Table Of Contents
Configuring Call Control for Cisco Unified MeetingPlace
Prerequisites for Configuring Call Control
How to Configure Call Control for Voice Conferencing
Configuring SIP on Cisco Unified MeetingPlace
Configuring Cisco Unified Communications Manager 6.x or a Later Release: SIP Trunk to Cisco Unified MeetingPlace
Configuring Cisco Unified Communications Manager 6.x or a Later Release: Route Patterns
Configuring SIP Trunks Between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a Later Release
Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager 4.x or 5.x and Cisco Unified Communications Manager 6.1 or a Later Release
Configuring Cisco Unified Communications Manager 4.x or 5.x: Route Patterns
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 6.1 or a Later Release: H.323 Endpoints
Configuring Call Control for Cisco Unified MeetingPlace
Release 7.1
Revised: April 3, 2011 8:30 pm
•
Prerequisites for Configuring Call Control
•How to Configure Call Control for Voice Conferencing
•How to Configure Call Control for Video Conferencing with H.323 Endpoints
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 and Compatibility Matrix 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.
•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 Installation, Upgrade, and Migration Guide for Cisco Unified MeetingPlace at http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_installation_guides_list.html.
Related Topics
•Configuring Access Phone Numbers and Notification Labels module
•Configuring Direct Inward Dial for Cisco Unified MeetingPlace module
How to Configure Call Control for Voice Conferencing
•Configuring SIP on Cisco Unified MeetingPlace
•Configuring Cisco Unified Communications Manager 6.x or a Later Release: SIP Trunk to Cisco Unified MeetingPlace
•Configuring Cisco Unified Communications Manager 6.x or a Later Release: Route Patterns
•Configuring SIP Trunks Between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a Later Release
•Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager 4.x or 5.x and Cisco Unified Communications Manager 6.1 or a Later Release
•Configuring Cisco Unified Communications Manager 4.x or 5.x: Route Patterns
•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 Log 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 Page in the Administration Center Page References for Cisco Unified MeetingPlace module
What to Do Next
Proceed to the "Configuring Cisco Unified Communications Manager 6.x or a Later Release: SIP Trunk to Cisco Unified MeetingPlace" section.
Configuring Cisco Unified Communications Manager 6.x or a Later Release: 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 telephone 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, then 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 Log in with your Cisco Unified Communications Manager administrator username and password.
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 6.x or a Later Release
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, then 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.
Note See "Configuring Cisco Unified Communications Manager: Music On Hold" in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.
|
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, then select the default Non Secure SIP Trunk Profile.
|
Rerouting Calling Search Space
|
Make sure you set this value appropriately to ensure that call transfers (out to attendant or other systems) are successful. Consult your Communications Manager administrator for the appropriate 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, then 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 module
•Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module
What to Do Next
Proceed to the "Configuring Cisco Unified Communications Manager 6.x or a Later Release: Route Patterns" section.
Configuring Cisco Unified Communications Manager 6.x or a Later Release: 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 6.x or a Later Release: SIP Trunk to Cisco Unified MeetingPlace" section.
•Write down each of the following 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/, where ccm-server is the fully-qualified domain name or IP address of the Cisco Unified Communications Manager server.
Step 2 Log in with your Cisco Unified Communications Manager administrator username and password.
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 6.x or a Later Release
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 Page in the Administration Center Page References for Cisco Unified MeetingPlace module
•Configuring Access Phone Numbers and Notification Labels module
•Configuring Direct Inward Dial for Cisco Unified MeetingPlace module
What to Do Next
If you are using Cisco Unified Communications Manager 6.1 or a later release to provide front-end signaling for Cisco Unified MeetingPlace, then proceed to one of the following sections:
•Configuring SIP Trunks Between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a Later Release
•Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager 4.x or 5.x and Cisco Unified Communications Manager 6.1 or a Later Release
Otherwise, proceed to the "Verifying the Call-Control Configuration" section.
Configuring SIP Trunks Between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a Later Release
Perform this task if you have already deployed Cisco Unified Communications Manager 5.x and are using Cisco Unified Communications Manager 6.1 or a later release to provide front-end signaling for Cisco Unified MeetingPlace.
Before You Begin
•You may instead choose to configure inter-cluster trunks (instead of SIP trunks) between Cisco Unified Communications Manager 5.x and Cisco Unified Communications Manager 6.1 or a later release. If this is the case, then do not perform this task. Instead, see the "Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager 4.x or 5.x and Cisco Unified Communications Manager 6.1 or a Later Release" section.
•Complete the "Configuring Cisco Unified Communications Manager 6.x or a Later Release: Route Patterns" section.
•Perform this task on both of the following servers:
–Cisco Unified Communications Manager 5.x
–Cisco Unified Communications Manager 6.1 or a later release
•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 telephone 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 Log in with your Cisco Unified Communications Manager administrator username and password.
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 5.x, then enter the name or IP address of the Cisco Unified Communications Manager 6.1 (or a later release) server that provides front-end signaling for Cisco Unified MeetingPlace.
•If you are configuring Cisco Unified Communications Manager Release 6.1 (or a later release), then enter the name or IP address of the Cisco Unified Communications Manager 5.x server.
|
Device Pool
|
If no device pools are defined, select Default.
If device pools are already defined, then 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, then 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.
Note See "Configuring Cisco Unified Communications Manager: Music On Hold" in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.
|
Media Termination Point Required
|
Uncheck this check box.
|
Destination Address
|
Enter the destination IP address, specifically:
•If you are configuring Cisco Unified Communications Manager 5.x, then enter the IP address of the Cisco Unified Communications Manager 6.1 (or a later release) server that provides front-end signaling for Cisco Unified MeetingPlace.
•If you are configuring Cisco Unified Communications Manager 6.1 or a later release, then enter the name or IP address of the Cisco Unified Communications Manager 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, then 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 following servers are configured with SIP trunks that point to each other:
•Cisco Unified Communications Manager 5.x
•Cisco Unified Communications Manager 6.1 or a later release
Related Topics
•Configuring Operator Assistance in the Configuring Attendant Settings for Cisco Unified MeetingPlace module
What to Do Next
Proceed to the "Configuring Cisco Unified Communications Manager 4.x or 5.x: Route Patterns" section.
Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager 4.x or 5.x and Cisco Unified Communications Manager 6.1 or a Later Release
Perform this task if you already deployed Cisco Unified Communications Manager 4.x or 5.x and are using Cisco Unified Communications Manager 6.1 or a later release to provide front-end signaling for Cisco Unified MeetingPlace.
Before You Begin
•You may instead choose to configure SIP trunks (instead of inter-cluster trunks) between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a later release. If this is the case, then do not perform this task. Instead, see the "Configuring SIP Trunks Between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a Later Release" section.
•Complete the "Configuring Cisco Unified Communications Manager 6.x or a Later Release: Route Patterns" section.
•Perform this task on both of the following servers:
–Cisco Unified Communications Manager 4.x or 5.x
–Cisco Unified Communications Manager 6.1 or a later release
•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 telephone 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 Log in with your Cisco Unified Communications Manager administrator username and password.
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 4.x or 5.x, then enter the name or IP address of the Cisco Unified Communications Manager 6.1 (or a later release) server that provides front-end signaling for Cisco Unified MeetingPlace.
•If you are configuring Cisco Unified Communications Manager Release 6.1 (or a later release), then enter the name or IP address of the Cisco Unified Communications Manager 4.x or 5.x server.
|
Device Pool
|
If no device pools are defined, select Default.
If device pools are already defined, then 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, then 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.
Note See "Configuring Cisco Unified Communications Manager: Music On Hold" in the Integrating Cisco Unified MeetingPlace with Cisco Unified Communications Manager module.
|
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 4.x or 5.x, then specify the Cisco Unified Communications Manager 6.1 (or a later release) server that provides front-end signaling for Cisco Unified MeetingPlace.
•If you are configuring Cisco Unified Communications Manager Release 6.1 (or a later release), then specify the Cisco Unified Communications Manager 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, then 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 following servers are configured with SIP trunks that point to each other:
•Cisco Unified Communications Manager 4.x or 5.x
•Cisco Unified Communications Manager 6.1 or a later release
What to Do Next
Proceed to the "Configuring Cisco Unified Communications Manager 4.x or 5.x: Route Patterns" section.
Configuring Cisco Unified Communications Manager 4.x or 5.x: Route Patterns
Use this procedure to configure route patterns to enable Cisco Unified Communications Manager 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 a later release) server that provides front-end signaling for Cisco Unified MeetingPlace
•(Cisco Unified Communications Manager 5.x only) SIP trunk to the Cisco Unified Communications Manager Release 6.1 (or a later release) server that provides front-end signaling for Cisco Unified MeetingPlace
Before You Begin
•Complete the "Configuring SIP on Cisco Unified MeetingPlace" section.
•Complete one of the following items, depending on your deployment:
–Configuring SIP Trunks Between Cisco Unified Communications Manager 5.x and Cisco Unified MeetingPlace 6.1 or a Later Release
–Configuring Inter-Cluster Trunks Between Cisco Unified Communications Manager 4.x or 5.x and Cisco Unified Communications Manager 6.1 or a Later Release
•Write down each of the following 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 Log in with your Cisco Unified Communications Manager administrator username and password.
Step 3 Select Call Routing > Route/Hunt > Route Pattern.
Step 4 Select Add New.
Step 5 Configure the fields described in Table 5.
Table 5 Fields for Adding a Route Pattern in Cisco Unified Communications Manager 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 a later release) server that provides front-end signaling for Cisco Unified MeetingPlace.
•(Cisco Unified Communications Manager 5.x only) SIP trunk to the Cisco Unified Communications Manager Release 6.1 (or a later release) server that provides front-end signaling for Cisco Unified MeetingPlace.
|
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 Labels module
•Configuring Direct Inward Dial for Cisco Unified MeetingPlace module
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 Telephone 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 End-User Web Interface in the Quick Start Configuration: Cisco Unified MeetingPlace Basic Voice and Video Conferencing module
•Configuring Access Phone Numbers and Notification Labels module
•Configuring Direct Inward Dial for Cisco Unified MeetingPlace module
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 6.1 or a Later Release: H.323 Endpoints
How to Configure the Cisco IOS Gatekeeper
Perform one of the following 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.
Step 2 Enter global configuration mode.
Router# configure terminal
Step 3 Enter gatekeeper configuration mode.
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
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
What to Do Next
Proceed to one of the following 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 End-User Web Interface
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.
Step 2 Enter global configuration mode.
Router# configure terminal
Step 3 Enter gatekeeper configuration mode.
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
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
What To Do Next
Proceed to the "Configuring Cisco Unified Communications Manager 6.1 or a Later Release: H.323 Endpoints" section.
Configuring Cisco Unified Communications Manager 6.1 or a Later Release: 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 Log in with your Cisco Unified Communications Manager administrator username and password.
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 6.
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 the following 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 End-User Web Interface
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