Cisco Prime Collaboration Provisioning Guide, 9.5
Infrastructure Data Object Fields
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Table Of Contents

Infrastructure Data Object Fields


Infrastructure Data Object Fields


To create Configuration Templates, you add infrastructure data objects to the Configuration Template. Table B-1 through Table B-31 lists the infrastructure data objects that are available in Provisioning.

Not all fields in an infrastructure configuration template are applicable on all Cisco Unified Communications Manager versions.


Note All the data object fields, where you manually enter text, are case sensitive.


CTI Route Point Data Object Fields

Table B-1 CTI Route Point Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Device Pool

List of available device pools. The device pool specifies a collection of properties for this device, including Unified CM Group, Date/Time Group, Region, and Calling Search Space for auto-registration of devices.

Common Device Config

Configuration of common device settings, such as the softkey template and user locale.

Call Search Space

Specifies the collection of Route Partitions that are searched to determine how a collected (originating) number should be routed.

Location

Specifies the total bandwidth that is available for calls to and from this location. A location setting of None means that the location feature does not keep track of the bandwidth that this route point consumes.

Directory Numbers

Enter directory numbers. These directory numbers must not exist on the Cisco Unified Communications Manager.

Route Partition for Directory Numbers

Available route partitions.

Media Resource Group List

Provides a prioritized grouping of media resource groups. An application chooses the required media resource, such as a Music On Hold server, from the available media resources according to the priority order that is defined in a Media Resource Group List.

If this field is left blank, the Media Resource Group that is defined in the device pool is used.

User Locale

User location associated with the phone user. The user locale identifies a set of detailed information to support users, including language, font, date and time formatting, and alphanumeric keyboard text information.

User Hold MOH Audio Source

The audio source that plays Music On Hold when the user initiates a hold action.

Network Hold Audio Source

The audio source that plays when the network initiates a hold action.


Call Park Infrastructure Data Object Fields

Table B-2 Call Park Infrastructure Data Object Fields 

Field
Description

Number/Range

Enter the call park extension number or a range of numbers.

Description

Optional description.

Route Partition

List of available route partitions.

Unified CM

List of available Cisco Unified Communications Managers.


Call Pickup Group Infrastructure Data Object Fields

Table B-3 Call Pickup Group Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Number

Unique directory number (integers).

Description

Optional description.

Route Partition

List of available route partitions.

Calling Party Information

Enables the visual notification message to the call pickup group to include identification of the calling party. This setting is applicable only when the Call Pickup Group Notification Policy is set to Visual Alert or Audio and Visual Alert.

Available Member Call Pickup Groups

List of available call pickup groups. The Call Pickup Groups are listed by their names, not by directory number and partition.

Call Pickup Group Notification Policy

Sets the notification policy on the call pickup group.

Call Pickup Group Notification Timer (seconds)

Sets the delay between the time that the call first comes into the called party and the time that the notification is sent to the rest of the call pickup group.

Directory Number Info

List of directory numbers with route partition. Only directory numbers that are associated or linked to the subscribers can be added to a call pickup group.

Note You can add or delete (or a combination of the two) no more than 200 directory numbers at one time.

Called Party Information

Enables the visual notification message to the call pickup group to include identification of the called party. This setting is applicable only when the Call Pickup Group Notification Policy is set to Visual Alert or Audio and Visual Alert.


Call Search Space Infrastructure Data Object Fields

Table B-4 Call Search Space Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Available Route Partitions

List of available route partitions. The route partitions list is not strictly required, but you should provide at least one value.

You must reference a route partition that already exists on the Cisco Unified Communications Manager, or define one in the same Configuration Template before to this call search space.


Common Device Config Infrastructure Data Object Fields

Table B-5 Common Device Config Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Softkey Template

Softkey template that determines the configuration of the softkeys on Cisco IP Phones.

User Hold MOH Audio Source

The audio source that plays Music On Hold when the user initiates a hold action.

Network Hold Audio Source

The audio source that plays when the network initiates a hold action.

User Locale

User location associated with the phone user. The user locale identifies a set of detailed information to support users, including language, font, date and time formatting, and alphanumeric keyboard text information.

MLPP Indication

Specifies whether devices in the device pool that are capable of playing precedence tones will use the capability when the devices place an MLPP precedence call.

MLPP Preemption

Specifies whether devices in the device pool that are capable of preempting calls in progress will use the capability when the devices place an MLPP precedence call.

MLPP Domain

Multilevel Precedence and Preemption (MLPP) Domain that is associated with this device.


Unity Distribution List Infrastructure Data Object Fields

Table B-6 Unity Distribution List Infrastructure Data Object Fields

Field
Description

Alias

Alias name of the distribution list.

Display Name

Name of the distribution list.

Extension

Extension that the phone system uses to connect.

Owner

Owner of the Call Handler for any subscriber or distribution list.

Owner Type

Type of the owner.

Show Distribution List in Email Server Address Book

Displays the distribution list name in the email server's address book.

Member List

List of members associated with the distribution list. Use the format Alias/MemberType.


Note You cannot remove the default system distribution list.



Infrastructure Data Object Fields for Common Phone Profile

Table B-7 Infrastructure Data Object Fields for Common Phone Profile 

Field
Description

Common Phone Profile Information

Name

Enter a name to identify the common phone profile; for example, CPP_7905. The value can include 1 to 50 characters, including alphanumeric characters, dot, dash, and underscores.

Description

Identify the purpose of the common phone profile; for example, common phone profile for the 7905 phone. The description can include up to 50 characters in any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), back-slash (\), or angle brackets (<>).

Local Phone Unlock Password

Enter the password that is used to unlock a local phone. Valid values comprise 1 to 15 characters.

DND Option

When you enable Do Not Disturb (DND) on the phone, this parameter allows you to specify how the DND features handle incoming calls:

Call Reject—This option specifies that no incoming call information is presented to the user. Depending on how you configure the DND Incoming Call Alert parameter, the phone may play a beep or display a flash notification of the call.

Ringer Off—This option turns off the ringer, but incoming call information is presented to the device, so the user can accept the call.

Note For 7940/7960 phones that are running SCCP, you can only choose the Ringer Off option. For mobile devices and dual-mode phones, you can only choose the Call Reject option. When you activate DND Call Reject on a mobile device or dual-mode phone, no call information is presented to the device.

DND Incoming Call Alert

When you enable the DND Ringer Off or Call Reject option, this parameter specifies how a call displays on a phone.

From the drop-down list, choose one of the following options:

Disable—This option disables both beep and flash notification of a call, but for the DND Ringer Off option, incoming call information is still displayed. For the DND Call Reject option, no call alerts display, and no information is sent to the device.

Beep Only—For an incoming call, this option causes the phone to beep.

Flash Only—For an incoming call, this option causes the phone to display a flash alert.

Enable End User Access to Phone Background Image Setting

Check this check box to enable end users to change the background image on phones that use this common phone profile.

Feature Control Policy

You can choose a feature control policy that has already been configured in the Feature Control Policy configuration.

Secure Shell Information

Secure Shell User

Enter a user ID for the secure shell user. Cisco Technical Assistance Center (TAC) uses secure shell for troubleshooting and debugging. Contact TAC for further assistance.

See the Cisco Unified Communications Manager Security Guide for this release for information about how to configure encrypted phone configuration files to ensure that Cisco Unified Communications Manager does not send SSH credentials to the phone in the clear.

Secure Shell User Password

Enter the password for a secure shell user. Contact TAC for further assistance.

See the Cisco Unified Communications Manager Security Guide for this release for information about how to configure encrypted phone configuration files to ensure that Cisco Unified Communications Manager does not send SSH passwords to the phone in the clear.

Phone Personalization Information

Phone Personalization

The Phone Personalization setting allows you to enable a Cisco Unified IP Phone so it works with Phone Designer, a Cisco Unified Communications widget that allows a phone user to customize the wallpaper and ring tones on the phone.

From the Phone Personalization drop-down list box, choose one of the following options:

Disabled—The user cannot customize the Cisco Unified IP Phone by using Phone Designer.

Enabled—The user can use Phone Designer to customize the phone.

Default—The phone uses the configuration from the Phone Personalization enterprise parameter if you choose Default in both the Phone Configuration and Common Phone Profile Configuration windows. If you choose Default in the Common Phone Profile Configuration window but not in the Phone Configuration window, the phone uses the configuration that you specify in the Phone Configuration window.

You must install and configure Phone Designer so the phone user can customize the phone. Before you install and configure Phone Designer, identify which Cisco Unified IP Phone models work with Phone Designer, as described in the Phone Designer documentation. For more information on Phone Designer, see the Phone Designer documentation.

Always Use Prime Line

From the drop-down list box, choose one of the following options:

Off—When the phone is idle and receives a call on any line, the phone user answers the call from the line on which the call is received.

On—When the phone is idle (off hook) and receives a call on any line, the primary line is chosen for the call. Calls on other lines continue to ring, and the phone user must select those other lines to answer these calls.

Default—Cisco Unified Communications Manager uses the configuration from the Always Use Prime Line service parameter, which supports the Cisco Unified Communications Manager service.

Always Use Prime Line for Voice Message

From the drop-down list box, choose one of the following options:

On—If the phone is idle, the primary line on the phone becomes the active line for retrieving voice messages when the phone user presses the Messages button on the phone.

Off—If the phone is idle, pressing the Messages button on the phone automatically dials the voice-messaging system from the line that has a voice message. Cisco Unified Communications Manager always selects the first line that has a voice message. If no line has a voice message, the primary line is used when the phone user presses the Messages button.

Default—Cisco Unified Communications Manager uses the configuration from the Always Use Prime Line for Voice Message service parameter, which supports the Cisco Unified Communications Manager service.

Services Provisioning

From the drop-down list, choose how the phone will support the services:

Internal—The phone uses the phone configuration file to support the service. Choose this option or Both for Cisco-provided default services where the Service URL has not been updated; that is, the service URL indicates Application:Cisco/<name of service>; for example, Application:Cisco/CorporateDirectory. Choose Internal or Both for Cisco-signed Java MIDlets because Cisco-signed Java MIDlets are provisioned in the configuration file.

External URL—Choosing External URL indicates that the phone ignores the services in the phone configuration file and retrieves the services from a Service URL. If you configured a custom Service URL for a service, you must choose either External URL or Both; if you choose Internal in this case, the services that are associated with the custom URLs do not work on the phone.

Both—Choosing Both indicates that the phone supports both the services that are defined in the configuration file and external applications that are retrieved from custom service URLs. If you have phones in your network that can obtain the service information from the phone configuration file and phones in your network that can only use custom service URLs for obtaining the information, choose Both.

VPN Information

VPN Group

From the drop-down list, choose the VPN Group for the phone. For information about creating VPN groups, see the Virtual Private Network Configuration chapter in the Cisco Unified Communications Manager Security Guide.

VPN Profile

From the drop-down list, choose the VPN profile for the phone. For information about creating VPN profiles, see the Virtual Private Network Configuration chapter in the Cisco Unified Communications Manager Security Guide.

Product Specific Configuration

Disable USB

Disable the USB ports on the device and dock.

This is a required field.

Default: False

Note A reset of the device is required for this parameter to take effect.

Back USB Port

Indicates whether the back USB port on the phone is enabled or disabled.

This is a required field.

Default: Enabled

Side USB Port

Indicates whether the side USB port on the phone is enabled or disabled.

This is a required field.

Default: Enabled

Enable/Disable USB Classes

Indicates which the USB classes on the phone are enabled or disabled.

Default: Audio Class

SDIO

Indicates whether the SDIO device on the phone is enabled or disabled.

This is a required field.

Default: Disabled

Bluetooth

Indicates whether the Bluetooth device on the phone is enabled or disabled.

This is a required field.

Default: Enabled

Bluetooth Profiles

Indicates which bluetooth profiles on the phone are enabled or disabled.

This is a required field.

Default: Hands-free

Cisco Camera

Indicates whether the Cisco Camera on the phone is enabled or disabled.

This is a required field.

Default: Disabled

Enable Power Save Plus

To enable the Power Save Plus feature, select the day(s) that you want the phone to power off on schedule. You can select multiple days by pressing and holding the Control key while clicking on the days that you want Power Save Plus to operate. Power Save Plus mode turns off the phone for the time period specified in the Phone Off Time and Phone On Time fields. This time period is usually outside of your organization's regular operating hours. Power Save Plus mode turns on the phone automatically when Phone On Time arrives. When you select day(s) in this field, you receive a message that warns about emergency (e911) concerns.

Default: Disabled (no days selected).

Enable Audible Alert

This check box, when enabled, instructs the phone to play an audible alert ten minutes prior to the time specified in the field, Phone Off Time. To also audibly alert the user, enable this checkbox. This checkbox only applies if the Enable Power Save Plus list box has one or more days selected.

This is a required field.

Default: False

Allow EnergyWise Overrides

This checkbox determines whether you will allow the EnergyWise domain controller policy to send power level updates to the phones. A few conditions apply; first, one or more days must be selected in the Enable Power Save Plus field. If the Enable Power Save Plus list box does not have any days selected, the phone will ignore the EnergyWise directive to turn off the phone. Second, the settings in Unified CM Administration will take effect on schedule even if EnergyWise sends an override. For example, assume the Display Off Time is set to 22:00 (10:00 p.m.), the value in the Display On Time field is 06:00 (6:00 a.m.), and the Enable Power Save Plus has one or more days selected. If EnergyWise directs the phone to turn off at 20:00 (8:00 p.m.), that directive will remain in effect (assuming no phone user intervention occurs) until the configured Phone On Time at 6:00 a.m. At 6:00 a.m., the phone will turn on and resume receiving its power level changes from the settings in Unified CM Administration. To change the power level on the phone again, EnergyWise must reissue a new power level change command. Also, any user interaction will take effect so if a user presses the select softkey after EnergyWise has directed the phone to power off, the phone will power on as a result of the user action.

This is a required field.

Default: False

EnergyWise Domain

This field defines the EnergyWise domain in which the phone is participating. An EnergyWise domain is required by the Power Save Plus feature. If you have chosen days in the Enable Power Save Plus list box, you must also provide an EnergyWise domain.

Maximum length: 127

Default: Blank

EnergyWise Endpoint Security Secret

This field defines the password (shared secret) used to communicate within the EnergyWise domain. An EnergyWise domain and secret is required by the Power Save Plus feature. If you have chosen days in the Enable Power Save Plus list box, you must also provide an EnergyWise domain and secret.

Maximum length: 127

Default: Blank

Phone On Time

This field determines the time that the phone turns on automatically on the days that are selected in the Enable Power Save Plus list box. Enter the time in 24-hour format, where 00:00 represents midnight. For example, to automatically turn the phone on at 7:00 a.m., (0700), enter 07:00. To turn the phone on at 2:00 p.m. (1400), enter 14:00. If this field is blank, the phone automatically turns on at 00:00.

Default: 00:00

Phone Off Time

This field determines the time of day that the phone will turn itself off on the days that are selected in the Enable Power Save Plus list box. Enter the time in the hours:minutes format. If this field is blank, the phone automatically turns off at midnight (00:00).

Note If Phone On Time is blank (or 00:00) and Phone Off Time is blank (or 24:00), the phone will remain on continuously, effectively disabling the Power Save Plus feature unless you allow EnergyWise to send overrides.

Default: 24:00

Phone Off Idle Timeout

This field represents the number of minutes that the device must be idle before it requests the power sourcing equipment (PSE) to power down the device. The value in this field takes effect:

When the device was in Power Save Plus mode as scheduled and was taken out of Power Save Plus mode via some user interactions.

When the phone is repowered by the attached switch.

When the Phone Off Time is met but the phone is in use.

Default: 60

Minimum: 20

Maximum: 1440

Days Display Not Active

This field allows the user to specify the days that the backlight is to remain off by default. Typically this would be Saturday and Sunday for US corporate customers. Saturday and Sunday should be the default. The list contains all of the days of the week. To turn off backlight on Saturday and Sunday the User would hold down Control and select Saturday and Sunday.

Display On Duration

This field indicates the amount of time the display is to be active for when it is turned on by the programmed schedule. No value indicates the end of the day. Maximum value is 24 hours. This value is in free form hours and minutes. "1:30" activates the display for one hour and 30 minutes.

Default: 10:30

Maximum length: 5

Display Idle Timeout

This field indicates how long to wait before the display is turned off when it was turned on by user activity. This inactivity timer will continually reset itself during user activity. Leaving this field blank will make the phone use a predetermined default value of one hour. Maximum value is 24 hours. This value can be in free form hours and minutes. "1:30" turns off the display after an hour and 30 minutes of inactivity.

Default: 01:00

Maximum length: 5

Display On When Incoming Call

When the device is in screen saver mode, this turns on the display when a call is ringing. This is a required field.

Default: Enabled

Incoming Call Toast Timer

This parameter specifies the maximum time in seconds that the toast displays a new incoming call notification.

This is a required field.

Default: 5

Enable Mute Feature

This parameter provides a Mute softkey on 7906/7911.

This is a required field.

Default: False

Join And Direct Transfer Policy

This field indicates join and direct transfer policy for same line and across lines.

This is a required field.

Default: Same line, across line enable

Advertise G.722 and iSAC Codecs

Indicates whether Cisco Unified IP Phones will advertise the G.722 codec to Cisco Unified Communications Manager. Codec negotiation involves two steps: first, the phone must advertise the supported codec(s) to Cisco Unified Communications Manager (not all endpoints support the same set of codecs). Second, when Cisco Unified Communications Manager gets the list of supported codecs from all phones involved in the call attempt, it chooses a commonly supported codec based on various factors, including the region pair setting. Valid values specify Use System Default (this phone will defer to the setting specified in the enterprise parameter, Advertise G.722 Codec), Disabled (this phone will not advertise G.722 to Cisco Unified Communications Manager) or Enabled (this phone will advertise G.722 to Cisco Unified Communications Manager).

This is a required field.

Default: Use System Default

Video Calling

When enabled, indicates that the phone will participate in video calls when connected to an appropriately equipped PC.

This is a required field.

Default: Enabled

Wifi

Indicates whether the Wi-Fi on the phone is enabled or disabled.

This is a required field.

Default: Enabled

PC Port

Indicates whether the PC port on the phone is enabled or disabled. The port labeled "10/100 PC" on the back of the phone connects a PC or workstation to the phone so they can share a single network connection.

This is a required field.

Default: Enabled

Span to PC Port

Indicates whether the phone will forward packets transmitted and received on the phone port to the PC port. Select Enabled if an application is being run on the PC port that requires monitoring of the IP Phone's traffic; for example, monitoring and recording applications (common in call center environments) or network packet capture tools used for diagnostic purposes. To use this feature, PC Voice VLAN access must be enabled.

This is a required field.

Default: Disabled

PC Voice VLAN Access

Indicates whether the phone will allow a device attached to the PC port to access the Voice VLAN. Disabling Voice VLAN Access will prevent the attached PC from sending and receiving data on the Voice VLAN. It will also prevent the PC from receiving data sent and received by the phone. Set this setting to Enabled if an application is being run on the PC that requires monitoring of phone traffic. Such applications could include monitoring and recording applications and network monitoring software for analysis purposes.

This is a required field.

Default: Enabled

PC Port Remote Configuration

Allows remote configuration of the speed and duplex for the PC port of the phone, which overrides any manual configuration at the phone.

This is a required field.

Default: Disabled

Switch Port Remote Configuration

Allows remote configuration of the speed and duplex for the switch port of the phone, which overrides any manual configuration at the phone. Be aware that configuring this port may cause the phone to lose network connectivity.

This is a required field.

Default: Disabled

Automatic Port Synchronization

Enables the phone to synchronize the PC and SW ports to the same speed and to duplex. Only ports configured for auto negotiate change speeds.

This is a required field.

Default: Disabled

Cisco Discovery Protocol (CDP) Switch Port

Allows administrator to enable or disable Cisco Discovery Protocol (CDP) on the switch port.

This is a required field.

Default: Enabled


Caution CDP should only be disabled on the network port if this phone is connected to a non-Cisco switch. For further details, see the Cisco Unified Communications Manager Administration Guide.

Cisco Discovery Protocol (CDP) PC Port

Allows administrator to enable or disable Cisco Discovery Protocol (CDP) on the PC port.

This is a required field.

Default: Enabled


Caution Disabling CDP on the PC port will prevent Cisco VT Advantage/Unified Video Advantage from working properly on this phone. For further details, see the Cisco Unified Communications Manager Administration Guide.

LLDP-MED- Switch Port

Allows administrator to enable or disable Link Layer Discovery Protocol (LLDP-MED) on the switch port.

This is a required field.

Default: Enabled

Link Layer Discovery Protocol (LLDP)- PC Port

Allows administrator to enable or disable Link Layer Discovery Protocol (LLDP) on the PC port.

This is a required field.

Default: Enabled

LLDP Asset ID

Allows administrator to set Asset ID for Link Layer Discovery Protocol.

Maximum length: 32

LLDP Power Priority

Allows administrator to set Power Priority for Link Layer Discovery Protocol.

This is a required field.

Default: Unknown

Power Negotiation

Allows administrator to enable or disable power negotiation.

This is a required field.

Default: Enabled

802.1x Authentication

Specifies the 802.1x authentication feature status.

This is a required field.

Default: User Controlled

FIPS Mode

This parameter sets the Federal Information Processing Standards (FIPS) mode for the phone. The phone is a FIPS 140-2 level 1 compliant device when this option is enabled.

This is a required field.

Default: Disabled

80-bit SRTCP

Enable 80-bit authentication tag for SRTCP.

This is a required field.

Default: Disabled

Always On VPN

Indicates whether the device will always start the VPN AnyConnect client and establish a connection with the configured VPN profile from the Cisco Unified Communications Manager.

This is a required field.

Default: False

Allow User-Defined VPN Profiles

This parameter controls whether the user can use the AnyConnect VPN client to create VPN profiles. If disabled, the user cannot create VPN profiles.

This is a required field.

Default: True

Require Screen Lock

This parameter indicates whether screen lock is required on the device. If User Controlled is selected, the device will not prompt for a PIN or password. The PIN and Password options require the user to enter a password to unlock the screen. A PIN is a numeric password that is at least four digits long. A password is an alphanumeric password, consisting of at least 4 alphanumeric characters, one of which must be a non-numeric number, and one capital letter.

This is a required field.

Default: PIN

Screen Lock Timeout

Maximum idle time, in seconds, before the device automatically locks the screen. After the screen is locked, the user password is required to unlock it.

This is a required field.

Default: 600

Minimum: 15

Maximum: 1800

Lock Device During Audio Call

When the device is in a charging state and an active voice call is in progress, an administrator can override the screen lock PIN enforcement timer to keep the screen active during an audio call. Screen lock timer takes effect after the audio call is completed and the timer is exceeded.

This is a required field.

Default: Disabled

Kerberos Server

Authentication server for web proxy Kerberos.

Maximum length: 256

Kerberos Realm

Realm for web proxy Kerberos.

Maximum length: 256

Detect Unified CM Connection Failure

This field determines the sensitivity that the phone application has for detecting a connection failure to Cisco Unified Communications Manager (Unified CM), which is the first step before device failover to a backup Unified CM/SRST occurs. Valid values specify Normal (detection of a Unified CM connection failure occurs at the standard system rate) or Delayed (detection of a Unified CM connection failover occurs at approximately one-fourth the rate of Normal). For faster recognition of a Unified CM connection failure, choose Normal. If you prefer failover to be delayed slightly to give the connection the opportunity to reestablish, choose Delayed. Note that the precise time difference between Normal and Delayed connection failure detection depends on many variables that are constantly changing. This only applies to the wired Ethernet connection.

This is a required field.

Default: Normal

Time to Wait for Seamless Reconnect After TCP Drop or Roaming

This field indicates a grace period to establish a new TCP connection via keep-alive registration after the original TCP connection is torn down. The Seamless Reconnect is disabled if the value is set to 0.

Default: 5

Minimum: 0

Maximum: 300

Load Server

Indicates that the phone will use an alternative server to obtain firmware loads and upgrades, rather than the defined TFTP server. This option enables you to indicate a local server to be used for firmware upgrades, which can assist in reducing install times, particularly for upgrades over a WAN. Enter the hostname or the IP address (using standard IP addressing format) of the server. The indicated server must be running TFTP services and have the load file in the TFTP path. If the load file is not found, the load will not install. The phone will not be redirected to the TFTP server. If this field is left blank, the phone will use the designated TFTP server to obtain its load files and upgrades.

Maximum length: 256

IPv6 Load Server

Indicates that the phone will use an alternative IPv6 server to obtain firmware loads and upgrades, rather than the defined TFTP server. This option enables you to indicate a local IPv6 server to be used for firmware upgrades, which can assist in reducing install times, particularly for upgrades over a WAN. Enter the hostname or the IPv6 address (using standard IPv6 addressing format) of the server. The indicated server must be running TFTP services and have the load file in the TFTP path. If the load file is not found, the load will not install, and the phone will not be redirected to the TFTP server. If this field is left blank, the phone will use the designated TFTP server to obtain its load files and upgrades.

Maximum length: 25

Peer Firmware Sharing

Enable Peer to Peer image distribution (PPID) to allow a single phone in a subnet to retrieve an image firmware file and distribute it to its peers, thus reducing TFTP bandwidth and providing for a faster firmware upgrade time.

This is a required field.

Default: Enabled

Log Server

Specifies an IP address and port of a remote system where log messages are sent.

Maximum length: 32

HTTPS Server

Allows Administrator to permit http and https or https only connections if Web Access is enabled. This is a required field.

Default: http and https enabled

Web Access

This parameter indicates whether the phone will accept connections from a web browser or other HTTP client. Disabling the Web Server functionality of a phone will block access to the phone's internal web pages. These pages provide statistics and configuration information. Features, such as Quality Report Tool (QRT) , will not function properly without access to the phone's web pages.

This is a required field.

Default: Disabled

Settings Access

Indicates whether the Settings button on the phone is functional. When Settings Access is enabled, you can change the phone's network configuration, ring type, and volume. When Settings Access is disabled, the Settings button is also disabled; no options appear when you press the button. Also, you cannot adjust the ringer volume or save any volume settings.

This is a required field.

Default: Enabled

SSH Access

This parameter indicates whether the device will accept SSH connections. Disabling the SSH server functionality of the device will block access to the device.

This is a required field.

Default: Disabled

Ring Locale

An IP Phone has distinctive ring for On-net/Off-net or line based, but its ring cadence is fixed, and it is based on the U.S. standard only. Ring cadence in the U.S. standard is opposite to that of the Japan standard.

This is a required field.

Default: Disabled

Android Debug Bridge or ADB

This parameter enables or disables the Android Debug Bridge (ADB) on the device.

This is a required field.

Default: Disabled

Allow Applications from Unknown Sources

This parameter controls whether the user can install Android applications on the device from a URL or from Android packages (APK) that are received through email, instant message (IM), or from a Secure Digital (SD) card.

This is a required field.

Default: Disabled

Allow Applications from Android Market

This parameter controls whether the user can install Android applications from the Google Android Market.

This is a required field.

Default: False

Allow Applications from Cisco AppHQ

This parameter controls whether the user can install Android applications from the Cisco AppHQ.

This is a required field.

Default: False

AppHQ Domain

The fully qualified domain name to use when users log into AppHQ. If empty, the user will specify their own domain name along with their username. The AppHQ domain is used to associate the user to a given Custom AppHQ store, if it exists. Example: cisco.com.

Maximum length: 256

Enable Cisco UCM App Client

This parameter controls whether the Application Client runs on the device. When the Application Client is enabled, users can select the applications they would like to install from the Cisco Unified Communications Manager.

This is a required field.

Default: False

Company Photo Directory

This parameter specifies a URL that the device can query for a user and get the image associated with that user.

Maximum length: 256

Voicemail Server (Primary)

Hostname or IP address of the primary mailstore voicemail server.

Maximum length: 256

Voicemail Server (Backup)

Hostname or IP address of the backup mailstore voicemail server.

Maximum length: 256

Presence and Chat Server (Primary)

Hostname or IP address of the primary Presence Server.

Maximum length: 256

Presence and Chat Server Type

This parameter indicates the type of server specified in the Presence and Chat Server field.

This is a required field.

Default: Cisco WebEx Connect

Presence and Chat Single Sign-On (SSO) Domain

The enterprise domain used by Cisco WebEx Connect Cloud to perform Single-Sign-On (SSO) authentication against an enterprise.

Maximum length: 256

PSTN Mode

Enable PSTN Mode for IP Phone 6921/6941/6961.

This is a required field.

Default: Disabled

Background Image

This parameter specifies the default wallpaper file. This parameter takes effect only if the administrator disables end-user access to the phone wallpaper list.

Maximum length: 64

Simplified New Call UI

This parameter specifies whether to use simplified call UI style when the phone is off-hook. Those who like the New Call Window can continue to use that at the same time that those who prefer the Simplified New Call Session can use that method.

This is a required field.

Default: Disabled

Revert to All Calls

When enabled, phone will revert to All Calls after any call is ended if the call is on a filter other than Primary line or All Calls.

This is a required field.

Default: Disabled

RTCP for Video

When disabled, video lipsync will be relying on free run mode.

This is a required field.

Default: Enabled

Provide Dial Tone from Release Button

Indicates whether Dial Tone is provided when Release button is pressed. If the value is true, a new Call Window will be displayed after Release button is pressed.

This is a required field.

Default: Disabled

Hide Video By Default

This field provides the additional flexibility of hiding the video window by default.

When this feature is enabled, the video window is initially hidden on video calls. If Auto Transmit Video is on, the phone will display a Hide Video View while the video is being transmitted to the remote party. This may make distinguishing video calls from voice calls more difficult for end users. The benefit of Hide Video by Default is that in work environments where users are more likely to mute their video or close the shutters on the camera, the far end user will see the audio call plane rather than a black mute box on their phone. Hide Video by Default is not recommended for work environments where video calling is used often with cameras open, enabled, and unmuted.

This is a required field.

Default: Disabled

VXC VPN Option

This field indicates how VXC VPN is supported. If Dual Tunnel is selected, phone establishes two VPN tunnels, one for Phone and another for VXC device. If Single Tunnel is selected, phone establishes only one VPN tunnel for phone and VXC-device to share. Where uncompromised voice/video quality is required the dual VPN tunnel solution is recommended.

Through the use of two VPN tunnels the host Cisco IP Phone is able to provide prioritization of its CPU and memory resources to the data associated with the Phones Voice/video functions over that of the data associated with the VXC VPN tunnel. This approach will require two manual login entries (dependent on security parameters), one for Phone's Voice/Video VPN and another for VXC VPN. The two tunnel approach also requires two VPN concentrator ports and two IP addresses adding potential costs.

A single VPN tunnel option is implemented for those customers willing to trade off potential voice/video quality for a simplified operating model. The solution consists of operating over a single VPN tunnel by sharing the available 89/99xx processor and memory resources across the voice, video, and VDI services. The IP Phone will be unable to prioritize data handing of one service over another. As a result, possible degradation of performance of the IP Phones voice/video media handling and/or Phone UI functions due to IP Phone CPU loading.

This is a required field.

Default: Dual Tunnel

VXC Challenge

This field indicates whether or not to challenge VXC device.

If No Challenge option is selected, VXC challenge will be bypassed.

This is a required field.

Default: Challenge

VXC-M Servers

VXC Management Server IP address list, separated with commas.

Maximum length: 255

Record Call Log from Shared Line

This field indicates whether or not to record the call log from a shared line.

This is a required field.

Default: Disabled

URLs to Block in File Transfer

Semicolon separated list of URLs to block during file transfer operations.

Maximum length: 1024

Default: Blank

Automatically Control Tethered Desk Phone

If enabled, the client will automatically control the tethered desktop phone.

This is a required field.

Default: Disabled

Extend and Connect Capability

Indicates if Extend and Connect capabilities are enabled for the client. This allows the client to monitor and control calls on a third party PBX, PSTN, and other remote phones.

This is a required field.

Default: Enabled

Display Contact Photos

Indicates if contact photo retrieval and display is enabled or disabled for the client.

This is a required field.

Default: Enabled

Number Lookups on Directory

Indicates if phone number lookups using the corporate directory are enabled or disabled for the client.

This is a required field.

Default: Enabled

Jabber For Windows Software Update Server URL

The URL of the software update server that the Jabber for Windows client will use when the user selects the Update Jabber link.

Maximum length: 1024

Default: Blank

Analytics Collection

Indicates if analytics collection is enabled or disabled for the client.

This is a required field.

Default: Disabled

Problem Report Server URL

The URL of the problem report server that will be used by the client.

Maximum length: 1024

Default: Blank

Analytics Server URL

The URL of the analytics server that will be used by the client.

Maximum length: 1024

Default: Blank

Cisco Support Field

Semicolon separated list of custom settings that will be used by the client to assist with deployment. This field should only be used with the assistance of Cisco Support personnel.

Maximum length: 1024

Default: Blank


Unity Connection Distribution List Infrastructure Data Object Fields

Table B-8 Unity Connection Distribution List Infrastructure Data Object Fields 

Field
Description

Alias

Alias name of the distribution list.

Display Name

Name of the distribution list.

Extension

Extension that the phone system uses to connect.

Partition

Partition that is used to define the scope of the distribution list that a user or outside caller can reach.

Allow Contacts

Specifies whether contacts can be added as members of the distribution list.

Accept Messages from Foreign Systems

Allows users on remote voice messaging systems that are configured as VPIM locations to send messages to this distribution list.

Member List

List of users associated with the distribution list. Use the format Alias/MemberType.

You are allowed to add, modify, or delete only 200 members at a time.

For better performance, we recommend a maximum of 20 distribution lists, each with 500 members. If you want to manage more than 500 members, you can use a nested distribution list.


Device Pool Infrastructure Data Object Fields

Table B-9 Device Pool Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Cisco Unified CM Group

List of available Cisco Unified Communications Manager groups.

Date/Time Group

The date/time group to assign to devices in this device pool.

Region

The Cisco Unified Communications Manager region to assign to devices in this device pool.

Softkey Template

Softkey template that determines the configuration of the softkeys on Cisco IP Phones.

SRST Reference

A survivable remote site telephony (SRST) reference to assign to devices in this device pool.

Calling Search Space for Auto-Generation

The calling search space to assign to devices in this device pool that auto-registers with Cisco Unified Communications Manager.

Media Resource Group List

Provides a prioritized grouping of media resource groups. An application chooses the required media resource, such as a Music On Hold server, from the available media resources according to the priority order that is defined in a Media Resource Group List. If this field is left blank, the Media Resource Group that is defined in the device pool is used.

Network Hold MOH Audio Source

The audio source that plays when the network initiates a hold action.

User Hold MOH Audio Source

The audio source that plays Music On Hold when the user initiates a hold action.

Network Locale

The locale that is associated with phones and gateways.

User Locale

User location associated with the phone user. The user locale identifies a set of detailed information to support users, including language, font, date and time formatting, and alphanumeric keyboard text information.

Connection Monitor Duration

Defines the amount of time that the IP phone monitors its connection to Cisco Unified Communications Manager before it unregisters from SRST and re-registers to Cisco Unified Communications Manager.

MLPP Indication

Specifies whether devices in the device pool that are capable of playing precedence tones will use the capability when the devices place an MLPP precedence call.

MLPP Preemption

Specifies whether devices in the device pool that are capable of preempting calls in progress will use the capability when the devices place an MLPP precedence call.

MLPP Domain

Multilevel Precedence and Preemption (MLPP) Domain that is associated with this device.


H323 Gateway Infrastructure Data Object Fields

Table B-10 H323 Gateway Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Device Pool

List of available device pools. The device pool specifies a collection of properties for this device including Unified CM Group, Date/Time Group, Region, and Calling Search Space for auto-registration of devices.

Call Classification

Determines whether an incoming call that is using this gateway is considered off the network (OffNet) or on the network (OnNet).

Media Resource Group List

Provides a prioritized grouping of media resource groups.

Location

Location for this device.

Media Termination Point Required

If Media Termination Point is used to implement features that H.323 does not support (such as hold and transfer), select Yes.

Retry Video Call As Audio

Applies to video endpoints that receive calls.

Wait for Far End H.245 Terminal Capability Set

Specifies that Cisco Unified Communications Manager needs to receive the far-end H.245 Terminal Capability Set before it sends its H.245 Terminal Capability Set.

MLPP Domain

Multilevel Precedence and Preemption (MLPP) Domain to associate with this device.

Significant Digits Value

Represents the number of final digits that are retained on inbound calls.

Calling Search Spaces

Specifies the collection of Route Partitions that are searched to determine how a collected (originating) number should be routed.

AAR Calling Search Space

Specifies the collection of route partitions that are searched to determine how to route a collected (originating) number that is otherwise blocked due to insufficient bandwidth.

Prefix DN

The prefix digits that are appended to the called party number on incoming calls.

Redirecting Number IE Delivery - Inbound

Selecting Yes accepts the Redirecting Number IE in the incoming SETUP message to the Cisco Unified Communications Manager.

Calling Party Selection

Any outbound call on a gateway can send directory number information. Choose which directory number is sent.

Calling Party Presentation

Choose whether you want the Cisco Unified Communications Manager to allow or restrict the display of the calling party phone number.

Called Party IE Number Type Unknown

Choose the format for the number type in called party directory numbers.

Calling Party IE Number Type Unknown

Choose the format for the number type in calling party directory numbers.

Called Numbering Plan

Choose the format for the numbering plan in called party directory numbers.

Calling Numbering Plan

Choose the format for the numbering plan in calling party directory numbers.

Caller ID DN

Enter the pattern that you want to use for calling line ID, from 0 to 24 digits.

Display IE Delivery

Enables delivery of the display IE in SETUP, CONNECT, and NOTIFY messages for the calling and called party name delivery service.

Redirecting Number IE Delivery - Outbound

Includes the Redirecting Number IE in the outgoing SETUP message from the Cisco Unified Communications Manager to indicate the first redirecting number and the redirecting reason of the call when the call is forwarded.

Packet Capture Mode

Configure this field if you need to troubleshoot encrypted signaling information for the H.323 gateway.

Common Device Config

Configuration of common device settings, such as the softkey template and user locale.

SRTP Allowed

Select Yes if you want Cisco Unified Communications Manager to allow secure and nonsecure calls over the gateway.

Trace Flag

Not used.

Version Stamp

Not used.

CTI

Not used.

Enable Outbound FastStart

Select Yes to enable the H323 FastStart feature for outgoing calls.

AAR Group

Select an alternate routing group if there is insufficient bandwidth.

Packet Capture Duration

Configure this field if you need to troubleshoot encrypted signaling information for the H.323 gateway.


Hunt List Infrastructure Data Object Fields

Table B-11 Hunt List Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Cisco Unified CM Group

List of available Cisco Unified Communications Manager groups.

Enable this Hunt List

Select Yes to enable the hunt list.

Available Line Group

List of available line groups.


Hunt Pilot Infrastructure Data Object Fields

Table B-12 Hunt Pilot Infrastructure Data Object Fields 

Field
Description

Pattern Definition

Pattern

The hunt pilot, including numbers and wildcards (do not use spaces).

Route Partition

If you want to use a partition to restrict access to the hunt pilot, choose the desired partition.

Description

Optional description.

Numbering Plan

Choose a numbering plan.

Route Filter

If your hunt pilot includes the @ wildcard, you may choose a route filter.

MLPP Precedence

MLPP precedence setting.

Hunt List

Choose the hunt list for which you are adding a hunt pilot.

Urgent Priority

Select Yes to interrupt interdigit timing when Cisco Unified Communications Manager must route a call immediately.

Block Enabled

Enable or disable block.

Release Cause

Dependent on the Block Enabled field. If a release cause is selected, then Block Enabled must be set to True.

Calling Party Transformations

Use Calling Party's External Phone Number Mask

Select Yes if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls.

Calling Party Transformation Mask

Enter a transformation mask value.

Calling Party Prefix Digits (Outgoing Calls)

Enter the prefix digits.

Calling Line Presentation

Used as a supplementary service to allow or restrict the originating caller's phone number on a call-by-call basis.

Calling Name Presentation

Used as a supplementary service to allow or restrict the originating caller's name on a call-by-call basis.

Connected Party Transformations

Connected Line Presentation

Used as a supplementary service to allow or restrict the called party's phone number on a call-by-call basis.

Connected Name Presentation

Used as a supplementary service to allow or restrict the called party's name on a call-by-call basis.

Called Party Transformations

Called Party Discard Digits

Select the discard digits instructions that you want to associate with this hunt pilot.

Called Party Transformation Mask

Enter a transformation mask value.

Called Party Prefix Digits (Outgoing Calls)

Enter the prefix digits.

Queuing

 

Queue Calls

Check this check box to enable Call Queuing.

Network Hold MOH Source and Announcements

Choose the audio source file that contains the music on hold and announcement to be played when a call is held in a queue.

Maximum Number of Callers Allowed in a Queue

Enter a value that specifies the maximum number of callers to be queued per hunt pilot.

Call Queuing allows up to 100 callers to be queued per hunt pilot. Once this limit is reached on a particular hunt pilot, subsequent calls can be routed to an alternate number.

Enable This When Queue is Full

Check this check box to route the calls to an alternate number when the queue is full.

Route the Call to This Destination When the Queue is Full

Enter the directory number to which the calls are routed when the queue is full.

This field allows the following characters: numerals (0 to 9), uppercase X (X), asterisk (*), and hash symbol (#).

Full Queue Calling Search Space

Specify the calling search space that is used for diverting the calls when the queue is full.

Maximum Wait Time in Queue

Enter a value (in seconds) that specifies the maximum wait time for each call in a queue.

Each caller can be queued for up to 3600 seconds per hunt pilot. Once this limit is reached, that caller is routed to an alternate number.

Enable This When Max Wait Time is Met

Check this check box to route the calls to an alternate number when the maximum wait time is reached.

Route the Call to This Destination If Max Wait Time is Met

Enter the directory number to which the calls are routed when the maximum wait time is reached.

Maximum Wait Time Calling Search Space

Specify the calling search space that is used for diverting the calls when the maximum wait time is reached.

Enable This When No Hunt Members are Logged In

Check this check box to route the calls to an alternate number when none of the hunt members are logged in or registered.

Route the Call to This Destination If there is No Agent

Enter the directory number to which the calls are routed when none of the members of the hunt pilot are available or registered at the time of the call.

For Call Queuing, a hunt pilot member is considered available if that member has both deactivated do not disturb (DND) and logged into the hunt group. In all other cases, the line member is considered unavailable or logged off.

No Hunt Members Logged In or Registered Calling Search Space

Specify the calling search space that is used for diverting the calls when none of the hunt members are logged in or registered.


Line Group Infrastructure Data Object Fields

Table B-13 Line Group Infrastructure Data Object Fields 

Field
Description

Name

Object name.

RNA Reversion Timeout

Enter a time, in seconds, after which Cisco Unified Communications Manager will distribute a call to the next available or idle member of this line group or to the next line group if the call is not answered and if the first hunt option, "Try next member; then, try next group in Hunt List" is chosen.

Distribution Algorithm

Select a distribution algorithm, which applies at the line group level.

Hunt Algorithm No Answer

For a given distribution algorithm, select a hunt option for Cisco Unified Communications Manager to use if a call is distributed to a member of a line group that does not answer.

Hunt Algorithm Busy

For a given distribution algorithm, select a hunt option for Cisco Unified Communications Manager to use if a call is distributed to a member of a line group that is busy.

Hunt Algorithm Not Available

For a given distribution algorithm, select a hunt option for Cisco Unified Communications Manager to use if a call is distributed to a member of a line group that is not available.

Directory Numbers

Enter a directory number that already exists in Cisco Unified Communications Manager.


Location Infrastructure Data Object Fields

Table B-14 Location Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Audio Bandwidth

Enter the maximum amount of audio bandwidth (in kbps) that is available for all audio calls on the link between this location and other locations.

Note This option is available for Cisco Unified Communications Manager 9.0 or higher versions. For Cisco Unified Communications Manager 8.x and earlier verisons, Audio Kilobytes field will be displayed.

Video Bandwidth

Enter the maximum amount of video bandwidth (in kbps) that is available for all video calls on the link between this location and other locations. Use 0 for Unlimited and -1 for None.

Note This option is available for Cisco Unified Communications Manager 9.0 or higher versions. For Cisco Unified Communications Manager 8.x and earlier verisons, Video Kilobytes field will be displayed.


Media Resource Group Infrastructure Data Object Fields

Table B-15 Media Resource Group Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Available Devices

The available media resources that can be selected.

Is Multicast for MOH Audio

Click Yes to use multicast for Music On Hold Audio.


Media Resource Group List Infrastructure Data Object Fields

Table B-16 Media Resource Group List Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Available Media Resource Group Names

The available media resource groups that can be selected.


Meet-Me Number/Pattern Data Object Fields

Table B-17 Meet-Me Number/Pattern Data Object Fields 

Field
Description

Directory Number or Pattern

Enter Meet-Me number/pattern or a range of numbers.

To configure a range, the dash must appear within brackets and follow a digit; for example, to configure the range 1000 to 1050, enter 10[0-5]0.

This field allows up to 24 characters.

Description

The description can include up to 50 characters. The following characters are not allowed: double-quotes ("), backslash (\), dash (-), percentage sign (%), ampersand (&), or angle brackets (<>).

Partition

To use a partition to restrict access to the Meet-Me number/pattern, choose the desired partition from the drop-down list.

Minimum Security Level

Choose the minimum security level for this Meet-Me number/pattern from the drop-down list.

Choose Authenticated to block participants with nonsecure phones from joining the conference.

Choose Encrypted to block participants with nonsecure phones from joining the conference.

Choose Non Secure to allow all participants to join the conference.


Route Group Infrastructure Data Object Fields

Table B-18 Route Group Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Available Members

The available devices that can be chosen.

Ports

If the device supports individually configurable ports, choose the port.


Route List Infrastructure Data Object Fields

Table B-19 Route List Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Cisco Unified CM Group

List of available Cisco Unified Communications Manager groups.

Enable this Route List

Select Yes to enable the route list.

Available Member Route Group

List of available route groups.

Available Member Use Fully Qualified Calling Party Number

Determines if the available route groups must use fully qualified calling party numbers.

Member Calling Party Transformation Mask

Transformation mask value.

Member Calling Party Prefix Digits

Prefix digits.

Available Member Discard Digits Instruction

Determines the discard digits instructions that you want to associate with this route list.

Member Called Party Transformation Mask

Transformation mask value.

Member Called Party Prefix Digits

Prefix digits.


Route Partition Infrastructure Data Object Fields

Table B-20 Route Partition Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.


Route Pattern Infrastructure Data Object Fields

Table B-21 Route Pattern Infrastructure Data Object Fields 

Field
Description

Pattern

A valid route pattern, including numbers and wildcards.

Route Partition

If you want to use a partition to restrict access to the route pattern, select the desired partition.

Description

Optional description.

Numbering Plan

Numbering plan. The default setting is NANP (North American Numbering Plan).

Route Filter

If your route pattern includes the @ wildcard, you may choose a route filter.

MLPP Precedence

MLPP precedence setting.

Gateway, Route List, or SIP Trunk

Choose the gateway or route list for which you are adding a route pattern. You can also enter a value that does not appear in the list. If you enter a custom value, make sure to specify whether it is a gateway, route list, or SIP trunk. After the name, add one of the following:

[GW]—Gateway

[RL]—Route list

[ST]—SIP trunk

For example, gatewayname[GW].

Is Gateway Destination Type Gateway

Indicates whether the destination device is a gateway.

Urgent Priority

If Yes is selected, the interdigit timing is interrupted when Cisco Unified Communications Manager must route a call immediately.

Block Enabled

Enables or disables block.

Release Cause

Dependent on the Block Enabled field. If a release cause is selected, then Block Enabled must be set to True.

Call Classification

Indicates whether the call that is routed through this route pattern is considered either off (OffNet) or on (OnNet) the local network.

Allow Device Override

If Yes is selected, the system uses the Call Classification setting that is configured on the associated gateway or trunk to consider the outgoing call as OffNet or OnNet.

Provide Outside Dial Tone

If Yes is selected, an outside dial tone is provided.

Use Calling Party's External Phone Number Mask

Select Yes if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls.

Calling Party Transformation Mask

Transformation mask value.

Calling Party Prefix Digits (Outgoing Calls)

Prefix digits.

Calling Line ID Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the calling party's phone number on the called party's phone display for this route pattern.

Calling Name Presentation

Determines whether you want the Cisco Unified Communications Manager to allow or restrict the display of the calling party's name on the called party's phone display for this route pattern.

Connected Line ID Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the connected party's phone number on the calling party's phone display for this route pattern.

Connected Name Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the connected party's name on the calling party's phone display for this route pattern.

Called Party Discard Digits (Outgoing Calls)

Determines the discard digits instructions that you want to associate with this route pattern.

Called Party Transformation Mask

Transformation mask value.

Called Party Prefix Digits (Outgoing Calls)

Prefix digits.


Service Profile Infrastructure Data Object Fields

Table B-22 Service Profile Infrastructure Data Object Fields 

Field
Description

Name

Enter the name of the service profile.

Maximum characters: 50 (ASCII only).

Allowed Values: All characters allowed except quotes ("), angle brackets (< >), backslash (\), ampersand (&), and percent (%).

Description

(Optional) Enter a description that helps you to distinguish between service profiles when you have more than one configured.

Allowed Values: All characters allowed except quotes ("), angle brackets (< >), backslash (\), ampersand (&), and percent (%).

Default profile

Check this check box to make this service profile the default option for the system.

If you specify a default service profile, end users that do not have an associated service profile automatically inherit the default service profile settings.

Voicemail Profile

Primary

Select the primary voicemail server with which you want to associate this service profile.

Secondary

Select a secondary voicemail server, if applicable.

Tertiary

Select a tertiary voicemail server, if applicable.

Credentials source for voicemail service

If user credentials for the voicemail service are shared with another service, select the appropriate service. The user credentials automatically synchronize from the service that you select.

Default Setting: Not set

Mailstore Profile

Primary

Select the primary mailstore server with which you want to associate this service profile.

Secondary

Select a secondary mailstore server, if applicable.

Tertiary

Select a tertiary mailstore server, if applicable.

Inbox folder

The name of the folder on the mailstore server in which to store new messages. Only change this value if the mailstore server uses a different folder name from the default folder.

Default: INBOX

Trash folder

The name of the folder on the mailstore server in which to store deleted messages. Only change this value if the mailstore server uses a different folder name from the default folder.

Default: Deleted Items

Polling Interval (in seconds)

The time (in seconds) that can elapse between polls of the IMAP server for new voice messages, when IDLE is not supported by the mailstore or when a connection failure occurs.

Allowed values: 60 - 900

Default: 60

Allow dual folder mode

This dual folder setting is checked by default for use with mailstores that support the IMAP UIDPLUS extensions (RFC 2359 and 4315). By default, the Client Services Framework (CSF) detects if UIDPLUS is not supported and automatically reverts to Single Folder mode. Uncheck this check box if you know that UIDPLUS is not supported and you want to force the system to use Single Folder mode.

Default: True

Conferencing Profile

Primary

Select the primary conferencing server with which you want to associate this service profile.

Secondary

Select a secondary conferencing server, if applicable.

Tertiary

Select a tertiary conferencing server, if applicable.

Server Certificate Verification

Specify how the conferencing server associated with this profile supports TLS connections. This setting is for TLS verification of the conferencing servers listed for this conferencing profile.

Select from the following options:

Any Certificate

Cisco Jabber accepts all valid certificates.

Self Signed or Keystore

Cisco Jabber accepts the certificate if the certificate is self-signed, or the signing Certificate Authority certificate is in the local trust store.

Note A keystore is a file that stores authentication and encryption keys.

Keystore Only

Cisco Jabber accepts only certificates that are defined in the keystore. You must import the certificate or its Certificate Authority signing certificate into the local trust store.

Credentials source for web conferencing service

If user credentials for the meeting service are shared with another service, select the appropriate service. The user credentials automatically synchronize from the service that you select.

Default Setting: Not set

Directory Profile

Primary

Select the primary directory server with which you want to associate this service profile.

Secondary

Select a secondary conferencing server, if applicable.

If you do not set up any backup directory servers, you cannot perform directory searches for Cisco Jabber clients if the first server fails.

Tertiary

Select a tertiary conferencing server, if applicable.

If you do not set up any backup directory servers, you cannot perform directory searches for Cisco Jabber clients if the first server fails.

Use UDS for contact resolution

Check this check box if you want to use the UDS service provided in Cisco Unified CM for the directory lookup instead of external directory.

Use Logged On User Credential

Check this check box to prevent anonymous queries and force the user to enter credentials to sign in to the LDAP server.

Username

Enter the distinguished name for the user ID that is authorized to run queries on the LDAP server, in the format useraccount@domain.com.

Maximum length: 128

Password

Type the password for the Username that is authorized to run queries on your LDAP server.

Maximum length: 128

Search Base

This field allows you to narrow your Cisco Unified Personal Communicator contact search queries to a certain part of the LDAP directory. Enter the container or directory on the LDAP server where you have configured your LDAP users. Example for the search base wtih Microsoft Active Directory inetgration: cn=users,DC=EFT-LA,DC=cisco,DC=com.

Maximum length: 256

Recursive Search on All Search Bases

Check this check box to perform a recursive search of the directory starting at the search base. Recursive search allows for Cisco Unified Personal Communicator contact search queries to search all of the LDAP directory tree from a given search context (search base).

Search Timeout (seconds)

Set the default timeout for searches (default is 5 seconds).

Base Filter (Only used for Advance Directory)

Use this option only if the object type that you want to retrieve with queries that you execute against Active Directory is not a user object.

Maximum length: 256

IM and Presence Profile

Primary

Select the primary IM and Presence server with which you want to associate this service profile.

Secondary

Select a secondary conferencing server, if applicable.

Tertiary

Select a tertiary conferencing server, if applicable.

CTI Profile

Primary

Select the primary CTI server with which you want to associate this service profile.

Secondary

Select a secondary CTI server, if applicable.

Tertiary

Select a tertiary CTI server, if applicable.


SIP Trunk Infrastructure Data Object Fields

Table B-23 SIP Trunk Infrastructure Data Object Fields 

Field
Description

AAR Group

The AAR group (Automated Alternate Routing) provides the prefix digits that are used to route calls that are otherwise blocked because of insufficient bandwidth.

An AAR group setting of None specifies that no rerouting of blocked calls will be attempted.

Call Classification

Determines whether an incoming call that is using this trunk is considered off the network (OffNet) or on the network (OnNet), or should use the system default setting.

Common Device Config

Choose the common device configuration for which you want this trunk assigned.

The common device configuration includes the attributes (services or features) that are associated with a particular user. Common device configurations are configured in the Common Device Configuration page.

Connected Party Transformation CSS

Choose to transform the connected party number on the device in order to display the connected number in another format, such as a DID or E164 number.

Cisco Unified Communications Manager includes the transformed number in the headers of various SIP messages, including 200 OK and mid-call update/reinvite messages.

Make sure that the Connected Party Transformation CSS that you choose contains the connected party transformation pattern that you want to assign to this device.

If you configure the Connected Party Transformation CSS as None, the transformation does not match and is not applied. Ensure that you configure the Connected Party Transformation CSS in a non-null partition that is not used for routing.

Device Name

Object name.

Description

Optional description.

Device Pool

List of available device pools. The device pool specifies a collection of properties for this device, including Unified CM Group, Date/Time Group, Region, and Calling Search Space for auto-registration of devices.

Location

Specifies the total bandwidth that is available for calls between this location and the central location (or hub). A location setting of Hub_None specifies unlimited available bandwidth.

Media Resource Group List

Provides a prioritized grouping of media resource groups. An application chooses the required media resource, such as a Music On Hold server, from the available media resources according to the priority order that is defined in a Media Resource Group List.

Media Termination Point Required

Used to indicate whether a media termination point (MTP) is used to implement features that H.323 does not support (such as hold and transfer).

Check the Media Termination Point Required check box if you want to use a media termination point to implement features. Deselect the Media Termination Point Required check box if you do not want to use a media termination point to implement features.

Check this check box only for H.323 clients and those H.323 devices that do not support the H.245 Empty Capabilities Set, or if you want media streaming to terminate through a single source.

If you check this check box to require an MTP, and either device is a video endpoint, the call operates as audio only.

Retry Video Call as Audio

Applies to video endpoints that receive calls. For trunks, it pertains to calls that are received from Cisco Unified Communications Manager but not to calls that are received from the wide area network (WAN).

By default, the system checks this check box to specify that this device should immediately retry a video call as an audio call (if it cannot connect as a video call) prior to sending the call to call control for rerouting.

If you uncheck this check box, a video call that fails to connect as video will not try to establish an audio call. The call then fails to call control, and call control routes the call via Automatic Alternate Routing (AAR) and (or) route/hunt list.

Unattended Port

If selected, calls can be redirected, transferred, or forwarded to an unattended port, such as a voice mail port.

The default value is deselected.

SRTP Allowed

Select if you want Cisco Unified Communications Manager to allow secure and nonsecure calls over the trunk.

If you do not check this check box, Cisco Unified Communications Manager prevents SRTP negotiation with the trunk and uses RTP.

If you check this check box, it is recommended that you configure IPSec, so you do not expose keys and other security-related information during call negotiations.

If you do not configure IPSec correctly, you must consider signaling between Cisco Unified Communications Manager and the gateway as nonsecure.

Use Trusted Relay Point

From the list, enable or disable whether Cisco Unified Communications Manager inserts a Trusted Relay Point (TRP) device with this media endpoint. Choose one of the following values:

Default—The device uses the Use Trusted Relay Point setting from the common device configuration with which this device associates.

Off—Disables the use of a TRP with this device. This setting overrides the Use Trusted Relay Point setting in the common device configuration with which this device associates.

On—Enables the use of a TRP with this device. This setting overrides the Use Trusted Relay Point setting in the common device configuration with which this device associates.

A TRP device designates an MTP or transcoder device that is labeled as a TRP.

Cisco Unified Communications Manager places the TRP closest to the associated endpoint device if more than one resource is needed for the endpoint (for example, a transcoder or RSVPAgent).

If both TRP and MTP are required for the endpoint, TRP is used as the required MTP.

If both TRP and RSVPAgent are needed for the endpoint, Cisco Unified Communications Manager first tries to find an RSVPAgent that can also be used as a TRP.

If both TRP and transcoder are needed for the endpoint, Cisco Unified Communications Manager first tries to find a transcoder that is also designated as a TRP.

Incoming Calling Party Unknown Number Prefix

If this is set to Default, the Call Processor uses the prefix at the next level setting (Device Pool/Service Parameter). Otherwise, the value configured is used as the prefix unless the field is empty, in which case no prefix is assigned.

MLPP Domain

Choose an MLPP Domain to associate with this device. If you leave this field empty, the device inherits its MLPP Domain from the value that was set for the device pool.

If the device pool does not have an MLPP Domain setting, this device inherits its MLPP Domain from the value that was set for the MLPP Domain Identifier enterprise parameter.

Remote-Party-Id

Allows the SIP Trunk to send the Remote-Party-ID (RPID) header in outgoing SIP messages from Cisco Unified Communications Manager to the remote destination. If you select Yes, the SIP Trunk always sends the RPID header.

Asserted-Identity

Allows the SIP Trunk to send the Asserted-Type and SIP Privacy headers in SIP messages.

If you select Yes, the SIP Trunk always sends the Asserted-Type header. Whether the SIP Trunk sends the SIP Privacy header depends on the SIP Privacy configuration.

If you select No, the SIP Trunk does not include any Asserted-Type or SIP Privacy headers in its SIP messages.

For more information, see the descriptions of Asserted-Type and SIP Privacy in this table.

Asserted-Type

Specifies the type of Asserted Identity header that SIP Trunk messages should include.

Select one of the following values:

Default—Represents the default value. Screening indication information that the SIP Trunk receives from Cisco Unified Communications Manager Call Control determines the type of header the SIP Trunk sends.

PAI—The Privacy-Asserted Identity (PAI) header is sent in outgoing SIP Trunk messages. This value overrides the screening indication value that comes from Cisco Unified Communications Manager.

PPI—The Privacy Preferred Identity (PPI) header is sent in outgoing SIP Trunk messages. This value overrides the screening indication value that comes from Cisco Unified Communications Manager.

Note These headers are sent only if the Asserted Identity check box is checked.

SIP Privacy

Specifies the type of SIP privacy header for SIP Trunk messages to include.

Select one of the following values:

Default—Represents the default value. Name and number presentation values that the SIP Trunk receives from the Cisco Unified Communications Manager Call Control compose the SIP Privacy header.

For example:

If the name and number presentation is restricted, the SIP Trunk sends the SIP Privacy header.

If the name and number presentation is allowed, the SIP Trunk does not send the Privacy header.

None—The SIP Trunk includes the header Privacy:none, which means that presentation is allowed. This value overrides the Presentation information that comes from Cisco Unified Communications Manager.

ID—The SIP Trunk includes the header Privacy:id, which means that the presentation is restricted for both name and number.

This value overrides the presentation information that comes from Cisco Unified Communications Manager.

ID Critical—The SIP Trunk includes the header Privacy:id;critical, which means that presentation is restricted for both name and number.

The critical label means that privacy services that are requested for this message are critical, and if the network cannot provide these privacy services, this request should be rejected.

This value overrides the presentation information that comes from Cisco Unified Communications Manager.

Note These headers are sent only if the Asserted Identity check box is checked.

Significant Digits

Represents the number of final digits that are retained on inbound calls. It is used for the processing of incoming calls and to indicate the number of digits that are used to route calls that are coming in to the H.323 device.

Select the number of significant digits to collect (0 to 32). Cisco Unified Communications Manager counts significant digits from the right (last digit) of the number that is called.

Connected Party ID Presentation

Cisco Unified Communications Manager uses connected line ID presentation (COLP) as a supplementary service to provide the calling party with the connected party number. The SIP Trunk level configuration takes precedence over the call-by-call configuration.

The default value is Default, which translates to Allowed. Select Default if you want Cisco Unified Communications Manager to send connected line information.

Select Restricted if you do not want Cisco Unified Communications Manager to send connected line information.

Connected Name Presentation

Cisco Unified Communications Manager uses connected name ID presentation (CONP) as a supplementary service to provide the calling party with the connected party name. The SIP Trunk level configuration takes precedence over the call-by-call configuration.

The default value is Default, which translates to Allowed. Select Default if you want Cisco Unified Communications Manager to send connected name information.

Select Restricted if you do not want Cisco Unified Communications Manager to send connected name information.

Calling Search Space

Available calling search spaces.

AAR Calling Search Space

Automated alternate routing (AAR) calling search space. Specifies the collection of route partitions that are searched to determine how to route a collected (originating) number that is otherwise blocked due to insufficient bandwidth.

Prefix DN

The prefix digits that are appended to the called party number on incoming calls.

Redirecting Diversion Header Delivery - Inbound

Select Yes (the default) to accept the Redirecting Number in the incoming invite message to the Cisco Unified Communications Manager.

Select No to exclude the Redirecting Number in the incoming invite message to the Cisco Unified Communications Manager.

You use Redirecting Number for voice messaging integration only. If your configured voice messaging system supports Redirecting Number, you should select Yes.

Called Party Transformation CSS

Allows you to localize the called party number on the device. The Called Party Transformation CSS that you choose must contain the called party transformation pattern that you want to assign to this device.

If you configure the Called Party Transformation CSS as None, the transformation does not match and is not applied. Make sure that you configure the Called Party Transformation CSS in a non-null partition that is not used for routing.

Use Device Pool Called Party Transformation CSS

Select Yes to use the Called Party Transformation CSS that is configured in the device pool that is assigned to this device.

If you select No, the device uses the Called Party Transformation CSS that was configured for the device in the Trunk Configuration page.

Calling Party Transformation CSS

Enables you to localize the calling party number on the device. Make sure that the Calling Party Transformation CSS that you choose contains the calling party transformation pattern that you want to assign to this device.

Before the call occurs, the device must apply the transformation by using digit analysis. If you configure the Calling Party Transformation CSS as None, the transformation will not match and will not be applied.

Make sure that you configure the Calling Party Transformation Pattern in a non-null partition that is not used for routing.

Calling Party Selection

Select the directory number that is sent on an outbound call on a gateway.

The following options specify which directory number is sent:

Originator—Send the directory number of the calling device.

First Redirect Number—Sends the directory number of the redirecting device.

Last Redirect Number—Sends the directory number of the last device to redirect the call.

First Redirect Number (External)—Sends the external directory number of the redirecting device.

Last Redirect Number (External)—Sends the external directory number of the last device to redirect the call.

Calling Line ID Presentation

Cisco Unified Communications Manager uses calling line ID presentation (CLIP) as a supplementary service to control the display of the calling party number on the called party phone display screen.

Select one of the following options:

Default—If you do not want to change the presentation setting.

Allowed—If you want the calling number information to be displayed.

Restricted—If you do not want the calling number information to be displayed.

Calling Name Presentation

Cisco Unified Communications Manager uses calling name ID presentation (CNIP) as a supplementary service to provide the calling party name. The SIP Trunk level configuration takes precedence over the call-by-call configuration.

Select one of the following options:

Default—If you do not want to change the presentation setting.

Allowed—If you want Cisco Unified Communications Manager to send calling name information.

Restricted—If you do not want Cisco Unified Communications Manager to send the calling name information.

Caller ID DN

Enter the pattern (0 to 24 digits) that you want to use to format the caller ID on outbound calls from the trunk.

For example (in North America):

555XXXX—Variable Caller ID, where X represents an extension number. The central office appends the number with the area code if it is not specified.

5555000—Fixed Caller ID. Use this form when you want the corporate number to be sent instead of the exact extension from which the call is placed. The central office appends the number with the area code if it is not specified.

Caller Name

Enter a caller name to override the caller name that is received from the originating SIP device.

Redirecting Diversion Header Delivery - Outbound

If Yes is selected, the redirecting number is included in the outgoing invite message from Cisco Unified Communications Manager to indicate the original called party number and the redirecting reason for the call when the call is forwarded.

If No is selected, the first redirecting number and the redirecting reason are excluded from the outgoing invite message.

The redirecting number is used for voice messaging integration only. If your configured voice messaging system supports redirecting Number, you should select Yes.

Destination Address

The remote SIP peer with which this trunk will communicate. The allowed values for this field are a valid V4 IP address, a fully qualified Domain name, or a DNS SRV record (applies only if yes is selected in the Destination Address is an SRV field).

SIP trunks only accept incoming requests from the configured destination address and the incoming port that is specified in the SIP Trunk Security Profile that is associated with this trunk.

If the remote end is a Cisco Unified Communications Manager cluster, DNS SRV represents the recommended choice for this field. The DNS SRV record should include all Cisco Unified Communications Managers within the cluster.

Destination Address is an SRV

Specifies that the configured Destination Address is an SRV record.

Destination Port

Enter the destination port. Make sure that the value you enter specifies a port between 1024 and 65535 (the default value is 5060).

You can specify the same port number for multiple trunks.

Do not enter a value if the destination address is a DNS SRV port. The default port number 5060 indicates a SIP port.

Geolocation

An unspecified geolocation, which designates that this device does not associate with a geolocation. You can also select a geolocation that has been configured.

Geolocation Filter

Specifies the geolocation filter for the device.

Incoming Port

Incoming port number.

Outgoing Transport Type

Outgoing transport type (TCP or UDP).

MTP Preferred Originating Codec

Indicates the preferred outgoing codec.

To configure G.79 codecs for use with a SIP Trunk, you must use a hardware MTP or transcoder that supports the G.79 codec.

Send Geolocation Information

Sends the geolocation information for the associated device.

SIP Trunk Security Profile

Select the security profile to apply to the SIP Trunk.

You must apply a security profile to all SIP trunks that are configured in Cisco Unified Communications Manager Administration.

Installing Cisco Unified Communications Manager provides a predefined, nonsecure SIP Trunk security profile for autoregistration.

To enable security features for a SIP Trunk, configure a new security profile and apply it to the SIP Trunk. If the trunk does not support security, choose a nonsecure profile.

To identify the settings that the profile contains, on Cisco Unified Communications Manager choose System > Security Profile > SIP Trunk Security Profile.

For information on how to configure security profiles, see Cisco Unified Communications Manager Security Guide.

Rerouting Calling Search Space

Determines where a SIP user (A) can refer another user (B) to a third party (C). After the referral is completed, B and C connect. In this case, the rerouting calling search space that is used is that of the initial SIP user (A).

Out-Of-Dialog Refer Calling Search Space

Used when a Cisco Unified Communications Manager refers a call (B) coming in to a SIP user (A) to a third party (C) when there is no involvement of a SIP user (A). In this case, the system uses the out-of-dialog calling search space of the SIP user (A).

Packet Capture Mode

Exists only for troubleshooting encryption. Packet capturing may cause high CPU usage or call-processing interruptions.

Select one of the following options:

None—This option, which is the default setting, indicates that no packet capturing is occurring. After you complete packet capturing, configure this setting.

Batch Processing Mode—Cisco Unified Communications Manager writes the decrypted or non encrypted messages to a file, and the system encrypts each file.

On a daily basis, the system creates a new file with a new encryption key. Cisco Unified Communications Manager, which stores the file for seven days, also stores the keys that encrypt the file in a secure location. Cisco Unified Communications Manager stores the file in the PktCap virtual directory.

A single file contains the time stamp, source IP address, source IP port, destination IP address, packet protocol, message length, and message.

The IREC tool uses HTTPS, administrator username and password, and the specified day to request a single encrypted file that contains the captured packets.

Likewise, the tool requests the key information to decrypt the encrypted file.

You do not have to reset the trunk after enabling/disabling Packet Capture.

Packet Capture Duration

Exists only for troubleshooting encryption. Packet capturing may cause high CPU usage or call-processing interruptions.

This field specifies the maximum number of minutes allotted for one session of packet capturing. The default setting is 0, and the range is from 0 to 300 minutes.

To initiate packet capturing, enter a value other than 0 in the field. After packet capturing completes, the value "0" is displayed.

Presence Group

Configures the Unified Presence features. Select a Presence group for the SIP trunk. The selected group specifies the destinations that the device/application/server that is connected to the SIP trunk can monitor.

The default value for Presence Group specifies Standard Presence group, which is configured with installation. Presence groups that are configured in Cisco Unified Communications Manager Administration also appear in the drop-down list box.

Presence authorization works with presence groups to allow or block presence requests between groups.

PSTN Access

Indicates that the calls made through this trunk might reach the PSTN. Check this check box even if all calls through this trunk device do not reach the PSTN.

For example, check this check box for tandem trunks or an H.323 gatekeeper-routed trunk if calls might go to the PSTN.

When checked, this check box causes the system to create upload voice call records (VCRs) to validate calls made through this trunk device.

By default, this check box remains checked.

Route Class Signaling Enabled

From the drop-down list, enable or disable route class signaling for the port.

Select one of the following values:

Default—If you choose this value, the device uses the setting from the Route Class Signaling service parameter.

Off—Choose this value to enable route class signaling. This setting overrides the Route Class Signaling service parameter.

On—Choose this value to disable route class signaling. This setting overrides the Route Class Signaling service parameter.

Route class signaling communicates special routing or termination requirements to receiving devices. It must be enabled for the port to support the Hotline feature.

SUBSCRIBE Calling Search Space

Determines how Cisco Unified Communications Manager routes presence requests from the device, server, or application that connects to the SIP Trunk.

This setting allows you to apply a calling search space separate from the call-processing search space for presence (SUBSCRIBE) requests for the SIP Trunk.

Select a SUBSCRIBE calling search space to use for presence requests for the SIP Trunk. All calling search spaces that you configure in Cisco Unified Communications Manager Administration appear in the SUBSCRIBE Calling Search Space drop-down list box.

If you do not select a different calling search space for the SIP Trunk from the drop-down list, the SUBSCRIBE calling search space defaults to None.

To configure a SUBSCRIBE calling search space specifically for this purpose, you can configure a calling search space as you do all calling search spaces.

SIP Profile

Select the SIP profile that is to be used for this SIP Trunk.

Trunk Service Type

Specifies the type of the Trunk Service. Select one of the following options:

None—Select this option if the trunk will not be used for call control discovery, Extension Mobility Cross Cluster, or Cisco Intercompany Media Engine.

Call Control Discovery—Selecting this option enables the trunk to support call control discovery.

If you assign this trunk to the CCD advertising service in the Advertising Service window, the trunk handles inbound calls from remote call-control entities that use the SAF network.

If you assign this trunk to the CCD requesting service in the Requesting Service window, the trunk handles outgoing calls to learned patterns.

Extension Mobility Cross Cluster—Select this option to enable the trunk to support the Extension Mobility Cross Cluster (EMCC) feature.

Choosing this option causes the following settings to remain blank or unchecked and become unavailable for configuration, thus retaining their default values: Media Termination Point Required, Unattended Port, Destination Address, Destination Address IPv6, and Destination Address is an SRV.

Cisco Intercompany Media Engine—Ensure that the Cisco IME server is installed and available before you configure this field.

Transmit UTF-8 for Calling Party Name

Specifies the user locale setting of the device pool to determine whether to send unicode and whether to translate received Unicode information.

For the sending device, if you check this check box and the user locale setting in the device pool matches the terminating phone user locale, the device sends unicode. If the user locale settings do not match, the device sends ASCII.

The receiving device translates incoming unicode characters based on the user locale setting of the sending device pool. If the user locale setting matches the terminating phone user locale, the phone displays the characters.

The default value for Transmit UTF-8 for Calling Party Name leaves the check box unchecked.

Use Device Pool Connected Party Transformation CSS

Enables you to use the Connected Party Transformation CSS that is configured in the device pool that is assigned to this device.

If you do not check this check box, the device uses the Connected Party Transformation CSS that you configured for this device in the Trunk Configuration window.

DTMF Signaling Method

Select one of the following options:

No Preference (default)—Cisco Unified Communications Manager will pick the DTMF method to negotiate DTMF, so an MTP is not required for the call.

If Cisco Unified Communications Manager does not have a choice but to allocate an MTP (if the Media Termination Point Required check box is checked), SIP Trunk will negotiate DTMF to RFC 2833.

RFC 2833—Select this configuration if the preferred DTMF method to be used across the trunk is RFC 2833. Cisco Unified Communications Manager makes every effort to negotiate RFC 2833 regardless of MTP usage. Out-of-band provides the fallback method if the peer endpoint supports it.

OOB and RFC 2833—Select this configuration if both out-of-band and RFC 2833 should be used for DTMF.

If the peer endpoint supports both out-of-band and RFC 2833, Cisco Unified Communications Manager will negotiate both out-of-band and RFC 2833 DTMF methods.

As a result, two DTMF events are sent for the same DTMF keypress (one out-of-band and the other RFC 2833).



Note You can provision SIP Trunk infrastructure data objects in Session Management Edition (SME) devices if you add the SME device as a Call Processor in Provisioning.


SIP Profile Infrastructure Data Object Fields

Table B-24 SIP Profile Infrastructure Data Object Fields 

Field
Description

Name

Name of the SIP profile.

Description

Description of the SIP profile.

Default MTP Telephony Event Payload Type

Specifies the default payload type for RFC2833 telephony event.

Resource Priority Namespace List

Select a configured Resource Priority Namespace Network Domain list.

Early Offer for G Clear Calls

It supports both standards-based G.Clear (CLEARMODE) and proprietary Cisco Session Description Protocols (SDP).

SDP Session-level Bandwidth Modifier

Bandwidth needed when all the media streams are in use. There are three session level bandwidth modifiers: Transport Independent Application Specific (TIAS), Application Specific (AS), and Conference Total (CT).

Select one of the following options to specify which session level bandwidth modifier to include in the SDP portion of SIP Early Offer or Reinvite requests.

TIAS and AS

TIAS only

AS only

CT only

This option is supported only for Cisco Unified Communications Manager 8.6.2 and above.

User-Agent and Server header information

This feature indicates how Cisco Unified Communications Manager handles the User-Agent and Server header information in a SIP message.

Choose one of the following options:

Send Unified CM Version Information as User-Agent Header—For INVITE requests, the User-Agent header is included with the CM version header information. For responses, the Server header is omitted. Cisco Unified Communications Manager passes through any contact headers untouched. This is the default behavior.

Pass Through Received Information as Contact Header Parameters —If this option is selected, the User-Agent/Server header information is passed as Contact header parameters. The User-Agent/Server header is derived from the received Contact header parameters, if present. Otherwise, they are taken from the received User-Agent/Server headers.

Pass Through Received Information as User-Agent and Server Header—If this option is selected, the User-Agent/Server header information is passed as User-Agent/Server headers. The User-Agent/Server header is derived from the received Contact header parameters, if present. Otherwise, they are taken from the received User-Agent/Server headers.

Supported only for Cisco Unified Communications Manager 8.6.2 and above.

Accept Audio Codec Preferences in Received Offer

Select On to enable Cisco Unified Communications Manager to honor the preference of audio codecs in a received offer and preserve it while processing.

Select Off to enable Cisco Unified Communications Manager to ignore the preference of audio codecs in a received offer and apply the locally configured Audio Codec Preference List.

Dial String Interpretation

Cisco Unified Communications Manager uses the Dial String Interpretation policy to determine if the SIP identity header is a directory number or directory URI.

Because directory numbers and directory URIs are saved in different database lookup tables, Cisco Unified Communications Manager examines the characters in the SIP identity header's user portion, which is the portion of the SIP address that is before the @ sign (for example, user@IP address or user@domain).

To configure Dial String Interpretation, choose one of the following options from the drop-down list box:

Always treat all dial strings as URI addresses—Cisco Unified Communications Manager treats the address of an incoming call as if it were a URI address.

Phone number consists of characters 0-9, A-D, *, and + (others treated as URI addresses)—Cisco Unified Communications Manager treats the incoming call as a directory number if all the characters in the user portion of the SIP identity header fall within this range. If the user portion of the address uses any characters that do not fall within this range, the address is treated as a URI.

Phone number consists of characters 0-9, *, and + (others treated as URI addresses)—Cisco Unified Communications Manager treats the incoming call as a directory number if all the characters in the user portion of the SIP identity header fall within this range. If the user portion of the address uses any characters that do not fall within this range, the address is treated as a URI.

Note If the user=phone tag is present in the Request URI, Cisco Unified Communications Manager always treats the dial string as a number regardless of what option you choose for the Dial String Interpretation field.

Redirect by Application

Check this check box to configure this SIP Profile on the SIP trunk, which allows the Cisco Unified Communications Manager administrator to:

Apply a specific calling search space to redirected contacts that are received in the 3xx response.

Apply digit analysis to the redirected contacts to make sure that the call is routed correctly.

Prevent DOS attack by limiting the number of redirections (recursive redirections) that a service parameter can set.

Allow other features to be invoked while the redirection is taking place.

Disable Early Media on 180

Check this check box to play local ringback on the calling phone and connect the media upon receipt of the 2000K response.

Outgoing T.38 INVITE include audio mline

Allows the system to accept a signal from Microsoft Exchange that causes it to switch the call from audio to T.38 fax. To use this feature, you must configure a SIP trunk with this SIP profile.

Enable ANAT

This option allows a dual-stack SIP trunk to offer both IPv4 and IPv6 media.

Assured Services SIP conformance

This checkbox should be checked for third-party AS-SIP endpoints as well as AS-SIP trunks to ensure proper Assured Service behavior. This setting provides specific Assured Service behavior that affects services such as Conference factory and SRTP.

MLPP User Authorization

Check this box to enable MLPP User Authorization. MLPP User Authorization requires the phone to send in an MLPP username and password.

Timer Invite Expires

The time, in seconds, after which a SIP INVITE expires.

Timer Register Delta

Specifies the parameter is in conjunction with the Timer Register Expires setting. The phone re-reregisters Timer Register Delta seconds before the registration period ends. The registration period is determined by the value of the SIP Station KeepAlive Interval service parameter.

Timer Register Expires

The value that the phone that is running SIP sends in the Expires header of the REGISTER message. Valid values include any positive number; however, 3600 (1 hour) is the default value.

In the 200OK response to REGISTER, Cisco Unified Communications Manager will include an Expires header with the configured value of the SIP Station KeepAlive Interval service parameter.

This value in the 2000K determines the time, in seconds, after which the registration expires. The phone refreshes the registration Timer Register Delta seconds before the end of this interval.

Timer T1

The lowest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number.

Timer T2

The highest value, in milliseconds, of the retransmission timer for SIP messages. Valid values include any positive number.

Retry INVITE

The maximum number of times that an INVITE request will be transmitted. Valid values include any positive number.

Retry Non-INVITE

The maximum number of times that an INVITE request will be retransmitted. Valid values include any positive number.

Start Media Port

The start real-time protocol (RTP) port for media. The ranges is from 16384 to 32767.

Stop Media Port

The stop real-time protocol (RTP) port for media. The ranges is from 16384 to 32767.

Call Pickup URL

Specifies a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call pickup feature.

Call Pickup Group Other URI

Specifies a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call pickup group other feature.

Call Pickup Group URI

Specifies a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call pickup group feature.

Meet Me Service URI

Specifies a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the meet me conference feature.

User Info

Configures the user = parameter in the REGISTER message.

DTMF DB Level

Specifies in-band DTMF digit tone level.

Call Hold Ring Back

Allows the system to ring to let you know that you still have another party on hold.

Anonymous Call Block

Configures anonymous call block.

Caller ID Blocking

Configures the caller ID blocking.

Do No Disturb Control

Enables the Do Not Disturb feature.

Telnet Level for 7940 and 7960

Controls the telnet level configuration parameter for phones that support Telnet.

Timer Keep Alive Expires

Specifies the interval between keepalive messages that are sent to the backup Cisco Unified Communications Manager to ensure that it is available in the event that a failover is required.

Timer Subscribe Expires

Specifies the time, in seconds, after which a subscription expires. This value is inserted into the Expires header field.

Timer Subscribe Delta

Resubscribes Timer Subscribe Delta seconds before the subscription period ends, as governed by Timer Subscribe Expires.

Maximum Redirections

Specifies the maximum number of times that the phone will allow a call to be redirected before dropping the call.

Off Hook To First Digit Timer

Specifies the time, in microseconds, that passes when the phone goes off hook and the first digit timer is set. The range is from 0 - 150,000 microseconds.

Call Forward URI

Specifies a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the call forward feature.

Abbreviated Dial URI

Specifies a unique address that the phone that is running SIP will send to Cisco Unified Communications Manager to invoke the abbreviated dial feature.

Conference Join Enabled

Specifies whether the Cisco Unified IP Phones 7940 or 7960, when the conference initiator that is using that phone hangs up, should attempt to join the remaining conference attendees.

RFC 2543 Hold

Specifies whether to enable setting connection address to 0.0.0.0 per RFC2543 when call hold is signaled to Cisco Unified Communications Manager. This allows backward compatibility with endpoints that do not support RFC3264.

Semi Attended Transfer

Specifies whether the Cisco Unified IP Phones 7940 or 7960 caller can transfer the second leg of an attended transfer while the call is ringing. Check the check box if you want semi attended transfer enabled; leave it unchecked if you want semi attended transfer disabled.

Enable VAD

Specifies whether you want voice activation detection (VAD) enabled; leave it unchecked if you want VAD disabled. When VAD is enabled, no media are transmitted when voice is detected.

Stutter Message Waiting

Specifies whether you want stutter dial tone when the phone goes off hook and a message is waiting; leave unchecked if you do not want a stutter dial tone when a message is waiting.

Incoming Requests FROM URI Settings

Caller ID DN

Enter the pattern that you want to use for calling line ID, from 0 to 24 digits. For example, in North America:

555XXXX = Variable calling line ID, where X equals an extension number. The CO appends the number with the area code if you do not specify it.

55000 = Fixed calling line ID, where you want the Corporate number to be sent instead of the exact extension from which the call is placed. The CO appends the number with the area code if you do not specify it.

You can also enter the international escape character +.

Caller Name

Enter a caller name to override the caller name that is received from the originating SIP device.

Trunk Specific Configuration

Reroute Incoming Request to new Trunk based on

Specifies the method that Cisco Unified Communications Manager uses to identify the SIP trunk where the call is rerouted.

RSVP Over SIP

Specifies the method that Cisco Unified Communications Manager uses to configure RSVP over SIP trunks.

Fall back to local RSVP

Allows failed end-to-end RSVP calls to fall back to local RSVP to establish the call.

SIP Rel1XX Options

Configures SIP Rel1XX, which determines whether all SIP provisional responses (other than 100 Trying messages) are sent reliably to the remote SIP endpoint.

Video Call Traffic Class

Video Call Traffic Class determines the type of video endpoint or trunk that the SIP profile is associated with. From the drop-down list box, select one of the following options

Immersive—High-definition immersive video.

Desktop—Standard desktop video.

Mixed—A mix of immersive and desktop video.

Cisco Unified Communications Manager Locations Call Admission Control (CAC) reserves bandwidth from two Locations video bandwidth pools: Video Bandwidth and Immersive Bandwidth, depending on the type of call as determined by the Video Call Traffic Class. See the "Call Admission Control" chapter of the Cisco Unified Communications Manager System Guide for more information.

Calling Line Identification Presentation

Select Strict From URI presentation Only to select network-provided identity.

Select Strict Identity Headers presentation Only to select user-provided identity.

Deliver Conference Bridge Identifier

Check this check box for the SIP trunk to pass the b-number that identifies the conference bridge across the trunk instead of changing the b-number to the null value.

The terminating side does not require that this field be enabled.

Checking this check box is not required for Open Recording Architecture (ORA) SIP header enhancements to the Recording feature to work.

Enabling this check box allows the recorder to coordinate recording sessions where the parties are participating in a conference.

This option is supported only for Cisco Unified Communications Manager 8.5 and above.

Early Offer support for voice and video calls (insert MTP if needed)

Check this check box if you want to create a trunk that supports early offer.

This option is supported only for Cisco Unified Communications Manager 8.5 and above.

Send send-receive SDP in mid-call INVITE

Check this check box to prevent Cisco Unified Communications Manager from sending an INVITE a=inactive SDP message during call hold or media break during supplementary services.

This option is supported only for Cisco Unified Communications Manager 8.5 and above.

Allow Presentation Sharing using BFCP

If this box is checked, Cisco Unified Communications Manager is configured to allow supported SIP endpoints to use the Binary Floor Control Protocol to enable presentation sharing.

Allow iX Application

Check this check box to enable support for iX media channel.

Allow Passthrough of Configured Line Device Caller Information

Check this box to allow passthrough of configured line device caller information from the SIP trunk.

Reject Anonymous Incoming Calls

Check this box to reject anonymous incoming calls.

Reject Anonymous Outgoing Calls

Check this box to reject anonymous outgoing calls.

SIP OPTIONS Ping

Enable OPTIONS Ping to monitor destination status for Trunks with service type "None (Default)"

Check this check box if you want to enable the SIP OPTIONS feature.

Supported only for Cisco Unified Communications Manager 8.5 and above.

Ping Interval for In-service and Partially In-service Trunks (seconds)

This field configures the time duration between SIP OPTIONS requests when the remote peer is responding and the trunk is marked as In Service.

Supported only for Cisco Unified Communications Manager 8.5 and above versions.

Ping Interval for Out-of-service SIP Trunks (seconds)

This field configures the time duration between SIP OPTIONS requests when the remote peer is not responding and the trunk is marked as Out of Service.

Supported only for Cisco Unified Communications Manager 8.5 and above versions.

Ping Retry Timer (milliseconds)

This field specifies the maximum waiting time before retransmitting the OPTIONS request.

Supported only for Cisco Unified Communications Manager 8.5 and above versions.

Ping Retry Count

This field specifies the number of times that Cisco Unified Communications Manager resends the OPTIONS request to the remote peer.

Supported only for Cisco Unified Communications Manager 8.5 and above versions.


Translation Pattern Infrastructure Data Object Fields

Table B-25 Translation Pattern Infrastructure Data Object Fields 

Field
Description

Translation Pattern

Translation pattern, including numbers and wildcards.

Route Partition

Available route partitions.

Description

Optional description.

Dial Plan

Numbering plan.

Route Filter

Optional route filter.

MLPP Precedence

Multilevel Precedence and Preemption (MLPP) precedence settings.

Call Search Space

Available calling search spaces.

Block Enabled

Enables or disables block.

Release Cause

Dependent on the Block Enabled field. If a release cause is selected, then Block Enabled must be set to True.

Use Calling Party's External Phone Number Mask

Determines whether or not to use the calling party's external phone number mask.

Calling Party Transform Mask

Transformation mask value.

Calling Party Prefix Digits (Outgoing Calls)

Prefix digits.

Calling Line ID Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the calling party's phone number on the called party's phone display for this translation pattern.

Calling Name Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the calling party's name on the called party's phone display for this translation pattern.

Connected Line ID Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the connected party's phone number on the calling party's phone display for this translation pattern.

Connected Name Presentation

Determines whether you want Cisco Unified Communications Manager to allow or restrict the display of the connected party's name on the calling party's phone display for this translation pattern.

Called Party Discard Digits

The discard digits instructions that you want to be associated with this translation pattern.

Called Party Transform Mask

Transformation mask value.


Unified Call Manager Group Infrastructure Data Object Fields

Table B-26 Unified Call Manager Group Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Unified CMs

List of available Cisco Unified Communications Managers.

Auto-Registration Unified CM Group

Select Yes if you want this Cisco Unified Communications Manager group to be the default Cisco Unified Communications Manager group when auto-registration is enabled.


UC Service Infrastructure Data Object Fields

Table B-27 UC Service Infrastructure Data Object Fields 

Field
Description

Voicemail

Product Type

Select a product type. Available options are Unity and Unity Connection.

Default setting: Unity.

Name

Enter the name of the voicemail service. Ideally, the voicemail service name should be descriptive enough for you to instantly recognize it.

Description

(Optional) Enter a description that helps you to distinguish between voicemail services. You can change the description if required.

Maximum characters: 100.

Hostname/IP Address

Enter the address of the voicemail service in one of the following forms:

Hostname

IP address

Fully qualified domain name (FQDN)

This field value must exactly match the hostname, IP address, or FQDN of the associated voicemail service. If the hostname or IP address of the voicemail service changes, change this field value accordingly.

Port

Enter the port to connect with the voicemail service.

This field value must match the available port on the voicemail service. Change the port number only if it conflicts with other services.

Default port: 443

Protocol

Select the protocol to route voicemail messages securely.

Available options: HTTP, HTTPS

Tip Cisco recommends that you use HTTPS as the voicemail transport protocol for Cisco Unity and Cisco Unity Connection servers. Only change to HTTP if your network configuration does not support HTTPS.

Mailstore

UC Service Type

Specifies the UC service type as Mailstore.

Product Type

Specifies the product type as Exchange.

Name

Enter the name of the mailstore service. Ideally, the mailstore service name should be descriptive enough for you to instantly recognize it.

Maximum characters: 50 (ASCII only).

Description

(Optional) Enter a description that helps you to distinguish between mailstore services. You can change the description if required.

Hostname/IP Address

Enter the address of the mailstore service in one of the following

forms:

Hostname

IP address

FQDN

This field value must exactly match the hostname, IP address, or FQDN of the associated mailstore service. If the address of the mailstore service changes, change this field value accordingly.

Cisco Unity creates subscriber mailboxes for message storage on the Microsoft Exchange server.

Note Cisco Unity Connection usually provides a mailstore service, and hosts themailstore service on the same server.

Port

Specify the port number configured for the service.

Default Port: 143

Allowed Values: 1 - 65535

Note For secure voice messaging with Cisco Unity Connection, use port 7993.

Note This value must match the available port on the mailstore service. Change the port number only if it conflicts with other services.

Protocol

Select the corresponding protocol to use when Cisco Jabber clients contact this service.

Available Options: TCP, SSL, TLS, UDP

Default Setting: TCP, which is the most commonly used network configuration. Change this setting to suit your deployment,

Unified CM settings, and security needs.

Note For secure voice messaging with Cisco Unity Connection, use TLS.

Directory

Service Type

Specifies directory as the UC service type.

Product Type

Select a supported directory product type from this list that applies to your network configuration.

Available Options: Directory, Enhanced Directory

Default Setting: Directory

Name

Enter the name of the directory service. Ideally, the directory service name should be descriptive enough for you to instantly recognize it.

Maximum characters: 50 (ASCII only).

Allowed values: All characters allowed except quotes ("), angle brackets (< >), backslash (\), ampersand (&), and percent (%).

Description

(Optional) Enter a description that helps you to distinguish between directory services. You can change the description if required.

Allowed values: All characters allowed except quotes ("), angle brackets (< >), backslash (\), ampersand (&), and percent (%).

Hostname/IP Address

Enter the address of the directory service in one of the following forms:

Hostname

IP address

FQDN

This field must exactly match the hostname, IP address, or FQDN of the associated directory service. If the address of the directory service changes, change this field value accordingly.

Allowed values: Allowed characters include alphanumeric (a-z, A-Z, 0-9), period (.), backslash (\), dash (-), and underscore (_).

Port

Enter the port for the directory service.

Default Port: 389

Allowed Values: 1- 65535

This value must match the available port on the directory service.

Note Change the port number only if it conflicts with other services.

Protocol

Select the protocol to route communications between the directory service and Cisco Jabber clients.

Available Options: TCP, UDP, TLS

Default Setting: TCP. This is the most commonly used network configuration. Change this setting to suit your network configuration, Unified CM settings, and security needs.

Connection Type

Specifies the directory server type to connect to Global Catalog server that is optimized for searching or a Domain Controller (or any server running an Ldap service) which may not be optimized for searching.

This is a required field.

Default: Global Catalog server

Use Secure Connection

Define whether to send credentials in clear text (default is not to send in clear text i.e. use a secure connection).

This is a required field.

Default: True

Use Wildcards

Use wildcards when doing number lookups.

This is a required field.

Default: False

Disable Secondary Number Lookups

Disables queries using home, mobile and other numbers.

This is a required field.

Default: False

Uri Prefix

Specify the Uri scheme name e.g. 'im:' or 'sip:'

Maximum length: 32

Phone Number Masks

Allows a mask to be defined which can be used when doing resolution by telephone number. E.g. the mask +353|+(###) ## ## #### could be used to resolve numbers in the format +35311221234 to +(353) 11 22 1234. Multiple masks can be defined by using the '|' operator. For example: +353|+(###) ## ## ####|+44|+44 (## )## #####) ## ## #### could be used to resolve numbers in the format +35311221234 to +(353) 11 22 1234.

Maximum length: 1024

IM and Presence

Service Type

Specifies IM and Presence as the UC service type.

Product Type

Select a supported IM and Presence product type from this list that applies to your network configuration.

Available options: Unified CM (IM and Presence), WebEx (IM and Presence)

Default setting: Unified CM (IM and Presence)

Name

Enter the name of the IM and Presence service. Ideally, the IM and Presence service name should be descriptive enough for you to recognize it instantly.

Maximum characters: 50 (ASCII only).

Description

(Optional) Enter a description that helps you to distinguish between IM and Presence services. You can change the description if required.

Hostname/IP Address

Enter the address of the IM and Presence service in one of the following forms:

Hostname

IP address

DNS SRV

Allowed values: Allowed characters include alphanumeric (a-z, A-Z, 0-9), period (.), backslash (\), dash (-), and underscore(_).

Note This field value must exactly match the hostname, IP address, or DNS SRV of the associated IM and Presence service. If the address of the IM and Presence service changes, change this field value accordingly.

Tip Cisco recommends DNS SRV to help the client find the correct IM and Presence service for the user.

Conferencing

UC Service Type

Specifies conferencing as the UC service type.

Product Type

Select a product type that applies to your network configuration.

Available Options: MeetingPlace Classic, MeetingPlace Express, WebEx

Name

Enter the name of the coferencing service. Ideally, the service name should be descriptive enough for you to instantly recognize it.

Maximum characters: 50 (ASCII only).

Description

(Optional) Enter a description that helps you to distinguish between directory services. You can change the description if required.

Hostname/IP Address

Enter the address of the conferencing service in one of the following

forms:

Hostname

IP address

FQDN

This field must exactly match the hostname, IP address, or FQDN of the associated directory service. If the address of the directory service changes, change this field value accordingly.

Allowed values: Alphanumeric (a-z, A-Z, 0-9), period (.), backslash (\), dash (-), and underscore (_).

Port

Enter the port for the conferencing service so that users can contact the service when they sign in to web conferences.

Default Port: 80

Allowed Values: 1- 65535

Note Use port 80 for HTTP and port 443 for HTTPS communications.

This value must match the available port on the conferencing service. Change the port number only if it conflicts with other services.

Protocol

Select the protocol to route web conference communications.

Available Options: HTTP, HTTPS

Default Setting: HTTP.

CTI

Service Type

Specifies CTI as the UC service type.

Product Type

Specifies CTI as the product type.

Name

Enter the name of the CTI service. Ideally, the CTI service name should be descriptive enough for you to instantly recognize it.

Maximum characters: 50 (ASCII only).

Description

(Optional) Enter a description that helps you to distinguish between CTI services when you have more than one configured. You can change the description if required.

Hostname/IP Address

Enter the address of the CTI service in one of the following forms:

Hostname

IP address

FQDN

This field must exactly match the hostname, IP address, or FQDN of the associated CTI service. If the address of the CTI service changes, change this field value accordingly.

Port

Enter the port for the CTI service.

Default port: 2748

Allowed ports: 1-65535

Note This value must match the available port on the CTI service.

Tip Change the port number only if it conflicts with other services.

Protocol

Specifies TCP as the default protocol.


Voice Region Infrastructure Data Object Fields

Table B-28 Voice Region Infrastructure Data Object Fields

Field
Description

Name

Object name.

Audio Codec

Codec setting.

For Cisco Unified Communications Manager higher versions (4.1 and above) , the Default Codec field is set to the option selected.


Voicemail Pilot Infrastructure Data Object Fields

Table B-29 Voicemail Pilot Infrastructure Data Object Fields 

Field
Description

Number

Voicemail pilot number.

Description

Optional description.

Calling Search Space

Available calling search spaces.

Is Default

Indicates whether this pilot number is the default Voice Mail Pilot for the system.


Voicemail Profile Infrastructure Data Object Fields

Table B-30 Voicemail Profile Infrastructure Data Object Fields 

Field
Description

Name

Object name.

Description

Optional description.

Voicemail Pilot

Available voicemail pilots.

Voicemail Box Mask

The mask that is used to format the voice mailbox number for auto-registered phones.

Is Default

Indicates whether this voicemail profile is the default for the system.


VG202, VG204, and VG 224 Infrastructure Data Object Fields

Table B-31 VG202, VG204, and VG 224 Infrastructure Data Object Fields 

Field
Description

Gateway Name

Name of the gateway.

Protocol

Protocol associated with the gateway.

MAC Address (Last 10 Characters)

MAC address of the selected device.

Updating the MAC Address field will update all associated phones' MAC addresses. However, to update MAC addresses in the subscriber records, you must perform a Domain synchronization.

Description

Description of the device.

Cisco Unified Communications Manager Group

The group of the Cisco Unified Communications Manager.

Module in Slot <Number>

Module that is in the slot number.

Subunit <Number>

Subunit's number.

Modem Passthrough

Enables or disables the modem passthrough.

Cisco Fax Relay

Enables and disables the Cisco fax relay.

T38 Fax Relay

Enables and disables the T-38 fax relay.

RTP Package Capability

Enables or disables RTP Package Capability.

MT Package Capability

Enables or disables MT Package Capability.

RES Package Capability

Enables or disables RES Package Capability.

PRE Package Capability

Enables or disables PRE Package Capability.

SST Package Capability

Enables or disables SST Package Capability.

RTP Unreachable OnOff

Enables or disables RTP unreachable timeout.

RTP Unreachable timeout (ms)

RTP unreachable timeout in milliseconds.

RTP Report Interval (secs)

RTP Report Interval in seconds.

Simple SDP

Enables or disables simple SDP.