.

Procedure

Step 1

Choose the power source for the phone:

  • Power over Ethernet (PoE)

  • Cisco Unified IP Phone Power Injector

Step 2

Connect the phone to the switch.

  • If you use PoE, plug the Ethernet cable into the LAN port and plug the other end into the phone.
  • If you use the Cisco Unified IP Phone Power Injector:

    1. Plug the Power Injector into the LAN port with one Ethernet cable.

    2. Connect the power cord to the Injector.

    3. Plug the cord into an electrical outlet.

    4. Use another Ethernet cable to connect the Injector to the conference phone.

    In the EU and UK, your phone is shipped with a power switch cable that is required in order to provide power to your phone when using an AC-to-DC power supply. In this case:

    1. Plug the Power Injector into the LAN port with one Ethernet cable.

    2. Connect the power switch cable to the Injector.

    3. Connect the power cord to the other end of the power switch cable.

    4. Plug the cord into an electrical outlet.

    5. Use another Ethernet cable to connect the Injector to the conference phone.

Each phone ships with one Ethernet cable in the box.

Step 3

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

Step 4

If you do not use autoregistration, manually configure the network settings on the phone.

See Configure the Network from the Phone.

Step 5

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

Step 6

Provide information to end users about how to use their phones and how to configure their phone options. This step ensures that users have adequate information to successfully use their Cisco phones.

Install the Conference Phone (8832)

After the phone connects to the network, the phone startup process begins, and the phone registers with Third-party Call Control System. You need to configure the network settings on the phone if you disable the DHCP service.

After the phone connects, it determines if a new firmware load should be installed on the phone.

Procedure

Step 1

Choose the power source for the phone:

  • Power over Ethernet (PoE) deployment with a Cisco IP Conference Phone 8832 PoE Injector

  • Non-PoE Ethernet deployment with a Cisco IP Conference Phone 8832 Non-PoE Ethernet Injector

Step 2

Connect the phone to the switch.

  • If you use PoE:

    1. Plug the Ethernet cable into the LAN port.

    2. Plug the other end of the Ethernet cable into either the Cisco IP Conference Phone 8832 PoE Injector or the Cisco IP Conference Phone 8832 Ethernet Injector.

    3. Connect the injector to the conference phone with the USB-C cable.

  • If you do not use PoE:

    1. Connect the power adapter to the Cisco IP Conference Phone 8832 Ethernet Injector using a USB-C cable.

    2. If you are using the Cisco IP Conference Phone 8832 Ethernet Injector, plug the power adapter into an electrical outlet.

    3. Connect the power adapter to the Ethernet injector using a USB-C cable.

      OR

      If you are using the Cisco IP Conference Phone 8832 Non-PoE Ethernet Injector, plug it into an electrical outlet.

    4. Plug the Ethernet cable into the Ethernet injector.

    5. Plug the Ethernet cable into the Non-PoE Ethernet injector or the Ethernet injector.

    6. Plug the Ethernet cable into the LAN port.

    7. Connect the Ethernet injector to the conference phone using a second USB-C cable.

    8. Connect the Non-PoE Ethernet injector or the Ethernet injector to the conference phone using a USB-C cable.

Step 3

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

Step 4

If you do not use autoregistration, manually configure the security settings on the phone.

Step 5

Allow the phone to upgrade to the current firmware image.

Step 6

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

Step 7

Provide information to the users about how to use their phones and how to configure their phone options. This step ensures that users have adequate information to successfully use their Cisco conference phones.

Configure the Network from the Phone

The phone includes many configurable network settings that you may need to modify before it is functional for your users. You can access these setting through the phone menus.

The Network configuration menu provides you with options to view and configure a variety of network settings.

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

Procedure

Step 1

Press Settings.

Step 2

Select Network configuration.

Step 3

Use the navigation arrows to select the desired menu and edit.

Step 4

To display a submenu, repeat step 3.

Step 5

To exit a menu, press Back .

Network Configuration Fields

Table 1. Network Configurations Menu Options

Field

Field Type or Choices

Default

Description

Ethernet configuration

See the following Ethernet configuration submenu table.

IP mode

Dual mode

IPv4 only

IPv6 only

Dual mode

Select the Internet Protocol mode for which the phone operates.

In dual mode, the phone can have both IPv4 and IPv6 addresses.

IPv4 address settings

DHCP

Static IP

DHCP

See the IPv4 address submenu table in the following tables.

IPv6 address settings

DHCP

Static IP

DHCP

See the IPv6 address submenu table in the following tables.

DHCPv6 option to use

17, 160, 159

Indicates the order in which the phone uses the IPv6 addresses provided by DHCP server.

HTTP proxy settings

See the following HTTP proxy settings submenu table.

Web server

On

Off

On

Indicates whether the phone has web server enabled or disabled.
Table 2. Ethernet Configuration Submenu

Field

Field Type

or Choices

Default

Description

802.1x authentication

Device authentication

Off

Enables you to turn on or turn off the 802.1x authentication. Valid options are:

  • On

  • Off

Transaction status

Disabled

  • Transaction status—Indicates different authentication status when you turn on 802.1x in the Device authentication field.

    • Disabled—Default status.

    • Connecting—802.1x authentication started in the device.

    • Authenticated—802.1x authentication established in the device.

  • Protocol—Specifies the protocol of the server.

User certificate type

Manufacturing installed

Choose the certificate for the 802.1X authentication during the initial enrollment and certificate renewal.

  • Manufacturing installed—The Manufacturing Installed Certificate (MIC) and Secure Unique Device Identifier (SUDI) is used.

  • Custom installed—The Custom Device Certificate (CDC) is used. This type of certificate can be installed either by manual uploading on the phone web page or by installation from a Simple Certificate Enrollment Protocol (SCEP) server.

This parameter appears on the phone only when Device authentication is enabled.

Switch port config

Auto

10MB half

10MB full

100MB half

100MB full

1000 full

Auto

Select speed and duplex of the network port.

If the phone is connected to a switch, configure the port on the switch to the same speed/duplex as the phone, or configure both to autonegotiate.

CDP

On

Off

On

Enable or disable Cisco Discovery Protocol (CDP).

CDP is a device-discovery protocol that runs on all Cisco manufactured equipment.

Using CDP, a device can advertise its existence to other devices and receive information about other devices in the network.

LLDP-MED

On

Off

On

Enable or disable LLDP-MED.

LLDP-MED enables the phone to advertise itself to devices that use the discovery protocol.

Startup delay

3 seconds

Set a value that causes a delay for the switch to get to the forwarding state before the phone sends out the first LLDP-MED packet. For configuration of some switches, you might need to increase this value to a higher value for LLDP-MED to work. Configuring a delay can be important for networks that use the Spanning Tree Protocol.

Default delay is 3 seconds.

VLAN

On

Off

Off

Enable or disable VLAN.

Permits you to enter a VLAN ID when you use VLAN without CDP or LLDP. When you use a VLAN with CDP or LLDP, that associated VLAN takes precedent over the VLAN ID you manually entered.

VLAN ID

1

Enter a VLAN ID for the IP phone when you use a VLAN without CDP (VLAN enabled and CDP disabled). Note that only voice packets are tagged with the VLAN ID. Do not use the 1 value for the VLAN ID. If VLAN ID is 1, you cannot tag voice packets with the VLAN ID.

DHCP VLAN option

Enter a predefined DHCP VLAN option to learn the voice VLAN ID.

When you use a VLAN ID with CDP, LLDP, or manually select a VLAN ID, that VLAN ID takes precedent over the selected DHCP VLAN option.

Valid values are:

  • Null

  • 128 to 149

  • 151 to 158

  • 161 to 254

Default value is null.

Cisco recommends that you use DHCP Option 132.
Table 3. IPv4 Address Settings Submenu

Field

Field Type

or Choices

Default

Description

Connection type

DHCP

Indicates whether the phone has DHCP enabled.

  • DNS1—Identifies the primary Domain Name System (DNS) server that the phone uses.

  • DNS2—Identifies the secondary Domain Name System (DNS) server that the phone uses.

  • DHCP address released—Releases the IP address that DHCP assigned. You can edit this field if DHCP is enabled. To remove the phone from the VLAN and release the IP address for reassignment, set this field to Yes and press Set.

Static IP

When DHCP is disabled, you must set the Internet Protocol (IP) address of the phone.

  • Static IP address—Identifies the IP that you assign to the phone. The phone uses this IP address instead of acquiring an IP from the DHCP server on the network.

  • Subnet Mask—Identifies the subnet mask used by the phone. When DHCP is disabled, you must set the subnet mask.

  • Gateway address—Identifies the default router used by the phone.

  • DNS1—Identifies the primary Domain Name System (DNS) server that the phone uses. When DHCP is disabled, you must set this field manually.

  • DNS2—Identifies the primary Domain Name System (DNS) server that the phone uses. When DHCP is disabled, you must set this field manually.

When you assign an IP address using this field, you must also assign a subnet mask and a gateway address. See the Subnet Mask and Default Router fields in this table.

Table 4. IPv6 Address Settings Submenu

Field

Field Type

or Choices

Default

Description

Connection type

DHCP

Indicates whether the phone has Dynamic Host Configuration Protocol (DHCP) enabled.

  • DNS1—Identifies the primary DNS server that the phone uses.

  • DNS2—Identifies the secondary DNS server that the phone uses.

  • Broadcast Echo—Identifies if the phone responses to multicast ICMPv6 message with destination address of ff02::1.

  • Auto config— Identifies if the phone uses automatic configuration for the address.

Static IP

When DHCP is disabled, you must set the Internet Protocol (IP) address of the phone and must set the values of the fields:

  • Static IP—Identifies the IP that you assign to the phone. The phone uses this IP address instead of acquiring an IP from the DHCP server on the network.

  • Prefix length—Identifies how many bits of a Global Unicast IPv6 Address are there in the network part.

  • Gateway—Identifies the default router used by the phone.

  • Primary DNS—Identifies the primary DNS server that the phone uses. When DHCP is disabled, you must set this field manually.

  • Secondary DNS—Identifies the primary DNS server that the phone uses. When DHCP is disabled, you must set this field manually.

  • Broadcast Echo—Identifies if the phone responses to multicast ICMPv6 message with destination address of ff02::1.
Table 5. HTTP Proxy Settings Submenu

Field

Field Type or Choices

Description

Proxy mode

Auto

Auto discovery (WPAD)—Enables or disables the Web Proxy Auto-Discovery protocol to retrieve a Proxy Auto-Configuration (PAC) file. Valid options are:

  • On

  • Off

    If the value is set to Off, you need to further set the following field:

    • PAC URL—Specifies the URL address for the PAC file that you want to retrieve. For example:

      http://proxy.department.branch.example.com

    The default value of Auto discovery (WPAD) is On.

Manual

  • Proxy host—Specifies an IP address or hostname of the proxy server for the phone. The scheme (http:// or https://) is not required.

  • Proxy port—Specifies a port number of the proxy server.

  • Proxy authentication—Selects an option according to the actual situation of the proxy server. If the server requires authentication credentials to grant access to the phone, then select On. Otherwise, select Off. Options are:

    • Off

    • On

      If the value is set to On, you need to further set the following fields:

      • Username—Specifies the username of a credential user on the proxy server.

      • Password—Provides the specified user's password to pass the authentication of the proxy server.

    The default value of Proxy authentication is Off.

Off

Disables the HTTP proxy feature on the phone.

Text and Menu Entry From the Phone

When you edit the value of an option setting, follow these guidelines:

  • Use the arrows on the navigation pad to highlight the field that you wish to edit. Press Select in the navigation pad to activate the field. After the field is activated, you can enter values.

  • Use the keys on the keypad to enter numbers and letters.

  • To enter letters by using the keypad, use a corresponding number key. Press the key one or more times to display a particular letter. For example, press the 2 key once for "a," twice quickly for "b," and three times quickly for "c." After you pause, the cursor automatically advances to allow you to enter the next letter.

  • Press the softkey if you make a mistake. This softkey deletes the character to the left of the cursor.

  • Press Back before pressing Set to discard any changes that you made.

  • To enter a period (for example, in an IP address), press * on the keypad.


Note

The Cisco IP Phone provides several methods to reset or restore option settings, if necessary.

Verify Phone Startup

After the Cisco IP Phone has power connected to it, the phone automatically cycles through a startup diagnostic process.

Procedure

Step 1

If you are using Power over Ethernet, plug the LAN cable into the Network port.

Step 2

If you are using the power cube, connect the cube to the phone and plug the cube into an electrical outlet.

The buttons flash amber and then green in sequence during the various stages of bootup as the phone checks the hardware.

If the phone completes these stages successfully, it has started up properly.

Disable or Enable DF Bit

You can disable or enable Don't Fragment (DF) bit in the TCP, UDP, or ICMP messages to determine whether a packet is allowed to be fragmented.

Before you begin

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

Procedure

Step 1

Select Voice > System.

Step 2

In the Network Settings section, configure the parameter Disable DF.

  • If you set the Disable DF to Yes, the Don't Fragment (DF) bit is disabled. In this case, the network can fragment an IP packet. This is the default behaviour.

  • If you set the Disable DF to No, the Don't Fragment (DF) bit is enabled. In this case, the network can't fragment an IP packet. This setting doesn't allow fragmentation in cases where the receiving host doesn't have sufficient resources to reassemble internet fragments.

Step 3

Click Submit All Changes.

You can also configure the parameter in the phone configuration file (cfg.xml) with the following XML string:

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

Allowed values: Yes and No

Default: Yes

Configure Internet Connection Type

You can choose how your phone receives an IP address. Set the connection type to one of the following:

  • Static IP—A static IP address for the phone.

  • Dynamic Host Configuration Protocol (DHCP)—Enables the phone to receive an IP address from the network DHCP server.

The Cisco IP phone typically operates in a network where a DHCP server assigns IP addresses to devices. Because IP addresses are a limited resource, the DHCP server periodically renews the phone lease on the IP address. If a phone loses the IP address, or if the IP address is assigned to another device on the network, the following occurs:

  • Communication between the SIP proxy and the phone is severed or degraded.

The DHCP Timeout on Renewal parameter causes the phone to request renewal of its IP address if the following occurs:

  • The phone doesn't receive an expected SIP response within programmable length of time after it sends a SIP command.

If the DHCP server returns the IP address that it originally assigned to the phone, the DHCP assignment is presumed to be operating correctly. Otherwise, the phone resets to try to fix the issue.

Before you begin

Access the Phone Web Interface.

Procedure

Step 1

Select Voice > System.

Step 2

In the IPv4 Settings section, use the Connection Type drop-down list to choose the connection type:

  • Dynamic Host Configuration Protocol (DHCP)

  • Static IP

Step 3

In the IPv6 Settings section, use the Connection Type drop-down list to choose the connection type:

  • Dynamic Host Configuration Protocol (DHCP)

  • Static IP

Step 4

If you choose Static IP, configure these settings in the Static IP Settings section:

  • Static IP—Static IP address of the phone

  • NetMask—Netmask of the phone (IPv4, only)

  • Gateway—Gateway IP address

Step 5

Click Submit All Changes.

In the phone configuration XML file (cfg.xml), enter a string in this format: 
<Connection_Type ua="rw">DHCP</Connection_Type>
<!-- available options: DHCP|Static IP -->
<Static_IP ua="rw"/>
<NetMask ua="rw"/>
<Gateway ua="rw"/>

Configure VLAN Settings

The software tags your phone voice packets with the VLAN ID when you use a virtual LAN (VLAN).

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

  • LLDP-MED

  • Cisco Discovery Protocol (CDP)

  • Network Startup Delay

  • VLAN ID (manual)

  • DHCP VLAN Option

The multiplatform phones support these four methods to obtain VLAN ID information. The phone attempts to obtain the VLAN ID information in this order:

  1. LLDP-MED

  2. Cisco Discovery Protocol (CDP)

  3. VLAN ID (manual)

  4. DHCP VLAN Option

Before you begin

Procedure

Step 1

Select Voice > System.

Step 2

In the VLAN Settings section, configure the parameters as defined in the VLAN Settings Parameters table.

Step 3

Click Submit All Changes.

You can also configure the parameters in the phone configuration file with XML(cfg.xml) code. To configure each parameter, see the syntax of the string in the VLAN Settings Parameters table.

VLAN Settings Parameters

The following table defines the function and usage of each parameter in the VLAN Settings Parameters section under the System tab in the phone web page. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Parameter Name

Description and Default Value

Enable VLAN

Controls the VLAN feature.

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

    <Enable_VLAN ua="rw">No</Enable_VLAN>

  • In the phone web interface, set to Yes to enable VLAN.

The default value is Yes.

VLAN ID

If you use a VLAN without CDP (VLAN enabled and CDP disabled), enter a VLAN ID for the IP phone. Note that only voice packets are tagged with the VLAN ID. Do not use 1 for the VLAN ID.

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

    <VLAN_ID ua="rw">1</VLAN_ID>

  • In the phone web interface, enter an appropriate value.

Valid values: An integer ranging from 0 through 4095

Default: 1

Enable CDP

Enable CDP only if you are using a switch that has Cisco Discovery Protocol. CDP is negotiation based and determines which VLAN the IP phone resides in.

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

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

  • In the phone web page: set to Yes to enable CDP.

Valid values: Yes/No

Default: Yes

Enable LLDP-MED

Choose Yes to enable LLDP-MED for the phone to advertise itself to devices that use that discovery protocol.

When the LLDP-MED feature is enabled, after the phone has initialized and Layer 2 connectivity is established, the phone sends out LLDP-MED PDU frames. If the phone receives no acknowledgment, the manually configured VLAN or default VLAN will be used if applicable. If the CDP is used concurrently, the waiting period of 6 seconds is used. The waiting period will increase the overall startup time for the phone.

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

    <Enable_LLDP-MED ua="na">Yes</Enable_LLDP-MED>

  • In the phone web interface, set to Yes to enable LLDP-MED.

Valid values: Yes/No

Default: Yes

Network Startup Delay

Setting this value causes a delay for the switch to get to the forwarding state before the phone will send out the first LLDP-MED packet. The default delay is 3 seconds. For configuration of some switches, you might need to increase this value to a higher value for LLDP-MED to work. Configuring a delay can be important for networks that use Spanning Tree Protocol.

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

    <Network_Startup_Delay ua="na">3</Network_Startup_Delay>

  • In the phone web interface, enter the delay in seconds.

Valid values: An integer ranging from 1 through 300

Default: 3

DHCP VLAN Option

A predefined DHCP VLAN option to learn the voice VLAN ID. You can use the feature only when no voice VLAN information is available by CDP/LLDP and manual VLAN methods. CDP/LLDP and manual VLAN are all disabled.

Set the value to Null to disable DHCP VLAN option.

Cisco recommends that you use DHCP Option 132.

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

    <DHCP_VLAN_Option ua="na">132</DHCP_VLAN_Option>

  • In the phone web page: specify the DHCP VLAN option.

SIP Configuration

SIP settings for the Cisco IP Phone are configured for the phone in general and for the extensions.

Configure the Basic SIP Parameters

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Parameters section, set the parameters as described in the SIP Parameters table.

Step 3

Click Submit All Changes.

SIP Parameters

Parameter

Description

Max Forward

Specifies SIP Max Forward value.

Perform one of the following.

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

    <Max_Forward ua="na">70</Max_Forward>

  • In the phone web page, enter an appropriate value.

Value range: 1 to 255

Default: 70

Max Redirection

Specifies number of times an invite can be redirected to avoid an infinite loop.

Perform one of the following.

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

    <Max_Redirection ua="na">5</Max_Redirection>

  • In the phone web page, enter an appropriate value.

Default: 5

Max Auth

Specifies the maximum number of times (from 0 to 255) a request can be challenged.

Perform one of the following.

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

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

  • In the phone web page, enter an appropriate value.

Allowed value: 0 to 255

Default: 2

SIP User Agent Name

Used in outbound requests.

Perform one of the following.

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

    <SIP_User_Agent_Name ua="na">$VERSION</SIP_User_Agent_Name>

  • In the phone web page, enter an appropriate name.

Default: $VERSION

If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed

SIP Server Name

Server header used in responses to inbound responses.

Perform one of the following.

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

    <SIP_Server_Name ua="na">$VERSION</SIP_Server_Name>

  • In the phone web page, enter an appropriate name.

Default: $VERSION

SIP Reg User Agent Name

User-Agent name to be used in a REGISTER request. If this is not specified, the SIP User Agent Name is also used for the REGISTER request.

Perform one of the following.

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

    <SIP_Reg_User_Agent_Name ua="na">agent name</SIP_Reg_User_Agent_Name>

  • In the phone web page, enter an appropriate name.

Default: Blank

SIP Accept Language

Accept-Language header used.

Perform one of the following.

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

    <SIP_Accept_Language ua="na">en</SIP_Accept_Language>

  • In the phone web page, enter an appropriate language.

There is no default. If empty, the header is not included.

DTMF Relay MIME Type

MIME Type used in a SIP INFO message to signal a DTMF event. This field must match that of the Service Provider.

Perform one of the following.

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

    <DTMF_Relay_MIME_Type ua="na">application/dtmf-relay</DTMF_Relay_MIME_Type>

  • In the phone web page, enter an appropriate MIME type.

Default: application/dtmf-relay

Hook Flash MIME Type

MIME Type used in a SIPINFO message to signal a hook flash event.

Perform one of the following.

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

    <Hook_Flash_MIME_Type ua="na">application/hook-flash</Hook_Flash_MIME_Type>

  • In the phone web page, enter an appropriate MIME type for a SIPINFO message.

Default:

Remove Last Reg

Enables you to remove the last registration before registering a new one if the value is different.

Set to Yes to remove the last registration.

Perform one of the following.

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

    <Remove_Last_Reg ua="na">No</Remove_Last_Reg>

  • In the phone web page, Select Yes or No.

Allowed values: Yes or No

Default: No

Use Compact Header

If set to yes, the phone uses compact SIP headers in outbound SIP messages. If inbound SIP requests contain normal headers, the phone substitutes incoming headers with compact headers. If set to no, the phones use normal SIP headers. If inbound SIP requests contain compact headers, the phones reuse the same compact headers when generating the response, regardless of this setting.

Perform one of the following.

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

    <Use_Compact_Header ua="na">No</Use_Compact_Header>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Default: No

Escape Display Name

Enables you to keep the Display Name private.

Set to Yes if you want the IP phone to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages.

Perform one of the following.

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

    <Escape_Display_Name ua="na">No</Escape_Display_Name>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Default: Yes.

Talk Package

Enables support for the BroadSoft Talk Package that lets users answer or resume a call by clicking a button in an external application.

Perform one of the following.

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

    <Talk_Package ua="na">No</Talk_Package>

  • In the phone web page, select Yes to enable the Talk Package.

Allowed values: Yes or No

Default: No

Hold Package

Enables support for the BroadSoft Hold Package, which lets users place a call on hold by clicking a button in an external application.

Perform one of the following.

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

    <Hold_Package ua="na">No</Hold_Package>

  • In the phone web page, select Yes to enable support for the Hold Package.

Allowed values: Yes or No

Default: No

Conference Package

Enables support for the BroadSoft Conference Package that enables users to start a conference call by clicking a button in an external application.

Perform one of the following.

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

    <Conference_Package ua="na">No</Conference_Package>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Default: No

RFC 2543 Call Hold

If set to yes, unit includes c=0.0.0.0 syntax in SDP when sending a SIP re-INVITE to the peer to hold the call. If set to no, unit will not include the c=0.0.0.0 syntax in the SDP. The unit will always include a=sendonly syntax in the SDP in either case.

Perform one of the following.

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

    <RFC_2543_Call_Hold ua="na">Yes</RFC_2543_Call_Hold>

  • In the phone web page, Yes or No.

Allowed values: Yes or No

Default: Yes

Random REG CID on Reboot

If set to yes, the phone uses a different random call-ID for registration after the next software reboot. If set to no, the Cisco IP phone tries to use the same call-ID for registration after the next software reboot. The Cisco IP phone always uses a new random Call-ID for registration after a power-cycle, regardless of this setting.

Perform one of the following.

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

    <Random_REG_CID_on_Reboot ua="na">No</Random_REG_CID_on_Reboot>

  • In the phone web page, select Yes or No.

Default: No.

SIP TCP Port Min

Specifies the lowest TCP port number that can be used for SIP sessions.

Perform one of the following.

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

    <SIP_TCP_Port_Min ua="na">5060</SIP_TCP_Port_Min>

  • In the phone web page, enter an appropriate value.

Default: 5060

SIP TCP Port Max

Specifies the highest TCP port number that can be used for SIP sessions.

Perform one of the following.

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

    <SIP_TCP_Port_Max ua="na">5080</SIP_TCP_Port_Max>

  • In the phone web page, enter an appropriate value.

Default: 5080

Caller ID Header

Provides the option to take the caller ID from PAID-RPID-FROM, PAID-FROM, RPID-PAID-FROM, RPID-FROM, or FROM header.

Perform one of the following.

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

    <Caller_ID_Header ua="na">PAID-RPID-FROM</Caller_ID_Header>

  • In the phone web page, select an option.

Allowed values: PAID-RPID-FROM, AID-FROM, RPID-PAID-FROM, RPID-FROM, and FROM

Default: PAID-RPID-FROM

Hold Target Before Refer

Controls whether to hold call leg with transfer target before sending REFER to the transferee when initiating a fully-attended call transfer (where the transfer target has answered).

Perform one of the following.

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

    <Hold_Target_Before_Refer ua="na">No</Hold_Target_Before_Refer>

  • In the phone web page, select Yes or No.

Default: No

Dialog SDP Enable

When enabled and the Notify message body is too big causing fragmentation, the Notify message xml dialog is simplified; Session Description Protocol (SDP) is not included in the dialog xml content.

Perform one of the following.

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

    <Dialog_SDP_Enable ua="na">No</Dialog_SDP_Enable>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Default: No

Keep Referee When Refer Failed

If set to yes, it configures the phone to immediately handle NOTIFY sipfrag messages.

Perform one of the following.

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

    <Keep_Referee_When_Refer_Failed ua="na">No</Keep_Referee_When_Refer_Failed>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Default: No

Display Diversion Info

Display the Diversion info included in SIP message on LCD or not.

Perform one of the following.

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

    <Display_Diversion_Info ua="na">No</Display_Diversion_Info>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Display Anonymous From Header

Show the caller ID from the SIP INVITE message “From” header when set to Yes, even if the call is an anonymous call. When the parameter is set to no, the phone displays "Anonymous Caller" as the caller ID.

Perform one of the following.

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

    <Display_Anonymous_From_Header ua="na">No</Display_Anonymous_From_Header>

  • In the phone web page, select Yes or No.

Allowed values: Yes or No

Default: No

Sip Accept Encoding

Supports the content-encoding gzip feature.

If gzip is selected, the SIP message header contains the string “Accept-Encoding: gzip”, and the phone is able to process the SIP message body, which is encoded with the gzip format.

Perform one of the following.

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

    <Sip_Accept_Encoding ua="na">none</Sip_Accept_Encoding>

  • In the phone web page, enter an appropriate MIME type for a SIPINFO message.

Allowed values: none and gzip

Default: none

SIP IP Preference

Sets if the phone uses IPv4 or IPv6.

Perform one of the following.

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

    <SIP_IP_Preference ua="na">IPv4</SIP_IP_Preference>

  • In the phone web page, select IPv4 or IPv6.

Allowed values: IPv4/IPv6

Default: IPv4.

Disable Local Name To Header

Controls the display name in “Directory”, “Call History”, and in the “To” header during an outgoing call.

Perform one of the following.

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

    <Disable_Local_Name_To_Header ua="na">No</Disable_Local_Name_To_Header>

  • In the phone web page, select Yes to disable the display name.

Allowed values: Yes/No

Default: No

Forced NAT64

 Sets the web parameter for Forced NAT64. Takes effect after a warm reboot.

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

    <Forced_NAT64 ua="na">Yes</Forced_NAT64>

  • In the phone web page, select Yes.

Allowed values: Yes/No

Default: No

Configure the SIP Timer Values

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Timer Values section, set the SIP timer values in seconds as described in SIP Timer Values (sec).

Step 3

Click Submit All Changes.

SIP Timer Values (sec)

Parameter

Description

SIP T1

RFC 3261 T1 value (RTT estimate) that can range from 0 to 64 seconds.

Default: 0.5 seconds

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses) that can range from 0 to 64 seconds.

Default: 4 seconds

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds.

Default: 5 seconds.

SIP Timer B

INVITE time-out value, which can range from 0 to 64 seconds.

Default: 16 seconds.

SIP Timer F

Non-INVITE time-out value, which can range from 0 to 64 seconds.

Default: 16 seconds.

SIP Timer H

INVITE final response, time-out value, which can from 0 to 64 seconds.

Default: 16 seconds.

SIP Timer D

ACK hang-around time, which can range from 0 to 64 seconds.

Default: 16 seconds.

SIP Timer J

Non-INVITE response hang-around time, which can range from 0 to 64 seconds.

Default: 16 seconds.

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000.

Default: 240 seconds

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Ranges from 0 to 2000000.

Default: 30

Reg Min Expires

Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used.

Reg Max Expires

Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used.

Reg Retry Intv

Interval to wait before the Cisco IP Phone retries registration after failing during the last registration.The range is from 1 to 2147483647

Default: 30

See the note below for additional details.

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match<Retry Reg RSC>, the Cisco IP Phone waits for the specified length of time before retrying. If this interval is 0, the phone stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0.

Default: 1200

See the note below for additional details.

Reg Retry Random Delay

Random delay range (in seconds) to add to <Register Retry Intvl> when retrying REGISTER after a failure. Minimum and maximum random delay to be added to the short timer. The range is from 0 to 2147483647.

Default: 0

Reg Retry Long Random Delay

Random delay range (in seconds) to add to <Register Retry Long Intvl> when retrying REGISTER after a failure.

Default: 0

Reg Retry Intvl Cap

Maximum value of the exponential delay. The maximum value to cap the exponential backoff retry delay (which starts at the Register Retry Intvl and doubles every retry). Defaults to 0, which disables the exponential backoff (that is, the error retry interval is always at the Register Retry Intvl). When this feature is enabled, the Reg Retry Random Delay is added to the exponential backoff delay value. The range is from 0 to 2147483647.

Default: 0

Sub Min Expires

Sets the lower limit of the REGISTER expires value returned from the Proxy server.

Sub Max Expires

Sets the upper limit of the REGISTER minexpires value returned from the Proxy server in the Min-Expires header.

Default: 7200.

Sub Retry Intvl

This value (in seconds) determines the retry interval when the last Subscribe request fails.

Default: 10.


Note

The phone can use a RETRY-AFTER value when it is received from a SIP proxy server that is too busy to process a request (503 Service Unavailable message). If the response message includes a RETRY-AFTER header, the phone waits for the specified length of time before to REGISTER again. If a RETRY-AFTER header is not present, the phone waits for the value specified in the Reg Retry Interval or the Reg Retry Long Interval.

Configure the Response Status Code Handling

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the Response Status Code Handling section, set the values as specified in the Response Status Code Handling Paramters table.

Step 3

Click Submit All Changes.

Response Status Code Handling Paramters

The following table defines the function and usage of the parameters in the Response Status Code Handling section under the SIP tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 6. Response Status Code Handling Paramters

Parameter

Description

Try Backup RSC

This parameter may be set to invoke failover upon receiving specified response codes.

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??

Perform one of the following:

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

    <Try_Backup_RSC ua="na"/>

  • In the phone web page, enter an appropriate value.

Default: Blank

Retry Reg RSC

Interval to wait before the phone retries registration after failing during the last registration.

For example, you can enter numeric values 500 or a combination of numeric values plus wild cards if multiple values are possible. For the later, you can use 5?? to represent all SIP Response messages within the 500 range. If you want to use multiple ranges, you can add a comma "," to delimit values of 5?? and 6??

Perform one of the following:

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

    <Retry_Reg_RSC ua="na"/>

  • In the phone web page, enter an appropriate value.

Default: Blank

Configure NTP Server

You can configure NTP servers with IPv4 and IPv6. You can also configure NTP server with DHCPv4 option 42 or DHCPv6 option 56. Configuring NTP with Primary NTP Server and Secondary NTP server parameters has higher priority over configuring NTP with DHCPv4 option 42 or DHCPv6 option 56.

Before you begin

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

Procedure

Step 1

Select Voice > Systems.

Step 2

In the Optional Network Configuration section, set the IPv4 or IPv6 address as described in the NTP Server Parameters table.

Step 3

Click Submit All Changes.

NTP Server Parameters

The following table defines the function and usage of NTP server parameters in the Optional Network Configuration section under the System tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 7. NTP Server Parameters

Parameter

Description

Primary NTP Server

IP address or name of the primary NTP server used to synchronize its time.

Perform one of the following:

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

    <Primary_NTP_Server ua="rw"/>

  • In the phone web page, enter the IP address of the priamry NTP server.

Default: Blank

Secondary NTP Server

IP address or name of the secondary NTP server used to synchronize its time.

Perform one of the following:

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

    <Secondary_NTP_Server ua="rw"/>

  • In the phone web page, enter the IP address of the secondary NTP server.

Default: Blank

Configure the RTP Parameters

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

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

Step 3

Click Submit All Changes.

RTP Parameters

The following table defines the function and usage of the parameters in the RTP Parameters section under the SIP tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 8. RTP Parameters

Parameter

Description

RTP Port Min

Minimum port number for RTP transmission and reception.

Perform one of the following:

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

    <RTP_Port_Min ua="na">16384</RTP_Port_Min>

  • In the phone web page, enter an appropriate port number.

Allowed values: 2048 to 49151

If the value range (RTP Port Max - RTP Port Min) is less than 16 or you configure the parameter incorrectly, the RTP port range (16382 to 32766) is used instead.

Default: 16384

RTP Port Max

Maximum port number for RTP transmission and reception.

Perform one of the following:

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

    <RTP_Port_Max ua="na">16482</RTP_Port_Max>

  • In the phone web page, enter an appropriate port number.

Allowed values: 2048 to 49151

If the value range (RTP Port Max - RTP Port Min) is less than 16 or you configure the parameter incorrectly, the RTP port range (16382 to 32766) is used instead.

Default: 16482

RTP Packet Size

Specifies packet size in seconds.

Perform one of the following:

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

    <RTP_Packet_Size ua="na">0.02</RTP_Packet_Size>

  • In the phone web page, enter an appropriate value to specify the packet size.

Allowed values: Ranges from 0.01 to 0.13. Valid values must be a multiple of 0.01 seconds.

Default: 0.02

Max RTP ICMP Err

Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the phone terminates the call. If value is set to 0, the phone ignores the limit on ICMP errors.

Perform one of the following:

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

    <Max_RTP_ICMP_Err ua="na">0</Max_RTP_ICMP_Err>

  • In the phone web page, enter an appropriate value.

Default: 0

RTCP Tx Interval

Interval for sending out RTCP sender reports on an active connection.

Perform one of the following:

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

    <RTCP_Tx_Interval ua="na">5</RTCP_Tx_Interval>

  • In the phone web page, enter an appropriate value.

Allowed values: 0 to 255 seconds

Default: 0

Call Statistics

Specifies whether the phone sends end-of-call statistics within SIP messages when a call terminates or is put on hold.

Perform one of the following:

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

    <Call_Statistics ua="na">No</Call_Statistics>

  • In the phone web page, select Yes to enable this feature.

Allowed values: Yes and No

Default: No

SDP IP Preferences

Select the preferred IP that the phone uses as RTP address.

If the phone is in dual-mode and has both ipv4 and ipv6 addresses, it will always include both addresses in SDP by attributes "a=altc …

If IPv4 address is selected, then ipv4 address has higher priority than ipv6 address in SDP and indicates that phone prefers using ipv4 RTP address.

If the phone has only ipv4 address or ipv6 address, SDP does not have ALTC attributes and RTP address is specified in “c=” line.

Perform one of the following:

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

    <SDP_IP_Preference ua="na">IPv4</SDP_IP_Preference>

  • In the phone web page, select the preferred IP .

Allowed values: IPv4 and IPv6

Default: IPv4

RTP Before ACK

Allows you to specify whether an RTP session starts before or after an ACK is received from the calling party.

Perform one of the following:

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

    <RTP_Before_ACK ua="na">No</RTP_Before_ACK>

  • In the phone web page select:

    • Yes: An RTP session doesn't await an ACK, but starts after a 200 OK message is sent.

    • No: An RTP session doesn't start until an ACK is received from the calling party.

Allowed values: Yes and No

Default: No

SSRC Reset on RE-INVITE

Controls whether to reset the Synchronization Source (SSRC) for the new RTP and SRTP sessions.

Perform one of the following:

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

    <SSRC_Reset_on_RE-INVITE ua="na">Yes</SSRC_Reset_on_RE-INVITE>

  • In the phone web page select:

    • Yes: the phone can avoid the call transfer error, where only one person on the call hears the audio. This occurs on calls of 30 minutes or longer, and often on three-way calls.

    • No: the SSRC still remains during a long duration call. In this case, this error might occur.

Allowed values: Yes and No

Default: No

Enable SSRC Reset for the New RTP and SRTP Sessions

You can enable the SSRC Reset on RE-INVITE to avoid a call transfer error, where only one person on the call hears the audio. This error occurs on calls of 30 minutes or longer, and often on three-way calls.

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the RTP Parameters section, set the parameter SSRC Reset on RE-INVITE to Yes.

You can also configure this parameter in the configuration file:

<SSRC_Reset_on_RE-INVITE ua="na">Yes</SSRC_Reset_on_RE-INVITE>

Allowed values: Yes and No.

Default: No

Note

 

If you set the parameter to No, the SSRC remains for the new RTP and SRTP sessions (SIP re-INVITEs). The call transfer error might occur during a long duration call.

Step 3

Click Submit All Changes.

Control SIP and RTP Behaviour in Dual Mode

You can control SIP and RTP parameters with SIP IP Preference and SDP IP Preference fields when phone is in dual mode.

SIP IP Preference parameter defines which IP address phone tries first when it is in dual mode.

Table 9. SIP IP Preference and IP Mode

IP Mode

SIP IP Preference

Address List from DNS, Priority, Result

P1 - First Priority Address

P2 - Second Priority Address

Failover Sequence

Dual Mode

IPv4

P1- 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 1.1.1.1 first.

1.1.1.1 ->2009:1:1:1:1 ->

2.2.2.2 -> 2009:2:2:2:2

Dual Mode

IPv6

P1- 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 2009:1:1:1::1 first.

2009:1:1:1:1 ->

1.1.1.1 -> 2009:2:2:2:2 ->

2.2.2.2

Dual Mode

IPv4

P1- 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 2009:1:1:1::1 first.

2009:1:1:1:1 -> 2.2.2.2 -> 2009:2:2:2:2

Dual Mode

IPv6

P1- 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 1.1.1.1 first.

2009:1:1:1:1 -> 2009:2:2:2:2

->2.2.2.2

IPv4 Only

IPv4

or

IPv6

P1 - 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 1.1.1.1 first.

1.1.1.1 -> 2.2.2.2

IPv6 Only

IPv4

or

IPv6

P1 - 1.1.1.1, 2009:1:1:1::1

P2 - 2.2.2.2, 2009:2:2:2::2

Result: Phone will send the SIP messages to 2009:1:1:1::1 first.

2009:1:1:1:1 -> 2009:2:2:2::2

SDP IP Preference - ALTC helps peers in dual-mode negotiate RTP address family.

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Parameters section, select IPv4 or IPv6 in the SIP IP Preference field.

For details, see SDP IP Preference field in the SIP Parameters table.

Step 3

In the RTP Parameters section, select IPv4 or IPv6 in the SDP IP Preference field.

For details, see SDP IP Preference in the RTP Parameters table.

Configure RTP IP and SIP IP Alignment

You can configure the phone so that the RTP IP always aligns with SIP IP. SIP INVITE packet will fill the SDP IP with the phone's registration IP (the IP used by the phone for SIP communication), regardless of whether it's IPv4 or IPv6. When the auto selection is enabled, SIP INVITE packet fills the SDP IP with the phone's registration IP that the phone uses for SIP communication.

Before you begin

Procedure

Step 1

Select Voice > SIP.

Step 2

In the RTP Parameters section, select Auto in the SDP IP Preference field.

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

<SDP_IP_Preference ua="na">Auto</SDP_IP_Preference>

Options: IPv4 | IPv6 | Auto

Default: IPv4

Step 3

Click Submit All Changes.

Configure the SDP Payload Types

Your Cisco IP Phone supports RFC4733. You can choose from three audio-video transport (AVT) options to send DTMF pulses to the server.

Configured dynamic payloads are used for outbound calls only when the Cisco IP Phone presents a Session Description Protocol (SDP) offer. For inbound calls with an SDP offer, the phone follows the caller’s assigned dynamic payload type.

The Cisco IP Phone uses the configured codec names in outbound SDP. For incoming SDP with standard payload types of 0-95, the phone ignores the codec names. For dynamic payload types, the phone identifies the codec by the configured codec names. The comparison is case-sensitive, so you need to set the name correctly.

You can also configure the parameters in the phone configuration file (cfg.xml). To configure each of the parameters, see the syntax of the string in SDP Payload Types.

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SDP Payload Types section, set the value as specified in SDP Payload Types.

  • AVT Dynamic Payload—is any nonstandard data. Both sender and receiver must agree on a number. The range is from 96 to 127. The default is 101.

  • AVT 16kHz Dynamic Payload —is any nonstandard data. Both sender and receiver must agree on a number. The range is from 96 to 127. The default is 107.

  • AVT 48kHz Dynamic Payload —is any nonstandard data. Both sender and receiver must agree on a number. The range is from 96 to 127. The default is 108.

Step 3

Click Submit All Changes.

SDP Payload Types

Parameter

Description

G722.2 Dynamic Payload

G722 Dynamic Payload type.

Perform one of the following:

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

    <Try_Backup_RSC ua="na"/>

  • In the phone web page, enter an appropriate value.

Allowed values:

Default: 96

iLBC Dynamic Payload

iLBC Dynamic Payload type.

Default: 97

OPUS Dynamic Payload

OPUS Dynamic Payload type.

Default: 99

AVT Dynamic Payload

AVT dynamic payload type. Ranges from 96-127.

Default: 101

INFOREQ Dynamic Payload

INFOREQ Dynamic Payload type.

H264 BP0 Dynamic Payload

H264 BPO Dynamic Payload type.

Default: 110

H264 HP Dynamic Payload

H264 HP Dynamic Payload type.

Default: 110

G711u Codec Name

G711u codec name used in SDP.

Perform one of the following:

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

    <G711u_Codec_Name ua="na">PCMU</G711u_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: PCMU

G711a Codec Name

G711a codec name used in SDP.

Perform one of the following:

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

    <G711a_Codec_Name ua="na">PCMU</G711a_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: PCMA

G729a Codec Name

G729a codec name used in SDP.

Perform one of the following:

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

    <G729a_Codec_Name ua="na">PCMU</G729a_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: G729a

G729b Codec Name

G729b codec name used in SDP.

Perform one of the following:

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

    <G729b_Codec_Name ua="na">PCMU</G729b_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: G729b

G722 Codec Name

G722 codec name used in SDP.

Perform one of the following:

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

    <G722_Codec_Name ua="na">PCMU</G722_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: G722

G722.2 Codec Name

G722.2 codec name used in SDP.

Perform one of the following:

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

    <G722.2_Codec_Name ua="na">PCMU</G722.2_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: G722.2

iLBC Codec Name

iLBC codec name used in SDP.

Perform one of the following:

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

    <iLBC_Codec_Name ua="na">iLBC</iLBC_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: iLBC

OPUS Codec Name

OPUS codec name used in SDP.

Perform one of the following:

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

    <OPUS_Codec_Name ua="na">OPUS</OPUS_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: OPUS

AVT Codec Name

AVT codec name used in SDP.

Perform one of the following:

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

    <AVT_Codec_Name ua="na">telephone-event</AVT_Codec_Name>

  • In the phone web page, enter an appropriate codec name.

Allowed values:

Default: telephone-event

AVT 16 kHz Dynamic Payload

AVT dynamic payload type for the 16 kHz clock rate.

Perform one of the following:

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

    <AVT_16kHz_Dynamic_Payload ua="na">107</AVT_16kHz_Dynamic_Payload>

  • In the phone web page, enter the payload.

Range: 96-127

Default: 107

AVT 48 kHz Dynamic Payload

AVT dynamic payload type for the 48 kHz clock rate.

Perform one of the following:

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

    <AVT_48kHz_Dynamic_Payload ua="na">108</AVT_48kHz_Dynamic_Payload>

  • In the phone web page, enter the payload.

Range: 96-127

Default: 108

Configure the SIP Settings for Extensions

Before you begin

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

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the SIP Settings section, set the parameter values as described in the Parameters for SIP Settings on Extensions table.

Step 3

Click Submit All Changes.

Parameters for SIP Settings on Extensions

The following table defines the function and usage of the parameters in the SIP Settings section under the Ext(n) tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 10. SIP Settings in Extensions

Parameter

Description

SIP Transport

Specifies the transport protocol for SIP messages.

Perform one of the following:

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

    <SIP_Transport_1_ ua="na">UDP</SIP_Transport_1_>

  • In the phone web page, select the transport protocol type.

    • UDP

    • TCP

    • TLS

    • AUTO

    AUTO allows the phone to select the appropriate protocol automatically, based on the NAPTR records on the DNS server. See Configure the SIP Transport for more details.

Default: UDP

SIP Port

The phone's port number for SIP message listening and transmission.

Note

 

Specify the port number here only when you are using UDP as the SIP transport protocol.

If you are using TCP, the system uses a random port within the range specified in SIP TCP Port Min and SIP TCP Port Max on the Voice > SIP tab.

If you need to specify a port of SIP proxy server, you can specify it using the Proxy field or the XSI Host Server field.

Perform one of the following:

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

    <SIP_Port_1_ ua="na">5060</SIP_Port_1_>

  • In the phone web page, enter an appropriate port number.

Default: 5060

SIP 100REL Enable

Individually enables the SIP 100REL feature.

When enabled, the phone supports the 100REL SIP extension for reliable transmission of provisional responses (18x) and uses the PRACK requests.

Perform one of the following:

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

    <SIP_100REL_Enable_1_ ua="na">Yes</SIP_100REL_Enable_1_>

  • In the phone web page, select Yes to enable the feature.

Allowed values: Yes and No

Default: No

Precondition Support

Determines whether the phone includes the precondition tag (defined in RFC 3312) in the Supported header field.

  • Disabled: The phone doesn't include the precondition tag in the Supported header filed. And the phone doesn't return the 183 response when it receives the INVITE request that contains the QoS precondition in the SDP description.

  • Enabled: The phone includes the precondition tag in the Supported header field.

Perform one of the following:

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

    <Precondition_Support_1_ ua="na">Enabled</Precondition_Support_1_>

  • In the phone web page, select Enabled to enable the feature.

Allowed values: Disabled and Enabled

Default: Disabled

EXT SIP Port

The external SIP port number.

Perform one of the following:

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

    <EXT_SIP_Port_1_ ua="na">5060</EXT_SIP_Port_1_>

  • In the phone web page, enter a port number.

Allowed values:

Default: 5060

Auth Resync-Reboot

The Cisco IP Phone authenticates the sender when it receives a NOTIFY message with the following requests:

  • resync

  • reboot

  • report

  • restart

  • XML-service

Perform one of the following:

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

    <Auth_Resync-Reboot_1_ ua="na">No</Auth_Resync-Reboot_1_>

  • In the phone web page, select Yes to enable the feature.

Allowed values: Yes and No

Default: Yes

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it receives the Proxy-Require header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported.

Perform one of the following:

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

    <SIP_Proxy-Require_1_ ua="na">header<SIP_Proxy-Require_1_>

  • In the phone web interface, enter the appropriate header in the field provided.

Default: Blank

SIP Remote-Party-ID

The Remote-Party-ID header to use instead of the From header. Select Yes to enable.

Default: Yes

Referor Bye Delay

Controls when the phone sends BYE to terminate stale call legs upon completion of call transfers. Multiple delay settings (Referor, Refer Target, Referee, and Refer-To Target) are configured on this screen.

Perform one of the following:

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

    <Referor_Bye_Delay_1_ ua="na">4</Referor_Bye_Delay_1_>

  • In the phone web page, enter the appropriate period of time in seconds.

Allowed values: An integer from 0 through 65535

Default: 4

Refer-To Target Contact

Indicates the refer-to target.

Perform one of the following:

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

    <Refer-To_Target_Contact_1_ ua="na">No</Refer-To_Target_Contact_1_>

  • In the phone web page, select Yes to send the SIP Refer to the contact.

Allowed values: Yes and No

Default: No

Referee Bye Delay

Specifies the Referee Bye Delay time in seconds.

Perform one of the following:

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

    <Referee_Bye_Delay_1_ ua="na">0</Referee_Bye_Delay_1_>

  • In the phone web page, enter the appropriate period of time in seconds.

Allowed values: An integer from 0 through 65535

Default: 0

Refer Target Bye Delay

Specifies the Refer Target Bye Delay time in seconds.

Perform one of the following:

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

    <Refer_Target_Bye_Delay_1_ ua="na">0</Refer_Target_Bye_Delay_1_>

  • In the phone web page, enter the appropriate period of time in seconds.

Allowed values: An integer from 0 through 65535

Default: 0

Sticky 183

Controls the first 183 SIP response for an outbound INVITE. To enable this feature,

Perform one of the following:

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

    <Sticky_183_1_ ua="na">No</Sticky_183_1_>

  • In the phone web page, select Yes to enable this feature.

    When enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE.

Allowed values: Yes and No

Default: No

Auth INVITE

Controls if authorization is required for initial incoming INVITE requests from the SIP proxy. To enable this feature.

Perform one of the following:

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

    <Auth_INVITE_1_ ua="na">No</Auth_INVITE_1_>

  • In the phone web page, select Yes to enable this feature.

    When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.

Allowed values: Yes and No

Default: No

Ntfy Refer On 1xx-To-Inv

If set to Yes, as a transferee, the phone will send a NOTIFY with Event:Refer to the transferor for any 1xx response returned by the transfer target, on the transfer call leg.

If set to No, the phone will only send a NOTIFY for final responses (200 and higher).

Perform one of the following:

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

    <Ntfy_Refer_On_1xx-To-Inv_1_ ua="na">Yes</Ntfy_Refer_On_1xx-To-Inv_1_>

  • In the phone web page, select Yes to enable this feature.

Allowed values: Yes and No

Default: Yes

Set G729 annexb

Configure G.729 Annex B settings.

Perform one of the following:

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

    <Set_G729_annexb_1_ ua="na">Yes</Set_G729_annexb_1_>

  • In the phone web page, select Yes to enable this feature.

Allowed values:

  • None

  • No

  • Yes

  • Follow silence supp setting

Default: Yes

User Equal Phone

When a tel URL is converted to a SIP URL and the phone number is represented by the user portion of the URL, the SIP URL includes the optional: user=phone parameter (RFC3261). For example:

To: sip:+12325551234@example.com; user=phone

Perform one of the following:

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

    <User_Equal_Phone_1_ ua="na">Yes</User_Equal_Phone_1_>

  • In the phone web page, select Yes to enable this feature.

Allowed values: Yes and No

Default: No

Call Recording Protocol

Determines the type of recording protocol that the phone uses. Options are:

  • SIPINFO

  • SIPREC

Perform one of the following:

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

    <Call_Recording_Protocol_1_ ua="na">SIPREC</Call_Recording_Protocol_1_>

  • In the phone web page, select a protocol from the list.

Allowed values: SIPREC|SIPINFO

Default: SIPREC

Privacy Header

Sets user privacy in the SIP message in the trusted network.

The privacy header options are:

  • Disabled (default)

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

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

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

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

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

Perform one of the following:

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

    <Privacy_Header_1_ ua="na">Disabled</Privacy_Header_1_>

  • In the phone web page, select an option from the list.

Allowed values: Disabled|none|header|session|user|id

Default: Disabled

P-Early-Media Support

Controls whether the P-Early-Media header is included in the SIP message for an outgoing call.

Perform one of the following:

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

    <P-Early-Media_Support_1_ ua="na">No</P-Early-Media_Support_1_>

  • In the phone web interface, to include the P-Early-Media header, select Yes.

Allowed values: Yes and No

Default: No

Configure the SIP Proxy Server

Before you begin

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

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the Proxy and Registration section, set the parameter values as described in the SIP Proxy and Registration for Extension Parameters table.

Step 3

Click Submit All Changes.

SIP Proxy and Registration for Extension Parameters

The following table defines the function and usage of the parameters in the Proxy and Registration section under the Ext(n) tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 11. SIP Proxy and Registration for Extension

Parameter

Description

Proxy

SIP proxy server and port number set by the service provider for all outbound requests. For example: 192.168.2.100:6060.

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

    <Proxy_1_ ua="na">64.101.154.134</Proxy_1_>

    <RTP_Port_Max ua="na">16482</RTP_Port_Max>

  • In the phone web page, enter SIP proxy server and port number.

When you need to refer to this proxy in another setting, for example, the speed dial line key configuration, use the $PROXY macro variable.

Default: The port number is optional. If you don't specify a port, the default port 5060 is used for UDP, and the default port 5061 is used for TLS.

Outbound Proxy

Specifies an IP address or domain name. All outbound requests are sent as the first hop.

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

    <Outbound_Proxy_1_ ua="na">10.79.78.45</Outbound_Proxy_1_>

  • In the phone web page, enter an IP address and a domain name.

Default: Empty

Proxy

Outbound Proxy

For Survivable Remote Site Telephony (SRST) support

These parameters can be configured with an extension that includes a statically-configured DNS SRV record or DNS A record. This allows for failover and fallback functionality with a secondary proxy server.

The format for the parameter value is as follows:

FQDN format: hostname[:port][:SRV=host-list OR :A=ip-list]

Where:

  • host-list: srv[|srv[|srv...]]

  • srv: hostname[:port][:p=priority][:weight][:A=ip-list]

  • ip-list: ip-addr[,ip-addr[,ip-addr...]]

Default:

  • Priority is 0.

  • Weight is 1.

  • Port is 5060 and 5061 for UDP and TLS respectively.

Alternate Proxy

Alternate Outbound Proxy

This feature provides fast fall back when there is network partition at the Internet or when the primary proxy (or primary outbound proxy) is not responsive or available. The feature works well in a Verizon deployment environment as the alternate proxy is the Integrated Service Router (ISR) with analog outbound phone connection.

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

    <Alternate_Proxy_1_ ua="na">10.74.23.43</Alternate_Proxy_1_><Alternate_Outbound_Proxy_1_ ua="na">10.74.23.44</<Alternate_Outbound_Proxy_1_>

  • In the phone web page, enter the proxy server addresses and port numbers in these fields.

After the phone is registered to the primary proxy and the alternate proxy (or primary outbound proxy and alternate outbound proxy), the phone always sends out INVITE and Non-INVITE SIP messages (except registration) via the primary proxy. The phone always registers to both the primary and alternate proxies. If there is no response from the primary proxy after timeout (per the SIP RFC spec) for a new INVITE, the phone attempts to connect with the alternate proxy. The phone always tries the primary proxy first, and immediately tries the alternate proxy if the primary is unreachable.

Active transactions (calls) never fall back between the primary and alternate proxies. If there is fall back for a new INVITE, the subscribe/notify transaction will fall back accordingly so that the phone's state can be maintained properly. You must also set Dual Registration in the Proxy and Registration section to Yes.

Default: Empty

Register

Enables periodic registration with the proxy. This parameter is ignored if a proxy is not specified.

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

    <Register_1_ ua="na">Yes</Register_1_>

  • In the phone web page, To enable this feature, select Yes.

Valid values: Yes and No

Default: Yes

Make Call Without Reg

Enables making outbound calls without successful (dynamic) registration by the phone.

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

    <Make_Call_Without_Reg_1_ ua="na">No</Make_Call_Without_Reg_1_>

  • In the phone web page, To enable this feature, select Yes. If set to No, the dial tone plays only when registration is successful.

Valid values: Yes and No

Default: No

Register Expires

Defines how often the phone renews registration with the proxy. If the proxy responds to a REGISTER with a lower expires value, the phone renews registration based on that lower value instead of the configured value.

If registration fails with an “Expires too brief” error response, the phone retries with the value specified in the Min-Expires header of the error.

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

    <Register_Expires_1_ ua="na">3600</Register_Expires_1_>

  • In the phone web page, enter a value in seconds to define how often the phone renews registration with the proxy.

Valid values: Numeric. The range is from 32 seconds to 2000000 seconds.

Default: 3600 seconds

Use DNS SRV

Enables DNS SRV lookup for the proxy and outbound proxy.

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

    <Use_DNS_SRV_1_ ua="na">Yes</Use_DNS_SRV_1_>

  • In the phone web page, To enable this feature, select Yes.

Valid values: Yes and No

Default: No

Proxy Fallback Intvl

Sets the delay after which the phone retries from the highest priority proxy (or outbound proxy) after it has failed over to a lower priority server.

The phone should have the primary and backup proxy server list from a DNS SRV record lookup on the server name. It needs to know the proxy priority; otherwise, it does not retry.

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

    <Proxy_Fallback_Intvl_1_ ua="na">3600</Proxy_Fallback_Intvl_1_>

  • In the phone web page, enter a value in seconds to set the duration in seconds after which the phone retries.

Valid values: Numeric. The range is from 0 seconds to 65535 seconds.

Default: 3600 seconds

Dual Registration

Controls both the dual registration and the fast fall back feature.

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

    <Dual_Registration_1_ ua="na">No</Dual_Registration_1_>

  • In the phone web page, set to Yes to enable the Dual registration/Fast Fall back feature. To enable the feature you must also configure the alternate proxy/alternate outbound proxy fields in the Proxy and Registration section.

Valid values: Yes and No

Default: No

TLS Name Validate

This field works only when SIP Transport is set to TLS for the phone line.

Specifies whether hostname verification is required when the phone line uses SIP over TLS. The options are:

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

    <TLS_Name_Validate_1_ ua="na">Yes</TLS_Name_Validate_1_>

  • In the phone web page, select Yes when hostname verification is required.

    Select No to bypass the hostname verification.

Valid values: Yes and No

Default: Yes

Add Outbound Proxy Survivability Support

You can configure a phone with the ability to register to the Site Survivability Gateway (SGW) nodes when WxC SSE nodes are unreachable.

Before you begin

Procedure

Step 1

Select Voice > Ext(n).

Step 2

In the Proxy and Registration section, set up Survivability Proxy and Survivability Proxy Fallback Intvl fields as described in Parameters for Outbound Proxy Survivability Support.

Step 3

Select Voice > System .

Step 4

In the System Configuration section, set up Survivability Test Mode field as described in Parameters for Outbound Proxy Survivability Support.

Step 5

Click Submit All Changes.

Parameters for Outbound Proxy Survivability Support

The following table defines the function and usage of WxC Outbound Proxy Survivability Support parameters in the Proxy and Registration section under the Ext(n) tab and the System Configuration section under the System tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 12. Conference Button Parameters

Parameter

Description

Survivability Proxy

The parameter can be configured with an extension that includes a statically-configured SRV record. This allows phone to perform a failover to a survivability gateway.

Perform one of the following:

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

    <Survivability_Proxy_n_>hostname[:port][:A=ip-list] [| hostname2[:port][:A=ip-list]]</Survivability_Proxy_n_>

  • In the phone web interface, enter the proxy server address as follows:

    hostname[:port][:A=ip-list] [| hostname2[:port][:A=ip-list]]

Where: ip-list: ip-addr[,ip-addr[,ip-addr…]]

Default: port=0

Example: wxclsg.example.com:8933:A=192.169.10.1

where,

wxclsg.example.com=Provisioned SGW hostname. It is used for TLS certificate validation when connecting to SGW nodes.

8933=SGW port

192.169.10.1=Provisioned SGW address

Compared to SGW, SSE nodes will always have high priority. If there are multiple SGW nodes, try one after the other.

Allowed values: String

Default: Blank

Survivability Proxy Fallback Intvl

The interval in seconds after which the phone will attempt to fallback to the SSE nodes

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

    <Survivability_Proxy_Fallback_Intvl_n_>30</Survivability_Proxy_Fallback_Intvl_n_>

  • in the phone web interface, specify the time interval in seconds.

Default: 30 secs

Survivability Test Mode

If set it to Yes, phone will always register to SGW nodes.

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

    <Survivability_Test_Mode>No</Survivability_Test_Mode>

  • in the phone web interface, select the test mode.

Options: Yes|No

Default: No

Configure the Subscriber Information Parameters

Before you begin

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

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

In the Subscriber Information section, set the parameter values as described in the Subscriber Information Parameters table.

Step 3

Click Submit All Changes.

Subscriber Information Parameters

The following table defines the function and usage of the parameters in the RTP Parameters section under the SIP tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 13. Subscriber Information

Parameter

Description

Display Name

Name displayed as the caller ID.

Perform one of the following:

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

    <Display_Name_1_ ua="na"/>

  • In the phone web page, enter a name that represents the caller ID.

User ID

Extension number for this line.

When you need to refer to this user ID in another setting, for example, the short name for a line key, use the $USER macro variable.

Perform one of the following:

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

    <User_ID_1_ ua="na">7001</User_ID_1_>

  • In the phone web page, enter an extension number

Password

Password for this line.

Perform one of the following:

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

    <Password_1_ ua="na">*************</Password_1_>

  • In the phone web page, enter a value to add password for the line.

Default: Blank (no password required)

Auth ID

Authentication ID for SIP authentication.

Perform one of the following:

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

    <Auth_ID_1_ ua="na"/>

  • In the phone web page, enter a value for an authentication ID.

Default: Blank

Reversed Auth Realm

The IP address for an authentication realm other than the proxy IP address.

Perform one of the following:

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

    <Reversed_Auth_Realm_1_ ua=”na”> </Reversed_Auth_Realm_1_>

    The parameter for extension 1 appears as defined in the phone configuration file.

  • In the phone web page, enter proxy IP address.

Default: Blank. The proxy IP address is used as the authentication realm.

Set Up Your Phone to Use OPUS Codec Narrowband

To improve bandwidth in your network, you can set up your phones to use the narrowband OPUS codec. The narrowband codec won't conflict with the wideband codec.

Before you begin

Access the Phone Web Interface

Procedure

Step 1

Select Voice > Ext <n> where (n) is the number of the extension to configure.

Step 2

In the SIP Settings section, set Use low-bandwidth OPUS to Yes.

Step 3

Click Submit All Changes.

NAT Transversal with Phones

Network Address Translation (NAT) allows multiple devices to share a single, public, routable, IP address to establish connections over the Internet. NAT is present in many broadband access devices to translate public and private IP addresses. For VoIP to coexist with NAT, NAT traversal is required.

Not all service providers provide NAT traversal. If your service provider does not provide NAT traversal, you have several options:

  • NAT Mapping with Session Border Controller: We recommend that you choose an service provider that supports NAT mapping through a Session Border Controller. With NAT mapping provided by the service provider, you have more choices in selecting a router.

  • NAT Mapping with SIP-ALG Router: NAT mapping can be achieved by using a router that has a SIP Application Layer Gateway (ALG). By using a SIP-ALG router, you have more choices in selecting an service provider.

  • NAT Mapping with a Static IP Address: NAT mapping with an external (public) static IP address can be acheived to ensure interoperability with the service provider. The NAT mechanism used in the router must be symmetric. For more information, see Determine Symmetric or Asymmetric NAT.

    Use NAT mapping only if the service provider network does not provide a Session Border Controller functionality. For more information on how to configure NAT mapping with a static IP, see Configure NAT Mapping with the Static IP Address.

  • NAT Mapping with STUN: If the service provider network does not provide a Session Border Controller functionality and if the other requirements are met, it is possible to use Session Traversal Utilities for NAT (STUN) to discover the NAT mapping. For information on how to configure NAT mapping with STUN, see Configure NAT mapping with STUN.

Enable NAT Mapping

You must enable NAT mapping to set NAT parameters.

Before you begin

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

Procedure

Step 1

Select Voice > Ext(n).

Step 2

Set up the fields as described in NAT Mapping Parameters.

Step 3

Click Submit All Changes.

NAT Mapping Parameters

The following table defines the function and usage of NAT Mapping parameters in the NAT Settings section under the Voice>Ext(n) tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 14. NAT Mapping Parameters

Parameter

Description

NAT Mapping Enable

To use externally mapped IP addresses and SIP/ RTP ports in SIP messages, select yes. Otherwise, select no.

Perform one of the following:

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

    <NAT_Mapping_Enable_1_ ua="na">Yes</NAT_Mapping_Enable_1_>

  • In the phone web page, set the parameter to Yes.

Allowed values: Yes|No

Default: No

NAT Keep Alive Enable

To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.

Perform one of the following:

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

    <NAT_Keep_Alive_Enable_1_ ua="na">Yes</NAT_Keep_Alive_Enable_1_>

  • In the phone web page, set the parameter to Yes.

Allowed values: Yes|No

Default: No

NAT Keep Alive Msg

Enter the keep alive message that should be sent periodically to maintain the current NAT mapping.

Perform one of the following:

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

    <NAT_Keep_Alive_Msg_1_ ua="na">$NOTIFY</NAT_Keep_Alive_Msg_1_>

  • In the phone web page, set the parameter to $NOTIFY or to $REGISTER.

If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent.

Allowed values: $NOTIFY and $REGISTER.

Default: $NOTIFY

Configure NAT Mapping with the Static IP Address

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

Before you begin

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

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

  • The NAT mechanism used in the router must be symmetric.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the NAT Support Parameters section, set the parameters as described in the NAT Mapping with Static IP Parameters table.

Step 3

Click the Ext(n) tab.

Step 4

In the NAT Settings section, set the parameters as described in the NAT Mapping from Ext Tab with Static IP Parameters table.

Step 5

Click Submit All Changes.

What to do next

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

NAT Mapping with Static IP Parameters

The following table defines the function and usage of NAT mapping with Static IP parameters in the NAT Support Parameters section under the Voice>SIP tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 15. NAT Mapping with Static IP Parameters

Parameter

Description

Handle VIA received

Enables the phone to process the received parameter in the VIA header.

Perform one of the following:

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

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

  • In the phone web page, set to Yes.

Default: No

Handle VIA rport

Enables the phone to process the rport parameter in the VIA header.

Perform one of the following:

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

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

  • In the phone web page, set to Yes.

Default: No

Insert VIA received

Enables to insert the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Perform one of the following:

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

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

  • In the phone web page, set to Yes.

Default: No

Insert VIA rport

Enables to insert the rport parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Perform one of the following:

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

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

  • In the phone web page, set to Yes.

Default: No

Substitute VIA Addr

Enables the user to use NAT-mapped IP:port values in the VIA header.

Perform one of the following:

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

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

  • In the phone web page, set to Yes.

Default: No

Send Resp To Src Port

Enables to send responses to the request source port instead of the VIA sent-by port.

Perform one of the following:

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

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

  • In the phone web page, set to Yes.

Default: No

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Perform one of the following:

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

    <NAT_Keep_Alive_Intvl ua="na">15</NAT_Keep_Alive_Intvl>

  • In the phone web page, enter an appropriate value.

Allowed values: Numeric ranges from 0 through 65535

Default: 15

EXT IP

External IP address to substitute for the actual IP address of phone in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, phone assumes this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line).

Perform one of the following:

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

    <EXT_IP ua="na">10.23.31.43</EXT_IP>

  • In the phone web page, enter an external static IP address.

Default: Blank

The following table defines the function and usage of NAT mapping with Static IP parameters in the NAT Support Parameters section under the Voice>Ext tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 16. NAT Mapping from Ext Tab

Parameter

Description

NAT Mapping Enable

Controls the use of externally mapped IP addresses and SIP/ RTP ports in SIP messages.

Perform one of the following:

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

    <NAT_Mapping_Enable_1_ ua="na">Yes</NAT_Mapping_Enable_1_>

  • In the phone web page, set to Yes to use externally mapped IP addresses.

Allowed values: Yes and No.

Default: No

NAT Keep Alive Enable

(Optional)

Configured NAT keep alive message periodically.

Perform one of the following:

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

    <NAT_Keep_Alive_Enable_1_ ua="na">Yes</NAT_Keep_Alive_Enable_1_>

  • In the phone web page, set to Yes to configure periodic NAT keep alive messages.

    Note

     

    The service provider might require the phone to send NAT keep alive messages to keep the NAT ports open.

    Check with your service provider to determine the requirements.

Allowed values: Yes and No.

Default: No

Configure NAT mapping with STUN

If the service provider network does not provide a Session Border Controller functionality and if the other requirements are met, it is possible to use Session Traversal Utilities for NAT (STUN) to discover the NAT mapping. The STUN protocol allows applications operating behind a network address translator (NAT) to discover the presence of the network address translator and to obtain the mapped (public) IP address (NAT addresses) and the port number that the NAT has allocated for the User Datagram Protocol (UDP) connections to remote hosts. The protocol requires assistance from a third-party network server (STUN server) located on the opposing (public) side of the NAT, usually the public Internet. This option is considered a last resort and should be used only if the other methods are not available. To use STUN:

  • The router must use asymmetric NAT. See Determine Symmetric or Asymmetric NAT.

  • A computer running STUN server software is available on the network. You can also use a public STUN server or set up your own STUN server.

Before you begin

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

Procedure

Step 1

Select Voice > SIP.

Step 2

In the NAT Support Parameters section, set the Handle VIA received, Insert VIA received, Substitute VIA Addr, Handle VIA rport, Insert VIA rport, and Send Resp To Src Port parameters as described in the NAT Mapping with Static IP Parameters table.

Step 3

Set the parameters as described in the NAT Mapping with STUN Parameters table.

Step 4

Click the Ext(n) tab.

Step 5

In the NAT Settings section, set the parameters as described in the NAT Mapping from Ext Tab with Static IP Parameters table.

Step 6

Click Submit All Changes.

What to do next

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

NAT Mapping with STUN Parameters

The following table defines the function and usage of NAT mapping with STUN parameters in the NAT Support Parameters section under the Voice>SIP tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 17. NAT Mapping with STUN Parameters

Parameter

Description

STUN Enable

Enables the use of STUN to discover NAT mapping.

Perform one of the following:

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

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

  • In the phone web page, set to Yes to enable the feature.

Allowed values: Yes and No.

Default: No

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery. You can use a public STUN server or set up your own STUN server.

Perform one of the following:

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

    <STUN_Server ua="na"/>

  • In the phone web page, enter an IP address or fully-qualified domain name of the STUN server.

Allowed values:

Default: Blank

Determine Symmetric or Asymmetric NAT

STUN does not work on routers with symmetric NAT. With symmetric NAT, IP addresses are mapped from one internal IP address and port to one external, routable destination IP address and port. If another packet is sent from the same source IP address and port to a different destination, a different IP address and port number combination is used. This method is restrictive because an external host can send a packet to a particular port on the internal host only if the internal host first sent a packet from that port to the external host.

This procedure assumes that a syslog server is configured and is ready to receive syslog messages.

To Determine Whether the Router Uses Symmetric or Asymmetric NAT:

Before you begin

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

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

Procedure

Step 1

Select Voice > System and navigate to Optional Network Configuration section.

Step 2

Enter the IP address for the Syslog Server, if the port number is anything other than the default, 514. It is not necessary to include the port number if it is the default.

The address and port number must be reachable from the Cisco IP phone. The port number appears on the output log file name. The default output file is syslog.514.log (if port number was not specified).

Step 3

Set the Debug Level to Error, Notice, or Debug.

Step 4

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

Step 5

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

Step 6

Click Voice > SIP and navigate to NAT Support Parameters.

Step 7

Set STUN Test Enable to Yes.

Step 8

Determine the type of NAT by viewing the debug messages in the log file. If the messages indicate that the device is using symmetric NAT, you cannot use STUN.

Step 9

Click Submit All Changes.

Configure Phone with DNS64/NAT64 Deployment

You can configure the phone to utilize DNS64 and NAT64 for IPv6 deployments. This configuration allows MPP phones to handle IPv4 addresses and connect to IPv4-only services via NAT64.


Note

For more information on SIP parameters, see SIP Parameters.

Before you begin

  • Phone onboards to Cisco Webex cloud successfully.

  • Set up NAT64 gateway for IPv4 to IPv6 transitions and co-existence of IPv4 and IPv6.

  • SIP Application Layer Gateway (ALG) or Session Border Controller (SBC) that co-exist with NAT64 gateway to translate SIP at the application layer.

  • Set up DNS64 to handle IPv6 DNS queries from the phone.

Procedure

Step 1

Select Voice > SIP.

Step 2

In the SIP Parameters section, select Yes in the Forced NAT64 field.

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

<Forced_NAT64>Yes</Forced_NAT64>

Options: Yes and No

Default: No

Step 3

Click Submit All Changes.

Dial Plan

Dial Plan Overview

Dial plans determine how digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls such as long distance or international.

Use the phone web user interface to configure dial plans on the IP phone.

This section includes information that you must understand about dial plans, and procedures to configure your own dial plans.

The Cisco IP Phone has various levels of dial plans and processes the digits sequence.

When a user presses the speaker button on the phone, the following sequence of events begins:

  1. The phone begins to collect the dialed digits. The interdigit timer starts to track the time that elapses between digits.

  2. If the interdigit timer value is reached, or if another terminating event occurs, the phone compares the dialed digits with the IP phone dial plan. This dial plan is configured in the phone web user interface in Voice > Ext(n) under the Dial Plan section.

Digit Sequences

A dial plan contains a series of digit sequences, separated by the | character. The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan consists of a series of elements that are individually matched to the keys that the user presses.

Dial plan is processed in conjunction with the Vertical Service Activation Codes (VSACs), hence, the digit analysis is done on both at the same time. Also, after a VSAC is processed, the dial plan rules then apply to the digits entered.

The minimum length specified in the dial plan and VSACs is not enforced, only the maximum length is enforced, which means the partial matches are processed and dialled out. For example, if the dial plan is xxxx, then x, xx, xxx and xxxx are allowed.

White space is ignored, but can be used for readability.

Digit Sequence

Function

0 1 2 3 4 5 6 7 8 9 0 * #

Characters that represent a key that the user must press on the phone keypad.

x

Numeric digit 0 to 9 on the phone keypad.

[sequence]

Characters within square brackets create a list of accepted key presses. The user can press any one of the keys in the list.

A numeric range, for example, [2-9] allows a user to press any one digit from 2 through 9.

A numeric range can include other characters. For example, [35-8*] allows a user to press 3, 5, 6, 7, 8, or *.

. 
(period)

A period indicates element repetition. The dial plan accepts 0 or more entries of the digit. For example, 01. allows users to enter 0, 01, 011, 0111, and so forth.

<dialed:substituted>

This format indicates that certain dialed digits are replaced by the substituted characters when the sequence is transmitted. The dialed digits can be zero to 9. For example: 


<8:1650>xxxxxxx

When the user presses 8 followed by a seven-digit number, the system automatically replaces the dialed 8 with the sequence 1650. If the user dials 85550112, the system transmits 16505550112.

If the dialed parameter is empty and there is a value in the substituted field, no digits are replaced and the substituted value is always prepended to the transmitted string. For example: 


<:1>xxxxxxxxxx

When the user dials 9725550112, the number 1 is added at the beginning of the sequence; the system transmits 19725550112.

, 
(comma)

An intersequence tone played (and placed) between digits plays an outside line dial tone. For example:


9, 1xxxxxxxxxx

An outside line dial tone plays after the user presses 9. The tone continues until the user presses 1.

! 
(exclamation point)

Prohibits a dial sequence pattern. For example:


1900xxxxxxx!

Rejects any 11-digit sequence that begins with 1900.

*xx

Allows a user to enter a 2-digit star code.

S0 or L0

For Interdigit Timer Master Override, enter S0 to reduce the short interdigit timer to 0 seconds, or enter L0 to reduce the long interdigit timer to 0 seconds.

P

To pause, enter P, the number of seconds to pause, and a space. This feature is typically used for implementation of a hotline and warm line, with a 0 delay for the hot line, and a nonzero delay for a warm line. For example: 

P5

A pause of 5 seconds is introduced.

Digit Sequence Examples

The following examples show digit sequences that you can enter in a dial plan.

In a complete dial plan entry, sequences are separated by a pipe character (|), and the entire set of sequences is enclosed within parentheses: 


( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )
  • Extensions on your system:
    
( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    [1-8]xx Allows a user to dial any three-digit number that starts with the digits 1 to 8. If your system uses four-digit extensions, enter the following string: [1-8]xxx

  • Local dialing with seven-digit number:
    
( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)

    9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can enter any seven-digit number, as in a local call.

  • Local dialing with 3-digit area code and a 7-digit local number:
    
( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required. After a user presses 9, an external dial tone sounds. The user must enter a 10-digit number that begins with a digit 2 through 9. The system automatically inserts the 1 prefix before it transmits the number to the carrier.

  • Local dialing with an automatically inserted 3-digit area code:
    
 ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    8, <:1212>xxxxxxx This example is useful where a local area code is required by the carrier but most calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number. The system automatically inserts the 1 prefix and the 212 area code before it transmits the number to the carrier.

  • U.S. long-distance dialing:
    
 ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The user can enter any 11-digit number that starts with 1 and is followed by a digit 2 through 9.

  • Blocked number:
    
 ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the U.S. After the user presses 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected.

  • U.S. international dialing:
    
 ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    9, 011xxxxxx After the user presses 9, an external dial tone sounds. The user can enter any number that starts with 011, as in an international call from the U.S.

  • Informational numbers:
    
 ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

    0 | [49]11 This example includes two-digit sequences, separated by the pipe character. The first sequence allows a user to dial 0 for an operator. The second sequence allows the user to enter 411 for local information or 911 for emergency services.

Acceptance and Transmission of the Dialed Digits

When a user dials a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As the user enters more digits, the set of candidates diminishes until only one or none is valid. When a terminating event occurs, the IP PBX either accepts the user-dialed sequence and initiates a call, or else rejects the sequence as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is invalid.

The following table explains how terminating events are processed.

Terminating Event

Processing

Dialed digits have not matched any sequence in the dial plan.

The number is rejected.

Dialed digits exactly match one sequence in the dial plan.

If the dial plan allows the sequence, the number is accepted and is transmitted according to the dial plan.

If the dial plan blocks the sequence, the number is rejected.

A timeout occurs.

The number is rejected if the dialed digits are not matched to a digit sequence in the dial plan within the time that the applicable interdigit timer specifies.

The Interdigit Long Timer applies when the dialed digits do not match any digit sequence in the dial plan.

Default: 10 seconds.

The Interdigit Short Timer applies when the dialed digits match one or more candidate sequences in the dial plan. Default: 3 seconds.

A user presses the # key or the dial softkey on the IP phone screen.

If the sequence is complete and is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan.

If the sequence is incomplete or is blocked by the dial plan, the number is rejected.

Dial Plan Timer (Off-Hook Timer)

You can think of the Dial Plan Timer as the off-hook timer. This timer starts when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated. Unless you have a special dial plan string to allow a null entry, the call is rejected.


Note

The timer before a number is dialed is whichever shorter of the dial plan default timer and the dial tone timer set in the Dial Tone field on the Regional tab.

Syntax for the Dial Plan Timer

SYNTAX: (Ps<:n> | dial plan)

  • s: The number of seconds; The timer before a number is dialed is whichever shorter of the dial plan default timer and the dial tone timer set in the Dial Tone field. With the timer set to 0 seconds, the call transmits automatically to the specified extension when the phone goes off hook.

  • n: (optional): The number to transmit automatically when the timer expires; you can enter an extension number or a DID number. No wildcard characters are allowed because the number is transmitted as shown. If you omit the number substitution, <:n>, the user hears a reorder (fast busy) tone after the specified number of seconds.

Examples for the Dial Plan Timer

Note

The actual timer before a number is dialed is whichever shorter of the dial plan default timer and the dial tone timer set in the Dial Tone field. In the following examples, the dial tone timer is assumed to be longer than the dial plan timer.

Allow more time for users to start dialing after taking a phone off hook:


 (P9 | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9 means that after taking a phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone. By setting a longer timer, you allow more time for users to enter digits.

To create a hotline for all sequences on the System Dial Plan:


 (P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9<:23> means that after taking the phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23.

To create a hotline on a line button for an extension:


 (P0 <:1000>)

With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook. Enter this sequence in the Phone Dial Plan for Ext 2 or higher on a client phone.

Interdigit Long Timer (Incomplete Entry Timer)

You can think of this timer as the incomplete entry timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. The default value is 10 seconds.

This section explains how to edit a timer as part of a dial plan. Alternatively, you can modify the Control Timer that controls the default interdigit timers for all calls.

Syntax for the Interdigit Long Timer

SYNTAX: L:s, (dial plan)

  • s: The number of seconds; if no number is entered after L:, the default timer is 5 seconds. With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook.

  • Note that the timer sequence appears to the left of the initial parenthesis for the dial plan.

Example for the Interdigit Long Timer

 L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

L:15 means that this dial plan allows the user to pause for up to 15 seconds between digits before the Interdigit Long Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.

Interdigit Short Timer (Complete Entry Timer)

You can think of this timer as the complete entry timer. This timer measures the interval between dialed digits. The timer applies when the dialed digits match at least one digit sequence in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated. If the entry is valid, the call proceeds. If the entry is invalid, the call is rejected.

Default: 3 seconds.

Syntax for the Interdigit Short Timer

SYNTAX 1: S:s, (dial plan)

Use this syntax to apply the new setting to the entire dial plan within the parentheses.

SYNTAX 2: sequence Ss

Use this syntax to apply the new setting to a particular dialing sequence.

s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies.

Examples for the Interdigit Short Timer

To set the timer for the entire dial plan:


 S:6, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

S:6 means that while the user enters a number with the phone off hook, the user can pause for up to 15 seconds between digits before the Interdigit Short Timer expires. This setting is especially helpful to users such as sales people, who are reading the numbers from business cards and other printed materials while dialing.

Set an instant timer for a particular sequence within the dial plan:


 (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxxS0 | 9,8,011xx. | 9,8,xx.|[1-8]xx)

9,8,1[2-9]xxxxxxxxxS0 means that with the timer set to 0, the call is transmitted automatically when the user dials the final digit in the sequence.

Edit the Dial Plan on the IP Phone


Note

You can edit the dial plan in the XML configuration file. Locate the Dial_Plan_n_ parameter in the XML configuration file, where n denotes the extension number. Edit the value of this parameter. The value must be specified in the same format as in the Dial Plan field on the phone administration web page, described below.

Before you begin

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

Procedure

Step 1

Select Voice > Ext(n), where n is an extension number.

Step 2

Scroll to the Dial Plan section.

Step 3

Enter the digit sequences in the Dial Plan field.

The default (US-based) systemwide dial plan appears automatically in the field.

Step 4

You can delete digit sequences, add digit sequences, or replace the entire dial plan with a new dial plan.

Separate each digit sequence with a pipe character, and enclose the entire set of digit sequences within parentheses. Example:

(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

Step 5

Click Submit All Changes.

The phone reboots.

Step 6

Verify that you can successfully complete a call with each digit sequence that you entered in the dial plan.

Note

 

If you hear a reorder (fast busy) tone, review your entries and modify the dial plan appropriately.

Regional Parameters Configuration

Regional Parameters

In the phone web user interface, use the Regional tab to configure regional and local settings, such as control timer values, dictionary server script, language selection, and locale to change localization. The Regional tab includes these sections:

  • Call Progress Tones—Displays values of all ringtones.

  • Distinctive Ring Patterns—Ring cadence defines the ringing pattern that announces a telephone call.

  • Control Timer Values—Displays all values in seconds.

  • Vertical Service Activation Codes (VSACs)—Includes Call Back Act Code and Call Back Deact Code. They are processed in conjunction with the dial plan rules, hence, the digit analysis is done on both at the same time. Also, after a VSAC is processed, the dial plan rules then apply to the digits entered.

    The minimum length specified in the dial plan and VSACs is not enforced, only the maximum length is enforced, which means the partial matches are processed and dialled out. For example, if the dial plan is xxxx, then x, xx, xxx and xxxx are allowed.

  • Outbound Call Codec Selection Codes—Defines the voice quality.

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

  • Language—Includes Dictionary Server Script, Language Selection, and Locale.

Set the Control Timer Values

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

Before you begin

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

Procedure

Step 1

Select Voice > Regional.

Step 2

Set the Reorder Delay, Interdigit Long Timer, and Interdigit Short Timer parameters as described in the Control Timer Values (sec) table.

Step 3

Click Submit All Changes.

Parameters for Control Timer Values (sec)

The following table defines the function and usage of Control Timer Values parameters in the Control Timer Values(s) Parameters section under the Voice>Regional tab in the phone web interface. It also defines the syntax of the string that is added in the phone configuration file with XML(cfg.xml) code to configure a parameter.

Table 18. Parameters for Control Timer Values (sec)

Parameter

Description

Reorder Delay

Delay after far end hangs up before reorder (busy) tone is played.

Perform one of the following:

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

    <Reorder_Delay ua="na">255</Reorder_Delay>

  • In the phone web page, set a value in seconds ranges from 0-255 secs.

0 = plays immediately, inf = never plays. Set to 255 to return the phone immediately to on-hook status and to not play the tone.

Allowed values: 0–255 seconds

Default: 255

Interdigit Long Timer

Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed.

Perform one of the following:

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

    <Interdigit_Long_Timer ua="na">10</Interdigit_Long_Timer>

  • In the phone web page, set a value in seconds ranges from 0-64 seconds.

Allowed values: 0–64 seconds

Default: 10

Interdigit Short Timer

Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences.

Perform one of the following:

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

    <Interdigit_Short_Timer ua="na">3</Interdigit_Short_Timer>

  • In the phone web page, set a value in seconds ranges from 0-64 seconds.

Allowed values: 0–64 seconds

Default: 3

Localize Your Cisco IP Phone

Before you begin

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

Procedure

Step 1

Select Voice > Regional.

Step 2

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

Step 3

Click Submit All Changes.

Configure Time and Date on Phone Web Page

You can manually set the time and date on the phone web page.
Before you begin
Access the Phone Web Interface. Review Time and Date Settings.
Procedure

Step 1

Select Voice > Regional.

Step 2

In the Time section, enter the time and date information.

Step 3

Select Voice > User.

Step 4

In the Supplementary Services, choose 12h or 24hr from the Time Format drop down list.

Default: 12hr

Step 5

Choose the date format from the Date Format drop down list.

Step 6

Click Submit All Changes

Configure Time and Date on the Phone

You can set the time and date manually on the phone.
Before you begin
Review the Time and Date Settings.
Procedure

Step 1

Press Applications .

Step 2

Select Device administration > Date/Time.

Step 3

Select Set current time manually.

Step 4

Set the date and time in the format requested on the screen:

YYYY MM DD HH MM

Step 5

Select the OK softkey.

Step 6

Select the Save softkey.

Time and Date Settings

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

  • NTP Server— NTP 24-hour time format takes priority over the time you set using the menu options on the phone or web page.

    When the phone boots up, it tries to contact the first Network Time Protocol (NTP) server to get and update the time. The phone periodically synchronizes its time with the NTP server, and between updates, it tracks time with its internal clock. The synchronization period is fixed at 64 seconds.

    If you manually enter a time, this setting takes effect for now, but on the next NTP synchronization, the NTP time is displayed.

  • Manual Setup—You can manually configure the local date and time by using one of the following methods:

    • On the phone web interface

    • On the phone itself

    The default format is 12-hour which is overwritten with the 24-hour format as soon as the phone synchronizes with the NTP server.

Table 19. Date and Time Parameters

Parameter

Description

Set Local Date (mm/dd/yyyy)

Sets the local date (mm represents the month and dd represents the day). The year is optional and uses two or four digits.

Default: Blank

Set Local Time (HH/mm)

Sets the local time (hh represents hours and mm represents minutes). Seconds are optional.

Default: Blank

Time Zone

Selects the number of hours to add to GMT to generate the local time for caller ID generation. Choices are GMT-12:00, GMT-11:00,…, GMT, GMT+01:00, GMT+02:00, …, GMT+13:00.

The time of the log messages and status messages are in UTC time and are not affected by the time zone setting.

Default: GMT-08:00

Time Offset (HH/mm)

This specifies the offset in 24-hour format from GMT to use for the local system time.

The NTP Server time is expressed in GMT time. The local time is obtained by offsetting the GMT according to the time zone of the region.

Default: 00/00

Ignore DHCP Time Offset

When used with some routers that have DHCP with time offset values configured, the IP phone uses the router settings and ignores the IP phone time zone and offset settings. To ignore the router DHCP time offset value, and use the local time zone and offset settings, choose yes for this option. If you choose no, the IP phone uses the router's DHCP time offset value.

Default: Yes.

Daylight Saving Time Rule

Enter the rule for calculating daylight saving time. This rule is comprised of three fields. Each field is separated by a semicolon (;). Optional values inside brackets [ ] are assumed to be 0 if they are not specified. Midnight is represented by colons. For example, 0:0:0 of the given date.

This is the format of the rule: Start = <start-time>; end=<end-time>; save = <save-time>.

The <start-time> and <end-time> values specify the start and end dates and times of daylight saving time. Each value is in this format: <month> /<day> / <weekday>[/HH:[mm[:ss]]]

The <save-time> value is the number of hours, minutes, and/or seconds to add to the current time during daylight saving time. The <save-time> value can be preceded by a negative (-) sign if subtraction is desired instead of addition. The <save-time> value is in this format: [/[+|-]HH:[mm[:ss]]]

The <month> value equals any value in the range 1-12 (January-December).

The <day> value equals [+|-] any value in the range 1-31.

If <day> is -1, it means the <weekday> on or before the end of the month (in other words the last occurrence of < weekday> in that month).

Daylight Saving Time Rule (continued)

The <weekday> value equals any value in the range 1-7 (Monday-Sunday). It can also equal 0. If the <weekday> value is 0, this means that the date to start or end daylight saving is exactly the date given. In that case, the <day> value must not be negative. If the <weekday> value is not 0 and the <day> value is positive, then daylight saving starts or ends on the <weekday> value on or after the date given. If the <weekday> value is not 0 and the <day> value is negative, then daylight saving starts or ends on the <weekday> value on or before the date given. Where:

  • HH stands for hours (0-23).

  • mm stands for minutes (0-59).

  • ss stands for seconds (0-59).

Default: 3/-1/7/2;end=10/-1/7/2;save=1.

Daylight Saving Time Enable

Enables Daylight Saving Time.

Default: Yes

Time Format

Choose the time format for the phone (12-hour or 24-hour).

Default: 12hr

Date Format

Choose the date format for the phone (month/day or day/month).

Default: month/day

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

<!--  Time  -->
<Set_Local_Date__mm_dd_yyyy_ ua="na"/>
<Set_Local_Time__HH_mm_ ua="na"/>
<Time_Zone ua="na">GMT-08:00</Time_Zone>
<!--
available options: GMT-12:00|GMT-11:00|GMT-10:00|GMT-09:00|
GMT-08:00|GMT-07:00|GMT-06:00|GMT-05:00|GMT-04:00|GMT-03:30|
GMT-03:00|GMT-02:00|GMT-01:00|GMT|GMT+01:00|GMT+02:00|GMT+03:00|
GMT+03:30|GMT+04:00|GMT+04:30|GMT+05:00|GMT+05:30|GMT+05:45|
GMT+06:00|GMT+06:30|GMT+07:00|GMT+08:00|GMT+09:00|GMT+09:30|
GMT+10:00|GMT+11:00|GMT+12:00|GMT+13:00|GMT+14:00 
-->
<Time_Offset__HH_mm_ ua="na"/>
<Ignore_DHCP_Time_Offset ua="na">Yes</Ignore_DHCP_Time_Offset>
<Daylight_Saving_Time_Rule ua="na">start=3/-1/7/2;end=10/-1/7/2;
save=1</Daylight_Saving_Time_Rule>
<Daylight_Saving_Time_Enable ua="na">Yes</Daylight_Saving_Time_Enable>
<Time_Format ua="na">12hr</Time_Format>
<!--  available options: 12hr|24hr  -->
<Date_Format ua="na">month/day</Date_Format>
<!--  available options: month/day|day/month  -->

Configure Daylight Saving Time

The phone supports automatic adjustment for daylight saving time.


Note

The time of the log messages and status messages are in UTC time. The time zone setting does not affect them.
Before you begin

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

Procedure

Step 1

Select Voice > Regional.

Step 2

Set the Daylight Saving Time Enable drop-down list box to Yes.

Step 3

In the Daylight Saving Time Rule field, enter the DST rule. This value affects the time stamp on the CallerID.

Step 4

Click Submit All Changes.

Daylight Saving Time Examples

The following example configures daylight saving time for the U.S, adding one hour starting at midnight on the second Sunday in March and ending at midnight on the first Sunday in November; add 1 hour (USA, North America): 


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

The following example configures daylight saving time for Finland, starting at midnight on the last Sunday in March and ending at midnight on the last Sunday in October: 


start=3/-1/7/03:0:0;end=10/-1/7/03:0:0;save=1 (Finland)

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


start=9/-1/7/02:0:0;end=4/1/7/02:0:0;save=1 (New Zealand)

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


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

Phone Display Language

The Cisco IP Phone supports multiple languages for the phone display.

By default, the phone is set up for English. To enable the use of another language, you must set up the dictionary for the language. For some languages, you must also set up the font for the language.

After the setup is complete, you or your users can specify the desired language for the phone display.

Supported Languages for the Phone Display

On the phone administration web page, go to Admin Login > Advanced > Voice > Regional. In the Language section, click the Locale drop-down list box to see the supported languages for the phone display.

  • ar-SA (Arabic)

  • bg-BG (Bulgarian)

  • ca-ES (Catalan)

  • cs-CZ (Czech)

  • da-DK (Danish)

  • de-DE (German)

  • el-GR (Greek)

  • en-GB (English-Great Britain)

  • en-US (English-United States)

  • es-CO (Spanish-Colombia)

  • es-ES (Spanish-Spain)

  • fi-FI (Finnish)

  • fr-CA (French-Canada)

  • fr-FR (French)

  • he-IL (Hebrew)

  • hr-HR (Croatian)

  • hu-HU (Hungarian)

  • it-IT (Italian)

  • ja-JP (Japanese)

  • ko-KR (Korean)

  • nl-NL (Dutch)

  • nn-NO (Norwegian)

  • pl-PL (Polish)

  • pt-PT (Portuguese)

  • ru-RU (Russian)

  • sk-SK (Slovak)

  • sl-SI (Slovenian)

  • sv-SE (Swedish)

  • tr-TR (Turkish)

  • zh-CN (Chinese)

  • zh-HK (Chinese-Hong Kong SAR)
Set Up Dictionaries and Fonts

Languages other than English require dictionaries. Some languages also require a font.


Note

To enable Latin and Cyrillic languages, you must not add a font file.
Procedure

Step 1

Download the locale zip file for your firmware version, from cisco.com. Place the file on your server, and unzip the file.

Dictionaries and fonts for all the supported languages are included in the zip file. Dictionaries are XML scripts. Fonts are standard TTF files.

Step 2

On the phone administration web page, go to Admin Login > Advanced > Voice > Regional. In the Language section, specify the necessary parameters and values in the Dictionary Server Script field as described below. Use a semicolon (;) to separate multiple parameter and value pairs.

  • Specify the location of the dictionary and font files with the serv parameter.

    For example: serv=http://server.example.com/Locales/

    Make sure to include the IP address of the server, the path, and folder name.

    Example: serv=http://10.74.128.101/Locales/

  • For each language that you want to set up, specify a set of parameters as described below.

    Note

     

    In these parameter specifications, n denotes a serial number. This number determines the sequential order in which the language options are displayed in the Settings menu of the phone.

    0 is reserved for US-English, which has a default dictionary. You can use it optionally, to specify your own dictionary.

    Use numbers starting with 1 for other languages.

    • Specify the language name with the dn parameter.

      Example for language name for Asian language: d1=Chinese-Simplified

      Example for language name for German (Latin and Cyrillic): d2=German

      Example for language name for French (Latin and Cyrillic): d1=French

      Example for language name for French (Canada) (Latin and Cyrillic) language: d1=French-Canada

      Example for language name for Hebrew (RTL language): d1=Hebrew

      Example for language name for Arabic (RTL language): d1=Arabic

      This name is displayed as a language option in the Settings menu of the phone.

    • Specify the name of the dictionary file with the xn parameter.

      Example for Asian language:

      Example for French (Latin and Cyrillic) languages:

      Example for Arabic (RTL language) language:

      Example for French (Canada) language: x1=fr-CA_78xx_68xx-11.3.6.0006.xml;

      Ensure to specify the correct file for the language and phone model that you use.

    • If a font is required for the language, specify the name of the font file with the fn parameter.

      For example:

      Make sure to specify the correct file for the language and phone model that you use.

    See Setup for Latin and Cyrillic Languages for specific details on setting up Latin languages.

    See Setup for an Asian Language for specific details on setting up an Asian language.

    See Setup for RTL Languages for specific details on setting up RTL languages.

Step 3

Click Submit All Changes.

Setup for Latin and Cyrillic Languages

If you use Latin and Cyrillic languages such as French or German, you can configure up to four language options for the phone. List of Latin and Cyrillic languages:

  • Bulgarian

  • Catalan

  • Croatian

  • Czech

  • Danish

  • Dutch

  • English (UK)

  • Finnish

  • French (France)

  • French (Canada)

  • German

  • Greek

  • Hungarian

  • Italian

  • Portuguese (Portugal)

  • Norwegian

  • Polish

  • Russian

  • Slovak

  • Slovenian

  • Spanish (Columbia)

  • Spanish (Spain)

  • Swedish

  • Turkish

  • Ukraine

To enable the options, set up a dictionary for each language that you want to include. To enable the language, specify a pair of dn and xn parameters and values in the Dictionary Server Script field, for each language that you want to include.

Example for including French and German:

Example for including French (Canada):

serv=http://10.74.128.101/Locales/;d1=French-Canada;x1=fr-CA_78xx_68xx-11.3.6.0006xml;
serv=http://10.74.128.101/Locales/;d1=French-Canada;x1=fr-CA_88xx-11.3.6.0006xml;

Note

In the above examples http://10.74.128.101/Locales/ is a web folder. The dictionary files are extracted in this web folder and are used in the examples.

To configure this option in the phone configuration XML file (cfg.xml), enter a string in this format:

<!-- Language -->
<Dictionary_Server_Script ua="na">serv=http://10.74.10.215/lcdapi/resync_files/;d1=French-Canada;x1=fr-CA_88xx-11.3.6.0006.xml;</Dictionary_Server_Script>

<Language_Selection ua="na">French-Canada</Language_Selection>
<Locale ua="na">fr-CA</Locale>

Add values for:

  • Language Selection Parameter as appropriate

    For French: French

    For French (Canada): French-Canada

    For German: German

  • Locale parameter list as appropriate

    For French: fr-FR

    For French (Canada): fr-CA

    For German: de-DE

After the successful configuration, the user can see the configured language option on the phone under the Language menu. User can access the Language menu from Applications > Device administration.

Setup for an Asian Language

If you use an Asian language such as Chinese, Japanese, or Korean, you can only set up one language option for the phone.

You must set up the dictionary and the font for the language. To do this, specify the d1, x1 and f1 parameters and values in the Dictionary Server Script field.

Example for setting up Chinese-Simplified:

Setup for RTL Languages

If you use a Right-to-Left (RTL) language such as Arabic and Hebrew, you can only set up one language option for the phone.

You must set up the dictionary and the font for the language. To do this, specify the d1, x1, and f1 parameters and values in the Dictionary Server Script field.

Example for Arabic:
serv=http://server.example.com/Locales;d1=Arabic;x1=ar-SA_88xx-11.3.4.xml;f1=ar-SA_88xx-11.3.4.ttf
Example for Hebrew:
serv=http://server.example.com/Locales;d1=Hebrew;x1=he-IL_88xx-11.3.4.xml;f1=he-IL_88xx-11.3.4.ttf

Values for Language Selection parameter must be Arabic or Hebrew as appropriate.

Values for Locale parameter must be ar-SA for Arabic and he-IL for Hebrew.

Specify a Language for the Phone Display

Note

Your users can select the language on the phone, from Settings > Device Administration > Language.

Before you begin
The dictionaries and fonts required for the language are set up. See Set Up Dictionaries and Fonts for details.
Procedure

Step 1

On the phone administration web page, go to Admin Login > Advanced > Voice > Regional, Language section. In the Language Selection field, specify the value of the appropriate dn parameter value from the Dictionary Server Script field, for the language of your choice.

Step 2

Click Submit All Changes.

Vertical Service Activation Codes

Parameter

Description

Call Return Code

This code calls the last caller.

Defaults to *69.

Blind Transfer Code

Begins a blind transfer of the current call to the extension specified after the activation code.

Defaults to *95.

Cfwd All Act Code

Forwards all calls to the extension specified after the activation code.

Defaults to *72.

Cfwd All Deact Code

Cancels call forward of all calls.

Defaults to *73.

Cfwd Busy Act Code

Forwards busy calls to the extension specified after the activation code.

Defaults to *90.

Cfwd Busy Deact Code

Cancels call forward of busy calls.

Defaults to *91.

Cfwd No Ans Act Code

Forwards no-answer calls to the extension specified after the activation code.

Defaults to *92.

Cfwd No Ans Deact Code

Cancels call forward of no-answer calls.

Defaults to *93.

CW Act Code

Enables call waiting on all calls.

Defaults to *56.

CW Deact Code

Disables call waiting on all calls.

Defaults to *57.

CW Per Call Act Code

Enables call waiting for the next call.

Defaults to *71.

CW Per Call Deact Code

Disables call waiting for the next call.

Defaults to *70.

Block CID Act Code

Blocks caller ID on all outbound calls.

Defaults to *61.

Block CID Deact Code

Removes caller ID blocking on all outbound calls.

Defaults to *62.

Block CID Per Call Act Code

Removes caller ID blocking on the next inbound call.

Defaults to *81.

Block CID Per Call Deact Code

Removes caller ID blocking on the next inbound call.

Defaults to *82.

Block ANC Act Code

Blocks all anonymous calls.

Defaults to *77.

Block ANC Deact Code

Removes blocking of all anonymous calls.

Defaults to *87.

DND Act Code

Enables the do not disturb feature.

Defaults to *78.

DND Deact Code

Disables the do not disturb feature.

Defaults to *79.

Secure All Call Act Code

Makes all outbound calls secure.

Defaults to *16.

Secure No Call Act Code

Makes all outbound calls not secure.

Defaults to *17.

Secure One Call Act Code

Makes a secure call.

Default: *18.

Secure One Call Deact Code

Disables secure call feature.

Default: *19.

Paging Code

The star code used for paging the other clients in the group.

Defaults to *96.

Call Park Code

The star code used for parking the current call.

Defaults to *68.

Call Pickup Code

The star code used for picking up a ringing call.

Defaults to *97.

Call Unpark Code

The star code used for picking up a call from the call park.

Defaults to *88.

Group Call Pickup Code

The star code used for picking up a group call.

Defaults to *98.

Referral Services Codes

These codes tell the IP phone what to do when the user places the current call on hold and is listening to the second dial tone.

One or more *code can be configured into this parameter, such as *98, or *97|*98|*123, and so on. Max total length is 79 chars. This parameter applies when the user places the current call on hold (by Hook Flash) and is listening to second dial tone. Each *code (and the following valid target number according to current dial plan) entered on the second dial-tone triggers the phone to perform a blind transfer to a target number that is prepended by the service *code.

For example, after the user dials *98, the IP phone plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the phone sends a blind REFER to the holding party with the Refer-To target equals to *98<target_number>. This feature allows the phone to hand off a call to an application server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally processed by the IP phone. You can empty the corresponding *code that you do not want to the phone to process.

Feature Dial Services Codes

These codes tell the phone what to do when the user is listening to the first or second dial tone.

One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, and so forth. The maximum total length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone). Enter *code (and the following target number according to current dial plan) entered at the dial tone triggers the phone to call the target number prepended by the *code. For example, after user dials *72, the phone plays a prompt tone awaiting the user to enter a valid target number. When a complete number is entered, the phone sends a INVITE to *72<target_number> as in a normal call. This feature allows the proxy to process features like call forward (*72) or BLock Caller ID (*67).

The *codes should not conflict with any of the other vertical service codes internally processed by the phone. You can empty the corresponding *code that you do not want to the phone to process.

You can add a parameter to each *code in Features Dial Services Codes to indicate what tone to play after the *code is entered, such as *72‘c‘|*67‘p‘. Below are a list of allowed tone parameters (note the use of back quotes surrounding the parameter without spaces)

• c = Cfwd Dial Tone

• d = Dial Tone

• m = MWI Dial Tone

• o = Outside Dial Tone

• p = Prompt Dial Tone

• s = Second Dial Tone

• x = No tones are place, x is any digit not used above

If no tone parameter is specified, the phone plays Prompt tone by default.

If the *code is not to be followed by a phone number, such as *73 to cancel call forward, do not include it in this parameter. In that case, simple add that *code in the dial plan and the phone sends INVITE *73@..... as usual when user dials *73.