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
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
The information in this document is based on these software and
Cisco Unified Communications Manager: 4.1(3)SR3b
CallManager Express: Cisco IOS®
12.4(9)T2, CME Version
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
Technical Tips Conventions for more information on document
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
In order to create a new region, go to System >
Region from the Cisco Unified Communication Manager Administration
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
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
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
In order to save the location information in the database, click
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
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
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
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
ip unnumbered Loopback1
service-module ip address 10.252.107.6 255.255.255.252
service-module ip default-gateway 10.252.107.5
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
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
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
In order to create a new route group for the new H.323 gateway,
choose Route Plan > Route/Hunt > Route
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
In order to create a new route list for the new dial pattern,
choose Route Plan > Route/Hunt > Route
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
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
Click Add a New Route
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
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.
Output Interpreter Tool
(registered customers only)
(OIT) supports certain
show commands. Use the OIT to view an analysis of
show command output.
Check the CallManager for Call Admission Control (CAC) through the
locations parameter. Verify that Call Admission Control monitors the bandwidth
Go to Start > Programs > Administrative Tool >
Performance > Cisco CallManager >
There is currently no specific troubleshooting information available
for this configuration.