Document ID: 99792
This document provides a sample configuration of the deployment of Cisco Unified CallManager Express (Cisco Unified CME) for branch offices in conjunction with a Cisco Unified CallManager deployed at a central office site. In this situation, the central Cisco Unified CallManager site can communicate with the remote CME with a H.323 gateway. In H.323 networks, Cisco Unified CME provides supplementary service interworking (H.450) with Voice over IP (VoIP) hairpin call routing when needed for intersite call transfer and forwarding.
Note: Direct MGCP integration between Cisco Unified CME IP phones and Cisco Unified CallManager is not supported.
Ensure that you meet these requirements before you attempt this configuration:
Knowledge of Cisco Unified Communications Manager (CallManager)
Basic knowledge of Cisco Unified CME
The information in this document is based on these software and hardware versions:
Cisco Unified Communications Manager: 4.1(3)SR3b
CallManager Express: Cisco IOS® 12.4(9)T2, CME Version 4.0(0)
The information in this document was created from the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.
Refer to the Cisco Technical Tips Conventions for more information on document conventions.
Both Cisco Unified CallManager and Cisco Unified CME support H.323, which you can use to create Cisco Unified CallManager-to-Cisco Unified CME links. Cisco Unified CME also supports SIP for VoIP interconnect. SIP has also been introduced as a WAN trunking interface on Cisco Unified CallManager. This document focuses only on the H.323 interconnect option. The information contained in this document applies to the Cisco Unified CME 3.1 and 3.2 releases and the Cisco Unified CallManager 3.3(3) and 4.0. Newer versions can have different behaviors and options than those described here.
When you create a new CME site, it can require a new region (for Codec selection), a new location (for bandwidth control), and a new device pool. Some sites can also create local media resources. In this section, you are presented with the information to configure the features described in this document.
In order to create a new region, go to System > Region from the Cisco Unified Communication Manager Administration page.
In the Region Name field, enter the name that you want to assign to the new region. Choose a value from the drop-down list box for the default codec to use between this region and other regions. Click Insert.
In the Audio Codec column, use the drop-down list boxes to choose the audio codec to use for calls within the new region and between the new region and existent regions. The audio codec determines the type of compression and the maximum amount of bandwidth that is allocated for these calls.
This section describes how to add a new location to the Cisco CallManager database. Use locations to implement call admission control in a centralized call-processing system. Call admission control enables you to regulate audio quality and video availability because it limits the amount of bandwidth that is available for audio and video calls over links between the locations.
Perform the procedure below to add a new location.
Choose System > Location.
In order to add a location, use one of these methods:
If a location already exists with settings that are similar to the one that you want to add, choose the existent location to display its settings. Click Copy, and modify the settings as needed.
In order to add a location without the need to copy an existent one, continue with Step 3.
In the upper, right corner of the window, click the Add a New Location link. Enter the appropriate settings.
In order to save the location information in the database, click Insert.
Note: When calls cannot use the link for a location, it is possible that bandwidth leakage has occurred that can reduce the allotted bandwidth for the location. You can resynchronize the location bandwidth to the maximum amount that is assigned to this location without the need to reset the Cisco CallManager server. Find the location and click ReSync Bandwidth to resynchronize the bandwidth for the chosen location.
Use the Device Pool Settings to define sets of common characteristics for devices such as the Date/Time Group, Region, SRST Reference, Media Resource Group List, etc.
Follow this procedure to add a new device pool.
Choose System > Device Pool.
Use one of these methods to add a device pool:
If a device pool already exists with settings that are similar to the one that you want to add, choose the existent device pool to display its settings; click Copy, and modify the settings, as needed.
In order to add a device pool without copying an existent one, continue with Step 3.
In the upper right corner of the window, click the Add a New Device Pool link.
Enter or edit the appropriate fields and click Insert to save the device pool information in the database.
Note: If the local IPT gateway provides DSP (Transcoding or Conferencing) services to local devices, they must also be configured with Media Resources , MRG, and MRGL.
Before you add the gateway, you need to check the interface IP address used by the CME router. Issue these commands in the CME Router to validate the IP address in use by the IOS Telephony-Service.
CMErouter#sh telephony-service | inc ^ip ip source-address 10.252.107.5 port 2000
This gateway uses 10.252.107.5 as the IP address.
Inspect which interfaces use the above IP address, as well as the status of the interfaces.
CMERouter#sh ip int brief | inc 10.252.107.5 Service-Engine0/0 10.252.107.5 YES TFTP up up Loopback1 10.252.107.5 YES TFTP up up
Note: The Service-Engine 0/0 slot in use by Cisco Unity Express runs in the Unnumbered mode.
In order to learn more information about the interface service-engine 0/0, use this command.
CMERouter#show runnning intferace service-engine0/0 ! interface Service-Engine0/0 ip unnumbered Loopback1 service-module ip address 10.252.107.6 255.255.255.252 service-module ip default-gateway 10.252.107.5 end
Follow this procedure to create a H.323 gateway.
In order to create a H.323 gateway from the CallManager Administration page, choose Device> GatewayClick Add a New Gateway.
Choose H.323 Gateway and click Next.
Enter a unique name for the Cisco CallManager to use to identify the device. Use either the IP address or the host name as the device name. The new Gateway needs to use distinct site settings, such as Device Pool or Location.
Note: After all configuration settings are validated, the H.323 gateway should be Updated and Reset.
Follow this procedure to create a new route group for the new H.323 gateway.
In order to create a new route group for the new H.323 gateway, choose Route Plan > Route/Hunt > Route Group.
Assign a new name for the Route Group and add the H.323 gateway to the route group.
The order in which to add the call routing is this:
Follow this procedure to create a new route list for the new dial pattern.
In order to create a new route list for the new dial pattern, choose Route Plan > Route/Hunt > Route List.
Click Add a New Route List.
Use concise and descriptive names for your route lists. The CompanynameLocationCalltype format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a route list.
Note: Two route groups are associated with this route list: one for OnNet calls from the H.323 gateway to the CME router and another for OffNet calls to the CME router through PSTN. OffNet calls need to translate the called number to use the PSTN circuits.
The route list details that are associated with the Failover route group look like this with the calling-party and called-party transformations.
Follow this procedure to add a new route pattern.
In order to add a new route pattern, choose Route Plan > Route/Hunt > Route Pattern from the CallManager Administration page.
Click Add a New Route Pattern.
Note: Make sure the route pattern is in an appropriate partition and any needed Calling Search Spaces (CSS). In this example, we put the route pattern in the same partition as the phones so that no additional CSS configuration is required to make this pattern reachable.
This section of the document explains how verify the details of active calls and dial-peers.
Use this section to confirm that your configuration works properly.
Verify the dial-peer configured on the CME.
shanghailab1#sh dial-peer voice summary | inc 5678 AD PRE PASS OUT TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT 5000 voip up up .. 1 syst ipv4:172.21.21.21 5001 pots up up .. 2 up 0/2/0 5003 pots up up .. 4 up 0/2/2 5004 pots up up .. 5 up 0/2/3 5002 pots up up .. 3 up 0/2/1
Note: Make sure that the VoIP dial-peer session target points to the CallManager IP address.
Check the CallManager for Call Admission Control (CAC) through the locations parameter. Verify that Call Admission Control monitors the bandwidth in use.
Go to Start > Programs > Administrative Tool > Performance > Cisco CallManager > Location.
There is currently no specific troubleshooting information available for this configuration.
- How to apply CAC based on CPU & Memory Utilization to prevent gateway overload
- Voice Technology Support
- Voice and Unified Communications Product Support
- Troubleshooting Cisco IP Telephony
- Technical Support & Documentation - Cisco Systems
|Updated: Dec 10, 2007||Document ID: 99792|