Cisco Hosted Collaboration Solution, Release 11.5/12.5 Dial Plan Management for Cisco Unified Communications Domain Manager, Release 11.5(1)

Cisco Hosted Collaboration Solution Roles and Privileges

Depending on the role assigned, an administrator has the following dial plan privileges.

Note

Administrators can perform all tasks associated with their roles, as well as all dial plan tasks that are lower on the navigation hierarchy. Hierarchy is shown from left (highest) to right (lowest) in the table below.

Table 1. Cisco HCS Dial Plan Roles and Privileges
Tasks	HCS Admin	Provider / Reseller Admin	Customer Admin	Site Admin
Create a Customer Dial Plan	X (Customer level)	X (Customer level)	X (Customer level)
Create a Site Dial Plan	X (Site level)	X (Site level)	X (Site level)
Configure Class of Service	X (Site level)	X (Site level)	X (Site level)
Configure Short Code	X (Site level)	X (Site level)	X (Site level)	X
Configure Directory Number Routing	X (Site level)	X (Site level)	X (Site level)	X
Add Directory Numbers	X (Customer level)	X (Customer level)	X
View Directory Number Inventory	X (Site level)	X (Site level)	X (Site level)
Configure SIP Route Patterns	X (Site level)	X (Site level)	X (Site level)
Create Voice Mail Service	X (Provider/Reseller level)	X (Provider/Reseller level)
Associate Voice Mail Services to Customer	X (Customer level)	X (Customer level)
Define a Voice Mail Pilot Number	X (Customer level)	X (Customer level)	X (Customer level)
Associate Pilot Numbers to a Site	X (Site level)	X (Site level)	X (Site level)
Configure SIP Trunks	X	X	X
Reset SIP Trunks	X	X	X
Restart SIP Trunks	X	X	X
Configure Route Groups	X	X	X
Configure Route Lists	X (Customer or Site level)	X (Customer or Site level)	X
Configure Device Pools	X (Customer or Site level)	X (Customer or Site level)	X
Provision Emergency Calls	X
Create Schemas	X	X
Modify Site Defaults	X (Site level)	X (Site level)	X (Site level)
Assign Custom Schemas to Customers	X (Customer level)	X (Customer level)
Configure Unified CM Groups	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Regions	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Route Patterns	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Route Partitions	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Calling Search Spaces	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Translation Patterns	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Calling Party Transformation Patterns	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)
Configure Called Party Transformation Patterns	X (Customer or Site level)	X (Customer or Site level)	X (Customer or Site level)

Note

For more information on bulk loading and Cisco HCS Intelligent Loader, see Cisco Unified Communications Domain Manager Bulk Loader Provisioning Guide.

Create a Customer Dial Plan

This procedure determines the type of Cisco HCS dial plan schema (Type 1 to 4) to be used, depending on how you fill in the form.

To create a shell (Type 5) dial plan, refer to Cisco Hosted Collaboration Solution Release 12.5 Dial Plan Management Guide .

Note

You can have only one dial plan per customer. If you try to add a second dial plan, the dial plan will fail. Once you have created the customer dial plan, Enable CSS filtering is the only setting that you can modify.

Procedure

Step 1

Sign in as the customer administrator or the provider administrator. For a list of the roles and tasks that can be done at each level, see Cisco Hosted Collaboration Solution Roles and Privileges.

Step 2

Select Dial Plan Management > Customer > Dial Plan.

Step 3

Click Add to add a customer dial plan.

Step 4

Perform one of the following:

If a site location code is required for this customer, click the Site-Location Code (SLC) based dial plan? box, OR
If an SLC is not required, go to Step 8.

Step 5

Perform one of the following:

To add an extension prefix for the dial plan, click the Use extension prefix? box. Enter the extension prefix in the form and go to Step 8.
To add an ISP for the dial plan, click the Inter-Site Prefix required for inter-site dialing? box. Enter the inter-site prefix (ISP). The ISP can be one digit in length.

Step 6

If the ISP should be included in the directory number, click the Is ISP included in directory number? box. If not, go to Step 8.

Step 7

To include the ISP as part of the Voice Mail ID, click the Is ISP included in Voice Mail ID? box. If not, go to the next step.

Step 8

Check Enable CSS filtering to filter the calling search spaces available when configuring a subscriber, phone, or line to site-level class of service calling search spaces. Filtering is disabled by default, which results in all available Cisco Unified Communications Manager calling search spaces being available when configuring a subscriber, phone, or line.

Step 9

Click Save to add the customer dial plan you defined.

Note

The customer ID is a unique, auto-generated, read-only number allocated to the customer. The customer ID is particularly useful in shared deployments (where a cluster may be shared across multiple customers) to correlate specific elements to a customer. It appears in the Unified CM as a prefix to elements (for example, Cu2Si7 identifies Customer 2, Site 7).

Note

The Cisco HCS dial plan schemas are configured such that the customer-level dial plan elements are not pushed to the Unified CM until the first site for the customer is deployed. Therefore, you will not see any dial plan elements provisioned on the Unified CM until at least one site is deployed for the customer. See Create a Site Dial Plan.

Note

When adding lines (DNs) at the site level, you must remember to define your DNs appropriately. That is, you are responsible for using ISP+SLC+EXT if you deploy a type 2 dial plan. Otherwise your inter- or intra-site calls will not route. To define your directory numbers, refer to Add Directory Number Inventory.

Create a Site Dial Plan

A site dial plan does not get created automatically for a site when a site is created. Perform this procedure to associate a site dial plan with the site. After the first site for a specific customer is deployed, the customer-level dial plan elements are provisioned on Cisco Unified Communications Manager (Unified CM), followed by the site-specific dial plan elements. Each subsequent site only has site-specific dial plan elements to provision, so it takes less time to create. If there is more than one site for a customer, do not forget to apply the site dial plan to each site.

Note

Step 13 of this procedure takes a few minutes to provision the site dial plan, especially for the first site.

Note

Each site can have one site dial plan only.

Before you begin

A site dial plan cannot be created until a customer dial plan is created for the customer. There are attributes that are defined in the customer dial plan that are needed when creating a site dial plan.

Procedure

Step 1

Sign in as the customer administrator or provider administrator. For a list of the roles and tasks that can be done at each level, see Cisco Hosted Collaboration Solution Roles and Privileges.

Step 2

Set the hierarchy path to the site for which you want to create a site dial plan.

If the hierarchy path is not set to a site, you are prompted to select a site.

Step 3

Select Dial Plan Management > Site > Dial Plan.

Step 4

Click Add to add a site dial plan.

Step 5

Modify the External Breakout Number, if desired. The external breakout number is the PSTN prefix that is used when deploying a country dial plan. For Cisco HCS Type 1 to 4 dial plan schemas, you deploy country dial plans at the customer level. The country dial plan is not pushed to Unified CM until the first site associated with a given country is deployed. For example, if a site is associated with the United States, and it is the first site dial plan being created for the U.S., the U.S. country dial plan is deployed as part of creating the site's dial plan. The default is 9. The external breakout number is one digit in length.

Note

We support only one external breakout number for each country. For example, all sites within the US have the same external breakout as the first site within US.

Step 6

Enter the site location code using a maximum of eight digits. The SLC must be unique across sites for a customer.

Note

If the customer dial plan does not use SLCs, this field does not appear.

Step 7

Enter the extension length. Values can be 1 to 30 characters in length. Default is 4; for example, 2000.

Note

When adding DNs for a site, extension length is not currently enforced. Therefore, the administrator must be conscious of extension length when adding DNs for a particular site; otherwise DNs may not be dialable.

Step 8

Perform one of the following for sites without inter-site prefixes (ISPs).

Note

This field appears if your customer dial plan does not use ISPs. For example, HCS type 3 dial plans (SLC, no ISP, DN=SLC+EXT).

Check Use extension prefix? if your customer dial plan has an extension prefix defined and you want this site to use the extension prefix, OR
If an extension prefix is not defined in the customer dial plan for this site, go to the next step.

Step 9

Enter the area code. Enter zero or more valid local area codes for the site. Specify the length of the subscriber part of the PSTN number for each area code. The area code is used to generate the PSTN local route patterns for the site. For example, in the U.S., if area codes are added for Dallas, Texas, the area codes could be specified for local dialing as 214, 469, and 972 with a subscriber length of 7.

Step 10

Enter the local number length. The local number length is the length for the subscriber section of the entire E.164 number.

Step 11

Check Area Code used for Local Dialing if the area code is needed for local dialing from this site. In the U.S., this setting determines whether you use 7-digit or 10-digit local dialing.

Step 12

Select the published number from the drop-down of available E.164 inventory numbers, or enter a custom number.

The site published number is the default E.164 mask when a line is associated to a phone at a particular site.

Step 13

Select the emergency call-back number for the site from the drop-down of available E.164 inventory numbers, or enter a custom number.

The site emergency call-back number is the calling number when initiating an outgoing emergency call. It can be used when you use Extension Mobility and make an emergency call from a site other than your own. It can be used when the emergency call goes out to the PSTN network, when the system includes the site emergency number so that the origin of the call is known. The system adds this calling party transformation to the DN2DDI4Emer-PT partition.

Note

The emergency number is not the number to dial for an emergency. Rather, it is the number used to identify the calling party for emergency calls originating from a particular site.

Note

Under the Emergency Number field, there is the Site ID read-only field. The site ID is a unique, auto-generated, read-only number for each customer site that is prefixed to elements as an identifier (for example, Cu4Si2 indicates customer 4, site 2).

Step 14

Click Save to add the site dial plan you defined.

The site information is loaded on the Unified CM, and is identifiable by its customer ID-site ID prefix.

Update a Site Dial Plan

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Set the hierarchy path to the site for which you want to update the site dial plan.

Step 3

Select Dial Plan Management > Site > Dial Plan.

Step 4

Click the site dial plan you want to update.

Step 5

In the Dial Plan window, you can update the following fields:

Field	Description
Area Code	An area code associated with the site.
Local Number Length	The length of a locally dialed number for the specified area code.
Area Code Used for Local Dialing	Set this flag if the area code is included in locally dialed calls.
Published Number	The site published number is the default E.164 mask when a line is associated to a phone at a particular site.
Emergency Call Back Number	The site emergency call-back number is the calling number when initiating an outgoing emergency call.

Note

You can also add or delete area codes.

Step 6

Click Save.

Area Code Changes

For the Cisco type 1-4 dial plans, area code changes result in the affected local dialing translation patterns getting reapplied for the site. For new area codes, new translation patterns are deployed to the site based on the country dial plan schema associated with the site. Any translation patterns related to deleted area codes are undeployed from Unified CM based on the site's country dial plan schema. For updated area codes, related translation patterns are undeployed from Unified CM, then new translation patterns based on the updated area codes are deployed.

For the Cisco type 1-4 dial plan schema groups, area code changes generate LBO IOS area code events. If you change the area code for a site associated with one or more local SIP gateways, area code IOS commands are generated. If an area code is:

Added - The area code add IOS command is generated.
Deleted - The area code delete IOS command is generated if no other sites associated with the same SIP local gateway are using the deleted area code. If another site still references the same gateway's area code, the delete area code IOS command is not generated. This prevents invalidating the other site's local dialing behavior.
Updated - The area code delete and add IOS commands are generated as necessary based on the added and deleted logic.

Published Number Changes

If you changed the published number, the following site defaults are updated if they used the previous published number:

Default CUCM Phone Line E164 Mask
Default CUCM Device Profile Line E164 Mask
Line E164 Mask

If you changed the published number, then phone line masks, device profiles, and remote destination profiles that use the previous published number are updated. Any phone line masks, device profiles, and remote destination profiles that use a number other than the previous published number are not updated.

If you changed the published number, previously generated E164 IOS commands for a SIP local gateway associated with the site are automatically regenerated.

Emergency Call-Back Number Changes

If you have configured a type 1 to 4 dial plan, two calling party transformations are created automatically with the emergency call-back number. Changing the emergency call-back number updates the calling party mask in these calling party transformation patterns if it used the previous emergency call-back number:

"*{{ macro.HcsDpSiteId}}*!"
"*{{ macro.HcsDpSiteId}}*\+!"

If the calling party mask has been manually changed, the fields are untouched.

These calling party transformation patterns insert the emergency call-back number as the caller ID for any emergency calls placed from phones within the site.

What to do next

Apply any generated or regenerated IOS commands to your IOS gateway.

Configure Class of Service

Use this procedure to create a new calling search space (CSS) or edit an existing CSS that is tied to a site. The CSS can be used as a class of service (COS) for a device or line, or any of the other templates that rely on COS to filter different features.

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Warning

When adding class of service, ensure that you select a valid site under the customer in the hierarchy node breadcrumb at the top of the view. If you attempt to add a class of service at any other node in the hierarchy, an error appears stating that you must be at a site.

Step 2

Select Dial Plan Management > Site > Class of Service.

Note

There is one default internal calling line identification presentation (CLIP) class of service that appears in the list. The default COS is provisioned automatically based on the criteria you selected when you added the site.

Step 3

Perform one of the following:

To add a class of service, click Add.
To edit an existing class of service, choose the COS to be updated by clicking on its box in the far left column, then click Edit.
To clone an existing class of service, choose the COS to be cloned by clicking on its box in the far left column, then click Clone.

Step 4

Enter a unique name for the class of service in the Class of Service Name field. Try to make the name as descriptive as possible using up to 50 alphanumeric characters, including spaces, periods, hyphens (-), and underscore characters (_). You can also use macros that are available in the system to create a class of service name. For a list of possible macros, refer to Macros. Macros allow you to dynamically add site IDs, customer IDs, and other types of information to the CSS.

Example:

Cu1-24HrsCLIP-PT-{{macro.HcsDpSiteName}}

Note

The actual CSS that is sent to the Cisco Unified Communications Manager (based on the macros entered) is mirrored in the Actual Calling Search Space field. For example, the macro example above changes to Cu1-24HrsCLIP-PT-SiteABC.

Step 5

Add a description for the class of service in the Description field if desired.

Step 6

Choose route partition members to include in the class of service by performing the following:

Click + to add route partitions.
From the drop-down list, select a route partition member.
Repeat this step as required until you have selected all desired members for this class of service.

Note

To remove a member from the class of service, click –.

Step 7

Click Save to add the class of service that you defined.

The new class of service appears in the CoS table and it can be edited or deleted as required.

Clone a Class of Service

Use this procedure to clone an existing class of service (CoS) to the same site hierarchy node with a new name.

Procedure

Step 1

Sign in as provider, reseller, customer, or site administrator.

Note

When cloning a class of service, ensure that you select a valid site under the customer in the hierarchy node breadcrumb at the top of the view. If you attempt to clone a class of service at any other node in the hierarchy, an error appears stating that you must be at a site.

Step 2

Select Dial Plan Management > Site > Class of Service.

Step 3

Choose the class of service to be cloned by clicking on its box in the far left column.

Step 4

Click Action > Clone.

Step 5

Enter a unique name for the class of service in the Class of Service Name field. Make the name as descriptive as possible using up to 50 alphanumeric characters, including spaces, periods, hyphens (-), and underscore characters (_).

Step 6

(Optional) Add a description for the class of service in the Description field.

Step 7

Click Save to save the new class of service.

Note

You must save the cloned class of service to the same site hierarchy node as the original CoS. You cannot save the cloned CoS to a different site, or to a different hierarchy node.

The new class of service appears in the CoS table and it can be edited or deleted as required.

Configure Short Code

Use this procedure to configure short codes. Short codes are used for abbreviated dialing to other extensions and services.

Before you begin

You must add a site dial plan before configuring short codes. Refer to Create a Site Dial Plan.

Procedure

Step 1

Sign in to the server as the provider, reseller, customer, or site administrator.

Warning

When adding a short code, ensure that you select a valid site under your customer in the hierarchy node breadcrumb at the top of the view. If you attempt to add a short code at any other node in the hierarchy, an error appears stating that you must be at a site.

Step 2

Select Dial Plan Management > Site > Short Code.

Step 3

Click Add to add a short code.

Step 4

Enter a short code in the Short Code field using up to 16 characters with the following format:

The first character may be 0-9, or *.
The last character may be 0-9, #, or the wildcard character X.
All other characters may be 0-9, . (period), or the wildcard character X. Only one . (period) is allowed.

Example:

*2.XXX

Step 5

From the Short Code Type drop-down list, choose one of the following:

Option	Description
Called Mask	The called mask maps to the short code. Valid entries include the digits 0 through 9; the international escape character +; and the wildcard character X. For example, a called mask of 567XXX using short code *2.123 converts to 567123.
Directory Number	The directory number maps to the short code. Valid entries are digits 0 through 9.
Pre-dot with Called Prefix	The called prefix maps to the short code.

Step 6

Enter the value for the short code type in the Value field.

Step 7

Check the Use Originator's Calling Search Space check box to indicate that the short code will use the originator's calling search space for routing a call rather than an explicit customer CSS.

If the originating device is a phone, the originator's calling search space is a combination of:

the device calling search space configured on their phone, and
the line calling search space configured on the originating line.

Step 8

Click Save to add the short code that you defined.

The new short code appears in the table of short codes and it can be edited or deleted as required.

Configure Directory Number Routing

Use this procedure to define directory number routing. Directory number routing is a translation pattern that is put into the pre-ISR and ISR partitions to route intrasite and intersite calls to extensions (directory numbers). This is similar to the way site location codes (SLCs) are used as short codes for Type 1, 2, and 3 customer dial plans.

Typically, directory number routing is used for type 4 (flat dial plans). From a customer and site perspective, you can see which patterns are directory numbers because there are no SLCs available.

Procedure

Step 1

Sign in as the provider, reseller, customer, or site administrator.

Warning

When adding directory number routing, ensure that you select a valid site under your customer in the hierarchy node breadcrumb at the top of the view. If you attempt to add directory number routing at any other node in the hierarchy, an error message states that you must be at a site.

Step 2

Select Dial Plan Management > Site > Directory Number Routing.

Step 3

Click Add to add directory number routing.

Step 4

Enter a prefix in the Directory Number Routing Prefix field using up to 30 characters.

Example:

Enter 234

Step 5

Enter a DN mask length in the Directory Number Mask Length field.

Example:

Enter 4. For this example, the directory number routing would be 234XXXX, where XXXX is the mask.

Step 6

Click Save to add the directory number routing that you defined.

The new directory number routing appears in the table and it can be edited or deleted as required.

Create a Customer

Note

In Cisco Unified CDM 10.6(2) or later, if the customer name matches an existing customer previously configured in HCM-F, you can migrate the existing customer.
In Cisco Unified CDM 11.5(1) or later, you can choose to disable number management for the customer.

Procedure

Step 1

Sign in to the server as the provider or reseller administrator, depending on which organization manages the customer.

Sign in with the provider or reseller administrator's email address, which is case-sensitive. The provider administrator can find the reseller administrator's email address by selecting User Management > Local Admins and then clicking the reseller.

Step 2

If signed in as a provider, and the customer is to be added under a reseller, set the hierarchy path to the reseller.

Step 3

Select Customer Management > Customers.

Step 4

Click Add.

Step 5

Complete the following fields:

Field

Description

Customer Name

The name of the customer. This field is mandatory.

Note

Any spaces in the customer name are converted to underscores in the customer local administrator name and email, if Create Local Admin is checked.

Note

A customer that has been configured in HCM-F and synced into Cisco Unified Communications Domain Manager may exist at the sys.hcs hierarchy. If the Customer Name you enter matches this customer, the Migrate from HCM-F to CUCDM check box is displayed. Click Save to migrate this customer to the current hierarchy level. The fields are populated with the values that were configured in HCM-F. If you do not want to migrate the customer, enter a different Customer Name.

Description

Customer description

Extended Name

The Extended Name can be used to provide a more descriptive name of the customer. The Extended Name is also used by external clients to correlate their own customer records with the customer records stored in HCS. This Extended Name value is synced to the Customer record in the Shared Data Repository (SDR).

The Extended Name is not referenced by other components in HCS.

External Customer ID

The External Customer ID is used by the Service Inventory service. The External Customer ID is included as a column in the customer record of the service inventory report. Specify an External Customer ID in this field that matches the customer ID used by the external inventory tool which receives the Service Inventory reports. If the Service Inventory service is not being used, this field is not required. However, it can be used to correlate customer records in external systems with customer records in HCS.

Domain Name

Customer domain. This field is used to create email addresses for:

The customer default local administrator, for example: Customer1Admin@customer1.com
Site default local administrators under the customer, for example: Site1Admin@customer1.com

If the customer domain is omitted, the provider domain (or reseller domain, if the customer is under a reseller in the hierarchy and the reseller domain was provided) is used instead.

Create Local Admin

Controls whether a default local administrator is created for the customer.

Cloned Admin Role

The Provider or Reseller role used to create a new role prefixed with the customer name. The created customer role, shown in Default Admin Role field, is assigned to the default local administrator user. This field appears only if Create Local Admin is checked.

Default Admin Role

The created customer role that is assigned to the default local administrator. This field is read only and appears only if Create Local Admin is checked.

Default Admin Password

The password to assign to the default local administrator. This field appears and is mandatory only if Create Local Admin is checked.

Repeat Default Admin Password

Confirm the default local administrator password. This field appears and is mandatory only if Create Local Admin is checked.

Account ID

The Account ID is used by external clients to correlate their own customer records with the customer records stored in HCS. This Account ID value is synced to the Customer record in the Shared Data Repository.

Deal IDs

Deal IDs are used by the Hosted License Manager (HLM) service which can be activated on the Hosted Collaboration Management Fulfillment (HCM-F) server. HLM supports Point of Sales (POS) report generation. The report includes all customers on the system with aggregate license consumption at customer level. The optional Deal ID field associated with the customer is included in the report. Each customer can have zero or more Deal IDs. The Deal ID field is free text format and each deal ID is separated by a comma.

Prime Collaboration

Prime Collaboration is the application which monitors equipment used by this customer. Available Prime Collaboration applications must first be configured using the HCM-F User Interface, refer to Cisco Hosted Collaboration Mediation Fulfillment Install and Configure Guide, Release 11.5(1) for more information. Then HCM-F synchronization must be executed on Cisco Unified Communications Domain Manager. After the HCM-F data syncs into Cisco Unified Communications Domain Manager, available Prime Collaboration applications will appear in this dropdown. Select an available Prime Collaboration application to monitor Unified Communications applications and customer equipment configured for this customer.

To unassociate Prime Collaboration for this customer, select None.

Shared UC Applications

Indicates whether the customer can use Shared UC Apps. If checked, the customer sites can use Network Device Lists that contain Shared UC Apps. Shared UC Apps are UC Apps that are defined above the Customer hierarchy level.

Disable Number Management

Check to disable Number Management for this customer. If checked, you cannot add Directory Numbers and E164 Numbers to inventories for this customer.

Step 6

Click Save.

Note

When deleting a customer, remove any entities associated with the customer like LDAP, SSO providers, Devices, and NDLs.

Add Directory Number Inventory

Use this procedure to add a single directory number (DN) or range of DNs for your customer. The DNs (extensions) you specify are validated against the dial plan type (1 to 4). The extension length assigned to the site is enforced for site location code (SLC)-based dial plans. The maximum number of directory numbers you can add at a time is 1,000. For more information on type 1 to type 4 dial plans, see Directory Numbers.

Note

You cannot add directory numbers if number management has been disabled for the customer.
If you are a customer with multiple sites using a type 4 dialing plan, ensure that the directory numbers you specify are unique across sites.
This procedure creates the DN inventory only in Cisco Unified Communications Domain Manager. The numbers are not synced to Cisco Unified Communications Manager.
Directory numbers can only be added or deleted. You cannot edit the directory numbers once they are added. The usage and availability property for each DN is associated with a line or taken into use by a service.
Using the bulk loader sheet or API, you can create the DN inventory only at the customer hierarchy. The Details column of Sub Transactions shows whether the DN exists or it is creating a new DN. If any DNs exist in the range, the sub-transaction fails and the parent transaction shows the status Success with Async Failures.
Adding an existing DN will not generate an error. This action will not have any impact on the usage and available property of the DN. If there is an existing DN, it updates the transaction log with DN with internal number XXXX already exists.

Note

In Cisco Unified CDM 10.6(2) or later, you can specify an * before a directory number in a type 4 dial plan.

Before you begin

Deploy a customer and site dial plan before performing this procedure.

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Select an available customer from the hierarchy node breadcrumb at the top of the interface.

Step 3

Select Dial Plan Management > Customer > Add Directory Number Inventory or Dial Plan Management > Number Management > Add Directory Number Inventory.

Step 4

From the Site drop-down list, select the site for which you are adding directory numbers. Leave this field empty to add customer-level directory numbers.

Note

Customer-level directory numbers can only be created for dial plans that do not use site location codes (flat dial plans). Creating customer-level directory numbers for site location code-based dial plans results in an error instructing you to specify a site when adding new DN inventory.

Step 5

Using the site's Extension Length, Site Location Code, and ISP read-only fields as a guide, enter the DN range's first number in the Starting Extension field.

Note

For type 4 dial plans (no SLCs), each Starting and Ending Extension field must contain 16 or fewer digits, including + sign if used. For type 1 to 3 dial plans, the Starting and Ending Extension fields must be less than or equal to the site extension length. If the Starting or Ending Extension field length is less than the site extension length, the DN number uses zeroes to equal the site extension length.

For type 4 dial plans (no SLCs), the Starting and Ending Extension fields may contain a * prefix (asterisk) before the 15-digit directory number. The * prefix denotes DNs that are used with hunt groups, assistant lines, contact center lines, and so on. This type of directory number cannot be reached from an outside line and cannot be associated with E.164 numbers. Typically, a DN with the * prefix is not called from another line (user). Rather, it is tied to a service feature such as call pickup, hunt groups, or contact center.

Example:

If the Extension Length field shows four digits for a type 3 dial plan, enter a number containing four digits or less in the Starting Extension field, such as DN 1234. If you enter DN 123, the extension number is created as DN 0123.

Step 6

(Optional) Using the site's Extension Length, Site Location Code, and ISP read-only fields as a guide, enter the DN range's last number in the Ending Extension field. If you are adding a single DN, the ending number is the same as the starting number.

Note

The maximum number of directory numbers you can add is 1,000 at a time. If you need more than 1,000 directory numbers, repeat this procedure as required to add ranges.

Step 7

Enter a tag name for the entered range to allow for tag filtering of the inventory list available from Dial Plan Management > Number Management > Directory Number Inventory.

Step 8

Use the following fields to input additional information for the range:

Description
E164Number
Extra1 to Extra3

Except for E164Number, these fields allow free text.

Step 9

Click Save to save the single DN or DN range.

Note

You can verify that the directory number or numbers were added correctly by navigating to Subscriber Management > Directory Number Inventory or Dial Plan Management > Number Management > Directory Number Inventory.
Columns for the Tag, E164 Number, and other additional information fields are also shown.

View Directory Number Inventory

Use this procedure to view the range of directory numbers that have been defined for a site.

Note

In Cisco Unified Communications Domain Manager 10.6(2) or later, an * can appear before a directory number in a type 4 dial plan.

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Select an available site from the hierarchy node breadcrumb at the top of the interface.

Step 3

Select Dial Plan Management > Number Management > Directory Number Inventory.

The list of all directory numbers (DNs) configured for the site appears. You can view the list of DN numbers or delete a DN number from this page. To filter the list of directory numbers, click the up arrow beside the title of the Internal Number column. Enter the search string you want to locate, and all directory numbers that match the search string appear.

When a DN is first added to the inventory, the Used column is blank, and the Available column shows “true.” The Used column changes to “true” when the DN is put into use when a line is created and associated to a phone or subscriber. The Available column indicates that the DN is used by a device or service that does not allow a shared line (for example, a hunt pilot).

Directory numbers that begin with a * (asterisk) denote DNs that are used with hunt groups, assistant lines, contact center lines, and so on. This type of directory number cannot be reached from an outside line. Typically, a DN with the * prefix is not called from another line (user). Rather, it is tied to a service feature such as call pickup, hunt groups, or contact center.

Note

Adding a new DN to inventory on Cisco Unified Communications Domain Manager does not add a directory number on Cisco Unified Communications Manager until it is associated to a line on Unified CDM.

The directory number inventory entries appear in other end-user provisioning tasks in Cisco Unified Communications Domain Manager as described in the table that follows. For more information on provisioning each of these tasks, refer to Cisco Hosted Collaboration Solution End-User Provisioning Guide.


Task	Cisco Unified Communications Domain Manager Location	Notes
Lines	Subscriber Management > Lines	When lines are added through phones and subscriber, line details can be modified. The DN for the line cannot be modified; if you attempt to change the DN assigned to the line, the operation fails.
Phones	Subscriber Management > Phones > Lines tab > Dirn > Pattern	The Dirn > Pattern contains a list of available directory numbers. DNs that are used are marked as "true" in the directory number inventory. Only available DNs are listed.
Subscribers	Subscriber Management > Subscribers > Phones > Lines > Dirn	The Dirn > Pattern contains a list of available directory numbers. DNs that are used are marked as "true" in the directory number inventory. Only available DNs are listed.
Subscribers	Subscriber Management > Subscribers > Voicemail	The Voicemail Line list contains DNs provisioned to lines.
Quick Add Subscribers	Subscriber Management > Quick Add Subscriber > Lines > Directory Number.	The Directory Number list contains available directory numbers. DNs that are used are marked as "true" in the directory number inventory. Only available DNs are listed.
PLAR (Hotdial)	Subscriber Management > PLAR (Hotdial)	DNs provisioned to lines are displayed in the Hotdial Destination Pattern list.
Hunt Groups	Subscriber Management > Hunt Groups > Members > Directory Number	DNs provisioned to lines are displayed in the Pattern list.
Call Pickup Groups	Subscriber Management > Call Pickup Groups > Call Pickup Group > Line	DNs provisioned to member lines are displayed in the Pattern list.

Run the Directory Number Inventory Audit Tool

When you run the Directory Number Inventory (DNI) Audit Tool, the tool checks and updates your directory number inventory since the last data sync. A sync of device/cucm/Line from the Unified CM will result in various line types being brought in, which includes lines assigned to devices, CTI devices,etc that are seen under Directory Numbers in the Unified CM.

This tool creates a number inventory entry for all the device/cucm/Lines that are in the system and at the site level. Any lines that are not at the site level will not be processed in the audit tool.

Note

This process can result in inventory entries being created for lines that are not user device related (For example CTI ports, CTI route points, etc). The Audit will not remove number inventory entries, if a corresponding device/cucm/Line does not exist, for example if it is removed outside of Unified CDM.

The Number Inventory audit then includes logic to determine if a number is used or not and to set the number inventory value accordingly. The logic handles these cases:

If a line is assigned to at least one of the phone, device profile, or remote destination profile, then it will be marked used = true and available will be left as true.
If a number is used as a Hunt Pilot - it will mark the number used = true and available = false.
Any other usage of the line is not handled (e.g. CTI route point). So while these numbers are added to the inventory, they will not be marked used or unavailable through the audit process.

The tool is run manually from the Overbuild menu or the Dial Plan Management submenu. When you run the tool, it creates new DNs for lines that don't have them. You specify where you want the tool to run and create a new DN inventory:

Customer Create

This option always creates a new DN inventory at the customer level.
This option is only available for non-SLC dial plans (type 4 or non-SLC shell dial plans).
The DN inventory is created at the customer hierarchy if the line exists at the customer level or at a site. The reason is that even if the administrator moves the line to a site later, the DN inventory at the customer level still applies. If the DN inventory exists, it is updated.

Site Create

This option always creates a new DN inventory at the site level.
This option applies to any type of dial plan.
The DN inventory is created at the first site the line is encountered. If more than one line with the same pattern but different partitions exist, the DN inventory is created for the first line encountered with that pattern.
If the line exists at customer level, a warning message is logged and the DN inventory is not created. If the DN inventory exists, it is updated.

Smart Create

Creates a new DN inventory at the site hierarchy level.
If the line exists at a site and is not used by a phone in another site, this option creates a new DN inventory at the hierarchy where the line exists.
If the line exists at a site, one or more phones in other sites reference that line, and the dial plan type is non-SLC (type 4), this option creates a new DN inventory at the customer level.
If the line exists at a site, one or more phones in other sites reference that line, and the dial plan type is SLC (type 1-3), a new DN inventory is created at the site where the line exists.
If the line exists at customer level, a warning message is logged and the DN inventory is not created. If the DN inventory exists, it is updated.

For sites using SLC-based dial plans, DN inventories can be created only at the site hierarchy. The option to create DN inventories at the customer hierarchy is unavailable in this case.

The DN Inventory Audit Tool marks data/InternalNumberInventory instances as shared across sites if a line is associated with multiple devices.

From Dial Plan Management > Number Management > Directory Number Inventory, you can see a list of DNs and move, delete, and export them as desired.

Dial Plan Management > Number Management > Log Messages provides information and warning messages generated by the Directory Number Inventory Audit Tool.

Note

You cannot run the Directory Number Inventory Audit Tool if number management has been disabled for the customer.

Common Errors and Caveats

Duplicate device profiles (same profile name) in different clusters.
Ensure device profiles are not duplicated across the sites.
Duplicate phones (same MAC) in different clusters.
Ensure phones are not duplicated across the clusters
Same directory number in one or more clusters.
Ensure directory numbers (even in different partitions) are not duplicated across clusters.

Procedure

Step 1

Sign in to Cisco Unified Communications Domain Manager as a provider or reseller admin.

Step 2

Click Dial Plan Management > Number Management > Directory Number Inventory Audit Tool.

Step 3

If prompted, select the correct hierarchy and click OK.

Note

The tool can only be run from customer hierarchies. If you run the tool from a hierarchy that is not of type Customer, the tool automatically provides you with a valid customer hierarchy choice.

Step 4

Click the Directory Number Inventory Creation Policy drop-down and select an option.

Note

Customers with SLC-based dial plans will not see the Customer Create option.

Step 5

Click Save.

The DN inventory is updated at the hierarchy you specified and below.

E.164 Inventory Management

E.164 Inventory Management provides direct dial-in (DDI) and direct inward dialing (DID) mapping to directory numbers (DNs) using translation patterns in the Unified Communications Domain Manager. The DDI-to-DN mapping allows you to route incoming PSTN calls to the appropriate internal directory number.

E.164 Inventory Management includes the ability to:

Add, view, and delete E.164 number inventory
Associate a range of E.164 numbers to a range of DNs
View associated range of E.164 numbers to a range of directory numbers
Disassociate a range of E.164 numbers from a range of DNs
Associate a range or set of E.164 numbers to a single DN
Disassociate a range or set of E.164 numbers from a single DN
View single-directory number associations

The E.164 inventory is available in the drop-down lists for Site Published Number and Emergency Number when creating a site dial plan.

Add E.164 Inventory

Use this procedure to define an inventory of E.164 numbers available to users.

Important

Each addition to the E.164 inventory must contain a unique set of numbers. That is, you cannot assign the same number more than once (globally).
In Shared Architecture deployment, if you need to configure E.164 number as Directory number, enable the Use Calling Party's External Phone Number Mask checkbox in the Translation Pattern “+.!” of the <Customer ID>-dFONetNatl-PT partition. This configuration is required for national calls, international calls or any PSTN breakout for customers having E.164 number as directory number.

Note

You cannot add E.164 numbers if number management has been disabled for the customer.

Note

In Cisco Unified CDM 10.6(3), you can define the E.164 inventory at the customer level.

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Set the hierarchy path to point to the customer for whom you are adding the E.164 inventory.

Step 3

Select Dial Plan Management > Number Management > Add E164 Inventory.

Step 4

Provide the following information:

Fields	Description
Site	For a site-specific E.164 inventory, select the customer site. For a customer-wide E.164 inventory, leave this field unset.
Country	Select the country associated with the E.164 inventory. If a site was specified, this field is automatically populated with the country associated with the site. This field is required.
Country Code	The country code for the selected country. Refer to this read-only field when specifying the Starting Number and Ending Number fields, which must contain a valid country code.
Starting Number	Enter the starting number of the range of E.164 numbers. The field is populated with + followed by the country code for the selected country. Append the rest of the starting number after the country code. This field is required.
Ending Number	Enter the ending number of the range of E.164 numbers. The format is the same as the starting number. This field is optional. If not provided, the single E.164 number specified in the Starting Number field is added. If provided, the range of E.164 numbers is added: starting number – ending number, inclusive. A maximum of 1000 numbers can be added at a time.

Step 5

Click Save.

Associate a Range of E.164 Numbers to a Range of Directory Numbers

Use this procedure to associate a range of E.164 numbers with a range of directory numbers (DN) at a customer or site. These associations create direct dial inward (DDI) associations so that incoming PSTN numbers are routed to directory numbers.

If you create the association at a site, you can mix customer-level DNs and E.164 numbers with site-level DNs and E.164 numbers.

Note

In Cisco Unified CDM 10.6(2) and later, the event related to SIP local gateway is generated as a result.

Note

You cannot associate numbers if number management has been disabled for the customer.
Only DNs or E.164 numbers that are not already associated are available for association.

Procedure

Step 1

Sign in as provider, reseller, customer, or site administrator.

Step 2

Set the hierarchy path to point to the customer or site where you want to associate E.164 numbers with directory numbers.

Step 3

Select Dial Plan Management > Number Management > E164 Associations (N to N DN).

Step 4

Click Add.

Step 5

Provide the following information:

Field

Description

Range

Select one of these ranges:

Note

The range values you select map to the mask value when the association translation pattern is created. For example, when 10 is selected, all E.164 numbers and directory numbers that end in 0 are listed. The mask affects all digits 0 to 9, so you can't start the mask on a nonzero number. Likewise, when 100 is selected, the E.164 number and DN end in two zeros. This pattern results in a mask of XX.

1—To list all E.164 numbers and DNs
10—To list all E.164 numbers and DNs that end in one zero (0)
100—To list all E.164 numbers and DNs that end in two zeros (00)
1000—To list all E.164 numbers and DNs that end in three zeros (000)

This field is required and affects what appears in the fields that follow.

E164 Number

Select the starting number of the range of E.164 numbers from the drop-down list. For a customer-level association, only customer-level E.164 numbers are available. For a site-level configuration, both customer-level and site-level E.164 numbers are available.This field is required.

DN Number

Select the starting extension number from the drop-down list. This field is required.

Note

You cannot associate extension numbers that begin with the prefix * (asterisk).

Step 6

Click Save.

For a site-level association, a translation pattern that is used to route inbound PSTN calls to their associated DNs is created on the Unified CM. This pattern is the mapping between the E164 range and DN range.
For a customer-level association, a translation pattern is created on each Unified CM cluster that has a dial plan provisioned.
For a site-level association, if the site has one or more SIP local gateways associated with it, the HcsSipLocalGwAddE164AssociationEVT event is generated. If enabled, the IOS Command Builder generates the default IOS commands associated with the event for each SIP local gateway.
For a customer-level association, if the E.164 number has the same country as any SIP local gateway configured for the customer, the HcsSipLocalGwAddE164AssociationEVT event is generated. If enabled, the IOS Command Builder generates the default IOS commands associated with the event for each SIP local gateway.

Associate a Set of E.164 Numbers to One Directory Number

Use this procedure to associate a set of E.164 numbers with one Directory Number (DN) at a customer or site. For example, you could associate a set of E.164 numbers for the sales department with an attendant's DN.

If you create the association at a site, you can mix customer-level DNs and E.164 numbers with site-level DNs and E.164 numbers.

Note

In Cisco Unified CDM 10.6(2) and later, the event related to SIP Local Gateway may be generated as a result. Also you can designate a primary E.164 number.

You can optionally specify a primary E.164 number to associate with the DN. This step can be useful when you perform a DN-to-E.164 translation (for example, when provisioning translation rules for LBO gateways) and the DN is associated to more than one E.164 presentation.

Note

You cannot associate numbers if Number Management has been disabled for the customer.
Only DNs or E.164 numbers that are not currently associated are available for association.

Procedure

Step 1

Sign in as a provider, reseller, customer, or site administrator.

Step 2

Set the hierarchy path to the customer or site where you are associating a set of E.164 numbers with one DN.

Step 3

Select Dial Plan Management > Number Management > E164 Associations (N to 1 DN).

Step 4

Click Add.

Step 5

From the DN Number menu, select an extension number. This field is mandatory.

Step 6

In the E164 Ranges table, click + as required, to add multiple sets of E.164 numbers. The E.164 numbers do not need to be contiguous. Provide the following information for each association:

Field

Description

E164 Range

Select one of the following sets:

1—To list all E.164 numbers
10—To list all E.164 numbers that end in one zero (0)
100—To list all E.164 numbers that end in two zeros (00)
1000—To list all E.164 numbers that end in three zeros (000)

This field is mandatory and affects what appears in the E164 Number field.

E164 Number

Select the starting number of the set of E.164 numbers. For a customer-level association, customer-level E.164 numbers are available. For a site-level configuration, both customer-level and site-level E.164 numbers are available. This field is mandatory.

Step 7

In the Primary E164 field, enter the primary E.164 number to associate with the DN. Ensure the E.164 number you enter starts with \+ and falls within the range you specified in the E164 Range field. This field is optional.

Step 8

Click Save.

For a site-level association, one or more translation patterns that are used to route inbound PSTN calls to their proper DN are created on the Unified CM. These patterns are the mappings between the set of E.164 numbers and the single directory number. When you associate a set of E.164 numbers to a single DN, multiple translation patterns are created; that is, each DN-to-E.164 range association results in a translation pattern being created on Cisco Unified Communications Manager.
For a customer-level association, the translation patterns are created on each Unified CM cluster that has a dial plan provisioned.
For a site-level association, if the site has one or more SIP Local Gateways associated with it, the HcsSipLocalGwAddMultiE164AssociationEVT is generated. If enabled, the IOS Command Builder generates the default IOS commands associated with the event for each SIP Local Gateway.
For a customer-level association, if the E.164 number has the same country as any SIP Local Gateway configured for the customer, the HcsSipLocalGwAddE164AssociationEVT event is generated. If enabled, the IOS Command Builder generates the default IOS commands associated with the event for each SIP Local Gateway.

Manual Configuration for Intersite Cross-Cluster Support

In order to support intersite calls for customers that have sites that span multiple clusters, the following manual procedure is required.

Procedure

Step 1

Create a full-mesh network between clusters for customers. Create trunk, route group, and route list with Cisco Unified Communications Domain Manager for a given cluster to every other cluster owned by the customer. For a shared Cisco Unified Communications Manager, a SIP security profile is needed for each trunk. For procedures, see Configure SIP Trunks, , andConfigure Route Groups Configure Route Lists.

Step 2

For each site added to a cluster, a route pattern must be added to all the other clusters in the mesh network owned by the customer. The route pattern is added to the InterSiteRouting partition, the partition name in Cisco Unified Communications Manager is Cu<CustomerID>-ISR-PT, where <CustomerID> is the customer ID.

The pattern in the route pattern depends on the internal dial plan type:
- Type 1 and type 3 are the site location code (SLC) plus the extension mask of the site.
- Type 2 is the ISP plus the SLC plus the extension mask of the site.
- Type 4 is the DN range of the site.
The route list in the route pattern is the route list associated to the site cluster.

Manual Configuration for Local Breakout Support

Important

Use this procedure only if you are using Cisco Unified CDM 10.6(1). Unified CDM 10.6(2) or later supports local breakout (LBO).

To manually configure a local gateway for a site to support LBO, perform the following manual steps.

Procedure

Step 1

Ensure that the Cisco Unified Communications Domain Manager hierarchy is set to the site where the local gateway is to be added.

Step 2

Use Cisco Unified Communications Domain Manager to create trunks, route groups, and route lists to the local gateway. For procedures, see Configure SIP Trunks, Configure Route Groups, and Configure Route Lists.

Step 3

In Cisco Unified Communications Manager, create a partition Cu<CustomerID>Si<SiteID>-LBO-LBR-PT to be used in the class of service. Refer to Configure Class of Service.

Step 4

In Cisco Unified Communications Manager, create a partition and CSS to handle LBO routing for the site.

Step 5

Add the following translation patterns to the partition defined in step 3:

Add the ++061.0! translation pattern to handle calls without forced authorization code (FAC) and client matter codes (CMC).
Add the ++061.1! translation pattern to handle calls with FAC and without CMC.
Add the ++061.2! translation pattern to handle calls with CMC and without FAC.
Add the ++061.3! translation pattern to handle calls with CMC and FAC.

Step 6

Associate the patterns in step 5 with the CSS defined in step 4.

Step 7

For the patterns in step 5, ensure that the called number transformation is PreDot and add the ‘**’ prefix.

Step 8

Create a default translation pattern in the routing partition defined in step 4 with an ‘**X!’ pattern. Set the CSS to Cu<CustomerID>-<CC>DP-LBRRteSel-CSS. This is used to switch the call processing back to central breakout (CBO) for all call types that are not sent using LBO.

Step 9

Create a route pattern for each call type that breaks out from the local gateway in step 4. Use the route list created in step 3.

Note

The called and calling number between the local gateway and Cisco Unified Communications Manager is in +E.164 format. Therefore, all incoming and outgoing calls between the gateway and Cisco Unified Communications Manager conform to it. It is also assumed that you provide the IOS gateway configuration.

Step 10

Create a new CoS to be included in step 3 before LBRtg-PT.

Create Voicemail Service

Before you begin

To associate voicemail service with a Cisco Unified Communications Manager (Unified CM), you must know the SIP trunking endpoint information between the voicemail server and the Unified CM.

A Cisco Unity Connection server must be configured before performing this procedure. For more information, see "Set Up Cisco Unity Connection" in the Cisco Hosted Collaboration Solution Release 12.5 Customer Onboarding Guide.

Procedure

Step 1

Sign in as a provider or reseller administrator.

Step 2

Make sure that the hierarchy path is set to the correct provider or reseller node.

Step 3

Select Services > Voice Mail > Voice Mail Service.

Step 4

Click Add to add a voicemail service.

Step 5

Enter a Voice Mail Service Name if desired. Do not add spaces in the name.

Step 6

From the Voice Mail Cluster pulldown menu, select the name of the server for the voicemail service.

Note

The Cisco Unity Connection server must be previously defined under the provider level at Device Management > CUCs. This is also the location whether the voicemail server in a multitenant environment is categorized as dedicated or partitioned. This determines what elements are available to the voicemail server, whether to create another tenant on the voicemail server, and so on.

Step 7

To integrate the voicemail service with Unified CM, check the Integrate with CUCM box. Default is unchecked.

Step 8

If Unified CM manages the voicemail service, select the Cisco Unified Communications Manager to be paired with the Voicemail Unity Server from the CUCM Cluster pulldown menu.

Note

The Unified CM must be previously defined under the provider level at Device Management > CUCMs.

Step 9

Complete the SIP trunk provisioning information (between the SIP trunk and the Cisco Unity Connection server) in the following fields:

Enter the hostname or IP address of the voicemail server trunk in the Voicemail Trunk Address field.

Enter the voicemail server port number (1 to 65535) in the Voicemail Trunk Port field.

Note

Do not specify port 5061, which is reserved for secure SIP.

Enter the hostname or IP address for the voicemail server to reach the Unified CM in the Remote Trunk Address field.

Enter the Cisco Unified Communications trunk port number in the Remote Trunk Port field.

Note

Do not specify port 5061, which is reserved for secure SIP.

Note

Only one Unified CM and one Cisco Unity Connection can be specified here. To support redundancy and failover in a multinode configuration, you must manually update the trunk information on the UC applications.

Step 10

In the Voice Messaging Ports field, enter the number of voice messaging ports to be created for the voicemail service and associated with the appropriate port group on Cisco Unity Connection when the voicemail service is associated to a customer.

Valid values are 1–250. The default is 3. This field is required.

Note

The number of voice messaging ports that you add cannot bring the total number of voice messaging ports for all port groups to more than the maximum number of voice messaging ports enabled by the Cisco Unity Connection license files. If the license files do not enable the total number of ports, you will not be able to add the new ports.

Step 11

Click Save to add the voicemail service you defined.

When a shared voicemail service is created and Integrate with CUCM is selected, the following occurs:

In Unified CM: Cluster-level SIP trunk and route group are provisioned for the shared voicemail service.
In Cisco Unity Connect: Cluster-level port group appears on the phone system for the shared voicemail service.

What to do next

Perform Associate Voicemail Services with a Customer.

Associate Voicemail Services with a Customer

Before you begin

To associate voicemail service with a customer, create the voicemail service before starting this procedure.
If the Integrate with CUCM option was selected when the voicemail service was created, a customer dial plan and a site dial plan must be created before a voicemail service can be associated with a customer, or the association will fail.

Procedure

Step 1

Sign in as the provider or reseller administrator.

Step 2

Set the hierarchy path to the customer to which you want to associate the voicemail service.

Step 3

Select Services > Voice Mail > Associate Voice Mail Service to Customer.

Step 4

Click Add to associate voicemail service to a customer.

Step 5

From the Voice Mail Service pulldown menu, select the name of the voicemail service defined by the provider and available to this customer.

Step 6

Click Save to associate the voicemail service with the customer.

The association appears in the list. When voicemail service is associated with a customer and the Integrated with CUCM option was selected for the voicemail service, the following is provisioned based on the deployment mode of the voicemail server:


Voicemail Deployment Mode	Cisco Unified Communications Manager	Cisco Unity Connection
Dedicated	Creates integration at customer level: SIP trunk, route group, AllowVm route partition	Creates customer-specific port group, ports (3), route partition, calling search space, and user template.
Partitioned	Creates integration at customer level: SIP trunk, route group, AllowVm route partition	Creates new tenant (partition), port group, ports (3), route partition, calling search space, and user template.

Note

The deployment mode for the voicemail service is determined by the mode selected when the Cisco Unity Connection is first added to the Cisco Unified Communications Domain Manager using Device Management > CUCs.

Disassociate Voice Mail Services from Customers

Procedure

Step 1	Log in as the Provider Administrator.
Step 2	Set the hierarchy path to the customer from which you want to disassociate the Voice Mail Service.
Step 3	Select Services > Voice Mail > Associate Voice Mail Service to Customer.
Step 4	From the list of associations, choose the Voice Mail Service customer association to be deleted, by clicking on its box in the leftmost column.
Step 5	Click Delete to disassociate the Voice Mail Service from the customer.
Step 6	From the popup window, click Yes to confirm the change. When the delete action is complete, the Voice Mail Service association to the customer disappears from the list.

Define a Voicemail Pilot Number

Note

In Cisco Unified CDM 10.6(2) or later, the voicemail pilot number is selectable from a list of available DN inventory.

Before you begin

To create voicemail pilot numbers for voicemail services that have previously been associated with the customer, complete the following tasks before performing this procedure:

Voicemail service must be created.
Voicemail service must be associated with the customer. See Associate Voicemail Services with a Customer.

Procedure

Step 1

Sign in as the customer or provider administrator.

Step 2

Ensure that the hierarchy path is set to the customer or site that you are defining a voicemail pilot number for.

Step 3

Select Services > Voice Mail > Pilot Numbers.

Step 4

Click Add to associate a pilot number with the voicemail service that has been associated with the customer.

Step 5

From the Voice Mail Service pulldown menu, select the appropriate voicemail service from the list of voicemail services associated with the customer.

Step 6

From the Voice Mail Pilot Number pulldown menu, select a pilot number from the list of your available DN inventory, or type the pilot number you want to use in the field. This is the internal voicemail pilot number that can be dialed from the site.

Note

More than one pilot number can be created for a single voicemail service.

Step 7

Click Save to create the pilot number.

The pilot number appears in the list. When a pilot number is created for a voicemail service and the Integrated with CUCM option was selected for the voicemail service, the following is provisioned based on the deployment mode of the voicemail server:


Voicemail Deployment Mode	Cisco Unified Communications Manager
Dedicated	At customer level: route list, route pattern, CSS, voicemail pilot, voicemail profile
Partitioned	At customer level: route list, route pattern, CSS, voicemail pilot, voicemail profile

Creating DDIs for Voice Mail Pilot Numbers

Before you begin

To create a DDI for a voice mail pilot number, perform the following steps on Cisco Unified Communications Domain Manager. The voice mail pilot number must be created before performing this procedure. See Define a Voicemail Pilot Number.

Procedure

Step 1

Log in to Cisco Unified Communications Domain Manager as a Provider, Reseller, or Customer administrator.

Step 2

Use the breadcrumbs to navigate to the customer hierarchy node that contains the voice mail pilot number.

Step 3

Select Device Management > CUCM > Route Patterns.

Step 4

Select Add.

Step 5

Create a new route pattern instance with the following information:

On the Pattern Definition tab, complete the following items.
1. CUCM: Select the appropriate Cisco Unified Communications Manager cluster for this route pattern. This should be the cluster on which you created the voice mail pilot.
2. Route Pattern: \+<E.164 number>: Enter an appropriate DDI number.
3. Route Partition: Cu<customerId>-E164LookUp-PT
4. Route List: From the drop-down, choose the appropriate route list for the target voice mail pilot number. The pilot number will be in the route list name. Example: Cu<customerId>-<voicemail service name><target VM pilot number>-RL, Cu5-TestVmService1000-RL
On the Called Party Transforms tab, enter a pilot number in the Called Party Transform Mask field; for example, 1000.

Step 6

Select Save.

Step 7

Repeat these steps for each voice mail pilot number.

Note

This route pattern needs to be deleted from Device Management > CUCM > Route Patterns before the voice mail pilot can be deleted. This is because this new route pattern will still reference the pilot-specific route list, causing the voice mail pilot number delete workflow to fail. If this occurs, delete the route pattern and attempt to delete the voice mail pilot again.

Associate Pilot Number to a Site

Note

In Cisco Unified CDM 10.6(2) or later, the event related to SIP Local Gateway may be generated as a result. Also you can select an E164 number to associate with the Pilot Number.

Before you begin

To associate a Voice Mail Pilot number with a site, the Pilot Number must be created before starting this procedure. See Define a Voicemail Pilot Number.

Procedure

Step 1	Log in as a Customer or Provider administrator.
Step 2	Set the hierarchy path to the desired Site.
Step 3	Select Services > Voice Mail > Associate Pilot Number to Site.
Step 4	Click Add to associate a Voice Mail Pilot Number with a site.
Step 5	From the Voice Mail Service menu, select the name of the Voice Mail Service.
Step 6	From the Voice Mail Service Pilot Number menu, select the Pilot Number for the selected Voice Mail Service.
Step 7	From the E164 Number menu, select a E164 number from your site's inventory to associate with the Pilot Number, or type the E164 number you want to use. This step is optional.
Step 8	Click Save to associate the Voice Mail Service Pilot Number with the site.

The association appears in the list. When a pilot number is associated to a site, the Site Management > Defaults > CUC Defaults are updated so that the subscriber management templates can take advantage of this new voice mail service for the site.
If the site has one or more SIP Local Gateways associated with it and an E164 Number has been specified, the HcsSipLocalGwAddVoiceMailPilotNumberEVT is generated. If enabled, the IOS Command Builder generates the default IOS commands associated with the event for each SIP Local Gateway.

Disassociate Pilot Number from a Site

Note

In Cisco Unified CDM 10.6(2) or later, the event related to SIP Local Gateway is generated as a result.

Procedure

Step 1	Log in as the Customer Administrator. For a list of the roles and tasks that can be done at each level, see Cisco Hosted Collaboration Solution Roles and Privileges.
Step 2	Select Services > Voice Mail > Associate Pilot Number to Site.
Step 3	From the list of associations, choose the Pilot Number association to be deleted, by clicking on its box in the leftmost column.
Step 4	Click Delete to disassociate the Pilot Number from the site.
Step 5	From the popup window, click Yes to confirm the change.

When the delete action is complete, the Pilot Number association to the site disappears from the list.
If the site has one or more SIP Local Gateways associated with it, the HcsSipLocalGwDelVoiceMailPilotNumberEVT event is generated. If enabled, the IOS Command Builder generates the default IOS commands associated with the event for each SIP Local Gateway.

Adding Aggregation Trunk and Route Group and Associating to Existing Route List and SLRG

In Cisco Unified Communications Domain Manager, the dial plan creates a route list to route the calls to the aggregation for central breakout (CBO). However, the following procedure is required to enable calls to egress to the PSTN network using a SIP trunk. Before adding a SIP trunk using Configure SIP Trunks, you must provision the following:

Note

In step 11, navigate to Device Management > CUCM > Local Route Groups if you are using Cisco Unified CDM 10.6(1).

Procedure

Step 1

In Cisco Unified Communications Domain Manager, create a region for the trunk as follows:

Sign in as a provider or reseller and navigate to Device Management > CUCM > Regions Information > Region.
Click Add.
Provide a name in the format Cu<cid>-Trunk-<TrunkName>-Region.

Step 2

In Cisco Unified Communications Manager, navigate to CUCM System > Device Pool and provision a device pool as follows:

Click Add.
Enter a device pool a name in the format Name Cu<cid>-DP-Trunk.
Choose a CCM group from the dropdown or leave at the default group.
Ensure that the region is set to the name created in step 1.
Set the location to Hub None.

Step 3

To create an aggregation SIP trunk, sign in as a provider in Cisco Unified Communications Domain Manager and perform the following:

Navigate to Device Management > CUCM > SIP Trunks.
Click Add.

Step 4

In the Device Information tab, perform the following:

Choose the Unified CM from the drop-down list.
Provide a device name; for example, Cu<cid>-Trunk-<TrunkName>.
Set the device pool to the device pool name you created.
Set the region to the name created in step 1.
Set call classification to OffNet.
Click Redirecting Diversion Header Delivery - Inbound.
Click Run On All Active Unified CM nodes.

Step 5

In the Call Routing Inbound tab, perform the following:

Provide the Prefix for the Incoming Number in the Incoming Calling Party Settings.

Note

The Prefix must be +1 for North America.

Choose the calling search space by selecting Cu<cid>-IngressFromCBO-CSS from the drop-down list.
Choose the connected party transformation CSS by selecting Cu<cid>-CNPNTranform-CSS from the drop-down list.
Uncheck the Use Device Pool Connected Party Transformation CSS box.
Check the Redirecting Diversion Header Delivery - Outbound box.

Step 6

In the Call Routing Outbound tab, perform the following:

Choose the called party transformation CSS by selecting Cu<cid>-CDPNTransform-CSS from the drop-down list.
Uncheck the Use Device Pool Called Party Transformation CSS box.
Choose the calling party transformation CSS by selecting Cu<cid>-CGPNTransform-CSS from the drop-down list.
Uncheck the Use Device Pool Calling Party Transformation CSS box.

Step 7

In the SIP Info tab, provide the destination IP address. It is assumed that the default SIP profile and SIP trunk security profile are used.

Step 8

Once the aggregation SIP trunk is created, assign it to a route group as follows:

Navigate to Device Management > Route Groups.
Click Add.
Provide a name for the route group in the format Cu<cid>-RouteGroup-<Name>.
Set the distribution algorithm to Top Down.
Add the above trunk as a member of the route group.

Note

For line-based routing (LBR), perform steps 9 and 10.

Step 9

Associate the above route group to the route lists. The assumption is that there is one trunk or route group to the aggregation that is shared by the whole country. However, if there is a trunk per country, then repeat the above step to create trunk and route groups for each country.

The country dial plan automatically creates the following LBR route lists for each country for each customer:

Cu<cid>-<ISO>Intl-RL . (cid is the customer ID number and <ISO> is the 3-letter alpha code for the countries of the world. For more information on ISO, refer to http://en.wikipedia.org/wiki/ISO_3166-1).
Cu<cid>-<ISO>Natl-PL
Cu<cid>-<SIO>Mobl-PL
Cu<cid>-<ISO>Emer-RL
Cu<cid>-<ISO>Serv-RL
Cu<cid>-<ISO>Local-RL
Cu<cid>-<ISO>PRSN-RL
Cu<cid>-<ISO>FPHN-RL
Cu<cid>-<ISO>PCSN-RL
Cu<cid>-<ISO>SRSN-RL
Cu<Cid>-<ISO>Oper-RL

Note

The SLRG-Emer local route group must be provisioned even for line-based routing (see step 11).

Step 10

Update each of the route lists to include the above-created route group as follows:

Navigate to Device Management > CUCM > Route Lists.
Select and enter each of the route list pages from the step above.
Click on the Add Route Group Items and select the above route group.
Save and proceed to the next route list until all the route lists include the route group.

Step 11

For device-based routing (DBR), nothing is needed for DBR route lists because they already contain the correct “well-known” local route groups. For each location that uses DBR, update the device pool as follows:

Navigate to Device Management > CUCM > Device Pools.
Select and enter the device pool SLRG page.

Add the following “well-known” SLGs and associate them to the route group created above.

SLRG-Emer

Note

SLRG-Emer must be added regardless of whether DBR is used. Emergency call handling depends on this in order to work.

SLRG-Intl
SLRG-Mobl
SLRG-Serv
SLRG-Local
SLRG-PRSN
SLRG-FPHN
SLRG-PCSN
SLRG-SRSN
SLRG-Oper

Note

For more details on route selection (LBR and DBR), see Calling Search Spaces and Partitions.

Configure SIP Profiles

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Make sure that the hierarchy path is set to the node where the Cisco Unified Communications Manager is configured.

Step 3

Perform one of the following:

If you signed in as a provider or reseller administrator, select Device Management > CUCM > SIP Profiles.
If you signed in as a customer administrator, select Device Management > Advanced > SIP Profiles.

Step 4

Perform one of the following:

To add a new SIP profile, click Add, then go to step 5.
To edit an existing SIP profile, choose the SIP file to be updated by clicking it in the list of SIP profiles. Go to step 6.

Step 5

If the Network Device List pop-up window appears, select the NDL for the SIP profile from the drop-down list. The window appears when you are on a nonsite hierarchy node. If you are at a site hierarchy node, the NDL associated with the site is automatically used.

Note

The Network Device list only appears when a SIP profile is added. It does not appear when you edit a SIP profile.

Step 6

Enter a unique name for the new SIP profile in the Name field, or modify the existing name if desired.

Step 7

Complete the fields on each tab as appropriate.

The following fields on each tab are required:

SIP Profile Information tab
- Name
- User-agent and server header information
- Version in user agent and server header
- Dial string interpretation
SDP Information tab
- Allow RR/RS bandwidth modifier (RFC 3556)
Parameters Used in Phone tab: no required fields
Normalization Script tab: no required fields
Incoming Requests FROM URI Strings tab: no required fields
Trunk Specific Configuration tab
- Calling Line Identification Presentation
- Session Refresh Method
- Early offer support for voice and video calls
Trunk SDP Information tab: no required fields
Trunk SIP OPTIONS Ping tab: no required fields

For details on each tab's fields, see SIP Profile Field Descriptions.

Step 8

To save a new SIP profile, click Save. To save an updated SIP profile, click Update.

SIP Profile Field Descriptions

Table 2. SIP Profile Information Tab

Option

Description

Name (Mandatory)

Enter a name to identify the SIP profile; for example, SIP_7905. The value can include 1 to 50 characters, including alphanumeric characters, dot, dash, and underscores.

Description (Optional)

This field identifies the purpose of the SIP profile; for example, SIP for 7970. 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 (<>).

Default MTP Telephony Event Payload Type (Optional)

This field specifies the default payload type for RFC2833 telephony event. See RFC 2833 for more information. Usually, the default value specifies the appropriate payload type. Be sure that you have a good understanding of this parameter before changing it, as changes could result in DTMF tones not being received or generated.

Default=101

Range=96 to 127

This parameter's value affects calls with the following conditions:

An outgoing SIP call from Cisco Unified Communications Manager
For the calling SIP trunk, the Media Termination Point Required box is checked on the SIP Trunk Configuration window

Early Offer for G.Clear Calls (Optional)

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

To enable or disable Early Offer for G.Clear Calls, choose one of the following options:

Disabled
CLEARMODE
CCD
G.nX64
X-CCD

User-Agent and Server header information (Mandatory)

This feature indicates how Unified CM 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. Unified CM passes any contact headers through untouched.
Pass Through Received Information as Contact Header Parameters—If selected, the User-Agent and Server header information is passed as Contact header parameters. The User-Agent and Server header is derived from the received Contact header parameters, if present. Otherwise, they are taken from the received User-Agent and Server headers.
Pass Through Received Information as User-Agent and Server Header—If selected, the User-Agent and Server header information is passed as User-Agent and Server headers. The User-Agent and Server header is derived from the received Contact header parameters, if present. Otherwise, they are taken from the received User-Agent and Server headers.

Default: Send Unified CM Version Information as User-Agent Header

Version in User Agent and Server Header (Mandatory)

This field specifies the portion of the installed build version that is used as the value of the User Agent and Server Header in SIP requests. Possible values are:

Major and Minor; for example, Cisco-CUCM10.6
Major: for example, Cisco-CUCM10
Major, Minor and Revision; for example, Cisco-CUCM10.6.2
Full Build; for example, Cisco-CUCM10.6.2.98000-19
None; header is omitted

Default: Major and Minor

Dial String Interpretation (Mandatory)

Possible values are:

Phone number consists of characters 0-9, *, #, and + (others treated as URI addresses). This is the default value.
Phone number consists of characters 0-9, A-D, *, #, and + (others treated as URI addresses)
Always treat all dial strings as URI addresses

Redirect by Application (Optional)

If you check this box and configure this SIP Profile on the SIP trunk, the Unified CM administrator can:

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 calls get routed correctly.
Prevent a DOS attack by limiting the number of redirection (recursive redirection) that a service parameter can set.
Allow other features to be invoked while the redirection is taking place.

Getting redirected to a restricted phone number (such as an international number) means that handling redirection at stack level causes the call to be routed, not blocked. This behavior occurs if you leave the Redirect by Application box unchecked.

Disable Early Media on 180 (Optional)

By default, Unified CM signals the calling phone to play local ringback if SDP is not received in the 180 or 183 response. If SDP is included in these responses, instead of playing ringback locally, Unified CM connects media. The calling phone then plays whatever the called device is sending (such as ringback or busy signal). If you receive no ringback, the device you are connecting to may include SDP in the 180 response, but not send media before 200OK response. In this case, check this box to play local ringback on the calling phone and connect the media upon receipt of the 200OK response.

Note

Even though the phone that is receiving ringback is the calling phone, you need the configuration on the called device profile because it determines the behavior.

Outgoing T.38 INVITE include audio mline (Optional)

The parameter 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, configure a SIP trunk with this SIP profile.

Note

The parameter applies to SIP trunks only, not phones that are running SIP or other endpoints.

Use Fully Qualified Domain Name in SIP Requests (Optional)

This feature enables Unified CM to relay a caller's alphanumeric hostname by passing it to the called device or outbound trunk as SIP header information. Enter one of the following:

f—To disable this option. The IP address for Unified CM is passed to the line device or outbound trunk instead of the user’s hostname.

t—To enable this option. Unified CM relays an alphanumeric hostname of a caller by passing it through to the called endpoint as a part of the SIP header information. This enables the called endpoint to return the call using the received or missed call list. If the call originates from a line device on the Unified CM cluster, and is routed on a SIP trunk, then the configured Organizational Top-Level Domain (for example, Cisco.com) is used in the Identity headers, such as From, Remote-Party-ID, and P-Asserted-ID. If the call originates from a trunk on Unified CM and is being routed on a SIP trunk, then:

If the inbound call provides a host or domain in the caller’s information, the outbound SIP trunk messaging preserves the hostname in the Identity headers, such as From, Remote-Party-ID, and P-Asserted-ID.
If the inbound call does not provide a host or domain in the caller's information, the configured Organizational Top-Level Domain is used in the Identity headers, such as From, Remote-Party-ID, and P-Asserted-ID.

Default: f—Disabled

Assured Services SIP conformance (Optional)

Check this box for third-party AS-SIP endpoints and 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.

Table 3. SDP Information Tab

Option

Description

SDP Transparency Profile (Optional)

Displays the SDP Transparency Profile Setting (read only)

Accept Audio Codec Preferences in Received Offer (Optional)

Choose one of the following options:

On—Enables Unified CM to honor the preference of audio codecs in the received offer and preserve it while processing.
Off—Enables Unified CM to ignore the preference of audio codecs in a received offer and apply the locally configured Audio Codec Preference List. The default selects the service parameter configuration.
Default—Selects the service parameter configuration.

Default: Default

Require SDP Inactive Exchange for Mid-Call Media Change (Optional)

This feature determines how Unified CM handles midcall updates to codecs or connection information such as IP address or port numbers.

If you check the box, during midcall codec or connection updates Unified CM sends an INVITE a=inactive SDP message to the endpoint to break the media exchange. This is required if an endpoint is not capable of reacting to changes in the codec or connection information without disconnecting the media. This applies only to audio and video streams within SIP-SIP calls.

Note

For early offer enabled SIP trunks, the Send send-receive SDP in midcall INVITE parameter overrides this parameter.

If this box is unchecked, Unified CM passes the midcall SDP to the peer leg without sending a prior Inactive SDP to break the media exchange.

Default: Unchecked

Allow RR/RS bandwidth modifier (RFC 3556) (Mandatory)

Specifies the RR (RTDP bandwidth allocated to other participants in an RTP session) and RS (RTCP bandwidth allocated to active data senders) in RFC 3556. Options are:

Transport Independent Application Specific bandwidth modifier (TIAS) and AS
TIAS only
AS only
CT only

Default: TIAS and AS

Table 4. Parameters used in Phone Tab

Option

Description

Timer Invite Expires (seconds) (Optional)

This field specifies the time, in seconds, after which a SIP INVITE expires. The Expires header uses this value.

Valid values: Any positive number

Default: 180 seconds

Timer Register Delta (seconds) (Optional)

This field is intended to be used by SIP endpoints only. The endpoint receives this value through a TFTP config file. The endpoint reregisters Timer Register Delta seconds before the registration period ends. The registration period gets determined by the value of the SIP Station KeepAlive Interval service parameter.

Valid values: 0 to 32767

Default: 5 seconds

Timer Register Expires (seconds) (Optional)

This field is intended to be used by SIP endpoints only. The SIP endpoint receives the value through a TFTP config file. This field specifies 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) specifies the default value.

Valid values: Any positive number

Default: 3600 seconds (1 hour)

If the endpoint sends a shorter Expires value than the SIP Station Keepalive Interval service parameter, Unified CM responds with a 423 “Interval Too Brief.”

If the endpoint sends a greater Expires value than the SIP Station Keepalive Interval service parameter, Unified CM responds with a 200 OK with the Keepalive Interval value for Expires.

Note

For mobile phones running SIP, Unified CM uses this value instead of the SIP Station KeepAlive Interval service parameter to determine the registration period.

Note

For TCP connections, the value for the Timer Register Expires field must be lower than the value for the SIP TCP Unused Connection service parameter.

Timer T1 (msec) (Optional)

This field specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages.

Valid values: Any positive number

Default: 500 msec

Timer T2 (msec) (Optional)

This field specifies the highest value, in milliseconds, of the retransmission timer for SIP messages.

Valid values: Any positive number

Default: 4000 msec

Retry INVITE (Optional)

This field specifies the maximum number of times that an INVITE request gets retransmitted.

Valid values: Any positive number

Default: 6

Retry Non-INVITE (Optional)

This field specifies the maximum number of times that a SIP message other than an INVITE request gets retransmitted.

Valid values: Any positive number

Default: 10

Start Media Port (Optional)

This field designates the start real-time protocol (RTP) port for media.

Range: 2048 to 65535

Default: 16384

Stop Media Port (Optional)

This field designates the stop real-time protocol (RTP) port for media.

Range: 2048 to 65535

Default: 32766

Call Pickup URI (Optional)

This URI provides a unique address that the phone that is running SIP sends to Unified CM to invoke the call pickup feature.

Call Pickup Group URI (Optional)

This URI provides a unique address that the phone that is running SIP sends to Unified CM to invoke the call pickup group feature.

Meet Me Service URI (Optional)

This URI provides a unique address that the phone that is running SIP sends to Unified CM to invoke the meet me conference feature.

User Info (Optional)

This field configures the user= parameter in the REGISTER message. Valid values are:

None—No value is inserted
Phone—The value user=phone is inserted in the To, From, and Contact Header for REGISTER
IP—The value user=ip is inserted in the To, From, and Contact Header for REGISTER

Default: None

DTMF DB Level (Optional)

This field specifies the in-band DTMF digit tone level. Valid values are:

6 dB below nominal
3 dB below nominal
Nominal
3 dB above nominal
6 dB above nominal

Default: Nominal

Call Hold Ring Back (Optional)

This parameter causes the phone to ring in cases where you have another party on hold when you hang up a call. Valid values are:

Off—Off permanently and cannot be turned on and off locally by the user interface
On—On permanently and cannot be turned on and off locally by the user interface

Anonymous Call Block (Optional)

The field configures anonymous call block. Valid values are:

Off—Disabled permanently and cannot be turned on and off locally by the user interface
On—Enabled permanently and cannot be turned on and off locally by the user interface

Caller ID Blocking (Optional)

This field configures caller ID blocking. When blocking is enabled, the phone blocks its own number or email address from phones that have caller identification enabled. Valid values are:

Off—Disabled permanently and cannot be turned on and off locally by the user interface
On—Enabled permanently and cannot be turned on and off locally by the user interface

Do Not Disturb Control (Optional)

This field sets the Do Not Disturb (DND) feature. Valid values are:

User—The dndControl parameter for the phone specifies 0.
Admin—The dndControl parameter for the phone specifies 2.

Telnet Level for 7940 and 7960 (Optional)

Cisco Unified IP Phones 7940 and 7960 do not support SSH for sign-in access or HTTP that is used to collect logs. However, these phones support Telnet, which lets the user control the phone, collect debugs, and look at configuration settings. This field controls the telnet_level configuration parameter with the following possible values:

Disabled—No access
Limited—Some access but cannot run privileged commands
Enabled—Full access

Resource Priority Namespace (Optional)

This field enables the administrator to select one of the cluster's defined Resource Priority Namespace network domains for assignment to a line using its SIP Profile.

Timer Keep Alive Expires (seconds) (Optional)

Unified CM requires a keepalive mechanism to support redundancy. This field specifies the interval between keepalive messages sent to the backup Unified CM to ensure its availability for failover.

Default: 120 seconds

Timer Subscribe Expires (seconds) (Optional)

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

Valid values: Any positive number

Default: 120 seconds

Timer Subscribe Delta (seconds) (Optional)

Use this parameter with the Timer Subscribe Expires setting. The phone resubscribes Timer Subscribe Delta seconds before the subscription period ends, as governed by Timer Subscribe Expires.

Range: 3 to 15 seconds

Default: 5 seconds

Maximum Redirections (Optional)

Use this configuration variable to determine the maximum number of times that the phone allows a call to be redirected before dropping the call.

Default: 70 redirections

Off hook To First Digit Timer (msec) (Optional)

This field specifies the time in microseconds that passes when the phone goes off hook and the first digit timer gets set.

Range: 0 to 15,000 microseconds

Default: 15,000 microseconds

Call Forward URI (Optional)

This URI provides a unique address that the phone that is running SIP sends to Unified CM to invoke the call forward feature.

Speed Dial (Abbreviated Dial) URI (Optional)

This URI provides a unique address that the phone that is running SIP sends to Unified CM to invoke the abbreviated dial feature.

Speed dials that are not associated with a line key (abbreviated dial indices) do not download to the phone. The phone uses the feature indication mechanism (INVITE with Call-Info header) to indicate when an abbreviated dial number has been entered. The request URI contains the abbreviated dial digits (for example, 14), and the Call-Info header indicates the abbreviated dial feature. Unified CM translates the abbreviated dial digits into the configured digit string and extends the call with that string. If no digit string has been configured for the abbreviated dial digits, a 404 Not Found response gets returned to the phone.

Conference Join Enabled (Optional)

Check this box to join the remaining conference participants when a conference initiator using a Cisco Unified IP Phone 7940 or 7960 hangs up. Leave it unchecked if you do not want to join the remaining conference participants.

Note

This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.

RFC 2543 Hold (Optional)

Check this box to enable setting connection address to 0.0.0.0 per RFC2543 when call hold is signaled to Unified CM. This allows backward compatibility with endpoints that do not support RFC3264.

Semi Attended Transfer (Optional)

This check box determines whether the Cisco Unified IP Phones 7940 and 7960 caller can transfer an attended transfer's second leg while the call is ringing. Check the box if you want semiattended transfer enabled; leave it unchecked if you want semiattended transfer disabled

Note

This check box applies to the Cisco Unified IP Phones 7941/61/70/71/11 when they are in SRST mode only.

Enable VAD (Optional)

Check this box if you want voice activation detection (VAD) enabled; leave it unchecked if you want VAD disabled. When VAD is enabled, no media is sent when voice is detected.

Stutter Message Waiting (Optional)

Check this box if 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.

This setting supports Cisco Unified IP Phones 7960 and 7940 that run SIP.

MLPP User Authorization (Optional)

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

Table 5. Normalization Script Tab

Option

Description

Normalization Script

From the drop-down list box, choose the script that you want to apply to this SIP profile.

To import another script from Unified CM, go to the SIP Normalization Configuration window (Device > Device Settings > SIP Normalization Script), and import a new script.

Enable Trace

Check this box to enable tracing within the script or uncheck this box to disable tracing. When checked, the trace.output API provided to the Lua scripter produces SDI trace.

Note

We recommend that you only enable tracing while debugging a script. Tracing impacts performance and is not recommended under normal operating conditions.

Script Parameters

Enter parameter names and parameter values in the Script Parameters box as comma-delineated key-value pairs. Valid values include all characters except equals signs (=), semicolons (;), and nonprintable characters, such as tabs. You can enter a parameter name with no value.

Alternatively, to add another parameter line from Unified CM, click the + (plus) button. To delete a parameter line, click the - (minus) button.

Table 6. Incoming Requests FROM URI Settings Tab
Option	Description
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.

Option

Description

Reroute Incoming Request to new Trunk based on

Unified CM only accepts calls from a SIP device whose IP address matches the destination address of the configured SIP trunk. In addition, the port on which the SIP message arrives must match the one that is configured on the SIP trunk. After Unified CM accepts the call, Unified CM uses the configuration for this setting to determine whether to reroute the call to another trunk.

From the drop-down list box, choose the method that Unified CM uses to identify the SIP trunk where the call gets rerouted:

Never—If the SIP trunk matches the IP address of the originating device, choose this option. Unified CM, which identifies the trunk by the incoming packet's source IP address and the signaling port number, does not route the call to a different (new) SIP trunk. The call occurs on the SIP trunk on which the call arrived.
Contact Info Header—If the SIP trunk uses a SIP proxy, choose this option. Unified CM parses the IP address or domain name and the signaling port number in the incoming request's header. Unified CM then reroutes the call to the SIP trunk using that IP address and port. If no SIP trunk is identified, the call occurs on the trunk where the call arrived.
Call-Info Header with purpose=x-cisco-origIP—If the SIP trunk uses a Customer Voice Portal (CVP) or a Back-to-Back User Agent (B2BUA), choose this option. When the incoming request is received, Unified CM performs the following:
- parses the Call-Info header
- looks for the parameter purpose=x-cisco-origIP
- uses the IP address or domain name and signaling port number in the header to reroute the call to the SIP trunk using the IP address and port
If the parameter is not in the header, or no SIP trunk is identified, the call occurs on the SIP trunk where the call arrived.

Default: Never

Note

This setting does not work for SIP trunks connected to:

A Unified CM IM and Presence Service proxy server.
Originating gateways in different Unified CM groups.

RSVP Over SIP

This field configures RSVP over SIP trunks. From the drop-down list box, choose the method that Unified CM uses to configure RSVP over SIP trunks:

Local RSVP—In a local configuration, RSVP occurs within each cluster, between the endpoint and the local SIP trunk, but not on the WAN link between the clusters.
E2E—In an end-to-end (E2E) configuration, RSVP occurs on the entire path between the endpoints, including within the local cluster and over the WAN.

Resource Priority Namespace List

Select a configured Resource Priority Namespace list from the drop-down menu. The Namespace List is configured in Unified CM in the Resource Priority Namespace List menu. You can access the menu in Unified CM from System > MLPP > Namespace.

Fall back to local RSVP

Check this box if you want to allow failed end-to-end RSVP calls to fall back to local RSVP to establish the call. If this box is not checked, end-to-end RSVP calls that cannot establish an end-to-end connection fail.

SIP Rel1XX Options

This field configures SIP Rel1XX, which determines whether all SIP provisional responses (other than 100 Trying messages) are sent reliably to the remote SIP endpoint. Valid values are:

Disabled—Disables SIP Rel1XX.
Send PRACK if 1XX contains SDP—Acknowledges a 1XX message with PRACK, only if the 1XX message contains SDP.
Send PRACK for all 1XX messages—Acknowledges all1XX messages with PRACK.

If you set the RSVP Over SIP field to E2E, you cannot choose Disabled.

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:

Immersive—High-definition immersive video
Desktop—Standard desktop video.
Mixed—A mix of immersive and desktop video.

Unified CM Locations Call Admission Control (CAC) reserves bandwidth from two Locations video bandwidth pools, Video Bandwidth and Immersive Bandwidth. The pool used depends on the type of call determined by the Video Call Traffic Class. Refer to the “Call Admission Control” chapter of the Cisco Unified Communications Manager System Guide for more information.

Calling Line Identification Presentation (Mandatory)

Select one of:

Strict From URI presentation Only—To select the network-provided identity
Strict Identity Headers presentation Only—To select the user-provided identity
Default—To select the system default calling line identification

Default: Default

Session Refresh Method (Mandatory)

Session Timer with Update: The session refresh timer allows for periodic refresh of SIP sessions. This allows the Unified CM and remote agents to determine whether the SIP session is still active. Prior to Release 10.01, when the Unified CM received a refresh command, it supported receiving either Invite or Update SIP requests to refresh the session. When the Unified CM initiated a refresh, it supported sending only Invite SIP requests to refresh the session. With Release 10.01, this feature extends the refresh capability so that Unified CM can send both Update and Invite requests.

Specify whether to use Invite or Update as the Session Refresh Method.

Default: Invite

Note

Sending a midcall Invite request requires specifying an offer SDP in the request. This means that the far end must send an answer SDP in the Invite response.

Update: Unified CM requests a SIP Update if the SIP session's far end supports the Update method in the Supported or Require headers. When sending the Update request, the Unified CM includes an SDP. This simplifies the session refresh since no SDP offer or answer exchange is required.

Note

If the far end of the SIP session does not support the Update method, the Unified CM continues using the Invite method for session refresh.

Enable ANAT

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

Checking the Enable ANAT and MTP Required boxes sets Unified CM to insert a dual-stack MTP and send an offer with two m-lines, for IPv4 and IPv6. If a dual- stack MTP cannot be allocated, Unified CM sends an INVITE without SDP.

When you check the Enable ANAT check box and the Media Termination Point Required check box is unchecked, Unified CM sends an INVITE without SDP.

When the Enable ANAT and MTP Required boxes are unchecked (or when an MTP cannot be allocated), Unified CM sends an INVITE without SDP.

When you uncheck the Enable ANAT box but you check the MPT Required box, consider the information, which assumes that an MTP can be allocated:

Unified CM sends an IPv4 address in the SDP for SIP trunks with an IP Addressing Mode of IPv4 Only.
Unified CM sends an IPv6 address in the SDP for SIP trunks with an IP Addressing Mode of IPv6 Only.
For dual-stack SIP trunks, Unified CM determines which IP address type to send in the SDP based on the configuration for the IP Addressing Mode Preference for Media enterprise parameter.

Deliver Conference Bridge Identifier

When checked, the SIP trunk passes the b-number identifying the conference bridge across the trunk instead of changing the b-number to the null value.

The terminating side does not require this field.

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.

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.

Send ILS Learned Destination Route String

When this box is checked, for calls routed to a learned directory URI, learned number, or learned pattern, Unified CM:

adds the x-cisco-dest-route-string header to outgoing SIP INVITE and SUBSCRIBE messages
inserts the destination route string into the header

When this box is left unchecked, Unified CM does not add the x-cisco-dest-route-string header to any SIP messages.

The x-cisco-dest-route-string header allows Unified CM to route calls across a Session Border Controller.

Table 8. Trunk SIP OPTIONS Ping Tab
Option	Description
Enable OPTIONS Ping to monitor destination status for Trunks with Service Type "None (Default)"	Check this box if you want to enable the SIP OPTIONS feature. SIP OPTIONS are requests to the configured destination address on the SIP trunk. If the remote SIP device is unresponsive or returns a SIP error response such as 503 Service Unavailable or 408 Timeout, Unified CM reroutes the calls by using other trunks or a different address. If this box is not checked, the SIP trunk does not track the status of SIP trunk destinations. When this box is checked, you can configure two request timers.
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. If at least one IP address is available, the trunk is In Service; if all IP addresses are unavailable, the trunk is Out of Service. Default: 60 seconds Range: 5 to 600 seconds
Ping Interval for Out-of-service 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. The remote peer may be marked as Out of Service if: it fails to respond to OPTIONS it sends 503 or 408 responses the Transport Control Protocol (TCP) connection cannot be established If at least one IP address is available, the trunk is In Service; if all IP addresses are unavailable, the trunk is Out of Service. Default: 120 seconds Range: 5 to 600 seconds
Ping Retry Timer (msec)	This field specifies the maximum waiting time before retransmitting the OPTIONS request. Range: 100 to 1000 milliseconds Default: 500 milliseconds
Ping Retry Count	This field specifies the number of times that Unified CM resends the OPTIONS request to the remote peer. After the configured retry attempts are used, the destination is considered to have failed. To obtain faster failure detection, keep the retry count low. Range: 1 to 10 Default: 6

Table 9. Trunk SDP Information Tab

Option

Description

Send send-receive SDP in midcall INVITE

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

Note

This check box applies only to early offer enabled SIP trunks and has no impact on SIP line calls.

When you enable Send send-receive SDP in midcall INVITE for an early offer SIP trunk in tandem mode, Unified CM inserts MTP to provide sendrecv SDP when a SIP device sends offer SDP with a=inactive or sendonly or recvonly in audio media line. In tandem mode, Unified CM depends on the SIP devices to reestablish media path by sending either a delayed INVITE or midcall INVITE with send-recv SDP.

When you enable Send send-receive SDP in midcall INVITE and Require SDP Inactive Exchange for Mid-Call Media Change on the same SIP Profile, the Send send-receive SDP in midcall INVITE overrides the Require SDP Inactive Exchange for Mid-Call Media Change, so Unified CM does not send an INVITE with a=inactive SDP in midcall codec updates. For SIP line side calls, the Require SDP Inactive Exchange for Mid-Call Media Change check box applies when enabled.

Note

To prevent the SDP mode from being set to inactive in a multiple-hold scenario, set the Duplex Streaming Enabled clusterwide service parameter in Unified CM (System > Service Parameters) to True.

Allow Presentation Sharing using BFCP

If the box is checked, Unified CM allows supported SIP endpoints to use the Binary Floor Control Protocol (BFCP) to enable presentation sharing.

The use of BFCP creates an added media stream in addition to the existing audio and video streams. This additional stream is used to stream a presentation, such as a PowerPoint presentation from someone’s laptop, into a SIP videophone.

If the box is unchecked, Unified CM rejects BFCP offers from devices associated with the SIP profile. The BFCP application line and associated media line ports are set to 0 in the answering SDP message.

Default: Unchecked

Note

BFCP is only supported on SIP networks. BFCP must be enabled on all SIP trunks, lines, and endpoints for presentation sharing to work. BFCP is not supported if the SIP line or SIP trunk uses MTP, RSVP, TRP, or Transcoder.

For more information on BFCP, refer to the Cisco Unified Communications Manager System Guide.

Allow iX Application Media

Check this box to enable support for iX media channel.

Allow multiple codecs in answer SDP

This option applies when incoming SIP signals do not indicate support for multiple codec negotiation and Unified CM can finalize the negotiated codec.

When this box is checked, the endpoint behind the trunk can handle multiple codecs in the answer SDP.

For example, an endpoint that supports multiple codec negotiation calls the SIP trunk, and Unified CM sends a Delay Offer request to a trunk. The endpoint behind the trunk returns all support codecs without the Contact header to indicate the support of multiple codec negotiation.

In this case, Unified CM identifies that the trunk can handle multiple codec negotiation, and sends SIP response messages to both endpoints with multiple common codecs.

When unchecked, Unified CM identifies that the endpoint behind the trunk cannot handle multiple codec negotiation, unless SIP contact header URI states it can. Unified CM continues the call with single codec negotiation.

Configure SIP Trunk Security Profiles

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Make sure that the hierarchy path is set to the node where the Cisco Unified Communications Manager is configured.

Step 3

Perform one of the following:

If you signed in as the provider or reseller administrator, select Device Management > CUCM > SIP Trunk Security Profiles.
If you signed in as the customer administrator, select Device Management > Advanced > SIP Trunk Security Profiles.

Step 4

Perform one of the following:

To add a new SIP trunk security profile, click Add, then go to Step 5.
To edit an existing SIP trunk security profile, click the SIP trunk security profile to be updated. Go to Step 6.

Step 5

If the Network Device List pop-up window appears, select the NDL for the SIP trunk security profile from the pulldown menu. The window appears when you are on a non-site hierarchy node. If you are at a site hierarchy node, the NDL associated with the site is automatically used.

Note

The Network Device List pulldown menu appears when a SIP trunk security profile is added. It does not appear when you edit a SIP trunk security profile.

Step 6

Enter a unique name for the new SIP trunk security profile in the Name field, or modify the existing name if desired. This field is required.

Step 7

Complete the other fields as appropriate.

The V.150 Outbound SDP Offer Filtering field is required.

For details on each field, see SIP Trunk Security Profile Field Descriptions.

Step 8

To save a new SIP trunk security profile, click Save. To save an updated SIP trunk security profile, click Update.

SIP Trunk Security Profile Field Descriptions

Option

Description

Name (Mandatory)

Enter a name for the security profile. When you save the new profile, the name displays in the SIP Trunk Security Profile drop-down list in the Trunk Configuration window. The maximum length for the name is 64 characters.

Description (Optional)

Enter a description for the security profile. 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 (<>).

Device Security Mode (Optional)

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

Non Secure—No security features except image authentication apply. A TCP or UDP connection opens to Cisco Unified Communications Manager.
Authenticated—Unified CM provides integrity and authentication for the trunk. A TLS connection that uses NULL/SHA opens.
Encrypted—Unified CM provides integrity, authentication, and signaling encryption for the trunk. A TLS connection that uses AES128/SHA opens for signaling.

Incoming Transport Type (Optional)

Select one of

TCP+UDP
UDP
TLS
TCP

If you do not specify an incoming transport type, TCP+UDP is assigned.

When Device Security Mode is Non Secure, TCP+UDP specifies the transport type.

When Device Security Mode is Authenticated or Encrypted, TLS specifies the transport type.

Note

The Transport Layer Security (TLS) protocol secures the connection between Unified CM and the trunk.

Outgoing Transport Type (Optional)

From the drop-down list box, choose the outgoing transport mode. Select one of

TCP+UDP
UDP
TLS
TCP

When Device Security Mode is Non Secure, choose TCP or UDP.

When Device Security Mode is Authenticated or Encrypted, TLS specifies the transport type.

Note

TLS ensures signaling integrity, device authentication, and signaling encryption for SIP trunks.

Tip

Use UDP as the outgoing transport type when connecting SIP trunks between Unified CM systems and IOS gateways that do not support TCP connection reuse. See “Understanding Session Initiation Protocol (SIP)” in the Cisco Unified Communications Manager System Guide for more information.

Enable Digest Authentication (Optional)

Check this check box to enable digest authentication. If you check this check box, Unified CM challenges all SIP requests from the trunk.

Digest authentication does not provide device authentication, integrity, or confidentiality. Choose a security mode of Authenticated or Encrypted to use these features.

Tip

Use digest authentication to authenticate SIP trunk users on trunks that are using TCP or UDP transport.

Nonce Validity Time (mins) (Optional)

Enter the number of minutes (in seconds) that the nonce value is valid. When the time expires, Unified CM generates a new value.

Note

A nonce value (a random number that supports digest authentication) is used to calculate the MD5 hash of the digest authentication password.

Default is 600 minutes. If you do not specify a Nonce Validity Time, the default of 600 minutes is assigned.

X.509 Subject Name (Optional)

This field applies if you configured TLS for the incoming and outgoing transport type.

For device authentication, enter the subject name of the X.509 certificate for the SIP trunk device. If you have a Unified CM cluster or if you use SRV lookup for the TLS peer, a single trunk may resolve to multiple hosts. This situation results in multiple X.509 subject names for the trunk. If multiple X.509 subject names exist, enter one of the following characters to separate the names: space, comma, semicolon, or a colon.

You can enter up to 4096 characters in this field.

Tip

The subject name corresponds to the source connection TLS certificate. Ensure that subject names are unique for each subject name and port. You cannot assign the same subject name and incoming port combination to different SIP trunks.

Example:

SIP TLS trunk1 on port 5061 has X.509 Subject Names my_cm1, my_cm2.

SIP TLS trunk2 on port 5071 has X.509 Subject Names my_cm2, my_cm3.

SIP TLS trunk3 on port 5061 can have X.509 Subject Name my_ccm4 but cannot have X.509 Subject Name my_cm1.

Incoming Port (Optional)

Choose the incoming port. Enter a value that is a unique port number from 0 to 65535. The value that you enter applies to all SIP trunks that use the profile.

The default port value for incoming TCP and UDP SIP messages is 5060. The default SIP secured port for incoming TLS messages is 5061.

If the incoming port is not specified, the default port of 5060 is used.

Tip

All SIP trunks that use TLS can share the same incoming port; all SIP trunks that use TCP + UDP can share the same incoming port. You cannot mix SIP TLS transport trunks with SIP non-TLS transport trunk types on the same port.

Enable application level authorization (Optional)

Application-level authorization applies to applications that are connected through the SIP trunk.

If you check this box, also check the Enable Digest Authentication check box and configure digest authentication for the trunk. Unified CM authenticates a SIP application user before checking the allowed application methods.

When application level authorization is enabled, trunk-level authorization occurs first, and application-level authorization occurs second. Unified CM checks the methods authorized for the trunk (in this security profile) before the methods authorized for the SIP application user in the Application User Configuration window.

Tip

Consider using application-level authorization if you do not trust the identity of the application or if the application is not trusted on a particular trunk. Application requests may come from a different trunk than you expect.

For more information about configuring application level authorization at the Application User Configuration window, see the Cisco Unified Communications Manager Administration Guide.

Accept presence subscription (Optional)

If you want Unified CM to accept presence subscription requests that come through the SIP trunk, check this box.

If you checked Enable Application Level Authorization, go to the Application User Configuration window and check Accept Presence Subscription for any application users authorized for this feature.

When application-level authorization is enabled, if you check Accept Presence Subscription for the application user but not for the trunk, a 403 error message is sent to the SIP user agent connected to the trunk.

Accept out-of-dialog refer (Optional)

If you want Unified CM to accept incoming non-INVITE, Out-of-Dialog REFER requests that come through the SIP trunk, check this box.

If you checked Enable Application Level Authorization, go to the Application User Configuration window and check Accept Out-of-Dialog Refer for any application users authorized for this method.

Note

If this profile is associated with an EMCC SIP trunk, Accept Out-of-Dialog REFER is enabled regardless of the setting on this page.

Accept unsolicited notification (Optional)

If you want Unified CM to accept incoming non-INVITE, unsolicited notification messages that come through the SIP trunk, check this box.

If you checked Enable Application Level Authorization, go to the Application User Configuration window and check Accept Unsolicited Notification for any application users authorized for this method.

Accept replaces header (Optional)

If you want Unified CM to accept new SIP dialogs, which have replaced existing SIP dialogs, check this check box.

If you checked Enable Application Level Authorization, go to the Application User Configuration window and check Accept Header Replacement for any application users authorized for this method.

Transmit security status (Optional)

If you want Unified CM to send the security icon status of a call from the associated SIP trunk to the SIP peer, check this box.

Default=Not Checked

Allow charging header (Optional)

If you want to allow RFC 3455 SIP charging headers in transactions (for example, where billing information is passed in the headers for prepaid accounts), check this box. If unchecked, RFC 3455 SIP charging headers are not allowed in sessions that use the SIP profile.

Default=Not Checked

SIP V.150 Outbound SDP Offer Filtering (Mandatory)

Select one of the following filter options from the drop-down list:

Use Default Filter—The SIP trunk uses the default filter that is indicated in the SIP V.150 Outbound SDP Offer Filtering service parameter. To locate the service parameter, go to System > Service Parameters > Clusterwide Parameters (Device-SIP) in Unified CM Administration.
No Filtering—The SIP trunk performs no filtering of V.150 SDP lines in outbound offers.
Remove MER V.150—The SIP trunk removes V.150 MER SDP lines in outbound offers. Select this option to reduce ambiguity when the trunk is connected to a pre-MER V.150 Unified CM.
Remove Pre-MER V.150—The SIP trunk removes any non-MER compliant V.150 lines in outbound offers. Select this option to reduce ambiguity when your cluster is in a network of MER-compliant devices that cannot process offers with pre-MER lines.

Default: Use Default Filter

Configure SIP Trunks

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Ensure that the hierarchy path is set to the node where the Cisco Unified Communications Manager is configured.

Step 3

Perform one of the following:

If you logged in as the provider or reseller administrator, select Device Management > CUCM > SIP Trunks.
If you logged in as the Customer Administrator, select Device Management > Advanced > SIP Trunks.

Step 4

Perform one of the following:

To add a new SIP trunk, click Add.
To edit an existing SIP trunk, click the SIP trunk in the far left column. Then, click Modify to edit the selected SIP trunk.

Step 5

From the CUCM pulldown menu, select the hostname, domain name, or IP address of the Unified CM to which you want to add the SIP trunk.

Note

The CUCM pulldown menu only appears when a SIP trunk is added. It does not appear when you edit a SIP trunk.

Important

The CUCM pulldown menu shows, in addition to the Unified CM located at the node, ALL the Unified CM nodes in the hierarchies above the node you are adding the SIP trunk. To provision a Unified CM server, refer to the “Installation Tasks” section of Installing Cisco Unified Communications Manager.

Step 6

Enter a unique name for the new SIP trunk in the Device Name field, or modify the existing device name if desired.

Step 7

Complete the fields on each tab as appropriate.

The following fields on each tab are required:

Device Information tab
- Device Name
- Trunk Service Type
- Call Classification
- Location
- Use Trusted Relay Point
Call Routing General tab
- SIP Privacy
Call Routing Inbound tab
- Significant Digits
- Connected Line ID Presentation
- Connected Name Presentation
Call Routing Outbound tab
- Calling Party Selection
- Calling Line ID Selection
- Calling Name Presentation
- Calling and Connected Party Info Format
SP Info tab
- Destination - Destination IPv4
- Destination - Destination IPv6 (if Destination - Destination IPv4 field is not completed)
- Destination - Destination Port
- Sort Order
- MTP Preferred Originating Codec
- BLF Presence Group
- SIP Trunk Security Profile
- SIP Profile
- DTMF Signaling Method
GeoLocation tab: no required fields

For details on each tab's fields, see SIP Trunks Field Descriptions.

Step 8

To save a new SIP trunk, click Save. To save an updated SIP trunk, click Update.

The SIP trunk appears in the SIP trunk list. To view the SIP trunk and its characteristics, sign in to the Unified CM where the SIP trunk was added. Then, select Device > Trunk, and perform the find operation. When you click the name of the SIP trunk in the list, the trunk characteristics are displayed.

Note

The SIP trunk is automatically reset on the Unified CM when it is added. To reset the SIP trunk at any other time, perform Reset SIP Trunks. For more details on configuring SIP trunk, refer the contact center documentation at http://www.cisco.com/c/en/us/support/unified-communications/hosted-collaboration-solution-contact-center/tsd-products-support-series-home.html.

SIP Trunks Field Descriptions

Table 10. Device Information Tab

Option

Description

Device Name

(Mandatory)

Enter a unique identifier for the trunk using up to 50 alphanumeric characters: A-Z, a-z, numbers, hyphens (-) and underscores (_) only.

Default value: None

Trunk Service Type

(Mandatory)

Select one of from the following:

None—Choose this option if the trunk is not used for call control discovery, Extension Mobility Cross Cluster, or Cisco Intercompany Media Engine
Call Control Discovery—Choose this option to enable the trunk to support call control discovery.
Extension Mobility Cross Cluster—Choose 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.
Intercompany Media Engine—Ensure that the Cisco IME server is installed and available before you configure this field.
IP Multimedia Subsystem Service Control (ISC)—Choose this option to enable the trunk to support IP multimedia subsystem service control.

Default value: None (Default)

Description (Optional)

Enter a descriptive name for the trunk using up to 114 characters in any language, but not including double-quotes ("), percentage sign (%), ampersand (&), backslash (\), or angle brackets (<>).

Default value: empty

Device Pool

Choose the appropriate device pool for the trunk. For trunks, device pools specify a list of Cisco Unified Communications Managers (Unified CMs) that the trunk uses to distribute the call load dynamically.

Note

Calls that are initiated from a phone that is registered to a Unified CM that does not belong to the device pool of the trunk use different Unified CMs of this device pool for different outgoing calls. Selection of Unified CM nodes occurs in a random order. A call that is initiated from a phone that is registered to a Unified CM that does belong to the device pool of the trunk uses the same Unified CM node for outgoing calls if the Unified CM is up and running.

Default value: Default

Common Device Configuration

(Optional)

Choose the common device configuration to which you want this trunk assigned. The common device configuration includes the attributes (services or features) that are associated with a particular user.

Default value: None

Call Classification

(Mandatory)

This parameter determines whether an incoming call through this trunk is considered off the network (OffNet) or on the network (OnNet). When the Call Classification field is configured as Use System Default, the setting of the Unified CM clusterwide service parameter, Call Classification, determines whether the trunk is OnNet or OffNet. This field provides an OnNet or OffNet alerting tone when the call is OnNet or OffNet, respectively.

Default value: Use System Default

Media Resource Group List

(Optional)

This list provides a prioritized grouping of media resource groups. An application chooses the required media resource, such as a Music On Hold server, from among the available media resources according to the priority order that a Media Resource Group List defines.

Default value: None

Location

(Mandatory)

Use locations to implement call admission control (CAC) in a centralized call-processing system. CAC enables you to regulate audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls over links between locations. The location specifies the total bandwidth that is available for calls to and from this location.

Select the appropriate location for this trunk:

Hub_None—Specifies that the locations feature does not keep track of the bandwidth that this trunk consumes.
Phantom—Specifies a location that enables successful CAC across intercluster trunks that use H.323 protocol or SIP.
Shadow—Specifies a location for intercluster enhanced location CAC. Valid for SIP intercluster trunks (ICT) only.

Default value: Hub_None

AAR Group

(Optional)

Choose the automated alternate routing (AAR) group for this device. The AAR group provides the prefix digits that are used to route calls that are otherwise blocked due to insufficient bandwidth. An AAR group setting of None specifies that no rerouting of blocked calls is attempted.

Default value: None

Tunneled Protocol

Select the QSIG option if you want to use SIP trunks or SIP gateways to transport (tunnel) QSIG messages from Unified CM to other PINXs. QSIG tunneling supports the following features: Call Back, Call Completion, Call Diversion, Call Transfer, Identification Services, Path Replacement, and Message Waiting Indication (MWI).

Note

Remote-Party-ID (RPID) headers coming in from the SIP gateway can interfere with QSIG content and cause unexpected behavior with Call Back capabilities. To prevent interference with the QSIG content, turn off the RPID headers on the SIP gateway.

Default value: None

QSIG Variant

To display the options in the QSIG Variant drop-down list box, select QSIG from the Tunneled Protocol pulldown menu. This parameter specifies the protocol profile that is sent in outbound QSIG facility information elements.

From the pulldown menu, select one of the following:

No Changes—Default. Keep this parameter set to the default value unless a Cisco support engineer instructs otherwise.
Not Selected
ECMA—Select for ECMA PBX systems that use Protocol Profile 0x91
ISO—Select for PBX systems that use Protocol Profile 0x9F

Default value: No Changes

ASN.1 ROSE OID Encoding

To display the options in the ASN.1 ROSE OID Encoding pulldown menu, choose QSIG from the Tunneled Protocol pulldown menu. This parameter specifies how to encode the Invoke Object ID (OID) for remote operations service element (ROSE) operations.

From the pulldown menu, select one of the following:

No Changes—Keep this parameter set to the default value unless a Cisco support engineer instructs otherwise.
Not Selected
Use Global Value ECMA—If you selected the ECMA option from the QSIG Variant pulldown menu, select this option.
Use Global Value ISO—If you selected the ISO option from the QSIG Variant pulldown menu, select this option.
Use Local Value

Default value: No Changes

Packet Capture Mode

This setting exists for troubleshooting encryption only; packet capturing may cause high CPU usage or call-processing interruptions.

From the drop-down menu, select one of the following:

None—This option, which serves as the default setting, indicates that no packet capturing is occurring. After you complete packet capturing, configure this setting.
Batch Processing Mode—Unified CM writes the decrypted or nonencrypted 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. Unified CM, which stores the file for seven days, also stores the keys that encrypt the file in a secure location. Unified CM 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 the message. The TAC debugging 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. Before you contact TAC, you must capture the SRTP packets by using a sniffer trace between the affected devices.

Default value: None

Packet Capture Duration

(Optional)

This setting exists for troubleshooting encryption only; packet capturing may cause high CPU usage or call-processing interruptions. This field specifies the maximum number of minutes that is allotted for one session of packet capturing.

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

Default value: 0 (zero), Range is from 0 to 300 minutes

Media Termination Point Required

(Optional)

You can configure Unified CM SIP trunks to always use an Media Termination Point (MTP). Check this box to provide media channel information in the outgoing INVITE request. When this check box is checked, all media channels must terminate and reoriginate on the MTP device. If you uncheck the check box, the Unified CM can decide whether calls are to go through the MTP device or be connected directly between the endpoints.

Note

If the check box remains unchecked, Unified CM attempts to dynamically allocate an MTP if the DTMF methods for the call legs are not compatible. For example, existing phones that run SCCP support only out-of-band DTMF, and existing phones that run SIP support RFC2833. Because the DTMF methods are not identical, the Unified CM dynamically allocates an MTP. If, however, a new phone that runs SCCP, which supports RFC2833 and out-of band, calls an existing phone that runs SIP, Unified CM does not allocate an MTP because both phones support RFC2833. So, by having the same type of DTMF method supported on each phone, there is no need for MTP.

Default value: False (Unchecked)

Retry Video Call as Audio

(Optional)

This check box pertains to outgoing SIP trunk calls and does not impact incoming calls. 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 does not try to establish as an audio call. The call then fails to call control, and call control routes the call using Automatic Alternate Routing (AAR) and route list or hunt list.

Default value: True (Checked)

Path Replacement Support

(Optional)

This check box is relevant when you select QSIG from the Tunneled Protocol pulldown menu. This setting works with QSIG tunneling to ensure that non-SIP information gets sent on the leg of the call that uses path replacement.

Default value: False (Unchecked)

Transmit UTF-8 for Calling Party Name

(Optional)

This device uses 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.

Note

The phone may display malformed characters if the two ends of the trunk are configured with user locales that do not belong to the same language group.

Default value: False (Unchecked)

Transmit UTF-8 Names for QSIG APDU

(Optional)

This device uses 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 and encodes in UTF-8 format. If the user locale settings do not match, the device sends ASCII and encodes in UTF-8 format. If the configuration parameter is not set and the user locale setting in the device pool matches the terminating phone user locale, the device sends unicode (if the name uses 8-bit format) and encodes in ISO8859-1 format.

Default value: False (Unchecked)

Unattended Port

(Optional)

Check this check box if calls can be redirected and transferred to an unattended port, such as a voice mail port.

Default value: False (Unchecked)

SRTP Allowed

(Optional)

Check this check box if you want Unified CM to allow secure and nonsecure media calls over the trunk. Checking this check box enables Secure Real-Time Protocol (SRTP) SIP Trunk connections and also allows the SIP trunk to fall back to Real-Time Protocol (RTP) if the endpoints do not support SRTP. If you do not check this check box, Unified CM prevents SRTP negotiation with the trunk and uses RTP negotiation instead.

Caution

If you check this check box, Cisco strongly recommends that you use an encrypted TLS profile, so that keys and other security related information do not get exposed during call negotiations. If you use a non-secure profile, SRTP still works but the keys get exposed in signaling and traces. In that case, you must ensure the security of the network between Unified CM and the destination side of the trunk.

Default value: False (Unchecked)

Consider Traffic on This Trunk Secure

This field provides an extension to the existing security configuration on the SIP trunk, which enables a SIP trunk call leg to be considered secure if SRTP is negotiated, independent of the signaling transport.

From the pulldown menu, select one of the following:

When using both sRTP and TLS
When using sRTP Only—Displays when you check the SRTP Allowed check box

Default value: When using both sRTP and TLS

Route Class Signaling Enabled

From the pulldown menu, enable or disable route class signaling for the port. Route class signaling communicates special routing or termination requirements to receiving devices. It must be enabled for the port to support the Hotline feature.

From the pulldown menu, select one of the following:

Default—The device uses the setting from the Route Class Signaling service parameter
Off—Enables route class signaling. This setting overrides the Route Class Signaling service parameter
On—Disables route class signaling. This setting overrides the Route Class Signaling service parameter.

Default value: Default

Use Trusted Relay Point

(Mandatory)

From the drop-down menu, enable or disable whether Unified CM inserts a trusted relay point (TRP) device with this media endpoint. A Trusted Relay Point (TRP) device designates an MTP or transcoder device that is labeled as Trusted Relay Point. Unified CM 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 gets used as the required MTP. If both TRP and RSVPAgent are needed for the endpoint, Unified CM first tries to find an RSVPAgent that can also be used as a TRP. If both TRP and transcoder are needed for the endpoint, Unified CM first tries to find a transcoder that is also designated as a TRP.

Select one of the following:

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.

Default value: Default

PSTN Access

(Optional)

If you use the Cisco Intercompany Media Engine feature, check this check box to indicate that 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.

Default value: True (Checked)

Run On All Active Unified CM Nodes

(Optional)

Check this check box to enable the trunk to run on every node.

Default value: False (Unchecked)

Table 11. Call Routing General Tab

Option

Description

Remote-Party-ID

(Optional)

Use this check box to allow or disallow the SIP trunk to send the Remote-Party-ID (RPID) header in outgoing SIP messages from Unified CM to the remote destination. If you check this box, the SIP trunk always sends the RPID header. If you do not check this box, the SIP trunk does not send the RPID header.

Note

Be aware that Calling Name Presentation, Connected Line ID, and Connected Name Presentation are not available when QSIG tunneling is enabled.

Outgoing SIP Trunk Calls

The configured values of the Calling Line ID Presentation and Calling Name Presentation provide the basis for the construction of the Privacy field of the RPID header. Each of these two options can have the values of Default, Allowed, or Restricted. If either option is set to Default, the corresponding information (Calling Line ID Presentation and/or Calling Name Presentation) in the RPID header comes from the Call Control layer (which is based on call-by-call configuration) within Unified CM. If either option is set to Allowed or Restricted, the corresponding information in the RPID header comes from the SIP trunk configuration window.

Incoming SIP Trunk Calls

The configured values of the Connected Line ID Presentation and Connected Name Presentation provide the basis for the construction of the Privacy field of the RPID header. Each of these two options can have the values of Default, Allowed, or Restricted.

Be aware that the Connected Line ID Presentation and Connected Name Presentation options are relevant for 180/200 messages that the SIP trunk sends in response to INVITE messages that Unified CM receives. If either option is set to Default, the corresponding information (Connected Line ID Presentation and/or Connected Name Presentation) in the RPID header comes from the Call Control layer (which is based on call-by-call configuration) within Unified CM. If either option is set to Allowed or Restricted, the corresponding information in the RPID header comes from the SIP trunk configuration window.

Note

The Remote-party ID and Asserted Identity options represent independent mechanisms for communication of display-identity information.

Default value: True (Checked)

Asserted-Identity

(Optional)

Use this check box to allow or disallow the SIP trunk to send the Asserted-Type and SIP Privacy headers in SIP messages. If you check this check box, the SIP trunk always sends the Asserted-Type header; whether or not the SIP trunk sends the SIP Privacy header depends on the SIP Privacy configuration.

Outgoing SIP Trunk Calls—P Headers

The decision of which Asserted Identity (either P-Asserted Identity or P-Preferred-Identity) header gets sent depends on the configured value of the Asserted-Type option. A non-default value for Asserted-Type overrides values that come from Unified CM Call Control. If the Asserted-Type option is set to Default, the value of Screening Identification that the SIP trunk receives from Unified CM Call Control dictates the type of Asserted-Identity.

Outgoing SIP Trunk Calls—SIP Privacy Header

The SIP Privacy header gets used only when you check the Asserted Identity check box and when the SIP trunk sends either a Privacy-Asserted Identity (PAI) or Privacy Preferred Identity (PPI) header. (Otherwise the SIP Privacy header neither gets sent nor processed in incoming SIP messages). The value of the SIP Privacy headers depends on the configured value of the SIP Privacy option. A non-default value for SIP Privacy overrides values that come from Unified CM Call Control.

If the SIP Privacy option is set to Default, the Calling Line ID Presentation and Calling Name Presentation that the SIP trunk receives from Unified CM Call Control determines the SIP Privacy header.

Incoming SIP Trunk Calls—P Headers

The decision of which Asserted Identity (either P-Asserted Identity or P-Preferred-Identity) header gets sent depends on the configured value of the Asserted-Type option. A non-default value for Asserted-Type overrides values that come from Unified CM Call Control. If the Asserted-Type option is set to Default, the value of Screening Identification that the SIP trunk receives from Unified CM Call Control dictates the type of Asserted-Identity.

Incoming SIP Trunk Calls—SIP Privacy Header

The SIP Privacy header gets used only when you check the Asserted Identity check box and when the SIP trunk sends either a PAI or PPI header. (Otherwise the SIP Privacy header neither gets sent nor processed in incoming SIP messages.) The value of the SIP Privacy headers depends on the configured value of the SIP Privacy option. A non-default value for SIP Privacy overrides values that come from Unified CM Call Control.

If the SIP Privacy option is set to Default, the Connected Line ID Presentation and Connected Name Presentation that the SIP trunk receives from Unified CM Call Control determine the SIP Privacy header.

Note

The Remote-party ID and Asserted Identity options represent independent mechanisms for communication of display-identity information.

Default value: True (Checked)

Asserted-Type

From the pulldown menu, select one of the following values to specify the type of Asserted Identity header that SIP trunk messages should include:

Default—Screening information that the SIP trunk receives from Unified CM Call Control determines the type of header that the SIP trunk sends.
PAI—The Privacy-Asserted Identity header gets sent in outgoing SIP trunk messages; this value overrides the Screening indication value that comes from Unified CM.
PPI—The Privacy Preferred Identity header gets sent in outgoing SIP trunk messages; this value overrides the Screening indication value that comes from Unified CM.

Note

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

Default value: Default

SIP Privacy

(Mandatory)

From the pulldown menu, select one of the following values to specify the type of SIP privacy header for SIP trunk messages to include:

Default—This option represents the default value; Name/Number Presentation values that the SIP trunk receives from the Unified CM Call Control compose the SIP Privacy header. For example, if Name/Number presentation specifies Restricted, the SIP trunk sends the SIP Privacy header; however, if Name/Number presentation specifies Allowed, the SIP trunk does not send the Privacy header.
None—The SIP trunk includes the Privacy: none header and implies Presentation allowed; this value overrides the Presentation information that comes from Unified CM.
ID—The SIP trunk includes the Privacy: id header and implies Presentation restricted for both name and number; this value overrides the Presentation information that comes from Unified CM.
ID Critical—The SIP trunk includes the Privacy: id;critical header and implies Presentation restricted for both name and number. The label critical implies that privacy services that are requested for this message are critical, and, if the network cannot provide these privacy services, this request should get rejected. This value overrides the Presentation information that comes from Unified CM.

Note

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

Default value: Default

Table 12. Call Routing Inbound Tab

Option

Description

Significant Digits

(Mandatory)

Significant digits represent the number of final digits that are retained on inbound calls. Use 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 SIP device.

Choose the number of significant digits to collect, from 0 to 32, or choose 99 to indicate all digits.

Note

Unified CM counts significant digits from the right (last digit) of the number that is called.

Default value: 99

Connected Line ID Presentation

(Mandatory)

Unified CM 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.

Select one of the following:

Default—Allowed. Choose Default if you want Unified CM to send connected line information. If a call that originates from an IP phone on Unified CM encounters a device, such as a trunk, gateway, or route pattern, that has the Connected Line ID Presentation set to Default, the presentation value is automatically set to Allowed.
Restricted—Choose Restricted if you do not want Unified CM to send connected line information.

Note

Be aware that this service is not available when QSIG tunneling is enabled.

Default value: Default

Connected Name Presentation

(Mandatory)

Unified CM 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.

Select one of the following:

Default—Allowed. Choose Default if you want Unified CM to send connected name information.
Restricted—Choose Restricted if you do not want Unified CM to send connected name information.

Note

Be aware that this service is not available when QSIG tunneling is enabled.

Default value: Default

Calling Search Space

(Optional)

From the pulldown menu, choose the appropriate calling search space for the trunk. The calling search space specifies the collection of route partitions that are searched to determine how to route a collected (originating) number.

You can configure the number of items that display in this pulldown menu by using the Max List Box Items enterprise parameter. If more calling search spaces exist than the Max List Box Items enterprise parameter specifies, the Find button displays next to the drop-down list box. Click the Find button to display the Find and List Calling Search Spaces window. Find and choose a calling search space name.

Note

To set the maximum list box items, choose System > Enterprise Parameters and choose CCMAdmin Parameters.

Default value: None

AAR Calling Search Space

(Optional)

Choose the appropriate calling search space for the device to use when performing automated alternate routing (AAR). The 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.

Default value: None

Prefix DN

(Optional)

Enter the prefix digits that are appended to the called party number on incoming calls. Unified CM adds prefix digits after first truncating the number in accordance with the Significant Digits setting. You can enter the international escape character +.

Default value: None

Redirecting Diversion Header - Delivery Inbound

(Optional)

Check this check box to accept the Redirecting Number in the incoming INVITE message to the Unified CM.

Uncheck the check box to exclude the Redirecting Number in the incoming INVITE message to the Unified CM.

You use Redirecting Number for voice messaging integration only. If your configured voice-messaging system supports Redirecting Number, you should check the check box.

Default value: False (Unchecked)

Incoming Calling Party - Prefix

(Optional)

Unified CM applies the prefix that you enter in this field to calling party numbers that use Unknown for the Calling Party Numbering Type. You can enter up to 8 characters, which include digits, the international escape field, you cannot configure the Strip Digits field. In this case, Unified CM takes the configuration for the Prefix and Strip Digits fields from the device pool that is applied to the device. If the word, Default, displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming calling party prefix, which supports both the prefix and strip digit functionality.

Default value: None

Incoming Calling Party - Strip Digits

(Optional)

Enter the number of digits, up to the number 24, that you want Unified CM to strip from the calling party number of Unknown type before it applies the prefixes.

Default value: None

Incoming Calling Party - Calling Search Space

(Optional)

This setting allows you to globalize the calling party number of Unknown calling party number type 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 CSS as None, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern in a non-null partition that is not used for routing.

Default value: None

Incoming Calling Party - Use Device Pool CSS

(Optional)

Check this check box to use the calling search space for the Unknown Number field that is configured in the device pool that is applied to the device.

Default value: True (Checked)

Incoming Called Party - Prefix

(Optional)

Unified CM applies the prefix that you enter in this field to called numbers that use Unknown for the Called Party Number Type. You can enter up to 16 characters, which include digits, the international escape character (+), asterisk (*), or the pound sign (#). You can enter the word, Default, instead of entering a prefix.

Tip

If the word Default displays in the Prefix field, you cannot configure the Strip Digits field. In this case, Unified CM takes the configuration for the Prefix and Strip Digits fields from the device pool that is applied to the device. If the word Default displays in the Prefix field in the Device Pool Configuration window, Unified CM does not apply any prefix or strip digit functionality.

Default value: None

Incoming Called Party - Strip Digits

(Optional)

Enter the number of digits that you want Unified CM to strip from the called party number of Unknown type before it applies the prefixes.

Tip

To configure the Strip Digits field, you must leave the Prefix field blank or enter a valid configuration in the Prefix field. To configure the Strip Digits fields in these windows, do not enter the word, Default, in the Prefix field.

Default value: None

Incoming Called Party - Calling Search Space

(Optional)

This setting allows you to transform the called party number of Unknown called party number type on the device. If you choose None, no transformation occurs for the incoming called party number. Make sure that the calling search space that you choose contains the called party transformation pattern that you want to assign to this device.

Default value: None

Incoming Called Party - Use Device Pool CSS

(Optional)

Check this check box to use the calling search space for the Unknown Number field that is configured in the device pool that is applied to the device.

Default value: True (Checked)

Connected Party Transformation CSS

(Optional)

This setting is applicable only for inbound calls. This setting allows you to transform the connected party number on the device to display the connected number in another format, such as a DID or E164 number. Unified CM includes the transformed number in the headers of various SIP messages, including 200 OK and mid-call update and 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.

Note

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

Default value: None

Use Device Pool Connected Party Transformation CSS

(Optional)

To use the Connected Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. 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.

Default value: True (Checked)

Table 13. Call Routing Outbound Tab

Option

Description

Called Party Transformation CSS

(Optional)

This setting allows you to send the transformed called party number in an INVITE message for outgoing calls made over SIP Trunk. Make sure that the Called Party Transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device.

Note

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

Default value: None

Use Device Pool Called Party Transformation CSS

(Optional)

To use the Called Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Called Party Transformation CSS that you configured for this device in the Trunk Configuration window.

Default value: True (Checked)

Calling Party Transformation CSS

(Optional)

This setting allows you to send the transformed calling party number in an INVITE message for outgoing calls made over a SIP Trunk. Also when redirection occurs for outbound calls, this CSS is used to transform the connected number that is sent from Unified CM side in outgoing reINVITE / UPDATE messages. 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.

Tip

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

Default value: None

Use Device Pool Calling Party Transformation CSS

(Optional)

To use the Calling Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Calling Party Transformation CSS that you configured in the Trunk Configuration window.

Default value: True (Checked)

Calling Party Selection

(Mandatory)

Choose the directory number that is sent on an outbound call. Select one of the following options to specify which directory number is sent:

Originator—Send the directory number of the calling device
First Redirect Number—Send the directory number of the redirecting device.
Last Redirect Number—Send the directory number of the last device to redirect the call.
First Redirect Number (External)—Send the external directory number of the redirecting device
Last Redirect Number (External)—Send the external directory number of the last device to redirect the call.

Default value: Originator

Calling Line ID Presentation

(Mandatory)

Unified CM uses calling line ID presentation (CLIP) as a supplementary service to provide the calling party number. The SIP trunk level configuration takes precedence over the call-by-call configuration.

Select one of the following:

Default—Allowed. Choose Default if you want Unified CM to send calling number information.
Restricted—Choose Restricted if you do not want Unified CM to send the calling number information.

Default value: Default

Calling Name Presentation

(Mandatory)

Unified CM used 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:

Default—Allowed. Choose Default if you want Unified CM to send calling name information.
Restricted—Choose Restricted if you do not want Unified CM to send the calling name information.

Note

This service is not available when QSIG tunneling is enabled.

Default value: Default

Calling and Connected Party Info Format

(Mandatory)

This option allows you to configure whether Unified CM inserts a directory number, a directory URI, or a blended address that includes both the directory number and directory URI in the SIP identity headers for outgoing SIP messages.

From the drop-down menu, select one of the following:

Deliver DN only in connected party—In outgoing SIP messages, Unified CM inserts the calling party’s directory number in the SIP contact header information.
Deliver URI only in connected party, if available—In outgoing SIP messages, Unified CM inserts the sending party’s directory URI in the SIP contact header. If a directory URI is not available, Unified CM inserts the directory number instead.
Deliver URI and DN in connected party, if available—In outgoing SIP messages, Unified CM inserts a blended address that includes the calling party's directory URI and directory number in the SIP contact headers. If a directory URI is not available, Unified CM includes the directory number only.

Note

You should set this field to Deliver URI only in connected party or Deliver URI and DN in connected party only if you are setting up URI dialing between Unified CM systems of Release 9.0 or greater, or between a Cisco Unified Communications Manager system of Release 9. 0 or greater and a third-party solution that supports URI dialing. Otherwise, you must set this field to Deliver DN only in connected party.

Default value: Deliver DN only in connected party

Redirecting Diversion Header Delivery - Outbound

(Optional)

Check this check box to include the Redirecting Number in the outgoing INVITE message from the Unified CM to indicate the original called party number and the redirecting reason of the call when the call is forwarded.

Uncheck the check box to exclude the first Redirecting Number and the redirecting reason from the outgoing INVITE message. Use Redirecting Number for voice-messaging integration only. If your configured voice messaging system supports Redirecting Number, check the check box.

Default value: False (Unchecked)

Use Device Pool Redirecting Party Transformation CSS (Optional)

Select this check box to use the Redirecting Party Transformation CSS that is configured in the device pool that is assigned to this device. If you do not select this check box, the device uses the Redirecting Party Transformation CSS that you configured for this device (see field below).

Redirecting Party Transformation CSS (Optional)

Allows you to localize the redirecting party number on the device.

Make sure that the Redirecting Party Transformation CSS that you enter contains the redirecting party transformation pattern that you want to assign to this device.

Caller Information Caller ID DN

(Optional)

Enter the pattern, from 0 to 24 digits that you want to use to format the Called 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 (CO) appends the number with the area code if you do not specify it.
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 CO appends the number with the area code if you do not specify it.

You can also enter the international escape character +.

Default value: None

Caller Information - Caller Name

(Optional)

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

Default value: None

Caller Information - Maintain Original Caller ID DN and Caller Name in Identity Headers

(Optional)

This check box is used to specify whether you will use the caller ID and caller name in the URI outgoing request. If you check this check box, the caller ID and caller name is used in the URI outgoing request. If you do not check this check box, the caller ID and caller name is not used in the URI outgoing request.

Default value: False (Unchecked)

Table 14. SP Info Tab

Option

Description

Destination Address is an SRV

(Optional)

This field specifies that the configured Destination Address is an SRV record.

Default value: False (Unchecked)

Destination - Destination Address IPv4

(Mandatory)

The Destination Address IPv4 represents the remote SIP peer with which this trunk communicates. The allowed values for this field are an IP address, a fully qualified domain name (FQDN), or DNS SRV record only if the Destination Address is an SRV field is checked.

Tip

For SIP trunks that can support IPv6 or IPv6 and IPv4 (dual stack mode), configure the Destination Address IPv6 field in addition to the Destination Address field.

Note

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

Note

For configuring SIP trunks when you have multiple device pools in a cluster, you must configure a destination address that is a DNS SRV destination port. Enter the name of a DNS SRV port for the Destination Address and check the Destination Address is an SRV Destination Port check box.

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

Default value: None

Destination - Destination Address IPv6

(Mandatory if Destination - Destination Address IPv4 field above is not completed)

The Destination IPv6 Address represents the remote SIP peer with which this trunk communicates. You can enter one of the following values in this field:

A fully qualified domain name (FQDN)
A DNS SRV record, but only if the Destination Address is an SRV field is checked.

SIP trunks only accept incoming requests from the configured Destination IPv6 Address and the specified incoming port that is specified in the SIP Trunk Security Profile that is associated with this trunk.

If the remote end is a Unified CM cluster, consider entering the DNS SRV record in this field. The DNS SRV record should include all Unified CMs within the cluster.

Tip

For SIP trunks that run in dual-stack mode or that support an IP Addressing Mode of IPv6 Only, configure this field. If the SIP trunk runs in dual-stack mode, you must also configure the Destination Address field.

Default value: None. If the above IPv4 field is completed, this field can be left blank.

Destination - Destination port

(Mandatory)

Choose the destination port. Ensure that the value that you enter specifies any port from 1024 to 65535, or 0.

Note

You can now have the same port number that is specified for multiple trunks.

You do not need to enter a value if the destination address is a DNS SRV port. The default 5060 indicates the SIP port.

Default value: 5060

Sort Order

(Mandatory)

Indicate the order in which the prioritize multiple destinations. A lower sort order indicates higher priority. This field requires an integer value.

Default value: Empty

MTP Preferred Originating Codec

(Mandatory)

Indicate the preferred outgoing codec by selecting one of:

711ulaw
711alaw
G729/G729a
G729b/G729ab

Note

To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec.

This field is used only when the MTP Termination Point Required check box is checked.

Default value: 711ulaw

BLF Presence Group

(Mandatory)

Configure this field with the Presence feature. From the drop-down menu, 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.

Standard Presence group is configured with installation. Presence groups that are configured in Unified CM Administration also appear in the drop-down menu.
Presence authorization works with presence groups to allow or block presence requests between groups.

Tip

You can apply a presence group to the SIP trunk or to the application that is connected to the SIP trunk. If a presence group is configured for both a SIP trunk and SIP trunk application, the presence group that is applied to the application overrides the presence group that is applied to the trunk.

Default value: Standard Presence Group

SIP Trunk Security Profile

(Mandatory)

Select the security profile to apply to the SIP trunk.

You must apply a security profile to all SIP trunks that are configured in Unified CM 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.

Default value: Non Secure SIP Trunk Profile

Rerouting Calling Search Space

(Optional)

Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The rerouting calling search space gets used to determine where a SIP user (A) can refer another user (B) to a third party (C). After the refer 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).

The rerouting CSS is used for 3xx and REFER processing. The administrator need to select the CSS according to restrictions that needs to be applied for rerouting a 3xx response, and REFER request.

Calling Search Space also applies to 3xx redirection and INVITE with Replaces features.

Call Transfer Call back toward aggregation is very similar to a line-originated call, except that the trunk Rerouting CSS is used for Call routing. There are no Call Rerouting-specific CSSs, but any of the line CSSs can be assigned to Call Transfer back to aggregation. It is used when Unified CM receives a 3xx response to an Invite or receives an in-dialog-REFER, Unified CM uses this CSS to route the call.

Default value: None

Out-Of-Dialog Refer Calling Search Space

(Optional)

Calling search spaces determine the partitions that calling devices can search when they attempt to complete a call. The out-of-dialog calling search space gets used when a Unified CM refers a call (B) that is coming into SIP user (A) to a third party (C) when no involvement of SIP user (A) exists. In this case, the system uses the out-of dialog calling search space of SIP user (A).

Default value: None

SUBSCRIBE Calling Search Space

(Optional)

Supported with the Presence feature, the SUBSCRIBE calling search space determines how Unified CM routes presence requests from the device/server/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.

From the drop-down menu, choose the SUBSCRIBE calling search space to use for presence requests for the SIP trunk. All calling search spaces that you configure in Unified CM Administration display in the SUBSCRIBE Calling Search Space drop-down menu.

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

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

Default value: None

SIP Profile

(Mandatory)

From the drop-down list box, select the SIP profile that is to be used for this SIP trunk.

Default value: Standard SIP Profile

DTMF Signaling Method

(Mandatory)

Select one of the following:

No Preference—Unified CM picks the DTMF method to negotiate DTMF, so the call does not require an MTP. If Cisco Unified Communications Manager has no choice but to allocate an MTP (if the Media Termination Point Required check box is checked), SIP trunk negotiates DTMF to RFC2833.
RFC 2833—Choose this configuration if the preferred DTMF method to be used across the trunk is RFC2833. Unified CM makes every effort to negotiate RFC2833, regardless of MTP usage. Out of band (OOB) provides the fallback method if the peer endpoint supports it.
OOB and RFC 2833—Choose this configuration if both out of band and RFC2833 should be used for DTMF.

Note

If the peer endpoint supports both out of band and RFC2833, Unified CM negotiates both out-of-band and RFC2833 DTMF methods. As a result, two DTMF events are sent for the same DTMF keypress (one out of band and the other, RFC2833).

Default value: No Preference

Normalization Script

(Optional)

From the pulldown menu, choose the script that you want to apply to this trunk.

To import another script, on Unified CM go to the SIP Normalization Script Configuration window (Device > Device Settings > SIP Normalization Script), and import a new script file.

Default value: None

Normalization Script - Enable Trace

(Optional)

Check this check box to enable tracing within the script or uncheck the check box to disable tracing. When checked, the trace.output API provided to the Lua scripter produces SDI trace.

Note

Cisco recommends that you only enable tracing while debugging a script. Tracing impacts performance and should not be enabled under normal operating conditions.

Default value: False (Unchecked)

Script Parameters

(Optional)

Enter parameter names and values in the format Param1Name=Param1Value; Param2Name=Param2Value where Param1Name is the name of the first script parameter and Param1Value is the value of the first script parameter. Multiple parameters can be specified by putting semicolon after each name and value pair . Valid values include all characters except equal signs (=), semi-colons (;); and non-printable characters, such as tabs. You can enter a parameter name with no value.

Recording Information

(Optional)

Enter one of the following:

0—None (default)
1— This trunk connects to a recording-enabled gateway
2— This trunk connects to other clusters with recording-enabled gateways

Table 15. GeoLocation Tab
Option	Description
Geolocation (Optional)	From the drop-down list box, choose a geolocation. You can choose the Unspecified geolocation, which designates that this device does not associate with a geolocation. On Unified CM, you can also choose a geolocation that has been configured with the System > Geolocation Configuration menu option. Default value: None
Geolocation Filter (Optional)	From the pulldown menu, choose a geolocation filter. If you leave the <None> setting, no geolocation filter gets applied for this device. On Unified CM, you can also choose a geolocation filter that has been configured with the System > Geolocation Filtermenu option. Default value: None
Send Geolocation Information (Optional)	Check this check box to send geolocation information for this device. Default value: False (Unchecked)

Reset SIP Trunks

Use this procedure to shut down a SIP trunk and bring it back into service. This procedure does not physically reset the hardware; it only reinitializes the configuration that is loaded by the Cisco Unified Communications Manager cluster. To restart a SIP trunk without shutting it down, use Restart SIP Trunks.

Procedure

Step 1	Log in as the Provider/Reseller or Customer Administrator.
Step 2	Perform one of If you logged in as the Provider or Reseller Administrator, select Device Management > CUCM > SIP Trunks. If you logged in as the Customer Administrator, select Device Management > Advanced > SIP Trunks.
Step 3	From the list of trunks, choose the SIP trunk to be reset, by clicking on its box in the leftmost column.
Step 4	Click Edit to open the SIP trunk information.
Step 5	Select Action > Reset.

Restart SIP Trunks

Use this procedure to restart a SIP trunk without shutting it down first. To shut down a SIP trunk prior to the reset, see Reset SIP Trunks.

Note

If the SIP trunk is not registered with Cisco Unified Communications Manager, you cannot restart it.

Warning

Restarting a SIP trunk drops all active calls that are using the trunk.

Procedure

Step 1	Log in as the Provider/Reseller or Customer Administrator.
Step 2	Perform one of If you logged in as the Provider/Reseller Administrator, select Device Management > CUCM > SIP Trunks. If you logged in as the Customer Administrator, select Device Management > Advanced > SIP Trunks.
Step 3	From the list of trunks, choose the SIP trunk to be restarted, by clicking on its box in the leftmost column.
Step 4	Click Edit to open the SIP trunk information.
Step 5	Select Action > Restart.

Configure SIP Route Patterns

Cisco Unified Communications Manager uses SIP route patterns to route or block both internal and external calls.

The domain name or IP address provides the basis for routing. The administrator can add domains, IP addresses, and IP network (subnet) addresses and associate them to SIP trunks (only). This method allows requests that are destined for these domains to be routed through particular SIP trunk interfaces.

Before you begin

Configure at least one SIP profile and SIP trunk before configuring a SIP route pattern.

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Ensure that the hierarchy path is set to a customer or site level.

Step 3

If prompted, select the NDL that contains the Cisco Unified CM on which you are configuring the SIP route pattern.

Step 4

Perform one of the following:

If you signed in as a provider or reseller administrator, select Device Management > CUCM > SIP Route Patterns.
If you signed in as a customer administrator, select Device Management > Advanced > SIP Route Patterns.

Step 5

Click Add.

Step 6

On the Pattern Definition tab, provide the following information:

Field

Description

Pattern Usage

From the drop-down list, choose either Domain Routing or IP Address Routing. This field is mandatory.

IPv4 Pattern

Enter the domain, subdomain, IPv4 address, or IP subnetwork address. This field is required.

For domain routing pattern usage, enter a domain name IPv4 pattern field that can resolve to an IPv4 address. The domain name can contain the following characters: -, ., 0-9, A-Z, a-z, *, ], and [.

For IP address routing pattern usage, enter an IPv4 address with the format X.X.X.X, where X represents a number between 0 and 255.

For the IP subnetwork address, in classless interdomain routing (CIDR) notation, X.X.X.X/Y; where Y is the network prefix that denotes the number of bits in the network address.

Tip

If the SIP trunk supports IPv6 or both IPv4 and IPv6 (dual-stack mode), configure the IPv6 pattern in addition to the IPv4 pattern.

IPv6 Pattern

Unified CM uses SIP route patterns to route or block both internal and external calls. The IPv6 address in this field provides the basis for routing internal and external calls to SIP trunks that support IPv6.

Tip

If the SIP trunk supports IPv6 or both IPv4 and IPv6 (dual-stack mode), configure the IPv4 pattern in addition to the IPv6 pattern.

Description

Enter a description of the SIP route pattern. The description can include up to 50 characters in any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), or angle brackets (<>).

Route Partition

If you want to use a partition to restrict access to the SIP route pattern, choose the desired partition from the drop-down list box. If you do not want to restrict access to the SIP route pattern, leave the Route Partition value empty.

SIP Trunk/Route List

Choose the SIP trunk or route list to which the SIP route pattern is associated. This field is required.

Block Pattern

Check if you want this pattern to be used for blocking calls.

Step 7

Click the Calling Party Transformations tab, and provide the following information:

Field

Description

Use Calling Party's External Phone Mask

Select On if you want the full, external phone number to be used for calling line identification (CLID) on outgoing calls. Select Default to use the default external phone number mask. This field is required.

Calling Party Transformation Mask

Enter a transformation mask value. Valid entries include the digits 0 to 9 and the wildcard characters X, asterisk (*), and octothorpe (#). If this field is blank and the preceding field is not checked, no calling party transformation takes place.

Prefix Digits (Outgoing Calls)

Enter prefix digits in the Prefix Digits (Outgoing Calls) field. Valid entries include the digits 0 to 9 and the wildcard characters asterisk (*) and octothorpe (#).

Note

The appended prefix digit does not affect which directory numbers route to the assigned device.

Calling Line ID Presentation

Calling line ID presentation (CLIP/CLIR) is a supplementary service that allows or restricts the originating caller phone number on a call-by-call basis.

Choose whether you want to allow or restrict the display of the calling party phone number on the called party phone display for this SIP route pattern.

Choose Default if you do not want to change calling line ID presentation. Choose Allowed if you want to allow the display of the calling number. Choose Restricted if you want to block the display of the calling number.

Calling Line Name Presentation

Calling line name presentation (CNIP/CNIR) is a supplementary service that allows or restricts the originating caller name on a call-by-call basis.

Choose whether you want to allow or restrict the display of the calling party name on the called party phone display for this SIP route pattern.

Choose Default if you do not want to change calling name presentation. Choose Allowed if you want to allow the display of the caller name. Choose Restricted if you want to block the display of the caller name.

Step 8

Click the Connected Party Transformations tab, and provide the following information:

Field

Description

Connected Line ID Presentation

Connected line ID presentation (COLP/COLR) is a supplementary service that allows or restricts the called party phone number on a call-by-call basis.

Choose whether you want to allow or restrict the display of the connected party phone number on the calling party phone display for this SIP route pattern.

Choose Default if you do not want to change the connected line ID presentation. Choose Allowed if you want to display the connected party phone number. Choose Restricted if you want to block the display of the connected party phone number.

If a call originating from an IP phone on Unified CM encounters a device, such as a trunk, gateway, or route pattern, that has the connected line ID presentation set to Default, the presentation value is automatically set to Allowed.

Connected Line Name Presentation

Connected name presentation (CONP/CONR) is a supplementary service that allows or restricts the called party name on a call-by-call basis.

Choose whether you want to allow or restrict the display of the connected party name on the calling party phone display for this SIP route pattern.

Choose Default if you do not want to change the connected name presentation. Choose Allowed if you want to display the connected party name. Choose Restricted if you want to block the display of the connected party name.

Step 9

Click Save.

Configure Route Groups

A route group allows you to designate the order in which gateways are selected. It allows you to prioritize a list of gateways and ports for outgoing trunk selection.

For example, if you use two long distance carriers, you could add a route group so that long distance calls to the less expensive carrier are given priority. Calls only route to the more expensive carrier if the first trunk is unavailable.

Use this procedure to add or modify route groups.

Note

Each gateway or gateway and port combination can only belong to one route group and can only be listed once within that route group. All gateways in a route group must have the same route pattern. The pattern is assigned to the route list containing the route group (not the route group itself).

Route groups are optional. If a proposed route group only contains one gateway or one gateway and port combination and that route group is not to be included in a route list, the route group is not needed.

Before you begin

You must define one or more gateway or SIP trunks before you add a route group.

Procedure

Step 1

Sign in as the provider/reseller or customer administrator.

Step 2

Perform one of the following:

If you logged in as the Provider or Reseller Administrator, select Device Management > CUCM > Route Groups.
If you logged in as the Customer Administrator, select Device Management > Advanced > Route Groups.

Step 3

Perform one of the following:

To add a new route group, click Add.
To edit an existing route group, choose the group to be updated by clicking on its box in the leftmost column, then click Update to edit the selected route group.

Step 4

From the CUCM pulldown menu, select or modify the Cisco Unified Communications Manager that corresponds to the route group.

Step 5

Enter a unique name for the new route group in the Route Group Name field, or modify the existing route group name if desired. The name can comprise up to 50 alphanumeric characters and can contain any combination of spaces, period(s), hyphens (-), and underscore characters (_). Ensure that each route group name is unique to the route plan.

Tip

Use concise and descriptive names for the route group. The CompanynameLocationGroup format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a route group. For example, “CiscoDallasAA1” identifies a Cisco Access Analog route group for the Cisco office in Dallas.

Step 6

From the pulldown menu, select or modify the Distribution Algorithm options for the route group. Default value is Circular.

Option

Description

Top Down

Select this option if you want Cisco Unified Communications Manager to distribute a call to idle or available members starting with the first idle or available member of a route group to the last idle or available member of a route group.

Note

Select Top Down to prioritize the order of devices in step 10.

Circular

Select this option if you want Cisco Unified Communications Manager to distribute a call to idle or available members starting from the (n+1)th member of a route group, where the nth member is the member to which the Cisco Unified Communications Manager most recently extended a call. If the nth member is the last member of a route group, Cisco Unified Communications Manager distributes a call starting from the top of the route group.

Step 7

Click + to open the Members box. Perform one or more of the following steps:

To add a device to the route group, perform step 8.
To modify the priority of a device, go to step 10.
To remove a device from the route group, go to step 11.

Step 8

To add a device to the route group:

From the Device Name pulldown menu, choose the device where the route group is added.

Note

When a SIP trunk or gateway is added, all ports on the device are selected.

For Device Selection Order, indicate the order in which to prioritize multiple devices. A lower selection order indicates higher priority. This field requires an integer value. The default is no setting. The device selection order, if specified, overrides the position of the device in the list.

Step 9

To add another device to the route group, click + at the top of the Members box, then repeat steps 8 and 9 for each additional device.

Step 10

If no device selection order is specified, you can change the priority of a device by moving the device up or down in the list by clicking the arrows on the right side of the Members box. Using the up arrow, move the device higher in the list to make it a higher priority in the route group, or using the down arrow, move the device lower in the list to make it a lower priority in the route group.

Note

The Top Down distribution algorithm must be selected in step 6 to prioritize the order of devices.

Step 11

To remove a device from the route group, select the device in the Members box and click the – on the right side.

Note

You must leave at least one device in the route group.

Step 12

To save a new or updated route group, click Save.

The route group appears in the Route Group list.

Configure Route Lists

Route lists are made up of route groups and are associated with route patterns. A route list associates a set of route groups with a route pattern and determines the order in which those route groups are accessed. The order controls the progress of the search for available trunk devices for outgoing calls.

A route list can contain only route groups. Each route list should have at least one route group. Each route group includes at least one device, such as a gateway, that is available. Based on device type, Cisco Unified Communications Manager (Unified CM) can choose some, or all, ports as resources in each route group. Some devices, such as digital access, only allow you to choose all ports.

You can add a route group to any number of route lists.

Use the following procedure to add route lists or to add, remove, or change the order of route groups in a route list.

Before you begin

Configure route groups before performing this procedure.

Procedure

Step 1

Sign in as the provider/reseller or customer administrator.

Note

When configuring a route list as a provider or reseller, ensure that you select a valid customer or site under your customer in the hierarchy node breadcrumb at the top of the view.

Step 2

Perform one of the following:

If you logged in as the provider or reseller administrator, select Device Management > CUCM > Route Lists.
If you logged in as the customer administrator, select Device Management > Advanced > Route Lists.

Step 3

Perform one of the following:

To add a new route list, click Add, then go to step 4.
To edit an existing route list, choose the list to be updated by clicking on its box in the far left column. Then, click Edit to update the selected route list. Go to step 5.

Step 4

Provide the following information:

Field

Description

CUCM

Select a Unified CM for the route list. This field is mandatory.

Name

Enter a unique name for the new route list. The name can contain up to 50 alphanumeric characters and can contain any combination of spaces, periods, hyphens (-), and underscore characters (_). This field is mandatory.

Tip

Use concise and descriptive names for the route list. The CompanynameLocationCalltype format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a route list. For example, “CiscoDallasMetro” identifies a route list for toll-free, inter-local access transport area (LATA) calls from the Cisco office in Dallas.

Description

A description of the route list.

Call Manager Group

Select a Unified CM Group. Default is the default field. You can choose from Default, None, or select a group. This field is required.

Note

The route list registers with the first Unified CM in the group (which is the Primary Unified CM).

Route List Enabled

Check to enable the route list. This is the default.

Uncheck to disable the route list. When disabling a route list, calls in progress do not get affected, but the route list does not accept more calls.

Run on Every Node

Check to enable the active route list to run on every node.

Step 5

To add a route group to this route list, click + on the right side of the Route Group Items box. Provide the following information:

Field

Description

Route Group Name

Select the route group. This field is required.

Selection Order

Indicate the order in which to prioritize multiple routes. A lower selection order indicates higher priority. This field requires an integer value. The default is no setting.

Use Calling Party's External Phone Number Mask

Choose On from the drop-down list if you want the full external phone number to be used for calling line identification (CLID) on outgoing calls. Choose Off or Default if you do not want to use the full external phone number for CLID on outgoing calls. You may also configure an external phone number mask on all phone devices.

Calling Party Digit Discards

Choose the discard digit instructions that you want to be associated with this calling party transformation pattern.

Calling Party Transformation Mask

Enter a transformation mask value. Valid entries include the digits 0 to 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +.

If:

the Digit Discards field is blank
the Prefix Digits field is blank
the Calling Party Transformation Mask field is blank
and Use Calling Party's External Phone Number Mask is set to Off or Default

then no calling party transformation takes place.

Calling Party Prefix Digits

Enter prefix digits in the field. Valid entries include the digits 0 to 9, the wildcard characters asterisk (*) and octothorpe (#), and the international escape character +.

Note

The appended prefix digit does not affect which directory numbers route to the assigned device.

Calling Party Number Type

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

Unified CM sets the calling directory number (DN) type. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Unified CM does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the calling directory number to be encoded to a non-national numbering plan.

Choose one of the following options:

Cisco CallManager - Use when the Unified CM sets the directory number type.
Unknown - Use when the dialing plan is unknown.
National - Use when you are dialing within the dialing plan for your country.
International - Use when you are dialing outside the dialing plan for your country.
Subscriber - Use when you are dialing a subscriber by using a shortened subscriber number.

Calling Party Numbering Plan

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

Unified CM sets the calling DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Unified CM does not recognize European national dialing patterns. You can also change this setting when you are connecting to PBXs by using routing as a non-national type number.

Choose one of the following options:

Cisco CallManager - Use when the Unified CM sets the Numbering Plan in the directory number.
ISDN - Use when you are dialing outside the dialing plan for your country.
National Standard - Use when you are dialing within the dialing plan for your country.
Private - Use when you are dialing within a private network.
Unknown - Use when the dialing plan is unknown.

Called Party Discard Digits

Choose the discard digit instructions that you want to be associated with this called party transformation pattern.

Called Party Transformation Mask

Enter a transformation mask value. Valid entries include the digits 0 to 9; the wildcard characters X, asterisk (*), and octothorpe (#); the international escape character +; and blank. If this field is blank and the preceding field is not checked, no transformation takes place.

Called Party Prefix Digits

Enter prefix digits in the field. Valid entries include the digits 0 to 9, the wildcard characters asterisk (*) and octothorpe (#), the international escape character +, and blank.

Note

The appended prefix digit does not affect which directory numbers route to the assigned device.

Called Party Number Type

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

Unified CM sets the called directory number (DN) type. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Unified CM does not recognize European national dialing patterns. You can also change this setting when connecting to a PBX that expects the called directory number to be encoded to a non-national numbering plan.

Choose one of the following options:

Cisco CallManager - Use when the Unified CM sets the directory number type.
Unknown - Use when the dialing plan is unknown.
National - Use when you are dialing within the dialing plan for your country.
International - Use when you are dialing outside the dialing plan for your country.
Subscriber - Use when you are dialing a subscriber by using a shortened subscriber number.

Called Party Numbering Plan

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

Unified CM sets the called DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Unified CM does not recognize European national dialing patterns. You can also change this setting when you are connecting to PBXs by using routing as a non-national type number.

Choose one of the following options:

Cisco CallManager - Use when the Unified CM sets the Numbering Plan in the directory number.
ISDN - Use when you are dialing outside the dialing plan for your country.
National Standard - Use when you are dialing within the dialing plan for your country.
Private - Use when you are dialing within a private network.
Unknown - Use when the dialing plan is unknown.

Step 6

To remove a route group from this route list, click – on the right side of its row in the Member box.

Step 7

To change a route group's priority, click the arrows on the right side of the Member box to move it up or down in the list. The up arrow moves the group higher in the list, making it a higher priority. The down arrow moves the group lower in the list, making it a lower priority.

Step 8

To save a new or updated route list, click Save.

Configure Date Time Groups

Use date time groups to define time zones for the various devices that are connected to Cisco Unified CM. Each device exists as a member of only one device pool, and each device pool has only one assigned date time group.

Unified CM automatically configures a default date time group that is called CMLocal. CMLocal synchronizes to the active date and time of the operating system on the server where Unified CM is installed. You can change the settings for CMLocal as desired. Normally, adjust server date and time to the local time zone date and time.

Tip

For a worldwide distribution of Cisco Unified IP Phones, create one named date time group for each of the time zones in which you deploy endpoints.

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Ensure that the hierarchy path is set to a customer or site level.

Step 3

If prompted, select the NDL that contains the Unified CM on which you are configuring the date time group.

Step 4

Perform one of the following:

If you logged in as a provider or reseller administrator, select Device Management > CUCM > Date Time Groups.
If you logged in as a customer administrator, select Device Management > Advanced > Date Time Groups.

Step 5

Click Add.

Step 6

Provide the following information:

Field	Description
Group Name	Enter the name that you want to assign to the new date time group. This field is required.
Time Zone	Choose the time zone for the group that you are adding. This field is required.
Separator	Choose the separator character to use between the date fields. This field is required.
Date Format	Choose the date format for the date that displays on the Cisco Unified IP Phones. This field is required.
Time Format	Choose a 12- or 24-hour time format. This field is required.
Selected Phone NTP References	Select the phone NTP references for the date time group. This ensures that a phone running SIP gets its date and time configuration from an NTP server.

Step 7

Click Save.

Configure Locations

Use locations to implement call admission control in a centralized call-processing system. Call admission control enables you to regulate audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls over links between the locations.

Important

Locations are different from sites. Locations are used by Cisco Unified CM to manage call admission control. Sites are used by Cisco Unified CDM to logically group resources.

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Make sure that the hierarchy path is set to a customer or site level.

Step 3

If prompted, select the NDL that contains the Unified CM on which you are configuring the location.

Step 4

Perform one of the following:

If you logged in as a provider or reseller administrator, select Device Management > CUCM > Locations.
If you logged in as a customer administrator, select Device Management > Advanced > Locations.

Step 5

Click Add.

Step 6

On the Location Information tab, enter the name of the location. This field is required.

Step 7

Click the Intra-Location tab, and provide the following information:

Field

Description

Audio Bandwidth

Enter the maximum amount of audio bandwidth (in kb/s) that is available for all audio calls on the link between this location and other locations. For audio calls, the audio bandwidth includes overhead. Valid values are 0 to 2147483647, where 0 means unlimited bandwidth. This field is required.

Note

To improve audio quality, lower the bandwidth setting, so fewer active calls are allowed on this link.

Video Bandwidth

Enter the maximum amount of video bandwidth (in kb/s) that is available for all video calls on the link between this location and other locations. For video calls, the video bandwidth does not include overhead. Valid values are –1 through 2147483647, where 0 means unlimited bandwidth and –1 means no bandwidth. Setting the value to –1 means you cannot make video calls within this location. This field is required.

Immersive Video Bandwidth

Enter the maximum amount of immersive video bandwidth (in kb/s) that is available for all immersive video calls on the link within this location. For video calls, the immersive video bandwidth does not include overhead. Valid values are –1 through 2147483647, where 0 means unlimited bandwidth and –1 means no bandwidth. Setting the value to –1 means you cannot make immersive video calls within this location. This field is required.

Step 8

Click the Between Locations tab, and provide the following information:

Field	Description
Location	Select a location from the list. This field is required.
Weight	Enter the relative priority of this link in forming the effective path between any pair of Locations. The effective path has the least cumulative weight of all possible paths. Valid values are 0-100. This field is required.
Audio Bandwidth	Enter the maximum amount of audio bandwidth (in kb/s) that is available for all audio calls on the link between this location and other locations. For audio calls, the audio bandwidth includes overhead. Valid values are 0 to 2147483647, where 0 means unlimited bandwidth. You can also select Unlimited Bandwidth. This field is required.
Video Bandwidth	Enter the maximum amount of video bandwidth (in kb/s) that is available for all video calls on the link between this location and other locations. For video calls, the video bandwidth does not include overhead. Valid values are –1 through 2147483647, where 0 means unlimited bandwidth and –1 means no bandwidth. You can also select Unlimited Bandwidth or None. Setting the value to None means you cannot make video calls between this location and other locations. This field is required.
Immersive Video Bandwidth	Enter the maximum amount of immersive video bandwidth (in kb/s) that is available for all immersive video calls on the link between this location and other locations. For video calls, the immersive video bandwidth does not include overhead. Valid values are –1 through 2147483647, where 0 means unlimited bandwidth and –1 means no bandwidth. You can also select Unlimited Bandwidth or None. Setting the value to None means you cannot make immersive video calls between this location and other locations. This field is required.

Step 9

Click the RSVP Settings tab, and provide the following information:

Field

Description

Location

To change the RSVP policy setting between the current location and a location that displays in this pane, choose a location in this pane. This field is required.

RSVP Setting

To choose an RSVP policy setting between the current location and the location that is chosen in the Location pane at left, choose an RSVP setting from the drop-down list. This field is required. Choose from the following available settings:

Use System Default – The RSVP policy for the location pair matches the clusterwide RSVP policy. See topics related to clusterwide default RSVP policy in the Cisco Unified Communications Manager System Guide for details.
No Reservation – No RSVP reservations can get made between any two locations.
Optional (Video Desired) – A call can proceed as a best-effort audio-only call if failure to obtain reservations for both audio and video streams occurs. RSVP Agent continues to attempt RSVP reservation and informs Unified CM if the reservation succeeds.
Mandatory – Unified CM does not ring the terminating device until RSVP reservation succeeds for the audio stream and, if the call is a video call, for the video stream too.
Mandatory (Video Desired) – A video call can proceed as an audio-only call if a reservation for the video stream cannot be reserved.

Step 10

Click Save.

Configure Device Pools

Device pools define sets of common characteristics for devices. The device pool structure supports the separation of user and location information. The device pool contains system, device, and location-related information.

After adding a new device pool, you can use it to configure devices such as Cisco Unified IP Phones, gateways, conference bridges, transcoders, media termination points, voice-mail ports, CTI route points, and so on.

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Perform one of these options:

If you signed in as the provider or reseller administrator, select Device Management > CUCM > Device Pools.
If you signed in as the customer administrator, select Device Management > Advanced > CUCM > Device Pools.

Step 3

Perform one of these options:

To add a new device pool, click Add, then go to step 5.
To edit an existing device pool, click the line item in the table. Go to step 5.

Step 4

In the pop-up, select from the drop-down the network device list (NDL) to which you are adding the device pool, and click OK.

Note

The NDL pop-up only appears when you add a new device pool instance. If you are updating an existing instance, go to step 5.

If you are adding the instance to a site hierarchy node, the NDL pop-up does not appear. You go right to the Add Device Pool form using the NDL associated to the site.

Step 5

From the Device Pool Settings tab, modify these fields as required.

Option

Description

Device Pool Name (Mandatory)

Enter the name of the new device pool that you are creating. You can enter up to 50 characters, which include alphanumeric characters, periods (.), hyphens (-), underscores (_), and blank spaces.

Default value: None

Cisco Unified CM Group (Mandatory)

Choose the Cisco Unified Communications Manager group to assign to devices in this device pool. A Unified CM group specifies a prioritized list of up to three Unified CMs. The first Unified CM in the list serves as the primary one for that group. The other members of the group serve as backup Unified CMs for redundancy.

Calling Search Space for Auto-registration

Choose the calling search space to assign to devices in this device pool that auto-register with Unified CM. The calling search space specifies partitions that devices can search when attempting to complete a call.

Adjunct CSS

From the drop-down list box, choose an existing calling search space (CSS) to use for the devices in this device profile as an adjunct CSS for the extension mobility cross cluster (EMCC) feature. (To configure a new CSS or modify an existing CSS, choose Call Routing > Class of Control > Calling Search Space in Unified CM Administration.)

When configuring the EMCC feature, the administrator must configure a device pool for each remote cluster. If the remote cluster is located in a different country, the adjunct CSS must embrace the partition with which the emergency patterns of that country associate. This configuration facilitates country-specific emergency call routing.

Default value: None

Reverted Call Focus Priority

Choose a clusterwide priority setting for reverted calls that the hold reversion feature invokes. This setting specifies which call type, incoming calls or reverted calls, have priority for user actions, such as going off hook.

Default-If you choose this option, incoming calls have priority.
Highest-If you choose this option, reverted calls have priority.

The Not Selected setting specifies the reverted call focus priority setting for the default device pool at installation. At installation, incoming calls have priority. You cannot choose this setting in Unified CM.

Note

This setting applies specifically to hold reverted calls; it does not apply to parked reverted calls.

Intercompany Media Services Enrolled Group

Select an intercompany media services enrolled group from the drop-down list.

Step 6

From the Local Route Groups tab, modify these fields as required.

Option	Description
Name	From the drop-down list, select the name of the local route group to associate with this device pool.
Value	From the drop-down list, select the value for the local route group to associate with this device pool.

Step 7

From the Roaming Sensitive Settings tab, modify these fields as required.

Option

Description

Date/Time Group (Mandatory)

Choose the date/time group to assign to devices in this device pool. The date/time group specifies the time zone and the display formats for date and time.

Default value: None

Region (Mandatory)

Choose the Unified CM region to assign to devices in this device pool. The Unified CM region settings specify voice codec that can be used for calls within a region and between other regions.

Default value: None

Media Resource Group List

From the drop-down list box, choose a media resource group list. A media resource group list specifies a prioritized list of media resource groups. An application selects the required media resource (for example, a music-on-hold server, transcoder, or conference bridge) from the available media resource groups according to the priority order defined in a media resource group list.

Default value: None

Location

Use locations to implement call admission control (CAC) in a centralized call-processing system. CAC enables you to regulate audio quality and video availability. It works by limiting the amount of bandwidth that is available for audio and video calls over links between locations. The location specifies the total bandwidth that is available for calls to and from this location.

From the drop-down list box, choose the appropriate location for this device pool.

A location setting of None or Hub_None means that the locations feature does not track the bandwidth that the devices in this pool consume. A location setting of Phantom specifies a location that enables successful CAC across intercluster trunks that use H.323 protocol or SIP.

Default value: None

Network Locale

From the drop-down list box, choose the locale that is associated with phones and gateways. The network locale contains a definition of the tones and cadences that the phones and gateways in the device pool in a specific geographic area use. Make sure that you select a network locale that all the phones and gateways that use this device pool can support.

Note

If the user does not choose a network locale, the locale that is specified in the Unified CM clusterwide parameters as the default network locale applies.

Note

Choose only a network locale already installed and supported by the associated devices. The list contains all available network locales for this setting, but not all are necessarily installed. When a device is associated with a network locale that it does not support in the firmware, the device fails to come up.

Default value: None

SRST Reference (Mandatory)

From the drop-down list box, choose a survivable remote site telephony (SRST) reference to assign to devices in this device pool. Choose from these options:

Disable - When you choose this option, devices in this device pool do not have SRST reference gateways that are available to them.
Use Default Gateway - When you choose this option, devices in this device pool use the default gateway for SRST.
Existing SRST references - When you choose an SRST reference from the drop-down list, devices in this device pool use this SRST reference gateway.

Default value: None

Connection Monitor Duration

This setting defines the time that the Cisco Unified IP Phone monitors its connection to Unified CM before it unregisters from SRST and reregisters to Unified CM.

To use the configuration for the enterprise parameter, you can enter 1 or leave the field blank. The default value for the enterprise parameter equals 120 seconds.

Tip

When you change the value of the connection monitor duration, it applies only to the device pool that is being updated. All other device pools use the value in their own connection monitor duration fields or use the value that is configured in the enterprise parameter.

Single Button Barge

This setting determines whether the devices or phone users in this device pool have single-button access for Barge and cBarge. From the drop-down list box, choose from these options:

Off - When you choose this option, the devices in this device pool have the single-button Barge/cBarge feature disabled.
Barge - When you choose this option, the devices in this device pool have the single-button Barge feature enabled.
CBarge - When you choose this option, the devices in this device pool have the single-button cBarge feature enabled.
Default - When you choose this option, the devices in this device pool use the service parameter setting for the single-button Barge/cBarge feature.

Default value: Default

Join Access Lines

This setting determines whether the Join Access Lines feature is enabled for the devices or phone users in this device pool. From the drop-down list box, choose from these options:

Off - When you choose this option, the devices in this device pool have the Join Access Lines feature disabled.
On - When you choose this option, the devices in this device pool have the Join Access Lines feature enabled.
Default - When you choose this option, the devices in this device pool use the service parameter setting for the Join Access Lines feature.

Default value: Default

Physical Location

Select the physical location for this device pool. The system uses physical location with the device mobility feature to identify the parameters that relate to a specific geographical location.

Default value: None

Device Mobility Group

Device mobility groups represent the highest level geographic entities in your network and are used to support the device mobility feature.

Default value: None

Wireless LAN Profile Group

Select a wireless LAN profile group from the drop-down list box.

Note

You can specify the wireless LAN profile group at the device pool level or the individual phone level.

Step 8

From the Device Mobility Related Information tab, modify these fields as required.

Option

Description

Device Mobility Calling Search Space

Choose the appropriate calling search space to be used as the device calling search space when the device is roaming and in the same device mobility group.

Default value: None

AAR Calling Search Space

Choose the appropriate calling search space for the device to use when automated alternate routing (AAR) is performed. The 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.

Default value: None

AAR Group

Choose the AAR group for this device. The AAR group provides the prefix digits that are used to route calls that are otherwise blocked due to insufficient bandwidth. An AAR group setting of None specifies that no rerouting of blocked calls is attempted.

Default value: None

Calling Party Transformation CSS

This setting allows 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 pool.

Tip

Before the call occurs, the device must apply the transformation by using digit analysis. If you configure the calling party transformation CSS as None for the device pool and you check the Use Device Pool Calling Party Transformation CSS check box in the device configuration window, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern in a non-null partition that is not used for routing.

Default value: None

Called Party Transformation CSS

This setting allows you to localize the called party number on the device. Make sure that the called party transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device pool.

Note

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

Default value: None

Step 9

From the Geolocation Configuration tab, modify these fields as required.

Option

Description

Geolocation

From the drop-down list box, choose a geolocation.

You can choose the Unspecified geolocation, which designates that the devices in this device pool do not associate with a geolocation.

Default value: None

Geolocation Filter

From the drop-down list box, choose a geolocation filter.

If you leave the <None> setting, no geolocation filter gets applied for the devices in this device pool.

Default value: None

Step 10

From the Incoming Calling Party Settings tab, modify these fields as required.

Option	Description
National Prefix	Unified CM applies the prefix that you enter in this field to calling party numbers that use national for the calling party numbering type. You can enter up to eight characters, which include digits, the international escape character (+), asterisk (*), or the pound sign (#). You can enter the word `Default` instead of entering a prefix. If the word `Default` displays in the Prefix field, Unified CM applies the service parameter configuration for the incoming calling party prefix, which supports both the prefix and strip-digit functionality.
National Strip Digits	Enter the number of digits, up to the number 24, that you want Unified CM to strip from the calling party number of National type before it applies the prefixes.
National Calling Search Space	This setting allows you to globalize the calling party number of National calling party number type on the device. Make sure that the calling search space 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 CSS as None, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern in a non-null partition that is not used for routing.
International Prefix	Unified CM applies the prefix that you enter in this field to calling party numbers that use International for the calling party numbering type. You can enter up to eight characters, which include digits, the international escape character (`+`), asterisk (`*`), or the pound sign (`#`). You can enter the word `Default` instead of entering a prefix. If the word `Default` displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming calling party prefix, which supports both the prefix and strip-digit functionality.
International Strip Digits	Enter the number of digits, up to the number 24, that you want Unified CM to strip from the calling party number of International type before it applies the prefixes.
International Calling Search Space	This setting allows you to globalize the calling party number of International calling party number type 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 CSS as None, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern in a non-null partition that is not used for routing.
Unknown Prefix	Unified CM applies the prefix that you enter in this field to calling party numbers that use Unknown for the calling party numbering type. You can enter up to eight characters, which include digits, the international escape character (`+`), asterisk (`*`), or the pound sign (`#`). If the word `Default` displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming calling party prefix, which supports both the prefix and strip-digit functionality.
Unknown Strip Digits	Enter the number of digits, up to the number 24, that you want Unified CM to strip from the calling party number of Unknown type before it applies the prefixes.
Unknown Calling Search Space	This setting allows you to globalize the calling party number of unknown calling party number type 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 CSS as None, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern in a non-null partition that is not used for routing.
Subscriber Prefix	Unified CM applies the prefix that you enter in this field to calling party numbers that use Subscriber for the calling party numbering type. You can enter up to eight characters, which include digits, the international escape character (`+`), asterisk (`*`), or the pound sign (`#`). If the word `Default` displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming calling party prefix, which supports both the prefix and strip-digit functionality.
Subscriber Strip Digits	Enter the number of digits, up to the number 24, that you want Unified CM to strip from the calling party number of Subscriber type before it applies the prefixes.
Subscriber Calling Search Space	This setting allows you to globalize the calling party number of Subscriber calling party number type on the device. Make sure that the 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 CSS as None, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern in a non-null partition that is not used for routing.

Step 11

From the Incoming Called Party Settings tab, modify these fields as required.

Option

Description

National Prefix

Unified CM applies the prefix that you enter in this field to called party numbers that use National for the called party numbering type. You can enter up to 16 characters, which include digits, the international escape character (+), asterisk (*), or the pound sign (#). You can enter the word Default instead of entering a prefix.

Tip

If the word Default displays in the Prefix field in the Gateway or Trunk Configuration window, you cannot configure the Strip Digits field in the Gateway or Trunk Configuration window. In this case, Unified CM takes the configuration for the Prefix and Strip Digits fields from the device pool that is applied to the device. If the word Default displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming called party prefix, which supports both the prefix and strip-digit functionality.

Tip

To configure the Strip Digits field, leave the Prefix field blank or enter a valid configuration in the Prefix field. To configure the Strip Digits fields, do not enter the word Default in the Prefix field.

National Strip Digits

Enter the number of digits that you want Unified CM to strip from the called party number of Unknown type before it applies the prefixes.

National Calling Search Space

This setting allows you to transform the called party number of Unknown called party number type on the device. If you choose None, no transformation occurs for the incoming called party number. Make sure that the calling search space that you choose contains the called party transformation pattern that you want to assign to this device.

International Prefix

Unified CM applies the prefix that you enter in this field to called party numbers that use National for the called party numbering type. You can enter up to 16 characters, which include digits, the international escape character (+), asterisk (*), or the pound sign (#). You can enter the word Default instead of entering a prefix.

Tip

If the word Default displays in the Prefix field in the Gateway or Trunk Configuration window, you cannot configure the Strip Digits field in the Gateway or Trunk Configuration window. In this case, Unified CM takes the configuration for the Prefix and Strip Digits fields from the device pool that is applied to the device. If the word Default displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming called party prefix, which supports both the prefix and strip-digit functionality.

Tip

To configure the Strip Digits field, leave the Prefix field blank or enter a valid configuration in the Prefix field. To configure the Strip Digits fields, do not enter the word Default in the Prefix field.

International Strip Digits

Enter the number of digits that you want Unified CM to strip from the called party number of International type before it applies the prefixes.

International Calling Search Space

This setting allows you to transform the called party number of International called party number type on the device. If you choose None, no transformation occurs for the incoming called party number. Make sure that the calling search space that you choose contains the called party transformation pattern that you want to assign to this device.

Unknown Prefix

Unified CM applies the prefix that you enter in this field to called numbers that use Unknown for the called party numbering type. You can enter up to 16 characters, which include digits, the international escape character (+), asterisk (*), or the pound sign (#). You can enter the word Default instead of entering a prefix.

Tip

If the word Default displays in the Prefix field in the Gateway or Trunk Configuration window, you cannot configure the Strip Digits in the Gateway or Trunk Configuration window. In this case, Unified CM takes the configuration for the Prefix and Strip Digits fields from the device pool that is applied to the device. If the word Default displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming called party prefix, which supports both the prefix and strip-digit functionality.

Tip

To configure the Strip Digits field, leave the Prefix field blank or enter a valid configuration in the Prefix field. To configure the Strip Digits fields, do not enter the word Default in the Prefix field.

Unknown Strip Digits

Enter the number of digits that you want Unified CM to strip from the called party number of Unknown type before it applies the prefixes.

Unknown Calling Search Space

This setting allows you to transform the called party number of Unknown called party number type on the device. If you choose None, no transformation occurs for the incoming called party number. Make sure that the calling search space that you choose contains the called party transformation pattern that you want to assign to this device.

Subscriber Prefix

Unified CM applies the prefix that you enter in this field to called numbers that use Subscriber for the called party numbering type. You can enter up to 16 characters, which include digits, the international escape character (+), asterisk (*), or the pound sign (#). You can enter the word Default instead of entering a prefix.

Tip

If the word Default displays in the Prefix field in the Gateway or Trunk Configuration window, you cannot configure the Strip Digits field in the Gateway or Trunk Configuration window. In this case, Unified CM takes the configuration for the Prefix and Strip Digits fields from the device pool that is applied to the device. If the word Default displays in the Prefix field in the Device Pool Configuration window, Unified CM applies the service parameter configuration for the incoming called party prefix, which supports both the prefix and strip-digit functionality.

Tip

To configure the Strip Digits field, leave the Prefix field blank or enter a valid configuration in the Prefix field. To configure the Strip Digits fields, do not enter the word Default in the Prefix field.

Subscriber Strip Digits

Enter the number of digits that you want Unified CM to strip from the called party number of Subscriber type before it applies the prefixes.

Subscriber Calling Search Space

This setting allows you to transform the called party number of Subscriber called party number type on the device. If you choose None, no transformation occurs for the incoming called party number. Make sure that the calling search space that you choose contains the called party transformation pattern that you want to assign to this device.

Step 12

From the Caller ID for Calls from This Phone tab, modify this field as required.

Option

Description

Calling Party Transformation CSS

From the drop-down list box, choose the calling search space that contains the calling party transformation pattern that you want to apply to devices in this device pool.

When Unified CM receives a call from a device in this device pool on an inbound line, Unified CM immediately applies the calling party transformation patterns in this CSS to the digits in the calling party number before it routes the call. This setting allows you to apply digit transformations to the calling party number before Unified CM routes the call. For example, a transformation pattern can change a phone extension to appear as an E.164 number.

Step 13

From the Connected Party Settings tab, modify this field as required.

Option

Description

Connected Party Transformation CSS

This setting is applicable for inbound calls only. This setting allows you to transform the connected party number on the device to display the connected number in another format, such as a DID or E164 number. Unified CM includes the transformed number in the headers of various SIP messages, including 200 OK and mid-call update/reinvite messages for SIP calls and in the connected number information element of CONNECT and NOTIFY messages for H.323 and MGCP calls. Make sure that the connected party transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device pool.

Note

If you configure the connected party transformation CSS as None, the transformation does not match and does not get applied. Ensure that you configure the calling party transformation pattern used for connected party transformation in a non-null partition that is not used for routing.

Step 14

From the Redirecting Party Settings tab, modify this field as required.

Option

Description

Redirecting Party Transformation CSS

This setting allows you to transform the redirecting party number on the device to E164 format. Unified CM includes the transformed number in the diversion header of invite messages for SIP trunks and in the redirecting number information element of setup message (for H.323 and MGCP) sent out of Unified CM. Make sure that the redirecting party transformation CSS that you choose contains the called party transformation pattern that you want to assign to this device pool.

Note

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

Step 15

Click Save.

The route partition appears in the device pool list.

To modify any of these characteristics, make your changes and click Save.

To delete a device pool, check the box to the left of the Name column in the group list, and click Delete.

Associate Local Route Groups to a Device Pool

Use this procedure to associate a local route group with an existing device pool for each site. This allows calls from a device that is tied to a device pool to go out on a specific route group based on the call type. You cannot use this procedure to add or delete device pools.

For example, you can associate multiple local route groups such as Emergency Route Group, Primary Local Route Group (for site A), Secondary Local Route Group (for site A), Primary Local Route Group (for site B), and Secondary Local Route Group (for site B). The Local Route Group feature enables you to specify different route groups for each site (site A and site B) for the respective device pool. Also, you can define a separate call routing option for emergency calls when you associate the Emergency Route Group with a different route group. Hence you can easily define separate call routing options for emergency calls and PSTN calls.

Procedure

Step 1

Sign in as the provider/reseller or customer administrator.

Warning

When associating a local route group, ensure that you select a valid site under your customer in the hierarchy node breadcrumb at the top of the view. If you attempt to associate a local route group at any other node in the hierarchy, a popup alerts you to select a site hierarchy node.

Step 2

Perform one of the following:

If you logged in as the provider or reseller administrator, select Device Management > CUCM > Device Pools.
If you logged in as the customer administrator, select Device Management > Advanced > Device Pools.

Step 3

Click the device pool to be associated.

Step 4

From the Call Manager Group pulldown menu, select a specific Cisco Unified Communications Manager group or leave the Call Manager Group as Default.

Step 5

Click the Local Route Group Settings tab.

Step 6

In the grid, from the Local Route Group pulldown menu, select the local route group.

Step 7

In the grid, from the Route Group pulldown menu, select the route group or gateway.

Step 8

To save the new local route association, click Save.

Emergency Handling

Emergency handling is device-based. It uses the device pool's local route group to handle call routing. When a phone has no direct inward dial (DDI) or the phone has DDI in a remote location, emergency handling uses the site's emergency number.

The implementation is as follows:

An emergency partition is created for each site.
For device-based routing (DBR), a dDeviceDBR CSS is created. For line-based routing (LBR), an EmerCSS is created. Both CSSs are country- and site-specific and contain the emergency partition.
Emergency number translation patterns are added to the emergency partition when a site dial plan is added. This translation pattern leverages the UseOriginatingCSS and prefixes the called number with **104. The calling number is prefixed with *1*LRID* to uniquely identify the calling site.
An emergency route pattern matching **104 is added to the emergency partition with the route list set to use the device pool's emergency local route group.

Provision Emergency Calls

Emergency calls require no additional provisioning. In Cisco Unified Communications Domain Manager, 911 is provisioned as part of the United States country scheme, and 999/112 is provisioned as part of the United Kingdom country scheme. For more information, see Emergency Handling.

Procedure

Step 1	When you Create a Site Dial Plan, enter the emergency number in the Emergency Number field. This is the site emergency published number that is sent if the line that makes the emergency call does not have DDI. Then, if there is a callback, the site emergency published number is dialed.
Step 2	Ensure that a local route group is set up with SLRG-Emer set to the route group. Refer to Associate Local Route Groups to a Device Pool.

Configure Cisco Unified Communications Manager Groups

In Cisco Unified Communications Domain Manager, use Cisco Unified CM Groups to configure Cisco Unified Communications Manager groups.

A Cisco Unified Communications Manager group specifies a prioritized list of up to three Cisco Unified Communications Managers. The first Cisco Unified Communications Manager in the list serves as the primary Cisco Unified Communications Manager for that group, and the other members of the group serve as secondary and tertiary (backup) Cisco Unified Communications Managers.

Each device pool has one Cisco Unified Communications Manager group that is assigned to it. When a device registers, it attempts to connect to the primary (first) Cisco Unified Communications Manager in the group that is assigned to its device pool. If the primary Cisco Unified Communications Manager is not available, the device tries to connect to the next Cisco Unified Communications Manager that is listed in the group, and so on.

Cisco Unified Communications Manager groups provide important features for your system:

Redundancy - This feature enables you to designate a primary and backup Cisco Unified Communications Managers for each group.
Call processing load balancing - This feature enables you to distribute the control of devices across multiple Cisco Unified Communications Managers.

For most systems, you need to have multiple groups, and you need to assign a single Cisco Unified Communications Manager to multiple groups to achieve better load distribution and redundancy.

Cisco Unified Communications Manager group configuration considerations

Before configuring a Cisco Unified Communications Manager group, you must configure the Cisco Unified Communications Managers that you want to assign as members of that group.

After you have configured the Cisco Unified Communications Manager group, you can use it to configure device pools. Devices obtain their Cisco Unified Communications Manager group list setting from the device pool to which they are assigned.

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Note

For Cisco HCS Shared Architecture deployment, you must sign in only as a provider or reseller admin so that only provider or reseller admins can add Unified CM.

Step 2

If you are adding a new instance, ensure the hierarchy path is set to the target node for the new instance.

Step 3

Perform one of the following:

If you signed in as the provider or reseller administrator, select Device Management > CUCM > Unified CM Groups.
If you signed in as the customer administrator, select Device Management > Advanced > CUCM > Unified CM Groups.

Step 4

Perform one of the following:

To add a new Cisco Unified Communications Manager group, click Add, then go to step 5.

To edit an existing Cisco Unified Communications Manager group, click on the line item in the list of existing instances. Go to step 5.

Step 5

Modify the following fields as required.

Option

Description

Name (Mandatory)

Enter the name of the new group.

Auto-registration Cisco Unified Communications Manager Group

Check the Auto-registration Cisco Unified Communications Manager Group check box if you want this Cisco Unified Communications Manager group to be the default Cisco Unified Communications Manager group when auto-registration is enabled.

Leave this check box unchecked if you do not want devices to auto-register with this Cisco Unified Communications Manager group.

Tip

Each Cisco Unified Communications Manager cluster can have only one default auto-registration group. If you choose a different Cisco Unified Communications Manager group as the default auto-registration group, that is, you check the Auto-registration Cisco Unified Communications Manager Group check box for a different Cisco Unified Communications Manager group, the previously chosen auto-registration group no longer serves as the default for the cluster; the Auto-registration Cisco Unified Communications Manager check box displays for the previously chosen group (the original default), and the check box gets disabled for the group that now serves as the default.

Unified CM Group Items (Mandatory)

Click the Add (+) button to select a Cisco Unified Communications Manager to add to the group. Repeat as necessary to add multiple Cisco Unified Communications Managers to the group.

Click the Remove (-) button to remove a Cisco Unified Communications Manager from the group.

Click the up and down arrow buttons to change the order of the Cisco Unified Communications Managers in the group.

Priority (Mandatory)

Enter the priority number for this Cisco Unified Communications Manager in the group. The smaller the integer, the higher the priority.

Selected Cisco Unified Communications Managers

This field displays the Cisco Unified Communications Managers that are in the Cisco Unified Communications Manager group.

Step 6

Click Save.

The group appears in the Call Manager Groups list. When you click on the name of the Cisco Unified Communications Manager group in the list, the group's characteristics are displayed.

To modify any of these characteristics, make your changes and click Save.

To delete a group, check the box to the left of the Name column in the group list, and click Delete.

Note

Verify if the Unified CM is deployed in Shared Mode. Sign in to the Unified CDM, and navigate to Device Management > CUCM > Servers. On the Publisher tab, verify the Multi-Tenant field is set to Shared.

Configure Route Partitions

A partition contains a list of route patterns (directory number (DN) and route patterns). Partitions facilitate call routing by dividing the route plan into logical subsets that are based on organization, location, and call type.

Partitions configuration tips

Timesaver

Use concise and descriptive names for your partitions. The CompanynameLocationCalltype format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a partition. For example, CiscoDallasMetroPT identifies a partition for toll-free, inter-local access and transport area (LATA) calls from the Cisco office in Dallas.

If you are updating a partition, use the Apply Config button as described in the procedure to synchronize a partition with affected devices. When you apply the configuration to devices that are associated with the partition, all calls on affected gateways drop.

Procedure

Step 1

Sign in as the provider, reseller, or customer administrator.

Step 2

Perform one of the following:

If you signed in as the provider or reseller administrator, select Device Management > CUCM > Route Partitions.
If you signed in as the customer administrator, select Device Management > Advanced > CUCM > Route Partitions.

Step 3

Make sure that the hierarchy path is set to the correct level.

Step 4

Perform one of the following:

To add a new route partition, click Add, then go to step 5.
To edit an existing route partition, click the line item in the table. Go to step 6.

Step 5

In the pop-up, select from the pull-down the network device list (NDL) to which you are adding the route partition, and click OK.

Note

The NDL pop-up only appears when you are adding a new route partition. If you are updating an existing partition, go to step 6.

If you are adding the partition to a site hierarchy node, the NDL pop-up will not appear. You will go right to the route partitions add page using the NDL associated to the site.

Step 6

From the Route Partitions page, modify the following fields as required.

Option

Description

Name (Mandatory)

Enter a name for the new partition that you are creating. Ensure that each partition name is unique to the route plan. Partition names can contain a-z, A-Z, and 0-9 characters, as well as spaces, hyphens (-), and underscore characters (_).

Note

The length of the partition names limits the maximum number of partitions that can be added to a calling search space (CSS). The CSS partition limitations table provides examples of the maximum number of partitions that can be added to a CSS with partition names of fixed length.

Description

Enter a description of the new partition that you are creating. The description can include up to 50 characters in any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), back-slash (\), angle brackets (<>), or brackets ([ ]).

If you do not enter a description, Cisco Unified Communications Manager automatically enters the partition name in this field.

Default value: None

Time Schedule

From the drop-down list box, choose a time schedule to associate with this partition. The associated time schedule specifies when the partition is available to receive incoming calls.

This field is empty by default, which indicates that time-of-day routing is not in effect and the partition remains active.

With the time zone value in the following field, association of a partition with a time schedule configures the partition for time-of-day routing. The system checks incoming calls to this partition against the specified time schedule.

Time Zone

Choose one of the following options to associate a partition with a time zone:

Use Originating Device Time Zone: If you choose this option, the system checks the partition against the associated time schedule with the calling device's time zone.
Time Zone: If you choose this option, choose a time zone from the drop-down list box. The system checks the partition against the associated time schedule at the time that is specified in this time zone.

These options all specify the time zone. When an incoming call occurs, the current time on the Cisco Unified Communications Manager gets converted into the specific time zone set when one option is chosen. The system validates this specific time against the value in the Time Schedule field.

The following table provides examples of the maximum number of partitions that can be added to a CSS if partition names are of fixed length.

Partition Name Length	Maximum Number of Partitions
2 characters	170
3 characters	128
4 characters	102
5 characters	86
...	...
10 characters	46
15 characters	32

Step 7

Click Save.

The route partition appears in the route partition list.

To modify any of these characteristics, click the item in the list, make your changes, and click Save.

To delete a route partition, check the box to the left of the Name column in the group list, and click Delete.

Configure Calling Search Spaces

A calling search space comprises an ordered list of route partitions that are typically assigned to devices. Calling search spaces determine the partitions that calling devices search when they are attempting to complete a call.

Procedure

Step 1

Log in as the Provider, Reseller, or Customer Administrator.

Step 2

Perform one of the following:

If you logged in as the Provider or Reseller Administrator, select Device Management > CUCM > Calling Search Spaces.
If you logged in as the Customer Administrator, select Device Management > Advanced > CUCM > Calling Search Spaces.

Step 3

Make sure that the hierarchy path is set to the correct level.

Step 4

Perform one of the following:

To add a new calling search space, click Add, then go to step 5.
To edit an existing calling search space, click on the line item in the table. Go to step 6.

Step 5

In the popup, select from the pull-down the network device list (NDL) to which you are adding the calling search space, and click OK.

Note

The NDL popup will only appear when you are adding a new calling search space. If you are updating an existing calling search space, go to Step 6.

If you are adding the calling search space to a Site hierarchy node, the NDL popup will not appear. You will go right to the Calling Search Spaces add page using the NDL associated to the site.

Step 6

From the Calling Search Spaces page, modify the following fields as required.

Option Description

Name (Mandatory)

Enter a name in the field. The name can comprise up to 50 alphanumeric characters and can contain any combination of spaces, periods (.), hyphens (-), and underscore characters (_). Ensure each calling search space name is unique to the system.

Note

Use concise and descriptive names for your calling search spaces. The CompanynameLocationCalltype format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a calling search space. For example, CiscoDallasMetroCS identifies a calling search space for toll-free, inter-local access and transport area (LATA) calls from the Cisco office in Dallas.

Default value: None

Description

Enter a description in the field. 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 (<>).

Default value: None

Route Partitions

Click the Add (+) button to add a partition to the calling search space. Repeat as necessary to add multiple partitions to the calling search space.

Partition Name

Click the drop-down list and select a partition to add to the calling search space.

Click the Add (+) button to add another partition to the Route Partitions list. Repeat as necessary to add multiple partitions to the list.

Click the Remove (-) button to remove a partition from the list.

Click the up and down arrow buttons to change the order of a partition in the list.

Partition Index

Enter the priority number for this partition in the calling search space. The smaller the integer, the higher the priority.

The following table provides examples of the maximum number of partitions that can be added to a calling search space if partition names are of fixed length.

Partition Name Length	Maximum Number of Partitions
2 characters	170
3 characters	128
4 characters	102
5 characters	86
...	...
10 characters	46
15 characters	32

Step 7

Click Save.

The calling search space appears in the list.

To modify any of these characteristics, click the item in the list, make your changes, and click Save.

To delete a calling search space, check the box to the left of the Name column in the group list, and click Delete.

Configure Calling Party Transformation Patterns

The parameters in the Calling Party Transformation Patterns window provide appropriate caller information using the Calling Party Transformation calling search space on the destination device. Be aware that calls through transformation patterns are not routable. When this pattern is matched, the call does not route to any device.

You use calling party transformation patterns with the calling party normalization feature.

Procedure

Step 1

Log in as the Provider, Reseller, or Customer Administrator.

Step 2

Perform one of the following:

If you logged in as the Provider or Reseller Administrator, select Device Management > CUCM > Calling Party Transformation Patterns.
If you logged in as the Customer Administrator, select Device Management > Advanced > CUCM > Calling Party Transformation Patterns.

Step 3

Make sure that the hierarchy path is set to the correct level.

Step 4

Perform one of the following:

To add a new calling party transformation pattern, click Add, then go to step 5.
To edit an existing calling party transformation pattern, click on the line item in the table. Go to step 6.

Step 5

In the popup, select from the pull-down the network device list (NDL) to which you are adding the calling party transformation pattern, and click OK.

Note

The NDL popup will only appear when you are adding a new calling party transformation pattern. If you are updating an existing pattern, go to Step 6.

If you are adding the calling party transformation pattern to a Site hierarchy node, the NDL popup will not appear. You will go right to the Calling Party Transformation Pattern add tabs using the NDL associated to the site.

Step 6

From the Pattern Definition tab, modify the following fields as required.

Option

Description

Pattern (Mandatory)

Enter the transformation pattern, including numbers and wildcards (do not use spaces); for example, for NANP, enter 9.@ for typical local access or 8XXX for a typical private network numbering plan. Valid characters include the uppercase characters A, B, C, and D and \+, which represents the international escape character +.

Note

Ensure that the pattern is unique. Check the transformation pattern, route pattern, translation pattern, directory number, call park number, call pickup number, message waiting on/off, or meet me number if you receive an error that indicates duplicate entries. You can also check the route plan report.

Default value: None

Partition

If you want to use a partition to restrict access to the transformation pattern, choose the desired partition from the drop-down list box.

Note

Configure transformation patterns in different non-null partitions rather than dialing patterns such as route patterns and directory numbers. For transformation pattern lookups, Cisco Unified Communications Manager ignores the patterns in null partitions.

Note

Make sure that the combination of pattern, route filter, and partition is unique within the Cisco Unified Communications Manager cluster.

Description

Enter a description of the transformation pattern.

Numbering Plan

Choose a numbering plan.

Route Filter

If your transformation pattern includes the @ wildcard, you may choose a route filter. The optional act of choosing a route filter restricts certain number patterns.

The route filters that display depend on the numbering plan that you choose from the Numbering Plan drop-down list box.

MLPP Preemption Disabled

Check this box to make the numbers in a transformation pattern non-preemptable.

Step 7

From the Calling Party Transformations tab, modify the following fields as required.

Option

Description

Use Calling Party's External Phone Number Mask

Choose On from the drop-down list if you want the full external phone number to be used for calling line identification (CLID) on outgoing calls. Choose Off or Default if you do not want to use the full external phone number for CLID on outgoing calls. You may also configure an External Phone Number Mask on all phone devices.

Digit Discards

Choose the discard digit instructions that you want to be associated with this calling party transformation pattern.

Calling Party Transformation Mask

Enter a transformation mask value. Valid entries include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); and the international escape character +.

If the Digit Discards field is blank, the Prefix Digits field is blank, the Calling Party Transformation Mask field is blank, and Use Calling Party's External Phone Number Mask is set to Off or Default, no calling party transformation takes place.

Prefix Digits

Enter prefix digits in the field. Valid entries include the digits 0 through 9, the wildcard characters asterisk (*) and octothorpe (#), and the international escape character +.

Note

The appended prefix digit does not affect which directory numbers route to the assigned device.

Calling Line ID Presentation

Cisco Unified Communications Manager uses calling line ID presentation (CLIP/CLIR) as a supplementary service to allow or restrict the originating caller phone number on a call-by-call basis.

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

Choose Default if you do not want to change calling line ID presentation. Choose Allowed if you want Cisco Unified Communications Manager to allow the display of the calling number. Choose Restricted if you want Cisco Unified Communications Manager to block the display of the calling number.

Calling Party Number Type

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

Cisco Unified Communications Manager sets the calling directory number (DN) type. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the calling directory number to be encoded to a non national type numbering plan.

Choose one of the following options:

Cisco CallManager - Use when the Cisco Unified Communications Manager sets the directory number type.
Unknown - Use when the dialing plan is unknown.
National - Use when you are dialing within the dialing plan for your country.
International - Use when you are dialing outside the dialing plan for your country.
Subscriber - Use when you are dialing a subscriber by using a shortened subscriber number.

Calling Party Numbering Plan

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

Cisco Unified Communications Manager sets the calling DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to PBXs by using routing as a non-national type number.

Choose one of the following options:

Cisco CallManager - Use when the Cisco Unified Communications Manager sets the Numbering Plan in the directory number.
ISDN - Use when you are dialing outside the dialing plan for your country.
National Standard - Use when you are dialing within the dialing plan for your country.
Private - Use when you are dialing within a private network.
Unknown - Use when the dialing plan is unknown.

Step 8

Click Save.

The calling party transformation pattern appears in the list.

To modify any of these characteristics, click the item in the list, make your changes, and click Save.

To delete a calling party transformation pattern, check the box to the left of the Name column in the group list, and click Delete.

Configure Called Party Transformation Patterns

The parameters in the Called Party Transformation Patterns window provide appropriate caller information by using the Called Party Transformation calling search space on the destination device. Be aware that calls through transformation patterns are not routable. When this pattern is matched, the call does not route to any device.

Procedure

Step 1

Log in as the Provider, Reseller, or Customer Administrator.

Step 2

Perform one of the following:

If you logged in as the Provider or Reseller Administrator, select Device Management > CUCM > Called Party Transformation Patterns.
If you logged in as the Customer Administrator, select Device Management > Advanced > CUCM > Called Party Transformation Patterns.

Step 3

Make sure that the hierarchy path is set to the correct level.

Step 4

Perform one of the following:

To add a new called party transformation pattern, click Add, then go to step 5.
To edit an existing called party transformation pattern, click on the line item in the table. Go to step 6.

Step 5

In the popup, select from the pull-down the network device list (NDL) to which you are adding the called party transformation pattern, and click OK.

Note

The NDL popup will only appear when you are adding a new called party transformation pattern. If you are updating an existing pattern, go to Step 6.

If you are adding the called party transformation pattern to a Site hierarchy node, the NDL popup will not appear. You will go right to the Called Party Transformation Pattern add tabs using the NDL associated to the site.

Step 6

From the Pattern Definition tab, modify the following fields as required.

Option

Description

Pattern (Mandatory)

Enter the transformation pattern, including numbers and wildcards (do not use spaces); for example, for NANP, enter 9.@ for typical local access, or 8XXX for a typical private network numbering plan. Valid characters include the uppercase letters A, B, C, and D and \+, which represents the international escape character +.

Note

Ensure that the pattern is unique. Check the transformation pattern, route pattern, translation pattern, directory number, call park number, call pickup number, message waiting on/off, or meet me number if you receive an error that indicates duplicate entries. You can also check the route plan report.

Default value: None

Partition

If you want to use a partition to restrict access to the transformation pattern, choose the desired partition from the drop-down list box. If you do not want to restrict access to the transformation pattern, choose <None> for the partition.

Note

Transformation patterns should be configured in different non- NULL partitions than dialing patterns such as route patterns and directory numbers. For transformation pattern lookups, the patterns in NULL partitions get ignored.

Note

Make sure that the combination of pattern, route filter, and partition is unique within the Cisco Unified Communications Manager cluster.

Description

Enter a description of the transformation pattern. The description can include up to 50 characters in any language, but it cannot include double-quotes ("), percentage sign (%), ampersand (&), or angle brackets (<>).

Numbering Plan

Choose a numbering plan.

Route Filter

If your transformation pattern includes the @ wildcard, you may choose a route filter. The optional act of choosing a route filter restricts certain number patterns.

The route filters that display depend on the numbering plan that you choose from the Numbering Plan drop-down list box.

MLPP Preemption Disabled

Check this box to make the numbers in a transformation pattern non-preemptable.

Step 7

From the Called Party Transformations tab, modify the following fields as required.

Option

Description

Digit Discards

Choose the discard digit instructions that you want to be associated with this called party transformation pattern.

Called Party Transformation Mask

Enter a transformation mask value. Valid entries include the digits 0 through 9; the wildcard characters X, asterisk (*), and octothorpe (#); the international escape character +; and blank. If this field is blank and the preceding field is not checked, no transformation takes place.

Prefix Digits

Enter prefix digits in the field. Valid entries include the digits 0 through 9, the wildcard characters asterisk (*) and octothorpe (#),the international escape character +, and blank.

Note

The appended prefix digit does not affect which directory numbers route to the assigned device.

Called Party Number Type

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

Cisco Unified Communications Manager sets the called directory number (DN) type. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to a PBX that expects the called directory number to be encoded to a non-national type numbering plan.

Choose one of the following options:

Cisco CallManager - Use when the Cisco Unified Communications Manager sets the directory number type.
Unknown - Use when the dialing plan is unknown.
National - Use when you are dialing within the dialing plan for your country.
International - Use when you are dialing outside the dialing plan for your country.
Subscriber - Use when you are dialing a subscriber by using a shortened subscriber number.

Called Party Numbering Plan

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

Cisco Unified Communications Manager sets the called DN numbering plan. Cisco recommends that you do not change the default value unless you have advanced experience with dialing plans such as NANP or the European dialing plan. You may need to change the default in Europe because Cisco Unified Communications Manager does not recognize European national dialing patterns. You can also change this setting when you are connecting to PBXs by using routing as a non-national type number.

Choose one of the following options:

Cisco CallManager - Use when the Cisco Unified Communications Manager sets the Numbering Plan in the directory number.
ISDN - Use when you are dialing outside the dialing plan for your country.
National Standard - Use when you are dialing within the dialing plan for your country.
Private - Use when you are dialing within a private network.
Unknown - Use when the dialing plan is unknown.

Step 8

Click Save.

The called party transformation pattern appears in the list.

To modify any of these characteristics, click the item in the list, make your changes, and click Save.

To delete a called party transformation pattern, check the box to the left of the Name column in the group list, and click Delete.

Configure CTI Route Points

A computer telephony integration (CTI) route point designates a virtual device that can receive multiple, simultaneous calls for application-controlled redirection.

Procedure

Step 1

Log in as a provider, reseller, or customer administrator.

Step 2

Set the hierarchy path to the site for which you want to configure CTI Route Points.

If the hierarchy path is not set to a site, you are prompted to choose a site.

Step 3

Perform one of the following:

If you signed in as a provider or reseller administrator, select Device Management > CUCM > CTI Route Points.
If you signed in as a customer administrator, select Device Management > Advanced > CTI Route Points.

Step 4

Click Add.

Step 5

Provide the following information:

Option

Description

Device Name

Enter a unique identifier for this device, from 1 to 15 characters, including alphanumeric, dot, dash, or underscores. This field is mandatory.

Description

Enter a descriptive name for the CTI route point. 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 (<>).

Device Pool

Choose the name of a Device Pool. The device pool specifies the collection of properties for this device, including Cisco Unified Communications Manager Group, Date Time Group, Region, and Calling Search Space for autoregistration. This field is mandatory.

Common Device Configuration

Choose the common device configuration to which you want this CTI route point assigned. The common device configuration includes the attributes (services or features) that are associated with a particular user. Configure common device configurations in the Common Device Configuration window.

Calling Search Space

From the drop-down list box, choose a calling search space. The calling search space specifies the collection of partitions that are searched to determine how a collected (originating) number is routed.

Location

From the drop-down list box, choose the appropriate location for this CTI route point. This field is mandatory.

Locations implement call admission control (CAC) in a centralized call-processing system. CAC regulates audio quality and video availability by limiting the amount of bandwidth that is available for audio and video calls between locations. The location specifies the total bandwidth that is available for calls to and from this location.

A location setting of Hub_None means that the locations feature does not track the bandwidth that this CTI route point consumes. A location setting of Phantom specifies a location that enables successful CAC across intercluster trunks that use H.323 protocol or SIP.

User Locale

From the drop-down list box, choose the locale that is associated with the CTI route point. The user locale identifies a set of detailed information to support users, including language and font.

Note

If no user locale is specified, Cisco Unified CM uses the user locale that is associated with the device pool

Media Resource Group List

Choose the appropriate Media Resource Group List. A Media Resource Group List is 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. The application chooses according to the priority order defined in a Media Resource Group List.

If you choose <none>, Cisco Unified CM uses the Media Resource Group that is defined in the device pool.

Network Hold MOH Audio Source

Choose the audio source that plays when the network starts a hold action.

If you do not choose an audio source, Cisco Unified CM uses the audio source that is defined in the device pool. If the device pool does not specify an audio source, the system default is used.

User Hold MOH Audio Source

Choose the audio source that plays when an application starts a hold action.

If you do not choose an audio source, Cisco Unified CM uses the audio source that is defined in the device pool. If the device pool does not specify an audio source, the system default is used.

Use Trusted Relay Point

Enable or disable whether Cisco Unified CM inserts a trusted relay point (TRP) device with this media endpoint. This field is mandatory. Choose one of the following values:

Default – If you choose this value, the device uses the Use Trusted Relay Point setting from the common device configuration with which this device associates.
Off – Choose this value to disable 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 – Choose this value to enable 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 Trusted Relay Point (TRP) device designates an MTP or transcoder device that is labeled as Trusted Relay Point.

Calling Party Transformation CSS

This setting allows 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.

Tip

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 does not match and does not get applied. Ensure that you configure the Calling Party Transformation Pattern in a non-null partition that is not used for routing.

Geolocation

From the drop-down list box, choose a geolocation.

You can choose the Unspecified geolocation, which designates that this device does not associate with a geolocation.

Use Device Pool Calling Party Transformation CSS

To use the Calling Party Transformation CSS that is configured in the device pool that is assigned to this device, check this check box. If you do not check this check box, the device uses the Calling Party Transformation CSS that you configured in the CTI Route Point Configuration window.

Step 6

In the Line field, click + to associate a line with the CTI Route Point. Provide the following information:

Field

Description

Dirn, Directory Number

Enter a dialable phone number. Values can include route pattern wildcards and numeric characters (0 to 9). Special characters such as a question mark (?), exclamation mark (!), backslash (\), brackets ([ ]), plus sign (+), dash (-), asterisk (*), caret (^), pound sign (#), and an X are also allowable. Special characters that are not allowed are a period (.), at sign (@), dollar sign ($), and percent sign (%). This field is mandatory.

At the beginning of the pattern, enter \+ if you want to use the international escape character +. For this field, \+ does not represent a wildcard; instead, entering \+ represents a dialable digit.

Dirn, Route Partition

Choose the partition to which the directory number belongs. Make sure that the directory number that you enter in the Directory Number field is unique within the partition that you choose. If you do not want to restrict access to the directory number, choose <None> for the partition.

Index

This field is the line position on the device. If left blank, an integer is automatically assigned.

External Phone Number Mask

Indicate phone number (or mask) that is used to send Caller ID information when a call is placed from this line.

You can enter a maximum of 24 number, the international escape character +, and "X" characters. The Xs represent the directory number and must appear at the end of the pattern. For example, if you specify a mask of 972813XXXX, an external call from extension 1234 displays a caller ID number of 9728131234.

Line Text Label

Use this field only if you do not want the directory number to show on the line appearance. Enter text that identifies this directory number for a line and phone combination.

Suggested entries include boss name, department name, or other appropriate information to identify multiple directory numbers to a secretary or assistant who monitors multiple directory numbers.

Display (Internal Caller ID)

Leave this field blank to have the system display the extension.

Use a maximum of 30 characters. Typically, use the username or the directory number. If using the directory number, the person receiving the call may not see the proper identity of the caller.

ASCII Display (Caller ID)

This field provides the same information as the Display (Internal Caller ID) field, but limit input to ASCII characters. Devices that do not support Unicode (internationalized) characters display the content of the ASCII Display (Internal Caller ID) field.

Ring Setting (Phone Active)

If applicable, the ring setting that is used when this phone has another active call on a different line. Choose one of the following options:

Use system default
Disable
Flash only
Ring once
Ring
Beep only

Ring Setting (Phone Idle)

If applicable, the ring setting for the line appearance when an incoming call is received and no other active calls exist on that device. Choose one of the following options:

Use system default
Disable
Flash only
Ring once
Ring

Recording Option

This field determines the recording option on the line appearance of an agent. The default recording option is Call Recording Disabled.

Choose one of the following options:

Call Recording Disabled – Calls made on this line appearance cannot be recorded.
Automatic Call Recording Enabled – Calls made on this line appearance are recorded automatically.
Selective Call Recording Enabled – Calls made on this line appearance can be recorded using a softkey or programmable line key that is:
- assigned to the device
- a CTI-enabled application
- both interchangeably

Recording Profile

This field determines the recording profile on the line appearance of an agent.

Recording Media Source

This field determines the recording media source option on the line appearance.

Choose one of the following options:

Gateway Preferred – Voice gateway is selected as the recording media source when the call is routed through a recording enabled gateway.
Phone Preferred – Phone is selected as the recording media source

Monitoring Calling Search Space

The monitoring calling search space of the supervisor line appearance must include the agent line or device partition to allow monitoring the agent.

Visual Message Waiting Indicator Policy

Use this field to configure the handset lamp illumination policy. Choose one of the following options:

Use System Policy (The directory number refers to the service parameter "Message Waiting Lamp Policy" setting.)
Light and Prompt
Prompt Only
Light Only
None

Audible Message Waiting Indicator Policy

Use this field to configure an audible message waiting indicator policy. Choose one of the following options:

Off
On – When you select this option, you receive a stutter dial tone when you take the handset off hook.
Default – When you select this option, the phone uses the default that was set at the system level.

Log Missed Calls

If checked, Cisco Unified CM logs missed calls in the call history for the shared line appearance on the phone.

Busy Trigger

This setting, working with Maximum Number of Calls and Call Forward Busy, determines the maximum call number for the line. Use this field with Maximum Number of Calls for CTI route points. The default specifies 4500 calls.

Maximum Number of Calls

For CTI route points, you can configure up to 10,000 calls for each port. The default specifies 5000 calls. Use this field with the Busy Trigger field.

Note

We recommend that you set the maximum number of calls to no more than 200 per route point. This prevents system performance degradation. If the CTI application needs more than 200 calls, we recommend that you configure multiple CTI route points.

Dialed Number

Check to display original dialed number upon call forward.

Redirected Number

Check to display the redirected number upon call forward.

Caller Number

Check to display the caller number upon call forward.

Caller Name

Check to display the caller name upon call forward.

End User, User ID

The User ID of a user associated with the line.

Step 7

Click Save.

Configure Time Periods

A time period specifies a time range that includes a start time and end time. Time periods also specify a repetition interval either as days of the week or a specified date on the yearly calendar. You define time periods and then associate the time periods with time schedules. A particular time period can be associated with multiple time schedules.

Note

Cisco Unified Communications Domain Manager provides one “All the time” time period. The “All the time” period is a special, default time period that includes all days and hours, and cannot be deleted.

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Make sure that the hierarchy path is set to the node where you want to configure the new time period.

Step 3

Perform one of these options as appropriate:

If you signed in as a provider or reseller administrator, select Device Management > CUCM > Time Periods.
If you signed in as a customer administrator, select Device Management > Advanced > Time Periods.

Step 4

Perform one of these options as appropriate:

To add a new time period, click Add, then go to Step 5.
To edit an existing time period, choose the time period to be updated by clicking it in the list of time periods. Go to Step 6.

Step 5

If the Network Device List popup window appears, select the NDL for the time period from the drop-down menu. The window appears when you are on a nonsite hierarchy node. If you are at a site hierarchy node, the NDL associated with the site is automatically used.

Note

The Network Device List drop-down menu only appears when a time period is added; it does not appear when you edit a time period.

Step 6

Enter a unique name for the new time period in the Name field, or modify the existing Name if desired. This field is mandatory. Enter a name in the Time Period Name field. The name can comprise up to 50 alphanumeric characters. It can contain any combination of spaces, periods (.), hyphens (-), and underscore characters (_).

Example:

Use concise and descriptive names for your time periods. The hours_or_days format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a time period. For example, office_M_to_F identifies a time period for the business hours of an office from Monday to Friday.

Step 7

Complete the other fields as appropriate.

Option

Description

Enter a description for the time period.

Time of Day Start

From the drop-down list box, choose the time when this time period starts. The available listed start times comprise 15-minute intervals throughout a 24-hour day.

Default: No Office Hours

Note

To start a time period at midnight, choose the 00:00 value.

Time of Day End

From the drop-down list box, choose the time when this time period ends. The available listed end times comprise 15-minute intervals throughout a 24-hour day.

Default: No Office Hours

Note

To end a time period at midnight, choose the 24:00 value.

Step 8

Choose one of these repetition periods and complete the required information:

Note

If you choose to repeat the time period by the week, the Repeat Every Year fields are dimmed and cannot be edited. If you choose to repeat the time period by the year, the Repeat Every Week fields are dimmed and cannot be edited.

Repeat Every Week—For time periods defined by the week
1. From the Start Day drop-down menu, choose a day of the week on which this time period starts.
2. From the End Day drop-down menu, choose a day of the week on which this time period ends.
Repeat Every Year—For time periods defined by the year
1. From the Start Month drop-down menu, choose a month of the year on which this time period starts.
2. Enter a number from 1 to 31 in the Start Date field to define the day of the month on which this time period starts.
3. From the End Month drop-down menu, choose a month of the year on which this time period ends.
4. Enter a number from 1 to 31 in the End Date field to define the day of the month on which this time period ends.

Example:

For weekly time intervals, choose a Start Day on Mon and End Day of Fri for a time period starting on Mondays and ending on Fridays.
For weekly time intervals, choose Start Day and End Day values of “Sat” to define a time period that applies only on Saturdays.
For yearly time intervals, choose Start values of “Jan” and “15” and End values of “Mar” and “15” to choose the days from January 15 to March 15.
For yearly time intervals, choose Start and End values of “Jan” and “1” to specify January 1 as the only day during which this time period applies.

Step 9

To save the new time period, click Save. To save an updated time period, click Update.

What to do next

Associate time periods with time schedules. See Configure Time Schedules.

Note

You cannot delete time periods that time schedules are using. Before deleting a time period that is currently in use, perform either or both of these tasks as appropriate:

Assign a different time period to any time schedule that is using the time period that you want to delete.
Delete the time schedules that are using the time period that you want to delete.

Configure Time Schedules

A time schedule includes a group of time periods. Time schedules are assigned to partitions to set up time-of-day call routing. Time schedules determine the partitions where calling devices search when they are attempting to complete a call during a particular time of day. Multiple time schedules can use a single time period.

Once you have configured a time period in Configure Time Periods, perform this procedure to assign the time period to a time schedule.

Note

Cisco Unified Communications Domain Manager provides one “All the time” schedule. The “All the time” schedule is a special, default time schedule that includes all days and hours, and cannot be deleted.

Procedure

Step 1

Sign in as a provider, reseller, or customer administrator.

Step 2

Make sure that the hierarchy path is set to the node where you want to create the new time schedule.

Step 3

Perform one of the following:

If you signed in as a provider or reseller administrator, select Device Management > CUCM > Time Schedules.
If you signed in as a customer administrator, select Device Management > Advanced > Time Schedules.

Step 4

Perform one of the following:

To add a new time schedule, click Add, then go to Step 5.
To edit an existing time schedule, choose the time schedule to be updated by clicking it in the list of time schedules. Go to Step 6.

Step 5

If the Network Device List popup window appears, select the NDL for the time schedule from the drop-down menu. The window appears when you are on a nonsite hierarchy node. If you are at a site hierarchy node, the NDL associated with the site is automatically used.

Note

The Network Device List drop-down menu only appears when a time schedule is added; it does not appear when you edit a time schedule.

Step 6

Enter a unique name for the new time schedule in the Name field, or modify the existing Name if desired. This field is mandatory. The name can comprise up to 50 alphanumeric characters. The name of the time schedule can contain any combination of spaces, periods (.), hyphens (-), and underscore characters (_).

Step 7

(Optional) Enter a description for the time schedule in the Description field.

Step 8

Click + to open the Time Periods form.

Step 9

From the Time Period drop-down box, choose a time period for the time schedule.

Step 10

Repeat Steps 8 and 9 to add another time period to the time schedule.

Note

If multiple time periods are associated with a schedule where the time periods overlap, time periods with Day of Year settings take precedence over time periods with Day of Week settings. Day of Year is applicable when Year Start value is set and the End value is left blank.

Example: If a Time Period configured for January 1 is configured as No Office Hours and another time period is configured for the same day of the week (for example, Sunday to Saturday) as 08:00 to 17:00, the time period for January 1 is used. In this example, No Office Hours takes precedence.
Time interval settings take precedence over No Office Hour settings for the same day of the year or day of the week.

Example: One time period specifies for Saturday as No Office Hours. Another time period specifies Saturday hours of 08:00 to 12:00. In this example, the resulting time interval specifies 08:00 to 12:00 for Saturday.
If multiple time periods are associated with a schedule where the time periods overlap, time periods with Day of Week settings take precedence over time periods with Range of Days settings. Range of Days applies to when Year Start and End values are set, even if they are configured for the same day.

Example: If a Time Period configured for Day of Week (for example, Sunday to Saturday) is configured as No Office Hours and another time period is configured for January 1 until December 31 as 08:00 to 17:00, the time period for Day of Week is used. In this example, No Office Hours takes precedence.

Step 11

To save the new time schedule, click Save. To save the updated time schedule, click Update.

Step 12

Repeat Steps 3 to 11 to configure another time schedule.

What to do next

You cannot delete time schedules that partitions are using. Before deleting a time schedule that is currently in use, perform either or both of the following tasks:

Assign a different time schedule to any partitions that are using the time schedule that you want to delete.
Delete the partitions that are using the time schedule that you want to delete.

Warning

Before you delete a time schedule, check carefully to ensure that you are deleting the correct time schedule. You cannot retrieve deleted time schedules. If you accidentally delete a time schedule, you must rebuild it.

Clone an Instance of a Cisco Unified CM Device Model

To save time, make a copy of an existing instance of a device model rather than adding a new one. To do this, use the clone operation. When you create a clone, give it a new unique name and modify other device model fields as needed before saving.

Note

You can clone an instance of a device model to the same Cisco Unified CM or to a different Cisco Unified CM.

If you clone to a different Cisco Unified CM, make sure that all device model fields have values that are appropriate for the target Cisco Unified CM. For example, make sure calling search spaces specified in the source instance exist on the target Cisco Unified CM.

Procedure

Step 1	Log in as a provider, reseller, or customer administrator.
Step 2	Do one of the following: If you signed in as a provider or reseller administrator, select Device Management > CUCM > {device_model_type}. If you signed in as a customer administrator, select Device Management > Advanced > {device_model_type}.
Step 3	From the device model list, click the instance to be cloned.
Step 4	Click Action > Clone.
Step 5	Depending on the device model, do one of the following: When prompted, select the NDL that contains the target Cisco Unified CM. Select the target Cisco Unified CM from the CUCM pulldown menu.
Step 6	Enter a unique name for the new instance of the device model in the Name field.
Step 7	Modify other fields as required. For more detailed information about the fields, see the corresponding topic on configuring a new instance of the device model. For example, if you are cloning a SIP trunk, see "Configure SIP Trunks" for field descriptions.
Step 8	Click Save to save the cloned instance.

The new instance appears in the list. The new instance is created on the target Cisco Unified CM.

Load Balancing

Cisco Unified Communications Manager (Unified CM) groups provide both call-processing redundancy and distributed call processing. You can distribute devices, device pools, and Unified CMs among the groups to improve redundancy and load balancing in your system.

A Cisco Unified Communications Manager Group specifies a prioritized list of up to three Unified CMs. The first Unified CM in the list serves as the primary Unified CM for that group, and the other members of the group serve as secondary and tertiary (backup) Unified CMs.

Each device pool has one Unified CM Group that is assigned to it. For example, Group 1 points to Device Pool 1, Group 2 points to Device Pool 2, and Group 3 points to Device Pool 3. When a device registers, it attempts to connect to the primary (first) Unified CM in the group that is assigned to its device pool. If the primary Unified CM is not available, the device tries to connect to the next Unified CM that is listed in the group, and so on.

Load balancing is a manual process on Unified CM requiring you to perform the following tasks:

Add new, custom Unified CM groups and device pools.
Synchronize the groups and device pools into Cisco Unified Communications Domain Manager.
Select the appropriate group and device pool in the Subscriber or Phone configuration for the site. To create more than one configuration for a site, create at least two Unified CM groups, then associate a device pool to the appropriate Unified CM group.

To determine if load balancing is required for your network, you can check the current device traffic load in Unified CM using the System > Device Pool menu path. When you click on the device configuration information for a specific device pool, the Device Pool Information field lists the number of members in the Device Pool. Compare different device pools to see if the members are evenly divided between pools.

To perform load balancing, see Load Balancing Using Site Default Device Pool.

Load Balancing Using Site Default Device Pool

A default device pool is created for each site when the site dial plan is deployed for the Type 1 through 4 dial plan schema groups. This procedure uses the default site device pools, so you do not need to create any additional device pools directly on Cisco Unified Communications Manager (Unified CM). Perform this procedure to load balance using the default site device pool. In this procedure, the default device pool is updated to point to the appropriate Cisco Unified Communications Manager group.

Note

Using this configuration, redundancy is gained within a site while load balancing is gained across multiple sites. Since there is one device pool per site, all devices at a site home to the same sequence of Cisco Unified Communications Managers, providing failover redundancy. Devices in different sites home to different sequences of Cisco Unified Communications Managers, providing load balancing across the sites.

Note

The default site device pool is not created until the Type 1 to 4 site dial plan has been deployed which updates the Site Defaults to use the default device pool. If the site dial plan has not been deployed, you will not see a site default device pool in the form Cu<customerId>Si<siteId>-DevicePool. You can determine the default device pool for a site in Cisco Unified Communications Domain Manager (Unified CDM) by selecting Site Management > Defaults.

Procedure

Step 1

Log in as the Provider, Reseller, or Customer administrator.

Step 2

Select the site from the hierarchy node breadcrumb at the top of the view in (Unified CDM).

Step 3

Follow the steps outlined in Create a Site Dial Plan if you have not already done so; the Create a Site Dial Plan procedure creates the default site device pool instance.

Step 4

Log in to Cisco Unified Communications Manager and create one or more Cisco Unified Communications Manager groups on Cisco Unified Communications Manager. See Cisco Unified Communications Manager Administration Guide.

Step 5

From Unified CDM, perform a sync operation of the Cisco Unified Communications Manager using the Administration Tools > Data Sync menu path. This sync updates the Unified CDM cache and makes the Cisco Unified Communications Manager groups that were added directly on Cisco Unified Communications Manager available to Unified CDM.

Step 6

Perform Associate Cisco Unified Communications Manager Group to a Device Pool, select a Unified CM group other than the default group in the Call Manager Group drop-down list.

Note

To verify that the phone or subscriber uses the device pool as expected, select a subscriber from the list of subscribers in Unified CDM (Subscriber Management > Subscribers) and check the Device Pool Name setting under the Phones tab.

Associate Cisco Unified Communications Manager Group to a Device Pool

Use this procedure to associate a Cisco Unified Communications Manager (Unified CM) group with an existing device pool for each site. This allows calls from a device that is tied to a device pool to go out on a specific Unified CM group based on the call type. You cannot use this procedure to add or delete device pools.

Procedure

Step 1

Log in as the Provider/Reseller or Customer administrator.

Warning

When associating a Unified CM group, ensure that you select a valid site under your customer in the hierarchy node breadcrumb at the top of the view. If you attempt to associate a Unified CM group at any other node in the hierarchy, a popup alerts you to select a site hierarchy node.

Step 2

Perform one of the following:

If you logged in as the Provider or Reseller Administrator, select Device Management > CUCM > Device Pools.
If you logged in as the Customer Administrator, select Device Management > Advanced > Device Pools.

Step 3

Click the device pool to be associated.

Step 4

From the Call Manager Group pulldown menu, select a specific Unified CM group or leave the Unified CM Group as Default.

Step 5

To save the new Unified CM group association, click Save.

Update the USA Device-Based Routing Dial Plan

Use this procedure if you deployed the United States country dial plan using Cisco Unified Communications Domain Manager 10.1(1) and are using Device-Based Routing. Update the calling search spaces (CSS) for each customer that uses the United States dial plan. Update one customer-level CSS and one site-level CSS for each USA site.

Note

Perform this procedure only once. For example, if you performed this procedure when you upgraded to Unified CDM 10.6(1), do not perform it again when upgrading to a later release.

Procedure

Step 1

Log in to Cisco Unified Communications Manager.

Step 2

Navigate to Call Routing > Class of Control > Calling Search Space.

Step 3

Find the calling search space where the CSS Name ends with USADP-DBRDevice-CSS.

Note

Records for each of your USA sites appear in the following format: Cu<customerId>Si<siteId>-USADP-DBRDevice-CSS.

Step 4

Edit each calling search space to include the pre-device-based route selection partition instead of the device-based route selection partition:

Remove the following partition: Cu<customerId>-USADP-DBRteSel-PT.
Add the following partition: Cu<customerId>-USADP-PreDBRteSel-PT.
Click Save.

Step 5

Find the calling search space where the CSS Name ends with USADP-DBRteSel-CSS.

Note

Records for each of your USA sites appear in the following format: Cu<customerId>Si<siteId>-USADP-DBRteSel-CSS.

Step 6

Edit this calling search space to include the device-based route selection partition instead of the line-based route selection partition:

Remove the following partition: Cu<customerId>-USADP-LBRteSel-PT.
Add the following partition: Cu<customerId>-USADP-DBRteSel-PT.
Click Save.

Step 7

After updating all calling search spaces, log in to Unified CDM and perform a Cisco Unified Communications Manager import operation:

Log in to Unified CDM as provider or customer administrator.
Navigate to Device Management > Advanced > Perform Publisher Actions.
Select Action Import, App Type CUCM Device and select the appropriate Cisco Unified Communications Manager cluster.
Click Save.

Sharing Lines Across Sites

This deployment model allows lines to be shared across sites, and is accomplished with the concept of an "Inventory Site" in addition to the normal real sites. The Inventory Site is used to provision the shared lines first, then the real sites make use of the shared lines by assigning them to phones. Devices are not provisioned in the Inventory Site; they are only provisioned on the real sites.

The following figure provides a basic Shared Line Across Sites configuration using one Inventory Site ("c2Inventory") and two real sites ("c2s1" and "c2s2"). In this example there are two shared DNs (9000 and 9001 shown in red) and four site-specific DNs (1000 and 1001 at c2s1, 2000 and 2001 at c2s2). The inventory for the shared DNs are provisioned at the Customer hierarchy level to make them visible to all the sites under the customer. This allows the sites to configure the associated line and assign the line to a device. The inventory for the non-shared-across-sites DNs is still configured at the real sites (in blue) as it was in previous Cisco HCS releases. Notice that both shared DNs and non-shared DNs can co-exist for the same customer.

It is important to understand that an Inventory Site is only an Inventory Site by name, not by type. An Inventory Site is just a regular site, is no different than any other site, does not have an "Inventory Site" checkbox, and is deployed exactly the same as any other site. It is only by convention that we're calling this an Inventory Site and designating this site as the repository for lines to be shared across sites.

Tip

If a line is potentially shareable, we recommend that you create the line in the Inventory Site, even if it will not be shared across sites immediately. The system does not support the ability to move a line from a real site to an Inventory Site, so to convert a line from site-local to cross-site shared, the line would need to be deleted from the real site and recreated in the Inventory Site.

For more information about sharing lines across sites, as well as configuration procedures for this deployment model, refer to Cisco Hosted Collaboration Solution Release Dial Plan Management Guide.

Shared Line Across Sites

This feature allows lines to be shared across sites, and is accomplished by introducing the concept of an "Inventory site" in addition to the normal real sites. The Inventory site is used to provision the shared lines first, then the real sites make use of the shared lines by assigning them to phones. Devices are not provisioned in the Inventory site; they are only provisioned on the real sites.

This feature also supports Hunt Groups and Call Pickup Groups across sites by leveraging the Inventory site to provision all of the lines to be included in the Hunt Group or Call Pickup Group. The lines used in Hunt Groups and Call Pickup Groups that are provisioned in the Inventory site can span multiple real sites (in other words, they are used by devices on the real sites). The key requirement is that all the lines to be used by a given Hunt Group or Call Pickup Group must be configured in the Inventory site, along with the Hunt Group and Call Pickup Group itself.

The Shared Line Across Sites deployment model is 100% backward compatible with the previous directory number (DN) and line configuration. Existing deployments are not impacted when the system is upgraded, and all existing dial plan configuration procedures are supported. The deployment configuration shown in Shared Line Across Sites Example is optional and is only required when sharing lines across sites.

Definitions for Shared Line Across Sites

Many of the terms used for the Shared Line Across Sites feature have a number of different meanings depending on the context. To help remove some ambiguity in the procedures documented in this section, please review the following definitions in the context of the Shared Line Across Sites feature:

Directory Number (DN)—This number can be assigned to a user and can be dialed. It may be composed of an extension prefix and/or a site location code and/or extension, but the DN is the final form of the internal dialable number. The DN is not the E.164 number, although they may coincide.
DN Inventory—A list of DNs configured in Cisco Unified Communications Domain Manager that can then be used in a line configuration. The DN inventory resides only in Cisco Unified Communications Domain Manager and is not pushed to Cisco Unified Communications Manager. DNs may also be used as feature pilot numbers (for example, Hunt Pilot or Call Pickup patterns). When used as a service number, the DN is marked as unavailable and it cannot be used in a line configuration. DN inventory is configured at the Site or Customer hierarchy level. However, to configure DN inventory at a customer hierarchy, the customer dial plan must be configured not to use site location codes ("flat dial plan").
E.164 Number—The globally routable phone number that includes country code and country-specific format. This number is used for offnet Public Switched Telephone Network (PSTN) calls.
E.164 Inventory—A list of E.164 numbers configured at a site hierarchy. This list only resides in Cisco Unified Communications Domain Manager and is not pushed to Cisco Unified Communications Manager.
Line or Line Relation—The line configured from menu item Subscriber Management > Lines which is pushed to Cisco Unified Communications Manager. A line is also pushed to Cisco Unified Communications Manager when it is referenced in a phone, extension mobility profile, or single number reach profile and doesn't already exists on the Cisco Unified Communications Manager. On Cisco Unified Communications Manager, a line corresponds with the items under Call Routing > Directory Number. It is also called a "line relation" because this is the technical term for the construct within Cisco Unified Communications Domain Manager.
Line Appearance—A line appearance is the assignment of a line to a phone. One line can have many line appearances. If a line has more than one line appearance, it is considered a shared line.
Class of Service (CoS)—This term refers to a Calling Search Space (CSS) that is specifically used to define call routing and feature processing for a line or a phone. Refer to Class of Service for Shared Line Across Sites for more information.
SLC-based Dial Plan—A site location code (SLC)-based dial plan is one that uses unique, site-specific dialable location codes that are embedded in the DN along with the extension. For example, the default Type 1 through Type 3 Cisco dial plans are SLC-based. Only the Type 4 dial plan is not SLC-based; Type 4 dial plan is commonly referred to as a "flat" dial plan because DNs are the actual extensions. This distinction between types of dial plans is important, because to support the Shared Line Across Sites feature, where devices at different sites can share a line that supports intra/intersite dialing from every site, an SLC would not allow a line to span multiple sites (because multiple sites can't have the same SLC). The Shared Line Across Sites feature requires the customer to deploy a non-SLC based dial plan.
DNR—Directory Number Routing allows an administrator to make their DN inventory inter- and intra-site routable by adding the necessary translation patterns on Cisco Unified Communications Manager when deploying a non-SLC-based dial plan. Normally, for the SLC-based dial plans, because each site requires a unique SLC, these translation patterns can automatically be deployed. This is not the case for non-SLC (flat) dial plans. In this case, DNR instances can be created when DN inventory is added to make these internally routable.
E.164 Associations—Allow the customer's DNs to be reachable from the PSTN network (DDI routing). The Administrator creates an E.164 (PSTN)-to-DN (internal extensions) association to provide the DDI mapping.

Shared Line Across Sites Example

The following figure provides a basic Shared Line Across Sites configuration using one Inventory site ("c2Inventory") and two real sites ("c2s1" and "c2s2"). In this example there are two shared DNs (9000 and 9001 shown in red) and four site-specific DNs (1000 and 1001 at c2s1, 2000 and 2001 at c2s2). The inventory for the shared DNs are provisioned at the Customer hierarchy level to make them visible to all the sites under the customer. This allows the sites to configure the associated line and assign the line to a device. The inventory for the non-shared-across-sites DNs is still configured at the real sites (in blue) as it was in previous Cisco HCS releases. Notice that both shared DNs and non-shared DNs can co-exist for the same customer.

Phones are always configured on the real sites, and can use both shared and site-local lines. In the example above, each phone has one site-local line (for example, 1000), and one cross-site shared line (for example, 9000). The following is a summary of the configuration that resides at each hierarchy type:

Customer Hierarchy
- DN inventory for the lines to be shared across sites. Note that the DN inventory is visible across all sites under the customer. Allowing DN Inventory to be configured at the customer hierarchy node is an enhancement for the Shared Line Across Sites feature. Note that DN inventory can only be created at the customer hierarchy node when a non-SLC-based customer dial plan has been deployed. A transaction error is sent if the administrator attempts to create customer level DN inventory with an SLC-based dial plan.
Inventory Site, includes
- Line relations for the DNs to be shared across sites.
- Directory Number Routing (DNR) entry for the line relations configured at this site to make the DNs inter/intra-site dialable.
- E.164 inventory for the line relations configured at this site.
- E.164 associations for the line relations configured at this site.
- Line Class of Service (CoS) for the lines configured at this site. CoS is discussed in more detail in Class of Service for Shared Line Across Sites.
- Short codes for the line relations configured at this site.
Real Site, includes
- DN inventory for lines to be used only at this site. Note that these DNs can be shared by multiple phones within the site.
- Subscribers, configured from Subscriber Management > Subscribers, or Subscriber Management > Quick Add Subscriber.
- Line relations for the DNs configured at this site. These line relations do not have to be configured first; they are configured automatically any time a phone, extension mobility profile, or remote destination profile references a line that doesn't exist in the inventory site.
- Directory Number Routing (DNR) for each of the line relations configured at this site.
- E.164 inventory for lines created at this site.
- E.164 associations for lines created at this site.
- Device Class of Service (CoS) to be used for the phones configured at this site.
- Phones. These phones can reference lines that were defined in the Inventory Site or the Real Site where the phone exists.
- Extension mobility. These profiles can also reference lines that were defined in the Inventory Site or the Real Site where the phone exists.
- Single Number Reach. These profiles can reference lines that were defined in the Inventory Site or the Real Site where the profile is defined.

Fields in Cisco Unified Communications Domain Manager which reference DNs, such as the Pattern field in the Line tab of a Phone, are in a drop-down list of DN inventory. The drop-down list of DNs includes inventory defined at the customer level, combined with the inventory defined at the current site context. The administrator can chose either a cross-site shared DN or a site-local DN.

Inventory Site

It is important to understand that an Inventory Site is only an Inventory Site by name, not by type. An Inventory Site is just a regular site, is no different than any other site, does not have an "Inventory Site" checkbox, and is deployed exactly the same as any other site. It is only by convention that we're calling this an Inventory Site and designating this site as the repository for lines to be shared across sites.

The Inventory Site is created from the Site Management > Sites menu. It requires an NDL and a Country, and requires a site dial plan to be deployed.

Note

There is no enforcement of configuration ensuring that, for example, only lines are configured at the Inventory Site and not phones. It is the responsibility of the administrator to ensure the proper procedures and conventions are followed as documented in this guide. Therefore, it is important to ensure a good understanding of how the Inventory Site is to be used, and how the Inventory Site configuration relates to the configuration of the "real sites".

There are several caveats and restrictions that must be followed when using the Inventory Site as summarized below. Detailed configuration procedures are provided later in this document. For the purposes of this discussion, the term Site Group is used to describe an Inventory Site combined with the "real sites" which use the shared lines defined in the Inventory Site. The following diagram shows a customer with three Site Groups.

All sites in a site group must conform to the following rules:

The sites must be configured with the same NDL and Country. Any site that has the same NDL and Country as the Inventory Site can participate in the same site group. In fact, the NDL and Country settings are what defines the site group.
Shared lines configured in the Inventory Site of a site group can only be used by other sites in the same group, not in other groups. This means that shared lines cannot span NDLs, and cannot span countries.

Tip

If a line is potentially shareable, we recommend that you create the line in the Inventory Site, even if it will not be shared across sites immediately. The system does not support the ability to move a line from a real site to an Inventory Site, so to convert a line from site-local to cross-site shared, the line would need to be deleted from the real site and recreated in the Inventory Site.

Dial Plan Type for Shared Line Across Sites

The Shared Lines Across Sites feature only works if you are using a flat dial plan (Type 4, or a custom dial plan that is not site-specific). The reason is that the other dial plans (Types 1 to 3) have site location codes in the DN which do not work if the DN is shared by multiple sites.

If you're using the predefined dial plans, do not check the Site Location Code checkbox when deploying the Customer dial plan.

Class of Service for Shared Line Across Sites

Class of Service (CoS) refers to a Calling Search Space (CSS) that is specifically used to define call routing and feature processing for a line or a phone. There are a number of CSSs defined when a customer and site dial plan are deployed, and some of the CSSs are only used internally and should not be selected in the CSS drop-down list on a line or phone configuration page.

The Class of Service CSSs are listed in the Dial Plan Management > Site > Class of Service menu item. A few example CoSs are predefined when a site dial plan is deployed, but the intent is for the administrator to create their own CoSs to meet the desired call routing and feature processing behavior. Below is a summary of Class of Service as it pertains to Shared Lines Across Sites feature.

COS is used in two places:

Line Calling Search Space which appears in Cisco Unified Communications Domain Manager at Subscriber Management > Lines > Directory Number Basic Information tab > Calling Search Space
Device Calling Search Space which appears in Cisco Unified Communications Domain Manager at
- Subscriber Management > Phones > Phone tab > Calling Search Space Name
- Subscriber Management > Subscribers > Phones tab > Calling Search Space Name

Additionally, CoS can provide line-based routing (LBR) or device-based routing (DBR). For each call made from a phone, the device CSS of the phone is combined with the line CSS of the line from which the call is being made, and the features and routing for the call are processed based on the combined list of partitions of these two CSSs. The default set of CoSs provided when a site dial plan is deployed includes a device CoS for emergency dialing only, and several line CoSs for feature processing, national dialing, and international dialing and that support either DBR and LBR. The following table shows the default allocation of feature and routing duties between the two sets of CoSs.

Table 16. Default Class of Service for Shared Line Across Sites Feature
	Default Device CoS	Default Line CoS
Emergency call routing	yes*	—
Intrasite routing	—	yes
Intersite routing	—	yes
Local PSTN call routing	—	yes**
National PSTN call routing	—	yes
International PSTN call routing	—	yes
Feature processing	—	yes

*Emergency call routing is dependent on the country configured for the site. The country is used to route to the correct emergency number for that country (for example, 911 routes to 112 in the United Kingdom). Emergency call routing is assigned to the Device CoS because it is location-dependent, and must be tied to the site where the phone/user actually resides.

**Local call routing is dependent on local area codes defined in the site dial plan. The local area codes configured in the site dial plan allow dialing local dialing (for example 7-digit dialing in the United States).

As shown in the table above, routing is weighted heavily toward the line CoS because when the CoS is assigned to the line, it applies equally to the phone, extension mobility, and single number reach, which all typically share the same line configuration and provide similar dialing behavior for a given user. However, this assumes that the lines and devices are all constrained to individual sites. When we open up lines to be shared across sites, the site-specific configuration becomes more important in order to determine what to put in the device CoS versus the line CoS.

Class of Service (CoS) management for Shared Lines Across Sites is heavily dependent on the customer's specific deployment scenario. The distribution of work between the device CoS and the line CoS depends on the type of country dial plan, and the dialing behavior the customer wants.

For example, if the country dial plan is flat and closed like the Swiss dial plan, meaning that the subscriber numbers are not variable length and there is no site-specific area codes (only national dialing), then most of the routing can occur in the line CoS because there is not much site-specific dialing behavior.

However, if the country dial plan uses area codes and the customer wants a local dialing experience (ability to dial a shorter number such as 7-digit dialing in the United States, and relying on the dial plan to fill in the local area code), then local call routing must be in the device CoS because the device context is needed to determine which area codes to apply to the dialed number. Feature processing partitions can almost always stay with the line CoS since there is usually no geographic dependencies for the feature processing. The exception to this is Time of Day (TOD) routing which may vary depending on the site.

In order to decide how to distribute routing and feature processing between the line COS and the device CoS, refer to the table that follows.

Table 17. Routing and Feature Processing between Line CoS and Device CoS
	Line CoS	Device CoS
Emergency call routing	—	Emergency routing should always be location-specific
Intrasite routing	Always using the PrelSR route partition	—
Intersite routing	Always using the PrelSR route partition	—
Local call routing	When full E.164 number is always dialed for offnet calls, for example, national dial plans with no local call routing	When site-specific area codes and/or variable length subscriber numbers (local dialing behavior) are defined
National call routing	If local dialing is line-specific, national dialing should be line-specific.	If local dialing is device-specific, national dialing should be device-specific.
Toll-free call routing	If local dialing is line-specific, toll-free dialing should be line-specific.	If local dialing is device-specific, toll-free dialing should be device-specific.
International call routing	If local dialing is line-specific, international dialing should be line-specific.	If local dialing is device-specific, international dialing should be device-specific.
Service call routing	If local dialing is line-specific, service number dialing should be line-specific.	If local dialing is device-specific, service number dialing should be device-specific.

To speed up the process of configuring lines and phones when you create new Classes of Service, set the site-specific default line CSS and site-specific default device CSS (Site Management > Defaults). These fields appear in the following tabs:

Device Defaults > Default CUCM Device CSS
Line Defaults > Default CUCM Line CSS

Call Forward Considerations for Shared Line Across Sites

As the administrator, you can create the Call Forward CSS as a CoS for a particular deployment scenario. Considerations must be made based on whether the local, national, and/or international dialing is configured on the device CoS or line CoS.

Be aware that if the Call Forward CSS allows national and local PSTN routing, you may need to consider call forward scenarios when a line is not associated to a device and PSTN dialing is in the device CoS.

Phone, Subscriber, and Quick Add Subscriber use for Shared Line Across Sites

Phones and Subscribers should only be created at real sites, not Inventory Sites. This is not enforced in the workflows, but will help facilitate ongoing management of the configuration data for the customer. Lines referenced in the Phone page, the Subscriber page, or the Quick Add Subscriber page are created automatically if they have not already been provisioned in the Inventory Site and pushed to Cisco Unified Communications Manager. This is acceptable as long as you intend for these lines to be only referenced within one site. If a line gets created on a real site that you intended to share across sites, it is recommended that you delete the line, and recreate it in the Inventory Site.

The fields of interest on the Phone page are on the Phone tab and the Lines tab. The Phone tab is where you specify the Calling Search Space Name; this is the device-based routing class of service (CoS). By default this is the emergency routing CSS. Depending on choices made above in the Class of Service section, you might chose a different CSS here.

The Lines tab is where you pick the DN (Pattern) from the drop-down list, and where you configure the E.164Mask used for line presentation. The DN drop-down list includes DNs from the Customer DN inventory combined with the current site DN inventory. The E.164Mask is a free-form field and is not tied to the E.164 inventory currently; it must be manually entered. These are the only fields that are pertinent to the Shared Line Across Sites feature.

The Route Partition Name is automatically populated with the correct directory number partition based on the Pattern (DN) that is selected. Similar fields exist in the Subscriber tabs.

Hunt Groups and Call Pickup Groups for Shared Line Across Sites

Hunt Groups and Call Pickup Groups can be configured in either the Inventory Site or the real sites. If configured in the Inventory Site, the Hunt Groups and Call Pickup Groups can include any line configured in the Inventory Site, but cannot include lines created in other sites. Likewise, if configured in the real site, the Hunt Groups and Call Pickup Groups can include any line configured in the real site but not other sites.

We recommend that you configure Hunt Groups and Call Pickup groups in the Inventory Site if they need to include lines that are not all isolated to one site. The following figure provides an example of a Hunt Group that uses lines spanning multiple sites.

Figure 5. Hunt Group Example with Shared Link Across Sites Feature

Note that lines 1000, 1001, 2000, and 2001 are not themselves shared across sites. However, because all lines in one Hunt Group must exist at the same site, all four lines must be configured in the Inventory Site to be included in the one Hunt Group with Hunt Pilot 5500.

Also note that the Hunt Pilot DN inventory is at the customer level. Once the Hunt Pilot is assigned, that DN is marked as unavailable for any other usage (that is, it cannot be assigned to a device as a line, nor can it be used for another service pilot number).

Site Short Codes

Site short codes work the same for deployments that use shared lines across sites as they do for "real site" deployments. That is, short codes can be added to a site to allow shorter, convenient numbers to be dialed that are transformed into longer directory numbers. Normally, short codes are added to real sites that contain devices in order to allow users of those devices to dial shorter numbers to reach exiting directory numbers.

Because the inventory site doesn't contain devices, but only line inventory, site codes don't need to be added to the inventory site. Short code translation patterns are created on a site's Allow Internal (AInt) route partition.

Handling Voice Mail to Secondary Shared Lines

To handle Voice Mail to secondary shared lines, create a separate user for each shared line at the Inventory Site level, then enable the voice mailbox for that user so that it can be managed by all shared lines.

This approach:

Offers the ability to differentiate between voice mail deposited for primary and secondary lines
Provides separate message waiting indication (MWI) notifications for voice mail in the phone's primary and secondary line
Allows all configuration to be done in Cisco Unified Communication Domain Manager. There are no separate manual configurations required in Cisco Unity Connection or Cisco Unified Communications Manager.

Note

One additional license is required for the shared line user mailbox.

Shared Line Across Sites Configuration Procedures

Most of the configuration for Shared Lines Across Sites is the same as with conventional lines, but this section provides procedures to highlight the differences.

For conventional site-local lines, the lines can be configured automatically as part of the Phone, Subscriber, or Quick Add Subscriber workflows; the lines do not need to be configured separately first.

For lines to be shared across sites, they must be configured first in the Inventory Site, then referenced from Phone, Subscriber, or Quick Add Subscriber workflows.

Refer to the following procedures to configure Shared Lines Across Sites:

Configure Shared Line Across Sites - Customer
Configure Shared Line Across Sites - Inventory Site
Configure Shared Line Across Sites - Real Site

Configure Shared Line Across Sites - Customer

The customer configuration is similar except that you create DN inventory at the customer hierarchy for lines you would like to share (or potentially share) across sites.

Procedure

Step 1	Configure the Cisco Unified Communications Manager and Cisco Unity Connection devices. These can be at the customer level (dedicated) or above (shared).
Step 2	Configure the customer normally (for example, c2).
Step 3	Configure the Network Device List (NDL) for the customer (for example, c2Ndl) that will be used for your site group (NDL/Country combination).
Step 4	Deploy the customer dial plan. This must be a flat dial plan (for example, Type 4) because shared lines across site dictates that DNs cannot be site-specific. The Type 4 dial plan does not impose site-specific structure (in other words, site location codes). When configuring the customer dial plan, ensure that the Site Location Code checkbox is unchecked.
Step 5	Configure the DN inventory to be used across sites for shared lines (Dial Plan Management > Number Management > Directory Number Inventory). Note that you should leave the site drop-down list empty to create the inventory on the Customer hierarchy node.

Configure Shared Line Across Sites - Inventory Site

The "Inventory" Site is only needed if you want to configure Shared Lines Across Sites. If you do not have this requirement you do not need an Inventory Site and configuration is exactly as it is done normally. Most of the Inventory Site configuration is the same as configuration for a real site (for example, deploy site dial plan, configure DN inventory, and so on). The areas that are unique to the Inventory Site are provided in Steps 1, 3, and 5.

Procedure

Step 1	Configure the Inventory Site and specify the NDL and Country (for example, c2InventorySite). A different Inventory Site is needed for each NDL/Country combination (site group). If the customer only has one NDL and one Country, they only need one Inventory Site.
Step 2	Deploy the site dial plan (Type 4 will automatically be used based on the customer dial plan that was deployed).
Step 3	Create the new Classes of Service to be used as the default line CSS and update the Site Defaults procedure for the Inventory Site. Refer to Class of Service for Shared Line Across Sites for more information.
Step 4	Configure Directory Number Routing (DNR) for the shared lines (Dial Plan Management > Site > Directory Number Routing).
Step 5	Create line relations for each shared line (Subscriber Management > Line).
Step 6	Create E.164 inventory (Dial Plan Management > Number Management > Add E164 Inventory).
Step 7	Associate E.164 to DN ()Dial Plan Management > Number Management > E164 Associations (N to N).
Step 8	Configure Hunt Groups that use shared lines (Subscriber Management > Hunt Groups).
Step 9	Configure Call Pickup Groups that use shared lines (Subscriber Management > Call Pickup Groups).

Configure Shared Line Across Sites - Real Site

Configuration at the real sites is almost exactly the same as in past Cisco HCS releases. The major difference is that the Shared Lines Across Sites exist at the Inventory Site and therefore any configuration associated with those lines (CoS, DNR, E.164 associations, and so on) exists at the Inventory Site.

Procedure

Step 1	Configure the real site (for example c2s1, c2s2, and so on). Use the same NDL and Country as the Inventory Site (same site group).
Step 2	Deploy the site dial plan on each of the real sites (again, the customer dial plan enforces that the flat dial plan is used).
Step 3	Create DN inventory for an DNs that will be used only at this site.
Step 4	Create Directory Number Routing (DNR) for any DNs created at this site.
Step 5	Create E.164 inventory and associations for an DNs created at this site.
Step 6	Create Device Class of Service if needed. Refer to Class of Service for Shared Line Across Sites.
Step 7	Create Line Class of Service if needed for your site-specific lines. Refer to Class of Service for Shared Line Across Sites.
Step 8	Configure subscribers and phones (Subscriber Management > Subscribers, Quick Add Subscriber, or Phones). When configuring normal lines (lines that are not shared across sites), select a line from the local site DN inventory, not the customer-level DN inventory. The line is created at the local site as normal; you can configure line CoS, DNR, E.164 associations at this site as normal. Note that this includes shared lines that are only shared within the site. When configuring a shared line across sites, select a customer-level DN from the drop-down list. Remember, the line should be configured at the Inventory Site first.
Step 9	Configure site-specific Hunt Groups that use lines local to the real site.
Step 10	Configure site-specific Call Pickup Groups that use lines local to the real site.

Notes and Limitations

The following summarizes some of the limitations concerning the Shared Lines Across Sites feature:

A new Inventory Site is required for each new combination of NDL and Country (a “site group”). In other words, the lines configured at the Inventory Site are specific to the NDL and Country defined for that site.
All real sites that reference lines in an Inventory Site must be defined with the same NDL and Country. Ensure that this requirement is met, as it is not enforced in Cisco Unified Communications Domain Manager.
Shared lines cannot span countries or NDLs. This is necessary because Cisco Unified Communications Manager doesn't support shared lines across clusters. The country must be consistent so that line CoSs (defined in the Inventory Site) are correct for each device referencing the line (defined in the real site). Ensure that the correct association is made between Inventory Sites and real sites, as it is not enforced in Cisco Unified Communications Domain Manager.
When configuring a phone or subscriber at a real site, any reference to a DN that does not exist in the Inventory Site results in a new line being created at the real site as it did prior to this Cisco HCS release. In other words, if the Inventory Site doesn't exist, or a line hasn't been configured in the Inventory Site first, the system behaves as it did in previous Cisco HCS releases (backwards compatible).
If a line can be potentially shared, create it in the Inventory Site before referencing it by any devices. If the DN is used in a device before it is configured in the Inventory Site, the line is created in the real site and may not have the desired CoS or other configuration desired for a shared line.
When a line has been created (either at the Inventory Site or a real site), it cannot be moved. To move the line, delete the line and re-add it. For example, if you forget to define the line at the Inventory Site first and configure a device with a line, the line is created at the real site. You would need to delete the line from the real site and add it to the Inventory Site, then reassign it to the phone.
An Site Administrator logged in to a real site is not able to see the line configuration that exists at the Inventory Site. A Customer Administrator or above can see the line configurations at all of the sites.
The Shared Lines Across Sites feature only works when using a flat dial plan. The reason is that other dial plans have site location codes in the DN which won't make sense if the DN is shared by multiple sites. The default CUCDM.template bundle includes a Type 4 flat dial plan, but other custom dial plans that are not site-specific can be used.
Self-provisioning does not work for DNs defined at the customer level.
Although an Administrator can delete Inventory Sites, we do not recommend it. If the Inventory Site is deleted, all hunt groups, call pickup groups, voice mail pilot associations, and lines that are part of the Inventory Site are deleted. If there are devices on the “real” sites that reference these lines, they will no longer reference these lines as they will have been deleted. The customer-level DN inventory is still intact, though no lines are associated with these DNs because they are deleted when the Inventory Site is deleted. The hunt groups and call pickup groups are self-contained to the Inventory Site and are therefore, deleted as part of the deletion of the Inventory Site.
When the inventory site is deleted, this deletes all shared lines, Classes of Service, DNR, and any other configuration added at that site. The shared lines are removed from all devices on “real” sites which may have referenced them.
If an emergency number is dialed from any shared line, the number displayed on the other end should be the Emergency Call Back Number of the corresponding site.

Configure Tail End Hop Off

Note

The following task is applicable only if you are using Cisco Unified Communications Domain Manager 10.6(2) or later.

Follow these steps to manually configure Tail End Hop Off (TEHO) in Cisco Unified CDM:

Procedure

Step 1

Configure route-group

Select the hierarchy up to customer level.

For procedures, see Configure Route Groups.

Note

While adding a new route group, enter a name and then select the sip_trunk added to the remote LBO site (for example, RouteGroup: TEHO-RG, Device: L1LBO-SIP).

Step 2

Configure route-list

Select the hierarchy up to customer level.

For procedures, see Configure Route Lists.

Note

While adding a new route list, enter a name and then add three route groups (for example, RoutList: TEHO-RL, 1stRouteGroup: TEHO-RG, 2ndRouteGroup: SLRG-Natl, 3rdRouteGroup: RG-AGGR).

Step 3

Configure route-pattern

Log in as Provider or Reseller Administrator.
Select the hierarchy up to customer level.
Select Device Management > CUCM > Route Patterns
Click the Route Pattern from the list.
Go to Action, and then click Clone.
In the Pattern Definition tab, select the CUCM.
Edit the Route Pattern name.
Select the Route List that is configured when configuring the route list for TEHO (for example, RoutePattern: **[0-3]0[236-9]1.608!, RouteList: TEHO-RL). In this route pattern, 608 is the area code for the remote location).

Bias-Free Language

Results

Chapter: Manage Dial Plan

Manage Dial Plan

Cisco Hosted Collaboration Solution Roles and Privileges

Create a Customer Dial Plan

Procedure

Create a Site Dial Plan

Before you begin

Procedure

Update a Site Dial Plan

Procedure

What to do next

Configure Class of Service

Procedure

Example:

Clone a Class of Service

Procedure

Configure Short Code

Before you begin

Procedure

Example:

Configure Directory Number Routing

Procedure

Example:

Example:

Create a Customer

Procedure

Add Directory Number Inventory

Before you begin

Procedure

Example:

View Directory Number Inventory

Procedure

Run the Directory Number Inventory Audit Tool

Procedure

E.164 Inventory Management

Add E.164 Inventory

Procedure

Associate a Range of E.164 Numbers to a Range of Directory Numbers

Procedure

Associate a Set of E.164 Numbers to One Directory Number

Procedure

Manual Configuration for Intersite Cross-Cluster Support

Procedure

Manual Configuration for Local Breakout Support

Procedure

Create Voicemail Service

Before you begin

Procedure

What to do next

Associate Voicemail Services with a Customer

Before you begin

Procedure

Disassociate Voice Mail Services from Customers

Procedure

Define a Voicemail Pilot Number

Before you begin

Procedure

Creating DDIs for Voice Mail Pilot Numbers

Before you begin

Procedure

Associate Pilot Number to a Site

Before you begin

Procedure

Disassociate Pilot Number from a Site

Procedure

Adding Aggregation Trunk and Route Group and Associating to Existing Route List and SLRG

Procedure

Configure SIP Profiles

Procedure

SIP Profile Field Descriptions

Configure SIP Trunk Security Profiles

Procedure

SIP Trunk Security Profile Field Descriptions

Configure SIP Trunks

Procedure

SIP Trunks Field Descriptions

Reset SIP Trunks