Configuration Guide for Cisco Unified MeetingPlace Release 8.5
Configuring Call Control
Downloads: This chapterpdf (PDF - 374.0KB) | Feedback

Configuring Call Control for Cisco Unified MeetingPlace

Table Of Contents

Configuring Call Control for Cisco Unified MeetingPlace

Prerequisites for Configuring Call Control

How to Configure Call Control on WebEx-Scheduled Deployments

How to Configure Cisco Unified Communications Manager for a Multinode Topology

Configuring SIP Trunks for a Node

Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region

Configuring a Route List to Point to a Route Group

Configuring a Route Pattern to Point to a Route List

Configuring a SIP Route Pattern for Each Node

How to Configure Inter-Cluster Trunks on WebEx-Scheduled Deployments

How to Configure CUCM Clusters with SME (with a Cisco Unified MeetingPlace Node Configured Behind Your Local CUCM Cluster)

How to Configure CUCM Clusters with SME (with a Cisco Unified MeetingPlace Node Configured Behind SME)

How to Configure CUCM Clusters without SME

How to Configure Call Control for Voice Conferencing

Configuring SIP on Cisco Unified MeetingPlace

Configuring Cisco Unified Communications Manager Release 6.x or Later: SIP Trunk to Cisco Unified MeetingPlace

Configuring Cisco Unified Communications Manager Release 6.x or Later: Route Patterns

Configuring SIP Trunks Between Cisco Unified Communications Manager Release 5.x and Cisco Unified MeetingPlace Release 6.1 or Later

Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager Release 4.x or 5.x and Cisco Unified Communications Manager Release 6.1 or Later

Configuring Cisco Unified Communications Manager Release 4.x or 5.x: H.323 Trunk to Cisco Unified Border Element

Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns

Configuring Cisco Unified Border Element: H.323 to SIP Conversion

Configuring the Cisco IOS Gateway: Dial Peers to Cisco Unified MeetingPlace

Verifying the Call-Control Configuration

How to Configure Call Control for Video Conferencing with H.323 Endpoints

How to Configure the Cisco IOS Gatekeeper

Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided By Gatekeeper

Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager

Configuring Cisco Unified Communications Manager Release 6.1 or Later: H.323 Endpoints

How to Configure Secure Conferencing Mode

Synchronizing Your Meeting Types

Creating a Secure SIP Trunk Profile

Creating a Secure SIP Trunk

Defining a Route Pattern for a Secure SIP Trunk

Configuring Certificates on a Cisco Unified MeetingPlace System

Configuring a Self-Signed Cisco Unified MeetingPlace Certificate

Configuring a Third-Party Generated Cisco Unified MeetingPlace Certificate

Configuring Certificates on SIP Proxies

Uploading Root Certificates of the Cisco Unified MeetingPlace Certificate to the Cisco Unified Communications Manager

Uploading Cisco Unified MeetingPlace or CA Root Certificates to the Cisco Unified Communications Manager.

Configuring SIP Proxy Certificates on a Cisco Unified MeetingPlace System

Sending Certificates to the Media Server

Enabling Secure Conferencing on your Cisco Unified MeetingPlace System


Configuring Call Control for Cisco Unified MeetingPlace


Release 8.5
Revised: June 14, 2012 4:46 pm

Prerequisites for Configuring Call Control

How to Configure Call Control on WebEx-Scheduled Deployments

How to Configure Call Control for Voice Conferencing

How to Configure Call Control for Video Conferencing with H.323 Endpoints

How to Configure Secure Conferencing Mode

Prerequisites for Configuring Call Control

Learn the benefits and restrictions of each supported call-control deployment option, and choose the best option for your Cisco Unified MeetingPlace system. Understand your deployment so that you know ahead of time which call-control devices you need to configure. See the Planning Guide for Cisco Unified MeetingPlace at http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_implementation_design_guides_list.html.

Verify that the versions of your call-control devices and Cisco Unified MeetingPlace are compatible. See the System Requirements for Cisco Unified MeetingPlace at http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_device_support_tables_list.html.

Install the call-control devices as described in the installation documentation for those devices.

Verify that the Cisco Unified IP Phones and other endpoints are connected and added to the database of your call-control devices. See the Compatibility Matrix for Cisco Unified MeetingPlace at http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_device_support_tables_list.html

Verify that you can place and receive internal and external calls on the Cisco Unified IP Phones and other endpoints.

Install Cisco Unified MeetingPlace as described in the Quick Start for Installing and Configuring Cisco Unified MeetingPlace Release 8.5 module.Install Cisco Unified MeetingPlace as described in the Quick Start for Installing and Configuring Cisco Unified MeetingPlace Release 8.5 module.

Related Topics

Configuring Access Phone Numbers and Notification Labelsmodule

Configuring Direct Inward Dial on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

How to Configure Call Control on WebEx-Scheduled Deployments

How to Configure Cisco Unified Communications Manager for a Multinode Topology

How to Configure Inter-Cluster Trunks on WebEx-Scheduled Deployments

How to Configure Cisco Unified Communications Manager for a Multinode Topology

Configuring Cisco Unified Communications Manager for a multinode topology is a multi-task procedure. Complete the following tasks in the order presented.

Configuring SIP Trunks for a Node

Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region

Configuring a Route List to Point to a Route Group

Configuring a Route Pattern to Point to a Route List

Configuring a SIP Route Pattern for Each Node


Note Do not perform these procedures if you are configuring a SIP control cluster.


Before You Begin

Make sure you configure DNS on CUCM. CUCM must be able to resolve the hostnames of the Cisco Unified MeetingPlace nodes by using DNS. If CUCM is unable to resolve the Cisco Unified MeetingPlace nodes during the REFER process by using DNS, the REFER process will fail.

Configuring SIP Trunks for a Node

To ensure proper call control to a specific Cisco Unified MeetingPlace node you must configure two SIP trunks:

The first SIP trunk handles straight dial-in calls into a Cisco Unified MeetingPlace node, and requires the SIP trunk to be associated with a SIP trunk security profile with the incoming port value set to 5060.

The second SIP trunk handles SIP REFER calls that are redirected to a Cisco Unified MeetingPlace node and requires the SIP trunk to be associated with a SIP trunk security profile with the incoming port value set to 5070.

Before You Begin

See Configuring a SIP Trunk Security Profile in Cisco Unified Communications Manager for Cisco Unified MeetingPlace in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.

Create two SIP trunk security profiles:

The first SIP trunk security profile is to be used with the dial-in SIP trunk. Configure the SIP trunk security profile properties so that the incoming port is set to 5060.

The second SIP trunk security profile is to be used with the REFER SIP trunk. Configure the SIP trunk security profile properties so that the incoming port is set to 5070.

Procedure


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Device > Trunk.

Step 4 Select Add New.

Step 5 Select SIP Trunk from the Trunk Type drop-down list.

The Device Protocol becomes SIP and the Trunk Service Type becomes None.

Step 6 Select Next.

Step 7 Configure the fields on the Trunk Configuration page as follows:

Field
Format

Device name

Enter the host name.

Description

Enter a description. For example, "SIP trunk for straight dial-in" or "Refer the SIP Route to host name."

Device pool

None.

Call Classification

Select Use System Default.

Retry Video Call as Audio

Select this checkbox.

PSTN Access

Select this checkbox.

Destination address

Enter the host name.

Presence Group

Select Standard Presence group.

SIP Trunk Security Profile

Select the SIP trunk security profile that you created for straight dial-in (the REFER SIP trunk).

SIP Profile

Select Standard SIP Profile.

DTMF Signaling Method

Select No Preference.


Step 8 Select Save.

Step 9 Repeat Steps 3 to 8 to create a second SIP trunk for this node.

Step 10 Repeat Steps 3 to 8 for each node on your system.


Related Topics

Configuring a SIP Trunk Security Profile in Cisco Unified Communications Manager for Cisco Unified MeetingPlace in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module

What to Do Next

Proceed to the "Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region" section.

Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region

This task describes how to associate the SIP trunk you configured with a route group for straight dial-in.

Before You Begin

Complete the "Configuring SIP Trunks for a Node" section.

This task assumes that you are still signed in to Cisco Unified Communications Manager.

Procedure


Step 1 Select Call Routing > Route/Hunt > Route Group.

Step 2 Select Add New.

Step 3 Enter a route group name (for example, "RG1").

Step 4 Select Circular from the Distribution Algorithm drop-down list.

Step 5 Select the devices you want in your route group, then click Add to Route Group.


Note If there is one dial-in number for a region or site, make sure to add SIP trunks for all the conferencing nodes in each site or region to a route group and associate that route group with a route list mapped to a pattern for the dial-in number for that site. This ensures that calls are distributed evenly across all the conferencing nodes. If you have more than one dial-in number for a site or region, you can map each one to the same route list.


Step 6 Select Save.


What to Do Next

Proceed to the "Configuring a Route List to Point to a Route Group" section.

Configuring a Route List to Point to a Route Group

Before You Begin

Complete the "Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region" section.

This task assumes that you are still signed in to Cisco Unified Communications Manager.

Procedure


Step 1 Select Call Routing > Route/Hunt > Route List.

Step 2 Select Add New.

Step 3 Enter a name and description (for example, "RL1").

Step 4 Select Default from the Cisco Unified Communications Manager Group drop-down list.

Step 5 Check Enable this Route List and select Add Route Group.

The Route List Detail Configuration page displays.

Step 6 Select your route group in the Route List Member Selection window and click Select your Route Group.

Step 7 Select Save.


What to Do Next

Proceed to the "Configuring a Route Pattern to Point to a Route List" section.

Configuring a Route Pattern to Point to a Route List

Before You Begin

Complete the "Configuring a Route List to Point to a Route Group" section.

This task assumes that you are still signed in to Cisco Unified Communications Manager.

Procedure


Step 1 Select Call Routing > Route/Hunt > Route Pattern.

Step 2 Select Add New.

Step 3 Configure the fields in the Route Pattern Configuration window:

Route Pattern: Enter your call route for the Route Group you configured.

Description: Enter a description (for example, "Route to RL1").

Gateway/Route List: Select your Route Group from the drop-down list.

Step 4 Select Save.


What to Do Next

Proceed to the "Configuring a SIP Route Pattern for Each Node" section.

Configuring a SIP Route Pattern for Each Node

This task describes how to associate the SIP REFER trunk you configured for a particular node to a SIP route pattern.

Restrictions

The Cluster Fully Qualified Domain Name (CFQDN) parameter in the Clusterwide Domain Configuration section of the Cisco Unified Communications Manager Enterprise Parameters (System > Enterprise Parameters) must either be blank or configured so that it does not match the hostname of any of the Cisco Unified MeetingPlace nodes. If a match occurs, SIP REFER will not function properly because the call will not be routed by a SIP route pattern.

Before You Begin

Complete the "Configuring a Route Pattern to Point to a Route List" section.

This task assumes that you are still signed in to Cisco Unified Communications Manager.

Procedure


Step 1 Select Call Routing > SIP Route Pattern.

Step 2 Select Add New.

Step 3 Configure the fields on the SIP Route Pattern Configuration page.

Pattern Usage: Select Domain Routing.

IPv4 Pattern: Enter the domain name (FQDN) of your node.

Description: Enter a description (for example, "Refer SIP Trunk to domain name.")

Route Partition: Select None.

Select the SIP REFER trunk of the corresponding node from the SIP Trunk drop-down list.

Step 4 Select Save.

Step 5 Repeat Steps 2 to 4 for each node on your system.


How to Configure Inter-Cluster Trunks on WebEx-Scheduled Deployments

You can configure inter-cluster trunks with Cisco Unified Communications Manager by using Session Manager Edition (SME) or by configuring multiple SIP trunks with associated route groups, lists, and patterns.

See Figure 1 for an illustration of a typical inter-cluster configuration in a deployment with multiple regions.

How to Configure CUCM Clusters with SME (with a Cisco Unified MeetingPlace Node Configured Behind Your Local CUCM Cluster)

How to Configure CUCM Clusters with SME (with a Cisco Unified MeetingPlace Node Configured Behind SME)

How to Configure CUCM Clusters without SME

Figure 1 Configuring Inter-Cluster Trunks on Multiple CUCMs

How to Configure CUCM Clusters with SME (with a Cisco Unified MeetingPlace Node Configured Behind Your Local CUCM Cluster)

You can configure CUCM clusters with Session Manager Edition (SME). An SME is essentially a CUCM installation. An SME typically does not have any devices configured. An SME primarily has SIP trunks pointing to each of the CUCM clusters that it connects with. Configure a trunk for each cluster to the SME that connects with the other clusters in your deployment and to the devices in those clusters.

Procedure


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Install and configure SME. Refer to the Cisco Unified Communications Manager documentation for more information.

Step 3 Perform the following on the SME.

a. Configure a SIP trunk to each CUCM cluster.

b. Configure two route patterns for each SIP trunk:

One route pattern to reach the devices on the cluster

One route pattern to reach the Cisco Unified MeetingPlace nodes on the cluster

c. Configure a SIP route pattern for each SIP trunk, one for each Cisco Unified MeetingPlace node in the deployment.

Step 4 Perform the following on each CUCM cluster:

Configure the following on your CUCM cluster to reach the local Cisco Unified MeetingPlace nodes:

a. A SIP trunk for each of the local Cisco Unified MeetingPlace nodes

b. A route group for the SIP trunks of the local Cisco Unified MeetingPlace nodes

c. A route list for the route group

d. A SIP route pattern for each local Cisco Unified MeetingPlace node

Configure the following in CUCM to reach the SME:

a. A SIP trunk to the SME

b. Multiple route patterns for the SME trunk:

To reach the devices on each CUCM cluster

To reach the Cisco Unified MeetingPlace nodes on each CUCM cluster

c. SIP route patterns for the SME trunk:

To reach each of the other non-local Cisco Unified MeetingPlace nodes


Related Topics

Cisco Unified Communications Manager documentation located at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html

Configuring SIP Trunks for a Node

Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region

Configuring a Route List to Point to a Route Group

Configuring a Route Pattern to Point to a Route List

Configuring a SIP Route Pattern for Each Node

Cisco Unified Communications Manager documentation

How to Configure CUCM Clusters with SME (with a Cisco Unified MeetingPlace Node Configured Behind SME)

Procedure


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Install and Configure SME. Refer to the Cisco Unified Communications Manager documentation for more information.

Step 3 Configure the following on the SME:

a. A SIP trunk to each CUCM cluster

b. A route pattern for each SIP trunk, to reach the devices on the cluster

c. A SIP trunk to reach each of the Cisco Unified MeetingPlace nodes in the deployment

d. A route pattern for each of the trunks to the Cisco Unified MeetingPlace nodes

e. A SIP route pattern for each SIP trunk that points to the Cisco Unified MeetingPlace nodes in the deployment

Step 4 Configure the following on each CUCM cluster:

a. A SIP trunk to SME

b. Multiple route patterns for the SME trunk:

To reach the devices on each CUCM cluster

To reach each of the Cisco Unified MeetingPlace nodes in the deployment

c. SIP route patterns for the SME trunk:

To reach each of Cisco Unified MeetingPlace nodes in the deployment


Related Topics

Cisco Unified Communications Manager documentation located at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html

Configuring SIP Trunks for a Node

Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region

Configuring a Route List to Point to a Route Group

Configuring a Route Pattern to Point to a Route List

Configuring a SIP Route Pattern for Each Node

Cisco Unified Communications Manager documentation

How to Configure CUCM Clusters without SME

Procedure

Perform the following procedure on each node in your CUCM cluster:


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Configure a SIP trunk for each Cisco Unified MeetingPlace node in the deployment.

Step 3 Configure route group to each SIP trunk.

Step 4 Configure a route list to point to each route group.

Step 5 Configure a SIP route pattern for each Cisco Unified MeetingPlace node in the deployment.


Related Topics

Cisco Unified Communications Manager documentation located at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html

Configuring SIP Trunks for a Node

Configuring a Route Group for Inbound Calls to a MeetingPlace Site or Region

Configuring a Route List to Point to a Route Group

Configuring a Route Pattern to Point to a Route List

Configuring a SIP Route Pattern for Each Node

Cisco Unified Communications Manager documentation

How to Configure Call Control for Voice Conferencing

Configuring SIP on Cisco Unified MeetingPlace

Configuring Cisco Unified Communications Manager Release 6.x or Later: SIP Trunk to Cisco Unified MeetingPlace

Configuring Cisco Unified Communications Manager Release 6.x or Later: Route Patterns

Configuring SIP Trunks Between Cisco Unified Communications Manager Release 5.x and Cisco Unified MeetingPlace Release 6.1 or Later

Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager Release 4.x or 5.x and Cisco Unified Communications Manager Release 6.1 or Later

Configuring Cisco Unified Communications Manager Release 4.x or 5.x: H.323 Trunk to Cisco Unified Border Element

Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns

Configuring Cisco Unified Border Element: H.323 to SIP Conversion

Configuring the Cisco IOS Gateway: Dial Peers to Cisco Unified MeetingPlace

Verifying the Call-Control Configuration

Configuring SIP on Cisco Unified MeetingPlace

Complete this task to connect Cisco Unified MeetingPlace to supported call-control devices.

Before You Begin

Complete the "Prerequisites for Configuring Call Control" section.

Procedure


Step 1 Sign in to the Administration Center.

Step 2 Select System Configuration > Call Configuration > SIP Configuration.

Step 3 Configure the fields on the SIP Configuration Page.

Step 4 Select Save.


Related Topics

Field Reference: SIP Configuration Pagein the Administration Center Page References for Cisco Unified MeetingPlace module

What to Do Next

Perform one of these actions:

If your call-control network includes Cisco Unified Communications Manager Release 6.x (or later), proceed to the "Configuring Cisco Unified Communications Manager Release 6.x or Later: SIP Trunk to Cisco Unified MeetingPlace" section.

If your call-control network does not include Cisco Unified Communications Manager, proceed to the "Configuring the Cisco IOS Gateway: Dial Peers to Cisco Unified MeetingPlace" section.

Configuring Cisco Unified Communications Manager Release 6.x or Later: SIP Trunk to Cisco Unified MeetingPlace

Before You Begin

Complete the "Configuring SIP on Cisco Unified MeetingPlace" section.

We recommend that you configure a Calling Search Space in Cisco Unified Communications Manager that does the following:

Allows dial-out calls to meeting participants and the help desk Attendant.

Prevents toll fraud by blocking unwanted dial-out calls, for example, to international or premium-rate phone numbers.

See the Administration Guide for your release of Cisco Unified Communications Manager at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html.

We recommend that you create a SIP trunk security profile in Cisco Unified Communications Manager specifically for Cisco Unified MeetingPlace.

See "Configuring a SIP Trunk Security Profile in Cisco Unified Communications Manager for Cisco Unified MeetingPlace" in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.

If you want to prevent conference disruption by music when a user places a call on hold, complete the "Configuring Cisco Unified Communications Manager: Music On Hold" task in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.

You perform this task in the Cisco Unified Communications Manager Administration pages. Because the pages and menus vary by release, you should check the Cisco Unified Communications Manager Administration online help for step-by-step instructions that are specific to your release.

Procedure


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Device > Trunk.

Step 4 Select Add New.

Step 5 In the Trunk type field, select SIP Trunk.

Step 6 Select Next.

Step 7 Configure the fields described in Table 1.

Table 1 Fields for Adding a SIP Trunk in Cisco Unified Communications Manager Release 6.x (or Later) 

Field
Action

Device Name

Enter a unique identifier for this trunk, such as the name or IP address of the Cisco Unified MeetingPlace server.

Device Pool

AAR Group

The device pool must use a codec that is compatible with the conferencing gateway (or bridge).

For security and toll-fraud prevention, use a device pool and an automatic alternate routing (AAR) group that will block any undesired phone numbers from being dialed out.

Media Resource Group List

(Optional) If Cisco Unified MeetingPlace-supported endpoints are registered to this Cisco Unified Communications Manager, we recommend that you choose one of the following to prevent conference calls from being disrupted by music whenever a user places a call on hold:

Default value of <None>.

A Media Resource Group List that does not contain music on hold resources.

Media Termination Point Required

Uncheck this check box.

Destination Address

The DNS hostname or IP address of the Cisco Unified MeetingPlace Application Server.

In an Application Server Failover deployment, make sure you enter the shared hostname and IP address of eth0.

Destination Port

Keep the default value of 5060.

SIP Trunk Security Profile

Select the SIP trunk security profile that you created specifically for Cisco Unified MeetingPlace.

If you did not create a SIP trunk security profile, select the default Non Secure SIP Trunk Profile.

Rerouting Calling Search Space

Make sure to set this field appropriately to ensure call transfers (out to attendant or to other systems) are successful. Consult the Cisco Unified Communications Manager administrator for the appropriate calling search space (CSS) to use.

DTMF Signaling Method

Select No Preference.


Step 8 Configure all other required fields appropriately for your current deployment.

If you configured a Calling Search Space to block unwanted dial-out calls, apply the Calling Search Space accordingly to the SIP trunk.


Tip For field descriptions, select Help > This Page.


Step 9 Select Save.


Related Topics

Configuring Application Server Failover for Cisco Unified MeetingPlace on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module

Configuring Cisco Unified Communications Manager: Music On Hold in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module

What to Do Next

Proceed to the "Configuring Cisco Unified Communications Manager Release 6.x or Later: Route Patterns" section.

Configuring Cisco Unified Communications Manager Release 6.x or Later: Route Patterns

Route patterns enable Cisco Unified Communications Manager to route calls to Cisco Unified MeetingPlace by associating phone numbers with the SIP trunk.

Before You Begin

Complete the "Configuring Cisco Unified Communications Manager Release 6.x or Later: SIP Trunk to Cisco Unified MeetingPlace" section.

Write down each of the phone numbers from the Cisco Unified MeetingPlace Administration Center:

(WebEx-managed deployments) Access phone numbers configured on the Cisco WebEx Site Configuration Page

(MeetingPlace-managed and audio-only deployments) Access phone numbers configured on the Usage Configuration Page

Direct Inward Dial (DID) numbers—only if you enable DID through the Route calls to meeting ID that matches DID field

You perform this task in the Cisco Unified Communications Manager Administration pages. Because the pages and menus vary by release, you should check the Cisco Unified Communications Manager Administration online help for step-by-step instructions that are specific to your release.

Procedure


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Call Routing > Route/Hunt > Route Pattern.

Step 4 Select Add New.

Step 5 Configure the fields described in Table 2.

Table 2 Fields for Adding a Route Pattern in Cisco Unified Communications Manager Release 6.x (or Later) 

Field
Action

Route Pattern

Enter the Cisco Unified MeetingPlace phone number.

Requirements:

This number must not conflict with any other route pattern defined in this Cisco Unified Communications Manager cluster.

Do not enter any spaces in this field.

Gateway/Route List

Select the Device Name of the SIP trunk to Cisco Unified MeetingPlace.

Call Classification

Select OnNet.

Provide Outside Dial Tone

Uncheck the check box.


Step 6 Configure all other required fields appropriately for your current deployment.


Tip For field descriptions, select Help > This Page.


Step 7 Select Save.

Step 8 Select OK to any pop-up dialog box messages that you see.

Step 9 Repeat this procedure as necessary to route calls to each access phone number and DID number for your Cisco Unified MeetingPlace system.


Related Topics

SIP Configuration Pagein the Administration Center Page References for Cisco Unified MeetingPlace module

Configuring Access Phone Numbers and Notification Labelsmodule

Configuring Direct Inward Dial on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

What to Do Next

Proceed to the "Verifying the Call-Control Configuration" section.

Configuring SIP Trunks Between Cisco Unified Communications Manager Release 5.x and Cisco Unified MeetingPlace Release 6.1 or Later

Perform this task if you have already deployed Cisco Unified Communications Manager Release 5.x and are using Cisco Unified Communications Manager Release 6.1 (or later) to provide front-end signaling for Cisco Unified MeetingPlace.

Before You Begin

You can instead choose to configure inter-cluster trunks (instead of SIP trunks) between Cisco Unified Communications Manager Release 5.x and Cisco Unified Communications Manager Release 6.1 (or later). If this is the case, do not perform this task. Instead, see the "Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager Release 4.x or 5.x and Cisco Unified Communications Manager Release 6.1 or Later" section.

Complete the "Configuring Cisco Unified Communications Manager Release 6.x or Later: Route Patterns" section.

Perform this task on both of these servers:

Cisco Unified Communications Manager Release 5.x

Cisco Unified Communications Manager Release 6.1 (or later)

We recommend that you configure a Calling Search Space in Cisco Unified Communications Manager that does the following:

Allows dial-out calls to meeting participants and the help desk Attendant.

Prevents toll fraud by blocking unwanted dial-out calls, for example, to international or premium-rate phone numbers.

See the Administration Guide for your release of Cisco Unified Communications Manager at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html.

You perform this task in the Cisco Unified Communications Manager Administration pages. Because the pages and menus vary by release, you should check the Cisco Unified Communications Manager Administration online help for step-by-step instructions that are specific to your release.

Procedure


Step 1 Go to http://ccm-server/ccmadmin/main.asp, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Device > Trunk.

Step 4 Select Add New.

Step 5 In the Trunk Type field, select SIP Trunk.

Step 6 Select Next.

Step 7 Configure the fields described in Table 3.

Table 3 Fields for Adding a SIP Trunk in Cisco Unified Communications Manager 

Field
Action

Device Name

Enter a unique identifier for this trunk, for example:

If you are configuring Cisco Unified Communications Manager Release 5.x, enter the name or IP address of the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

If you are configuring Cisco Unified Communications Manager Release 6.1 (or later), enter the name or IP address of the Cisco Unified Communications Manager Release 5.x server.

Device Pool

If no device pools are defined, select Default.

If device pools are already defined, either create a new device pool or choose an existing device pool for a region with a codec that is compatible with the conferencing gateway (or bridge).

Media Resource Group List

(Optional) If Cisco Unified MeetingPlace-supported endpoints are registered to this Cisco Unified Communications Manager, we recommend that you choose one of the following to prevent conference calls from being disrupted by music whenever a user places a call on hold:

Default value of <None>.

A Media Resource Group List that does not contain music on hold resources.

Media Termination Point Required

Uncheck this check box.

Destination Address

Enter the destination IP address, specifically:

If you are configuring Cisco Unified Communications Manager Release 5.x, enter the IP address of the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

If you are configuring Cisco Unified Communications Manager Release 6.1 (or later), enter the name or IP address of the Cisco Unified Communications Manager Release 5.x server.

Destination Port

Incoming Port

Keep the default value of 5060.


Step 8 Configure all other required fields appropriately for your current deployment.

If you configured a Calling Search Space to block unwanted dial-out calls, apply the Calling Search Space accordingly to the SIP trunk.


Tip For field descriptions, select Help > This Page.


Step 9 Select Save.

Step 10 Repeat this task so that both of these servers are configured with SIP trunks that point to each other:

Cisco Unified Communications Manager Release 5.x

Cisco Unified Communications Manager Release 6.1 (or later)


Related Topics

Configuring Operator Assistancein the Configuring Attendant Settings for Cisco Unified MeetingPlace module

Configuring Cisco Unified Communications Manager: Music On Hold in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module

What to Do Next

Proceed to the "Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns" section.

Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager Release 4.x or 5.x and Cisco Unified Communications Manager Release 6.1 or Later

Perform this task if you already deployed Cisco Unified Communications Manager Release 4.x or 5.x and are using Cisco Unified Communications Manager Release 6.1 (or later) to provide front-end signaling for Cisco Unified MeetingPlace.

Before You Begin

You can instead choose to configure SIP trunks (instead of inter-cluster trunks) between Cisco Unified Communications Manager Release 5.x and Cisco Unified MeetingPlace Release 6.1 (or later). If this is the case, do not perform this task. Instead, see the "Configuring SIP Trunks Between Cisco Unified Communications Manager Release 5.x and Cisco Unified MeetingPlace Release 6.1 or Later" section.

Complete the "Configuring Cisco Unified Communications Manager Release 6.x or Later: Route Patterns" section.

Perform this task on both of these servers:

Cisco Unified Communications Manager Release 4.x or 5.x

Cisco Unified Communications Manager Release 6.1 (or later)

We recommend that you configure a Calling Search Space in Cisco Unified Communications Manager that does the following:

Allows dial-out calls to meeting participants and the help desk Attendant.

Prevents toll fraud by blocking unwanted dial-out calls, for example, to international or premium-rate phone numbers.

See the Administration Guide for your release of Cisco Unified Communications Manager at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html.

You perform this task in the Cisco Unified Communications Manager Administration pages. Because the pages and menus vary by release, you should check the Cisco Unified Communications Manager Administration online help for step-by-step instructions that are specific to your release.

Procedure


Step 1 Go to http://ccm-server/ccmadmin/main.asp, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Device > Trunk.

Step 4 Select Add New.

Step 5 In the Trunk type field, select Inter-Cluster Trunk (Non-Gatekeeper Controlled).

Step 6 Select Next.

Step 7 Configure the fields described in Table 4.

Table 4 Fields for Adding an Inter-Cluster Trunk in Cisco Unified Communications Manager 

Field
Action

Device Name

Enter a unique identifier for this trunk, for example:

If you are configuring Cisco Unified Communications Manager Release 4.x or 5.x, enter the name or IP address of the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

If you are configuring Cisco Unified Communications Manager Release 6.1 (or later), enter the name or IP address of the Cisco Unified Communications Manager Release 4.x or 5.x server.

Device Pool

If no device pools are defined, select Default.

If device pools are already defined, either create a new device pool or choose an existing device pool for a region with a codec that is compatible with the conferencing gateway (or bridge).

Media Resource Group List

(Optional) If Cisco Unified MeetingPlace-supported endpoints are registered to this Cisco Unified Communications Manager, we recommend that you choose one of the following to prevent conference calls from being disrupted by music whenever a user places a call on hold:

Default value of <None>.

A Media Resource Group List that does not contain music on hold resources.

Media Termination Point Required

Uncheck this check box.

Server 1 IP Address/Host Name

Identify the target server, specifically:

If you are configuring Cisco Unified Communications Manager Release 4.x or 5.x, specify the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

If you are configuring Cisco Unified Communications Manager Release Release 6.1 (or later), specify the Cisco Unified Communications Manager Release 4.x or 5.x server.


Step 8 Configure all other required fields appropriately for your current deployment.

If you configured a Calling Search Space to block unwanted dial-out calls, apply the Calling Search Space accordingly to the SIP trunk.


Tip For field descriptions, select Help > This Page.


Step 9 Select Save.

Step 10 Repeat this task so that both of the servers are configured with SIP trunks that point to each other:

Cisco Unified Communications Manager Release 4.x or 5.x

Cisco Unified Communications Manager Release 6.1 (or later)


Related Topics

Configuring Cisco Unified Communications Manager: Music On Hold in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module

What to Do Next

Proceed to the "Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns" section.

Configuring Cisco Unified Communications Manager Release 4.x or 5.x: H.323 Trunk to Cisco Unified Border Element

Before You Begin

Complete the "Configuring SIP on Cisco Unified MeetingPlace" section.

This task is performed in the Cisco Unified Communications Manager administration interface. Because the pages and menus vary by release, you might need to see the Cisco Unified Communications Manager online help for more accurate step-by-step instructions than those provided in this procedure.

Procedure


Step 1 Go to http://ccm-server/ccmadmin/main.asp, where ccm-server is the fully qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Device > Trunk.

Step 4 Select Add New.

Step 5 In the Trunk type field, select Inter-Cluster Trunk (Non-Gatekeeper Controlled).

Step 6 Select Next.

Step 7 Configure the fields described in Table 5.

Table 5 Fields for Adding an Inter-Cluster Trunk in Cisco Unified Communications Manager Release 4.x or 5.x 

Field
Action

Device Name

Enter a unique identifier for this trunk, such as the name or IP address of the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

Device Pool

If no device pools are defined, select Default.

If the Cisco Unified Communications Manager deployment uses customer-defined device pools, either create a new device pool or choose an existing device pool for a region with a codec that is compatible with the conferencing gateway (or bridge).

Media Termination Point Required

Uncheck this check box.

Server 1 IP Address/Host Name

Identify the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.


Step 8 Configure all other required fields appropriately for your current deployment.


Tip For field descriptions, select Help > This Page.


Step 9 Select Save.


What to Do Next

Proceed to the "Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns" section.

Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns

Use this procedure to configure route patterns to enable Cisco Unified Communications Manager Release 4.x or 5.x to route calls that are placed to Cisco Unified MeetingPlace phone numbers. The route patterns associate the Cisco Unified MeetingPlace phone numbers with one of the following, depending on your deployment:

Inter-cluster trunk to the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace

(Cisco Unified Communications Manager Release 5.x only) SIP trunk to the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace

Cisco IOS gateway with Cisco Unified Border Element

Before You Begin

Complete the "Configuring SIP on Cisco Unified MeetingPlace" section.

Complete one of these items, depending on your deployment:

Configuring SIP Trunks Between Cisco Unified Communications Manager Release 5.x and Cisco Unified MeetingPlace Release 6.1 or Later

Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager Release 4.x or 5.x and Cisco Unified Communications Manager Release 6.1 or Later

Make sure that your Cisco Unified Communications Manager configuration database already includes a gateway entry for the Cisco Unified Border Element. See the Cisco Unified Communications Manager online help for information about finding or adding gateways in Cisco Unified Communications Manager.

Write down each of the phone numbers from the Cisco Unified MeetingPlace Administration Center:

Access phone numbers configured on the Usage Configuration Page

Direct Inward Dial (DID) numbers—only if you enable DID through the Route calls to meeting ID that matches DID field

You perform this task in the Cisco Unified Communications Manager Administration pages. Because the pages and menus vary by release, you should check the Cisco Unified Communications Manager Administration online help for step-by-step instructions that are specific to your release.

Procedure


Step 1 Go to http://ccm-server/ccmadmin/main.asp, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Call Routing > Route/Hunt > Route Pattern.

Step 4 Select Add New.

Step 5 Configure the fields described in Table 6.

Table 6 Fields for Adding a Route Pattern in Cisco Unified Communications Manager Release 4.x or 5.x 

Field
Action

Route Pattern

Enter the Cisco Unified MeetingPlace phone number.

Requirements:

This number must not conflict with any other route pattern defined in this Cisco Unified Communications Manager cluster.

Do not enter any spaces in this field.

Gateway/Route List

Select the Device Name of one of the following, depending on your deployment:

Inter-cluster trunk to the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

(Cisco Unified Communications Manager Release 5.x only) SIP trunk to the Cisco Unified Communications Manager Release 6.1 (or later) server that provides front-end signaling for Cisco Unified MeetingPlace.

Cisco IOS gateway with Cisco Unified Border Element

Call Classification

Select OffNet.


Step 6 Configure all other required fields appropriately for your current deployment.


Tip For field descriptions, select Help > This Page.


Step 7 Select Save.

Step 8 Select OK to any pop-up dialog box messages that you see.

Step 9 Repeat this procedure as necessary to route calls to each access phone number and DID number for your Cisco Unified MeetingPlace system.


Related Topics

Configuring Access Phone Numbers and Notification Labelsmodule

Configuring Direct Inward Dial on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

What to Do Next

Take one of these actions:

If your route patterns direct calls to Cisco Unified Communications Manager Release 6.1 (or later), proceed to the "Verifying the Call-Control Configuration" section.

If your route patterns direct calls to a Cisco IOS gateway with Cisco Unified Border Element, proceed to the "Configuring Cisco Unified Border Element: H.323 to SIP Conversion" section.

Proceed to the "Verifying the Call-Control Configuration" section.

Configuring Cisco Unified Border Element: H.323 to SIP Conversion

Perform this task to support H.323 endpoints if your call-control network does not include Cisco Unified Communications Manager 6.1 (or later).


Note Cisco Unified Border Element was previously known as the Cisco Multiservice IP-to-IP Gateway. Cisco Unified Border Element is available only on MeetingPlace-scheduled and audio-only systems.


Before You Begin

Complete the "Configuring Cisco Unified Communications Manager Release 4.x or 5.x: Route Patterns" section.

This task is performed in the Cisco IOS command-line interface (CLI) of the router. For more information about the Cisco IOS commands used in this procedure, see one of these documents:

Cisco Unified Border Element Configuration Guide for your Cisco IOS release

Cisco IOS Commands Master List for your Cisco IOS release

Restriction

CUBE is not compatible with Cisco Unified MeetingPlace systems that include one or more video blades.

Procedure


Step 1 On the Cisco router, enter privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted.

Router# enable 
 
   

Step 2 Enter global configuration mode.

Router# configure terminal 
 
   

Step 3 Enable basic CUBE functionality on the router. This functionality terminates an incoming VoIP call and re-originates it with the use of an outbound VoIP dial peer. The calls can be H.323 to SIP or SIP to SIP.

Router(config)# voice service voip 
Router(config-voi-serv)# allow-connections h323 to sip 
Router(config-voi-serv)# allow-connections sip to h323 
Router(config-voi-serv)# allow-connections sip to sip 
Router(config-voi-serv)# allow-connections h323 to h323 
 
   

What to Do Next

Proceed to the "Configuring the Cisco IOS Gateway: Dial Peers to Cisco Unified MeetingPlace" section.

Configuring the Cisco IOS Gateway: Dial Peers to Cisco Unified MeetingPlace

Use this procedure to enable the Cisco IOS Gateway to route calls to Cisco Unified MeetingPlace by configuring dial peers. Dial peers are used to identify call source and destination endpoints and to define the characteristics applied to each call leg in the call connection.

Before You Begin

Complete the "Configuring SIP on Cisco Unified MeetingPlace" section.

Make sure that your Cisco Unified Communications Manager configuration database already includes the Cisco IOS gateway. See the Cisco Unified Communications Manager online help for information about finding or adding gateways in Cisco Unified Communications Manager.

Write down each of the phone numbers from the Cisco Unified MeetingPlace Administration Center:

Access phone numbers configured on the Usage Configuration Page

Direct Inward Dial (DID) numbers—only if you enable DID through the Route calls to meeting ID that matches DID field

You perform this task in the Cisco IOS command-line interface (CLI) of the router. For more information about the Cisco IOS commands used in this procedure, see the Cisco IOS Commands Master List for your Cisco IOS software major release.

Procedure


Step 1 On the Cisco router, enter privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted.

Router# enable 
 
   

Step 2 Enter global configuration mode.

Router# configure terminal 
 
   

Step 3 Enter dial peer voice configuration mode and define a remote voice over IP (VoIP) dial peer.

Router(config)# dial-peer voice number voip
 
   

The number is one or more digits that identify the dial peer. Valid entries are from 1 to 2147483647.

Step 4 (Optional) Provide a comment or description to help you distinguish this particular dial peer from other dial peer configurations.

Router(config-dialpeer)# description string
 
   

Step 5 Route calls to the Cisco Unified MeetingPlace Application Server.

Router(config-dialpeer)# destination-pattern digits
 
   

For digits, enter the Cisco Unified MeetingPlace phone number.

Step 6 Configure the dial peer to use SIP.

Router(config-dialpeer)# session protocol sipv2 
 
   

Step 7 Configure the IP address of the Cisco Unified MeetingPlace Application Server.

Router(config-dialpeer)# session target ipv4:ip-address
 
   

In an Application Server failover deployment, enter the shared IP address of eth0.

Step 8 Configure the router to forward dual tone multifrequency (DTMF) tones by using Real-Time Transport Protocol (RTP) with the Named Telephone Event (NTE) payload type.

Router(config-dialpeer)# dtmf-relay rtp-nte
 
   

Step 9 Configure the router to use a particular codec.

Router(config-dialpeer)# codec [g711ulaw | g711alaw | g729 | g722-64 | ilbc]
 
   

Step 10 Disable voice activity detection (VAD) for the calls using this dial peer.

Router(config-dialpeer)# no vad 
 
   

Step 11 Exit the current mode.

Router(config-dialpeer)# exit
 
   

Step 12 Repeat this procedure as necessary to route calls to each access phone number and DID number for your Cisco Unified MeetingPlace system.


Example

This example displays dial peers that were configured to direct calls to two Cisco Unified MeetingPlace access phone numbers. The Cisco Unified MeetingPlace Application Server IP address is 10.10.10.4.

!
dial-peer voice 1 voip
 description Cisco Unified MeetingPlace access phone number 1
 destination-pattern 50111
 session protocol sipv2
 session target ipv4:10.10.10.4
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 2 voip
 description Cisco Unified MeetingPlace access phone number 2
 destination-pattern 50123
 session protocol sipv2
 session target ipv4:10.10.10.4
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!

Related Topics

Configuring Access Phone Numbers and Notification Labelsmodule

Configuring Direct Inward Dial on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

Configuring Application Server Failover for Cisco Unified MeetingPlace on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

What to Do Next

Proceed to the "Verifying the Call-Control Configuration" section.

Verifying the Call-Control Configuration

Procedure


Step 1 Call one of the Cisco Unified MeetingPlace access phone numbers configured on the Usage Configuration Page of the Administration Center.

Step 2 Verify that you hear the Cisco Unified MeetingPlace voice prompts.


Troubleshooting Tips

See the Troubleshooting Phone Issues for Cisco Unified MeetingPlace module.

Related Topics

Verifying Basic Voice and Video Conferencing Using the Telephone User Interface in the Quick Start Configuration: Cisco Unified MeetingPlace Basic Voice and Video Conferencing module

Verifying Basic Voice and Video Conferencing Using the Web User Portal in the Quick Start Configuration: Cisco Unified MeetingPlace Basic Voice and Video Conferencing module

Configuring Access Phone Numbers and Notification Labelsmodule

Configuring Direct Inward Dial on MeetingPlace-Scheduled and Audio-Only Deploymentsmodule

What To Do Next

If your network includes H.323 video endpoints, proceed to the "How to Configure Call Control for Video Conferencing with H.323 Endpoints" section.

For Cisco Unified Communications Manager environments, we recommend disabling the Music on Hold (MoH) feature for Cisco Unified MeetingPlace. See "Configuring Cisco Unified Communications Manager: Music On Hold" in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.

How to Configure Call Control for Video Conferencing with H.323 Endpoints

How to Configure the Cisco IOS Gatekeeper

Configuring Cisco Unified Communications Manager Release 6.1 or Later: H.323 Endpoints

How to Configure the Cisco IOS Gatekeeper

Perform one of these tasks, depending on which device provides dial plan resolution for your network:

Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided By Gatekeeper

Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager

Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided By Gatekeeper

Choose this configuration option if the Cisco IOS gatekeeper provides dial plan resolution for all devices in your network.

Before You Begin

Configure voice call control. See the "How to Configure Call Control for Voice Conferencing" section.

Configure the gatekeeper Cisco Unified Communications Manager. For instructions on adding the gatekeeper, see the Cisco Unified Communications Manager online help.

You perform this task in the Cisco IOS command-line interface (CLI) of the router. For more information about the Cisco IOS commands used in this procedure, see the Cisco IOS Commands Master List for your Cisco IOS release.

Procedure


Step 1 On the Cisco router, enter privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted.

Router# enable 
 
   

Step 2 Enter global configuration mode.

Router# configure terminal 
 
   

Step 3 Enter gatekeeper configuration mode.

Router# gatekeeper 
 
   

Step 4 Define the zone controlled by the gatekeeper.

Router(config-gk)# zone local gk-zone-name domain-name 
 
   

Step 5 Specify which subnets the gatekeeper will accept discovery and registration messages sent by endpoints in those subnets.

Router(config-gk)# no zone subnet gk-zone-name default enable 
Router(config-gk)# zone subnet gk-zone-name subnet1-address{/bits-in-mask | mask-address} 
enable 
Router(config-gk)# zone subnet gk-zone-name subnet2-address{/bits-in-mask | mask-address} 
enable 
 
   

Step 6 Define a technology prefix, which is stripped before checking for the zone prefix. Configure calls to hop off at the gatekeeper, regardless of the zone prefix in the destination address. The default-technology option specifies to use gateways registering with this prefix option as the default for routing any addresses that are otherwise unresolved.

Router(config-gk)# gw-type-prefix type-prefix hopoff gk-zone-name default-technology 
 
   

Step 7 Disable proxy communications with local terminals for calls between local and remote zones.

Router(config-gk)# no use-proxy gk-zone-name default inbound-to terminal 
Router(config-gk)# no use-proxy gk-zone-name default outbound-from terminal 
 
   

Step 8 Enable the gatekeeper.

Router(config-gk)# no shutdown 

Example

!
gatekeeper
 zone local mp2-video example.com
 no zone subnet mp2-video default enable
 zone subnet mp2-video 10.20.120.50/32 enable
 zone subnet mp2-video 10.10.1.0/24 enable
 gw-type-prefix 2#* hopoff mp2-video default-technology
 no use-proxy mp2-video default inbound-to terminal
 no use-proxy mp2-video default outbound-from terminal
 no shutdown
!

What to Do Next

Proceed to one of these sections in the Quick Start Configuration: Cisco Unified MeetingPlace Basic Voice and Video Conferencing module:

Verifying Basic Voice and Video Conferencing Using the Telephone User Interface

Verifying Basic Voice and Video Conferencing Using the Web User Portal

Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager

Choose this configuration option if the following are true:

All H.323 video endpoints register to this Cisco IOS gatekeeper.

Cisco Unified Communications Manager provides dial plan resolution and becomes the master call-control point for all devices in your network.

Before You Begin

Configure voice call control. See the "How to Configure Call Control for Voice Conferencing" section.

Add this gatekeeper to Cisco Unified Communications Manager. For instructions on adding the gatekeeper, see the Cisco Unified Communications Manager online help.

You perform this task in the Cisco IOS command-line interface (CLI) of the router. For more information about the Cisco IOS commands used in this procedure, see the Cisco IOS Commands Master List for your Cisco IOS release.

Procedure


Step 1 On the Cisco router, enter privileged EXEC mode or any other security level set by a system administrator. Enter your password if prompted.

Router# enable 
 
   

Step 2 Enter global configuration mode.

Router# configure terminal 
 
   

Step 3 Enter gatekeeper configuration mode.

Router# gatekeeper 
 
   

Step 4 Define the zone controlled by the gatekeeper. Specify which gatekeeper interface to use for Registration, Admission, and Status (RAS) signaling. Force all intra-zone calls, in addition to calls that enter and leave the zone, to use this gatekeeper.

Router(config-gk)# zone local gk-zone-name domain-name ras-IP-address invia gk-zone-name 
outvia gk-zone-name enable-intrazone 
 
   

Step 5 Define a technology prefix that matches what you configure for the H.323 endpoints in Cisco Unified Communications Manager. The default-technology option specifies to use gateways registering with this prefix option as the default for routing any addresses that are otherwise unresolved.

Router(config-gk)# gw-type-prefix type-prefix default-technology 
 
   

Step 6 Disable proxy communications with local terminals for calls between local and remote zones.

Router(config-gk)# no use-proxy local-zone-name default inbound-to terminal 
Router(config-gk)# no use-proxy local-zone-name default outbound-from terminal 
 
   

Step 7 Enable the gatekeeper.

Router(config-gk)# no shutdown 
 
   

Example

!
gatekeeper
 zone local MP-Zone1 example.net 192.168.2.50 invia MP-Zone1 outvia MP-Zone1 
enable-intrazone
 gw-type-prefix 1#* default-technology
 no use-proxy MP-Zone1 default inbound-to terminal
 no use-proxy MP-Zone1 default outbound-from terminal
 no shutdown
!

What To Do Next

Proceed to the "Configuring Cisco Unified Communications Manager Release 6.1 or Later: H.323 Endpoints" section.

Configuring Cisco Unified Communications Manager Release 6.1 or Later: H.323 Endpoints

Perform this task only if Cisco Unified Communications Manager provides dial plan resolution for your network.

Before You Begin

Complete the "Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager" section.

You perform this task in the Cisco Unified Communications Manager Administration pages. Because the pages and menus vary by release, you should check the Cisco Unified Communications Manager Administration online help for step-by-step instructions that are specific to your release.

Procedure


Step 1 Go to http://ccm-server/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.

Step 2 Sign in to Cisco Unified Communications Manager Administration.

Step 3 Select Device > Phone.

Step 4 (Optional) To display a list of existing phone entries, select Find.

Step 5 Select Add New.

Step 6 Select H.323 Client in the Phone Type field.

Step 7 Select Next.

Step 8 Configure the fields described in Table 7.

Table 7 Fields for Adding an H.323 Endpoint in Cisco Unified Communications Manager Release 6.1 (or Later) 

Field
Action

Device Name

Enter the IP address of the H.323 endpoint.

Device Pool

Select Default or your configured device pool.

Outgoing Caller ID Pattern

Enter the extension or phone number of the H.323 endpoint.

Gatekeeper Name

Select the gatekeeper that you configured in the "Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager" section.

E.164

Enter the E.164 phone number used by the H.323 endpoint.

Technology Prefix

Enter the technology prefix (type-prefix) that you configured in Step 5 in the "Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager" section.

Zone

Enter the zone name (gk-zone-name) that you configured in Step 4 in the "Configuring the Cisco IOS Gatekeeper: Dial Plan Resolution Provided by Cisco Unified Communications Manager" section.


Step 9 Configure all other required fields appropriately for your current deployment.


Tip For field descriptions, select Help > This Page.


Step 10 Select Save.

Step 11 Select OK to any pop-up dialog box messages that you see.

Step 12 Repeat this procedure to add each H.323 endpoint to Cisco Unified Communications Manager.


What to Do Next

Proceed to one of these sections in the Quick Start Configuration: Cisco Unified MeetingPlace Basic Voice and Video Conferencing module:

Verifying Basic Voice and Video Conferencing Using the Telephone User Interface

Verifying Basic Voice and Video Conferencing Using the Web User Portal

How to Configure Secure Conferencing Mode


Note Secure Conferencing is available only on MeetingPlace-managed and audio-only systems.


Restrictions

A maximum of 249 participants are allowed in any meeting on a server configured for secure conferencing.

Before You Begin

If you use Cisco Unified Communications Manager as your SIP proxy, you must configure it to operate in mixed mode before proceeding with configuration tasks.

To configure secure conferencing, you must have certificates on your Cisco Unified MeetingPlace system and all SIP proxies. Use the Secure Conferencing Certificate Management features to obtain your certificates.

You can configure secure conferencing modes by using the drop-down menus on the System Configuration > Call Configuration > SIP Configuration page:

Enable Secure Conferencing

Permit non-secure endpoints to connect to MeetingPlace

You can configure three different secure conferencing modes:

Non-secure—Non-secure mode is the default configuration in which no calls are secure. The RTP stream is not encrypted and signaling might be encrypted depending on how you configure port and transport. You can have secured signaling with this configuration (for example, TLS instead of TCP or UDP) with non-secure media.

Best effort—Best effort mode enables Cisco Unified MeetingPlace to attempt to establish secure connections with end points, but if this is not possible, non-secure connections are allowed. Best effort mode is intended as a short-term setting that you use during the transition from non-secure mode to secure only mode.

Secure only—Secure only mode ensures that non-secure end points will not be allowed to join a conference. When you have secure only mode configured, non-secure users will hear a fast busy signal if they attempt to dial into a conference.

You must configure certificates and secure SIP trunks on all SIP proxies if you want to use best effort or secure only modes. For non-secure mode these steps are not necessary because default non-secure SIP trunks are used.


Note All secure conferencing modes apply only to audio streams. Video streams will be unsecure in all secure modes.


Perform the procedures in the following sections in order to configure secure conferencing on your system:

Creating a Secure SIP Trunk Profile

Creating a Secure SIP Trunk

Defining a Route Pattern for a Secure SIP Trunk

Configuring Certificates on a Cisco Unified MeetingPlace System

Configuring Certificates on SIP Proxies

Enabling Secure Conferencing on your Cisco Unified MeetingPlace System

Related Topics

Generating a Certificate Signing Request and Obtaining the Certificate in the Configuring SSL for the Cisco Unified MeetingPlace Application Server module

Changing the Type of Media Server that Your System Uses in the Migrating from Cisco Unified MeetingPlace Release 7.x or 8.0 module

Synchronizing Your Meeting Types

Before you switch from non-secure mode to best effort or secure mode, you must first synchronize your meeting types.

Procedure


Step 1 Log in to the Cisco Unified MeetingPlace Administration Center.

Step 2 Select the Media Server Administration link in upper right corner of the Administration Center page.

Step 3 Log in to the Media Server Administration page. Your Media Server Administration login credentials should be the same as your Administration Center login credentials.

Step 4 Select Resource Management > Meeting Types.

Step 5 Select Synchronize in the lower right corner.

Step 6 Log out of the Media Server Administration page and return to the Administration Center.


What to Do Next

Synchronize your meeting types with your Audio Blades. See Synchronizing Audio Blade Meeting Types with the Template Audio Blade in the Changing Values for the Hardware Media Server module.

Create SIP trunk profiles on your SIP proxies as described in the "Creating a Secure SIP Trunk Profile" section.

Creating a Secure SIP Trunk Profile

Create a secure SIP trunk profile on all SIP proxies.

Before You Begin

See Configuring a SIP Trunk Security Profile in Cisco Unified Communications Manager for Cisco Unified MeetingPlace in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.

Procedure


Step 1 Select Security Profile > SIP Trunk Security Profile from the System menu in Cisco Unified Communications Manager Administration.

Step 2 Select Add New on the Find and List SIP Trunk Security Profiles page.

Step 3 On the SIP Trunk Security Profile Configuration page, enter the following settings under SIP Trunk Security:

Field
Setting

Name

Enter a name for your profile (for example, MeetingPlace SIP Trunk Security Profile).

Description

Enter a description for your profile (for example, SIP trunk security profile for Cisco MeetingPlace).

Device Security Mode

Encrypted

Note the following requirements for the Cisco Unified Communications server:

Make sure you have a TFTP server configured.

Configure the Cisco Unified Communications Manager server for security by using the Cisco CTL client. For more information, see "Configuring the Cisco CTL Client" in your release of the Cisco Unified Communications Security Guide at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html.

Incoming Transport Type

TLS

Outgoing Transport Type

TLS

Enable Digest Authentication

Leave unchecked.

X.509 Subject Name

Enter your X.509 subject name (for example, Connection). Make sure the name you enter matches the Subject Name field for the SIP application certificate on the Cisco Unified MeetingPlace server.

Incoming Port

5061


Step 4 Configure all other required fields appropriately for your current deployment.


Tip Select Help > This Page for field descriptions.


Step 5 Select Save.


Related Topics

Configuring a SIP Trunk Security Profile in Cisco Unified Communications Manager for Cisco Unified MeetingPlace in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module

Security Guide for your release of Cisco Unified Communications Manager at http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html

What To Do Next

Proceed to the Creating a Secure SIP Trunk.

Creating a Secure SIP Trunk

Before You Begin

Refer to the following URL for information on Cisco Unified Communications Manager 8.5 (CUCM) SIP trunk configuration:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b06trunk.html

Procedure


Step 1 On the Cisco Unified Communications Manager Administration page, select Device > Trunk.

Step 2 On the Find and List Trunks page, select Add New.

Step 3 On the Trunk Configuration page, select SIP Trunk in the Trunk Type field.

Step 4 In the Device Protocol field, select SIP and then select Next.

Step 5 Under Device Information, enter the following settings:

Field
Setting

Device Name

Enter your device name (for example, Connection_SIP_Trunk).

Description

Enter a description (for example, SIP trunk for Cisco Unified MeetingPlace).

SIP Trunk Security Profile

Select the name of the SIP trunk security profile that you created in the Creating a Secure SIP Trunk Profile procedure.

SRTP Allowed

Check the checkbox.

PSTN Access

Uncheck the checkbox.

Destination Port

5061


Step 6 Set the remaining fields in the SIP trunk configuration according to the "Configuring Cisco Unified Communications Manager Release 6.x or Later: SIP Trunk to Cisco Unified MeetingPlace" section.

Step 7 Select Save.


Related Topics

How to Configure Call Control for Voice Conferencing

What To Do Next

Proceed to the "Defining a Route Pattern for a Secure SIP Trunk" section.

Defining a Route Pattern for a Secure SIP Trunk

Before You Begin

Refer to the Cisco Unified Communications Manager Administration Guide for Cisco Unified Communications Manager for your release for more information on route pattern configuration.

Procedure


Step 1 On the Cisco Unified Communications Manager Administration page, select Call Routing > Route/Hunt > Route Pattern.

Step 2 On the File and List Route Patterns page, select Add New.

Step 3 Enter the following settings on the Route Pattern Configuration page:

Field
Setting

Route Pattern

Enter the Cisco Unified MeetingPlace phone number.

Requirements:

This number must not conflict with any route pattern defined in this Cisco Unified Communications Manager cluster.

Do not enter any spaces in this field.

Gateway/Route List

Select the device name of the secure SIP trunk to the Cisco Unified MeetingPlace created above.

Call Classification

Select OnNet.

Provide Outside Dial Tone

Uncheck the check box.


Step 4 Configure all other required fields appropriately for your current deployment.


Tip For field descriptions, select Help > This Page.


Step 5 Select Save.

Step 6 Select OK on any pop-up dialog box messages that appear.


What To Do Next

Proceed to the "Configuring Certificates on a Cisco Unified MeetingPlace System" section.

Configuring Certificates on a Cisco Unified MeetingPlace System

To configure secure conferencing, you must ensure that your certificates are correctly configured on both Cisco Unified Communication Manager and Cisco Unified MeetingPlace.

You can configure certificates for your Cisco Unified MeetingPlace system either by configuring a self-signed certificate or by having a certificate generated by a third party certificate authority (CA).

Configuring a Self-Signed Cisco Unified MeetingPlace Certificate

Configuring a Third-Party Generated Cisco Unified MeetingPlace Certificate

Configuring a Self-Signed Cisco Unified MeetingPlace Certificate

Procedure


Step 1 Log in to the Administration Center.

Step 2 Select Certificate Management > Secure Conferencing Certificate Management > Generate MeetingPlace Certificates.

Step 3 Select MeetingPlace Self-Signed certificates from the Type of certificate drop-down menu.

Step 4 Select Generate New Application Cert.

Step 5 A pop-up window appears. Set the following values:

Field
Setting

Subject Name

Enter the name you used for the X.509 Subject Name on the secure SIP profile page of the Cisco Unified Communications Manager.


Step 6 Configure all other required fields in the pop-up window.

Step 7 Select Generate New Application Cert.


Configuring a Third-Party Generated Cisco Unified MeetingPlace Certificate

Procedure


Step 1 Log in to the Administration Center.

Step 2 Select Certificate Management > Secure Conferencing Certificate Management > Generate MeetingPlace Certificates.

Step 3 Select MeetingPlace 3rd party certificates from the Type of certificate drop-down menu.

Step 4 Select Generate CSRs.

Step 5 A pop-up window appears. Set the following values:

Field
Setting

Subject Name

Enter the name the third-party CA generated for the X.509 Subject Name on the secure SIP profile page of Cisco Unified Communications Manager.


Step 6 Configure all other required fields in the pop-up window.

Step 7 Select Generate Certificate Signing Request (CSR).

Step 8 Select Download CSR.

Step 9 Select Save to save the CSR file.


Caution After you select Download CSR, do not select Generate CSRs again or you might generate an invalid certificate.


Related Topics

Generating a Certificate Signing Request and Obtaining the Certificate in the Configuring SSL for the Cisco Unified MeetingPlace Application Server module

Configuring Certificates on SIP Proxies

You can configure certificates for your SIP proxies either by configuring a self-signed certificate or by having a certificate generated by a third-party certificate authority (CA).

There are different ways to obtain SIP proxy certificates and to save them on the Cisco Unified MeetingPlace system depending on what kind of SIP proxy you have and whether the SIP proxy uses self-signed or third-party certificates.

Uploading Root Certificates of the Cisco Unified MeetingPlace Certificate to the Cisco Unified Communications Manager

Uploading Cisco Unified MeetingPlace or CA Root Certificates to the Cisco Unified Communications Manager.

Configuring SIP Proxy Certificates on a Cisco Unified MeetingPlace System

Sending Certificates to the Media Server

Uploading Root Certificates of the Cisco Unified MeetingPlace Certificate to the Cisco Unified Communications Manager

If Cisco Unified MeetingPlace is using self-signed certificates, download a root certificate.

Before You Begin

You must configure your self-signed certificate as shown in the "Configuring Certificates on a Cisco Unified MeetingPlace System" section.

Procedure


Step 1 Log in to the Administration Center.

Step 2 Select Certificate Management > Secure Conferencing Certificate Management > Generate MeetingPlace Certificates.

Step 3 Select MeetingPlace Self-Signed certificates in the Type of certificate drop-down menu.

Step 4 Select Download Root Certificate and choose to save your file with the Cisco Unified MeetingPlace root certificate on the local machine.


Uploading Cisco Unified MeetingPlace or CA Root Certificates to the Cisco Unified Communications Manager.

If Cisco Unified MeetingPlace is using third-party generated certificates then the root certificate should be obtained by the CA that generated the Cisco Unified MeetingPlace certificate. In addition to the root self-signed certificate from the CA, you should receive one or more intermediate certificates that are generated in the certificate chain. All of these certificates are used by the Cisco Unified Communications Manager to verify the Cisco Unified MeetingPlace application certificate.

Procedure


Step 1 On the Cisco Unified Communications Manager server, log on to Cisco Unified OS Administration.

Step 2 Select Certificate Management from the Security menu.

Step 3 Select an Upload Certificate on the Certificate List page.

Step 4 On the Upload Certificate pop-up window select the following options:

Field
Value

Certificate Name

Select CallManager-trust.

Description

Description of the certificate you are uploading.

Upload File

Path to the trusted certificate.

A trusted certificate is one of the following:

A Cisco Unified MeetingPlace self-signed root certificate

A CA self-signed root certificate and all intermediate certificates used in the chain to sign the Cisco Unified MeetingPlace application certificate.


Step 5 Select Upload File.

Step 6 Repeat this procedure for all certificates in the chain if there are more than one.


Note Cisco Unified MeetingPlace does not support sending a certificate chain in the TLS handshake (certificate chains can be used if you are using third-party certificates). Only the application certificate is sent. Therefore, all certificates in the chain ending with the CA root but excluding the application certificate must be uploaded as CallManager-trust using the procedure described above.


This procedure must be performed on all Cisco Unified Communications Manager systems that are configured on the SIP proxy page of the Cisco Unified MeetingPlace system.


Configuring SIP Proxy Certificates on a Cisco Unified MeetingPlace System

There are two ways to configure SIP proxy certificates on a Cisco Unified MeetingPlace system:

Manually Uploading a SIP Proxy Certificate

Automatically Downloading a SIP Proxy Certificate

Manually Uploading a SIP Proxy Certificate

Procedure


Step 1 Log in to the Administration Center.

Step 2 Select Certificate Management > Secure Conferencing Certificate Management > Add SIP proxy Certificates.

Step 3 Select Upload certificate authority (CA) root certificate from the Trust Certificate Type drop-down menu.

Step 4 In the For SIP Proxy drop-down menu, select the SIP proxy for which you want to upload a certificate. If all SIP proxies use the same CA root certificate, then you can select All entries.

Step 5 In the Certificate file field, select Browse, select the root certificate in the File Upload dialog box, and select Open.

Step 6 Select Upload Certificate.

The Cisco Unified MeetingPlace server does not support uploading certificate chains. If a SIP proxy has intermediate certificates between the CA root (self-signed) certificate and the application (SIP proxy) certificate, only the CA root (self-signed) certificate can be uploaded to Cisco Unified MeetingPlace using the procedure described above. The SIP proxy used in the TLS handshake must send all certificates in the chain in order to be correctly verified by Cisco Unified MeetingPlace.

If you add a new SIP proxy to the SIP proxy list, you must repeat this procedure to upload a certificate for the new SIP proxy to the Cisco Unified MeetingPlace. You must also send a certificate to the media server as explained in the "Sending Certificates to the Media Server" section.


Automatically Downloading a SIP Proxy Certificate


Note This feature is supported by Cisco Unified Communications Manager if it is using self-signed certificates. Self-signed certificates are copied to dedicated TFTP servers and Cisco Unified MeetingPlace downloads the certificates from the TFTP servers.


Before You Begin

Only use this procedure if Cisco Unified Communications Manager is using self-signed certificates. Only self signed certificates can be downloaded through a TFTP server. If Cisco Unified Communications Manager is using third-party certificates, a CA root certificate cannot be downloaded using TFTP servers. You must manually upload CA root certificates as described in this section.

In the SIP proxy list, specify any SIP proxies for which you want to download a certificate. Cisco Unified MeetingPlace does not save all certificates it finds on TFTP servers. It saves certificates only for SIP proxies that are specified in the SIP proxy list on the SIP Configuration page.

Procedure


Step 1 Log in to the Administration Center.

Step 2 Select Certificate Management > Secure Conferencing Certificate Management > Add SIP proxy Certificates.

Step 3 Select Download SIP proxy certificates from the Trust Certificate Type drop-down menu. .

Step 4 Select Add New TFTP Server.

Step 5 In the pop-up window, configure the following fields:

Field
Value

TFTP hostname

Host name or IP address of the machine on which the TFTP server is located.

Note If a Cisco Unified Communications Manager cluster is using a centralized TFTP server (where certificates for all Communications Managers in the cluster are located) then only that TFTP server address must be specified. Otherwise, if each Cisco Unified Communications Manager (from the SIP proxy list) is using its own TFTP server, then all the TFTP servers must be specified.

Port

Port on which the TFTP server is operating. The default value is 69.

Priority

Priority among TFTP servers that Cisco Unified MeetingPlace will honor when searching for SIP proxy certificates.


Step 6 Select Save.

Step 7 Repeat this procedure for all TFTP servers from which Cisco Unified MeetingPlace will download SIP proxy certificates.

Step 8 Select Download Certificates From TFTP servers.



Note Cisco Unified MeetingPlace will not save all certificates on TFTP servers. It will only save certificates for SIP proxies specified in the SIP proxy list.


What to Do Next

If you add a new SIP proxy to the SIP proxy list, you must repeat this procedure to upload a certificate for the new SIP proxy to the Cisco Unified MeetingPlace.

You must send a certificate to the media server as explained in the "Sending Certificates to the Media Server" section.

Related Topics

Configuring SIP on Cisco Unified MeetingPlace

Sending Certificates to the Media Server

After certificates are properly configured they need to be sent to the Media Server.

Procedure


Step 1 Log in to the Administration Center.

Step 2 Select Certificate Management > Secure Conferencing Certificate Management > View and Send Certificates.

Step 3 Verify that certificates for all SIP proxy servers and Cisco Unified MeetingPlace application certificates are in place and select Send Certificates to Media Server.

A dialog box appears indicating that you must restart your system for the changes to take place.

Step 4 Select OK to restart your system or Cancel if you do not want to restart your system.


Note You cannot proceed with the secure conferencing procedure until you have restarted your system and you relog in to the Administration Center.



Enabling Secure Conferencing on your Cisco Unified MeetingPlace System

Configure your secure conferencing setting based on the secure conferencing mode you want.


Note Whenever you change Enable Secure Conferencing from No to Yes on Express Media Server deployments, make sure to send your certificates to the media server as described in the "Sending Certificates to the Media Server" section.


Procedure


Step 1 Log in to the Administration Center.

Step 2 Select System Configuration > Call Configuration > SIP Configuration.

Step 3 Configure your secure conferencing mode:

To configure non-secure mode:

Field
Setting

Enable Secure Conferencing

No

Permit non-secure endpoints to connect to MeetingPlace

N/A


To configure best effort secure conferencing mode:

Field
Setting

Enable Secure Conferencing

Yes

Permit non-secure endpoints to connect to MeetingPlace

Yes


To configure secure-only secure conferencing mode:

Field
Setting

Enable Secure Conferencing

Yes

Permit non-secure endpoints to connect to MeetingPlace

No


Step 4 Select Save.


Note When you enable secure conferencing, the port is automatically set to 5061 and transport to TLS. If you disable secure conferencing, port and transport will not revert to their original values.