CIPT1 5.0 Lab Guide

CIPT1
Implementing Cisco
Unified CallManager
Part 1
Version 5.0
Lab Guide
Text Part Number: 97-2433-01
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CIPT1
Lab Guide
Overview
This guide presents the instructions and other information concerning the lab activities for this
course. You can find the solutions in the lab activity Answer Key.
Outline
This guide includes these activities:
Lab 2-1: Performing Platform Administration
Lab 2-2: Performing General Administration
Lab 2-3: Configuring Cisco Unified CallManager Release 5.0 Basic Settings
Lab 2-4: Backing Up and Restoring Cisco Unified CallManager Release 5.0
Lab 3-1: Configuring Auxiliary or Voice VLANs
Lab 3-2: Configuring Cisco Unified CallManager to Support Cisco IP Phones
Lab 3-3: Configuring SIP Endpoints
Lab 3-4: Using Cisco Unified CallManager BAT
Lab 4-1: Configuring Cisco Unified CallManager Trunks
Lab 4-2: Configuring Voice Gateways
Lab 4-3: Configuring Basic Dial Plan Elements
Lab 4-4: Configuring Hunt Groups and Call Coverage
Lab 4-5: Configuring Complex Dial Plan Elements
Lab 4-6: Implementing Calling Privileges and Restrictions
Lab 5-1: Configuring Call Admission Control
Lab 5-2: Configuring Cisco Unified SRST
Lab 6-1: Configuring Media Resources
Lab 6-2: Configuring User Features
Lab 6-3: Configuring Cisco Unified CallManager Extension Mobility
Lab 6-4: Configuring Presence and BLF Functionality
Lab 6-5: Configuring Additional Features
Lab 6-6: Configuring Cisco Unified CallManager Attendant Console
Answer Key
Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0
2006 Cisco Systems, Inc.
Lab 2-1: Performing Platform Administration

Complete this lab activity to practice what you learned in the related module.
Activity Objective
In this activity, you will administer the Cisco Unified CallManager platform using the Cisco
IPT Platform Administration application. After completing this activity, you will be able to
meet these objectives:
Check the status of various platform components
View IP, NTP, and SMTP settings
View security certificates
Upload files such as locales, dial plans, and TFTP server files
Use the ping interface and set up a limited-time remote account that Cisco support
personnel can use to access the system
Use the CLI to perform basic platform functions
Visual Objective
The figure illustrates what you will accomplish in this activity.
Visual Objective for Lab 2-1:

Performing Platform Administration
1.
Check platform status
2.
View platform settings
3.
View security certificates
4.
Upload TFTP files
5.
Check network connectivity

and setting up remote
support
6.
Use the CLI
PC-x
CCM1-x CCM2-x
HQ-x
HQ-x
POD-x
2006 Cisco Systems, Inc. All rights reserved.
CIPT1 v5.0-5
You will perform Cisco Unified CallManager platform administration from the administrator
PC (PC-x) using HTTPS and SSH access to the Cisco Unified CallManager appliance. In
addition, you will upload files to the Cisco Unified CallManager TFTP server directory.
Lab Guide
Required Resources
These are the resources and equipment that are required to complete this activity:
First node running Cisco Unified CallManager Release 5.0
PC running Windows Explorer
SSH client software on PC (for example, PuTTY)
Lab Devices and Their Roles

Device
Device Role in the Activity
CCM1-x
Cisco Unified CallManager Release 5.0 first node
CCM2-x
Cisco Unified CallManager Release 5.0 subsequent node
PC-x
PC with Internet Explorer browser

SSH client to connect to CLI of Cisco Unified CallManager
Release 5.0
Use the credentials in the table to log in to the lab devices that require authentication.
Credentials for CCM1-x Application Access
Device
Username / Password
Access
Cisco IPT Platform Administration

GUI
admin / adminpass
https://10.x.1.1/iptplatform
Cisco Unified CallManager

Platform Administration CLI
admin / adminpass
SSH to 10.x.1.1
Command List
The table describes the commands that are used in this activity. The table is a small sample of
the CLI commands available.
Cisco Unified CallManager Platform Administration CLI Commands
Command
Description
show status
Displays the basic platform status information:
Host name
Date
Time zone
Locale
Product version
Platform version
CPU usage
Memory and disk usage
show tech ccm_service
Displays information on all Cisco Unified CallManager

services that can run on the system.
show version active
Displays the software version on the active partition.
show network eth0
Displays information about the first Ethernet interface.
file list tftp *

file list install *
Shows a list of all TFTP files.

Shows a list of all installation log files.
file view install

install.log
Shows the content of the install.log installation log file.
file search activelog *

error
Searches for the string error in all active log files.
utils network capture eth0

src <ip_address> <packets>
Shows packets coming from the specified IP address. The

last command argument (packets) indicates how many
packets should be captured.
Job Aids
These job aids are available to help you complete the lab activity.
The addressing of lab devices uses the IP allocation scheme shown in the table.
Lab Guide
IP Allocation Scheme
Parameter
Value
Voice server network
10.x.1.0/24
Data network
10.x.3.0/24
Default gateways
HQ-x: Node address .101 (for all networks)
CCM1-x
10.x.1.1/24
CCM2-x
10.x.1.2/24
PC-x
DHCP assigned (10.x.3.0/24)
The figure shows the menu structure of the Cisco IPT Platform Administration.
Expanded Menus Reference
Help Menu Also
CIPT1 v5.0-6
Task 1: Checking Platform Status

In this task, you will check the status of various platform components.
Activity Procedure
Complete these steps:
Step 1
From PC-x, access the Cisco IPT Platform Administration GUI.
Step 2
Visit all pages available under the Show menu to obtain information about the
cluster, hardware, network software, and system.
Step 3
To what page in Cisco IPT Platform Administration would you go to find these
items?
Q1)
First node IP address

Show >
Q2)
NIC speed and duplex setting, MAC address of the Cisco Unified CallManager node,
DHCP enabled status
Show >
Q3)
CPU and memory resource utilization statistics of the Cisco Unified CallManager node
Show >
Note
Answers to questions are at the end of this document, in the lab activity Answer Key.
Activity Verification
You have completed this task when you attain this result:
You can navigate to pages within Cisco IPT Platform Administration to check platform
status.
Task 2: Viewing Platform Settings

In this task, you will view IP, NTP, and SMTP settings.
Activity Procedure
Step 1
Step 2
Visit all pages available under the Settings menu.
Lab Guide
Answer these questions:
Step 3
Q1)
To what page in Cisco IPT Platform Administration would you go to change the DHCP
status (enable to disable or vice versa)?
Settings >
Q2)
Can you modify the IP address of the first node from within Cisco IPT Platform
Administration?
You can navigate to pages within Cisco IPT Platform Administration to view or update
platform settings.
Task 3: Viewing Security Certificates

In this task, you will view the certificates that are used by your Cisco Unified CallManager.
Activity Procedure
Step 1
Step 2
Go to Security > Certificate Management > Display Certificate.
Step 3
From Cisco Unified CallManagers own certificates, select the certificate of the
tomcat service.
Step 4
Examine the certificate and write the fingerprints of the certificate:

SHA1: __ __ __ __ __ __ __ __ __ __ __ __ __ __ __ __ __ __ __ __
MD5: __ __ __ __ __ __ __ __ __ __ __ __ __ __ __ __
Note
When you browse to Cisco Unified CallManager web pages, Cisco Unified CallManager will
send the tomcat certificate to your browser. Because the certificate is self-signed, it is not
trusted automatically.
Therefore it is recommended that you write down the authentic fingerprints after installation.
To view the authentic fingerprints from a browser (using the GUI), make sure that the
network between your PC and Cisco Unified CallManager is trusted (ideally you are on a
server network that is physically secured).
From now on, you can access Cisco Unified CallManager web pages from any location
(even over untrusted network) because you can verify the received certificates authenticity
by comparing the received fingerprints against the fingerprints that you wrote down after
installation.
You have completed this task when you attain these results:
You viewed the tomcats certificate.
You found the fingerprint in the tomcat certificate and wrote it down.
Task 4: Uploading TFTP Server Files

In this task, you will upload TFTP server files using Cisco IPT Platform Administration.
Activity Procedure
Step 1
Step 2
Go to Software Upgrades > Upload TFTP Server File. You will see all files that
are already stored on the Cisco Unified CallManager TFTP server.
Step 3
Click Browse to select an international dial plan (UK). The file has been previously
downloaded from Cisco.com and is already stored on the desktop of your PC (PC-x)
in the CIPT1 directory. The filename of the UK dial plan is dp-ffr.1-1-4.GB.cop.
Step 4
Click Upload File to transfer the file to Cisco Unified CallManager.
Step 5
Now, use the same method to upload a locale file for German language. Again, the
file (cm-locale-de_DE-5.0.2.3-1.cop) is already stored to the CIPT1 directory of
your PC.
Step 6
Search for both files in the Uploaded Files list of the TFTP Server File Upload page.
You uploaded a dial plan and a locale file to Cisco Unified CallManager.
The new files are displayed in the list of uploaded files.
Task 5: Checking Network Connectivity and Setting Up Remote

Support
In this task, you will use the ping interface and set up a limited-time remote account that Cisco
support personnel can use to access the system.
Activity Procedure
Step 1
Step 2
Go to Services > Ping and ping the default gateway of PC-x (10.x.3.101).
Lab Guide
Step 3
Go to Services > Remote Support, add a remote support account for a duration of 1
day, and name it ciscotac. Write the passphrase that you would give to the Cisco
TAC representative.
Note
Support account names must be lowercase alphabet characters only. No numbers or special
characters are allowed. The account name has to have a length of at least six characters.
Only one support account can be configured at the same time.
You used the integrated ping utility to ping another device in your cluster, and the ping
output indicated that the ping was successful.
You set up a remote support account for Cisco TAC to access your system for a specified
duration. The account is shown under Current Remote Support Settings on the Remote
Support configuration page.
Task 6: Using the CLI

In this task, you will use the CLI to display basic platform statistics.
Activity Procedure
Step 1
Note
Step 2
Step 3
10
From PC-x, use an SSH client (PuTTY, which is stored in the CIPT1 directory on
the desktop of PC-x) to access the Cisco Unified CallManager Platform
Administration CLI.
Because the SSH connection uses a certificate at the server for device authentication, you
will see a pop-up window, displaying the thumbprint of the received certificate. This
thumbprint is different from the ones of the tomcat service, which you wrote down in task 3.
The SSH server uses its own certificate. Click Yes to add the certificate to the list of trusted
certificates of your SSH client application.
Use the CLI to gain information about the platform and system by entering these
show commands and any other show commands:
show status
show tech ccm_service
show version active
show network eth0
Use these file commands to list files, view files, display the content of files, and
search for a certain string within files or directories:
file list tftp *
file list install *
file view install install.log
Note
You will have to exit file viewing mode by entering the character q (for quit).
Note
Step 4
file search activelog * error
You will probably not find any hits when searching for this string on the newly installed Cisco
Unified CallManager.
Use this utils commands to capture packets sent out by your Cisco Unified
CallManager:
Note
utils network capture eth0 src 10.x.1.1 count 20
To generate traffic, you can press the Enter key after you see the confirmation that packets
will be captured. As you are connected to the server by SSH, you will see the packets of
your own SSH session.
You have completed this task when you can use the CLI to perform these actions:
Obtain status information about the system
List and view files
Search for text strings in logs
Capture network traffic
Lab Guide
11
Lab 2-2: Performing General Administration

Activity Objective
In this activity, you will perform general administration. You will add administrators with
different administrative privileges. After completing this activity, you will be able to meet these
objectives:
Access the Cisco Unified CallManager Administration page through the web interface
using the default administration account
Create two new users with different administration rights
Visual Objective

Performing General Administration
1.
2.
Access Cisco Unified

CallManager Administration
GUI from PC-x
Create users with different
administrative privileges
PC-x
CCM1-x CCM2-x
HQ-x
HQ-x
POD-x
CIPT1 v5.0-7
From your PC, you will log in to the Cisco Unified CallManager Administration GUI and set
up additional administrator accounts.
Required Resources
12

Device
CCM1-x
CCM2-x
PC-x
Device
Username / Password
Access

Administration GUI
CCMAdministrator /
appuserpass
https://10.x.1.1/ccmadmin
Job Aids
Parameter
Value
10.x.1.0/24
Data network
10.x.3.0/24
Default gateways
CCM1-x
10.x.1.1/24
CCM2-x
10.x.1.2/24
PC-x
Lab Guide
13
The figure shows the menu structure of the Cisco Unified CallManager Administration.
CIPT1 v5.0-8
Task 1: Accessing the Cisco Unified CallManager Release 5.0

Administration GUI
In this task, you will access Cisco Unified CallManager Administration pages using the default
administrator account (CCMAdministrator).
Activity Procedure
Step 1
From PC-x, browse to https://10.x.1.1/ccmadmin.
Step 2
Log in using the default administrator account CCMAdministrator. The password of

this account is shown in the Credentials for CCM1-x Application Access table at the
beginning of this lab exercise.
Note
Step 3
14
The initial password of the CCMAdministrator is set during the installation of Cisco Unified
CallManager and can be changed at any time. The account name is static.
Go to User Management > Application User and click Find to list all application
users.
Note
You will see all application user accounts that exist by default. These cannot be deleted or
renamed. The CCMAdministrator is one of these users. It exists all the time and refers to a
standard group, which refers to a standard role. Because you cannot change the group
membership of the CCMAdministrator application user, and standard groups and standard
roles cannot be modified either, there is no way to modify the permissions of the
CCMAdministrator user account. Only the password can be changed. However, do not
change the password in this lab exercise.
You logged in to Cisco Unified CallManager Administration using the default account
CCMAdministrator.
You verified that the CCMAdministrator account is an application user that cannot be
deleted or renamed.
You verified that the CCMAdministrators privileges cannot be changed because you
cannot change the group membership or the group or the role that the group refers to.
Task 2: Creating Users with Different Administrative Privileges

In this task, you will create two end-user accounts for two different administrators. One should
have full access; the other one should have read access only.
Activity Procedure
Step 1
From PC-x, access Cisco Unified CallManager Administration.
Step 2
Go to User Management > End User and click Add New.
Step 3
Configure a user with the attributes that follow, and save the newly created account
by clicking the Save button at the bottom of the page or the Save symbol at the top
of the End User Configuration window.
Step 4
Step 5
User ID:
Johnny
Password: password
PIN:
Last name: Jones
12345
Click Add New again and add a second user with these attributes.
User ID:
Jack
Password: password
PIN:
Last name: Smith
12345
At the upper-left corner of the End User Configuration window, ensure that the
selected related link is Back To Find List Users, and click the Go button to get to
the Find and List Users page.
Lab Guide
15
Step 6
Click Find and ensure that both users are listed.
Step 7
Go to User Management > User Group and click Find to list all default user
groups.
Step 8
Click the Standard CCM Super Users group.
Step 9
At the group configuration page, click Add End Users to Group to add one or more
users to the group.
Step 10
From the list of users, select user Johnny and click Add Selected. User Johnny now
has full access to Cisco Unified CallManager Administration because he is a
member of the Standard CCM Super Users group.
Note
Step 11
Verify permissions of user Johnny by clicking the Permission icon next to the
username at the group configuration page.
Step 12
At the related links, select Back to User Groups Find/List and click the Go button.
Step 13
From the list of groups, click the Standard CCM Read Only group.
Step 14
At the group configuration page, click Add End Users to Group to add one or more
users to the group.
Step 15
From the list of users, select user Jack and click Add Selected. User Jack now has
full access to Cisco Unified CallManager Administration because he is a member of
the Standard CCM Read Only group.
Note
16
For full access to Cisco Unified CallManager Administration, users need to have the
standard roles Standard CCM Admin Users (for access to the administration homepage in
general) and Standard CCMADMIN Administration (for read and write access to all menu
items of the administration homepage). The group Standard CCM Super Users is assigned
to these two standard roles (among others).
For read access to Cisco Unified CallManager Administration, users need to have the
standard roles Standard CCM Admin Users (for access to the administration homepage in
general) and Standard CCMADMIN Read Only (for read access to all menu items of the
administration homepage). The group Standard CCM Read Only is assigned to these two
standard roles.
Step 16
Verify permissions of user Jack by clicking the Permission icon next to the
username at the group configuration page.
Step 17
Click Log Off at the top right corner of the screen to get to the Cisco Unified
CallManager Administration login page.
Step 18
Log in using the two different usernames and access the Find and List Users page by
going to User Management > End User. Verify that user Jack has only read access
(no buttons to add or delete users are shown), while user Johnny has full access (all
buttons shown).
You created two end users.
The first end user is assigned to the standard group Standard CCM Super Users.
The second end user is assigned to the standard group Standard CCM Read Only.
When displaying the Permissions of the first user, you see that this user has read and update
privileges to all components of Cisco Unified CallManager Administration. When logging
in with the first user name, you have full access to Cisco Unified CallManager
Administration.
When displaying the Permissions of the second user, you see that this user has read-only
privileges to all components of Cisco Unified CallManager Administration. When logging
in with the second user name, you have read-only access to Cisco Unified CallManager
Administration.
Lab Guide
17
Lab 2-3: Configuring Cisco Unified CallManager

Release 5.0 Basic Settings
Activity Objective
In this activity, you will configure Cisco Unified CallManager Release 5.0 basic settings to
activate services and use the Cisco Unified CallManager IP address instead of its hostname.
After completing this activity, you will be able to meet these objectives:
Change the hostname of Cisco Unified CallManager Release 5.0 into its IP address
Activate required services
Visual Objective
Visual Objective for Lab 2-3: Configuring Cisco

Unified CallManager 5.0 Basic Settings
1.
Change hostnames to IP
addresses
2.
Activate Services
PC-x
CCM1-x CCM2-x
HQ-x
HQ-x
POD-x
CIPT1 v5.0-9
From your PC, you will eliminate the need for Domain Name System (DNS) and activate the
minimum services needed to support Cisco IP phones.
Required Resources
18

Device
CCM1-x
CCM2-x
PC-x
Device
Username / Password
Access

Administration GUI
CCMAdministrator /
appuserpass

Serviceability GUI
CCMAdministrator /
appuserpass
https://10.x.1.1/ccmservice
Job Aids
Parameter
Value
10.x.1.0/24
Data network
10.x.3.0/24
Default gateways
CCM1-x
10.x.1.1/24
CCM2-x
10.x.1.2/24
PC-x
Lab Guide
19
The figure shows the menu structure of the Cisco Unified CallManager Serviceability GUI.
CIPT1 v5.0-10
Task 1: Changing Hostnames to IP Addresses

In this task, you will eliminate the need for DNS by changing from DNS names to IP addresses.
Activity Procedure
20
Step 1
Step 2
Go to System > Server and click Find to list all servers of your cluster.
Step 3
Click CCM1-x and change the Host Name/IP Address field from CCM1-x to
10.x.1.1.
Step 4
Add a description for your server. Enter First Node in the Description field.
Step 5
Then click Save.
Step 6
From the related links, select Back To Find/List and click Go.
Step 7
From the list of servers, click CCM2-x and change the Host Name/IP Address field
from CCM2-x to 10.x.1.2.
Step 8
Again, configure a description for your server. Enter Subsequent Node in the
Description field.
Step 9
Then click Save.
Step 10
Go to System > Enterprise parameters. Check all entries for references to any
server name (CCM1-x or CCM2-x). Change all of them to the appropriate IP
addresses.
Note
Because your cluster has just been freshly installed and no services are activated at this
time, there will be no URL entries, and therefore no changes are needed. URLs, which are
autogenerated after some special services are activated, will not use server names any
more because you already changed from server names to IP addresses.
Your cluster does not rely on DNS any more because both servers are configured by IP
address instead of names.
No enterprise parameter URLs refer to server names any more; they all use IP addresses.
Task 2: Activating Services

In this task, you will activate the minimum services that are required by Cisco Unified
CallManager to serve Cisco IP phones.
Activity Procedure
Step 1
From PC-x, access Cisco Unified CallManager Serviceability.
Step 2
Go to Tools > Service Activation.
Step 3
At the Select Server page, select 10.x.1.1.
Step 4
From the list of services, check the Cisco CallManager and the Cisco TFTP services.
Step 5
Click Save to activate these services. There will be a pop-up window that informs
you that service activation will take a while. Confirm by clicking OK, and then wait
until the Status (on top left corner of the page) changes from Ready to Update
Completed.
Step 6
Using the related links or Tools > Control Center Feature Services, go to the
control center for feature services.
Step 7
Verify that the Cisco CallManager and the Cisco TFTP services are started and
activated.
Step 8
Use the same procedure to activate and verify the Cisco CallManager service at the
second node (10.x.1.2).
The Cisco CallManager service is activated and running on both servers of your cluster.
The Cisco TFTP service is activated and running on CCM1-x.
Lab Guide
21
Lab 2-4: Backing Up and Restoring Cisco Unified

CallManager Release 5.0
Activity Objective
In this activity, you will use Cisco Disaster Recovery System GUI to perform backup and
restore operations. After completing this activity, you will be able to meet these objectives:
Configure backup on Cisco Unified CallManager Release 5.0 and manually back it up to a
network directory
Restore Cisco Unified CallManager Release 5.0 from the previously created system backup
Visual Objective
From your PC, you will first back up your Cisco Unified CallManager cluster and then restore
from the previously created backup.
Visual Objective for Lab 2-4: Backing Up and

Restoring Cisco Unified CallManager 5.0
1.
2.
Back up Cisco Unified

CallManager cluster to
SFTP server on PC-x
Restore Cisco Unfied
CallManager cluster from
SFTP server on PC-x
PC-x
CCM1-x CCM2-x
HQ-x
HQ-x
POD-x
CIPT1 v5.0-11
Required Resources
22
SFTP server software installed on the PC
Job Aids
Use the credentials in the table to log in to the lab applications that require authentication.
Device
CCM1-x
CCM2-x
PC-x
PC with Internet Explorer browser and SFTP server
Device
Username / Password
Access

Administration GUI
CCMAdministrator /
appuserpass

Disaster Recovery System
GUI
admin / adminpass
https://10.x.1.1/drf

Platform Administration
CLI
admin / adminpass
SSH to 10.x.1.1
Job Aids
Parameter
Value
10.x.1.0/24
Data network
10.x.3.0/24
Default gateways
CCM1-x
10.x.1.1/24
CCM2-x
10.x.1.2/24
PC-x
Lab Guide
23
Task 1: Backing Up Cisco Unified CallManager Using Cisco

Disaster Recovery System GUI
In this task, you will manually back up your Cisco Unified CallManager cluster to your PC,
where an SFTP server is running.
Activity Procedure
Step 1
On PC-x, start the SFTP server from the CIPT1 directory located on the desktop.
Step 2
Configure the server as follows:
User:
user
Password: pass
Port:
Root path: C:\
22
Step 3
Start the server by clicking the Start button.
Step 4
Find out the IP address of your PC by going to Start > Run and then entering
cmd and clicking OK.
Step 5
In the command window, enter ipconfig and press Enter. Now write the IP address
of your PC:
_____ . _____ . _____. _____
Note
The IP address must be one of network 10.x.3.0/24.
Step 6
Access Cisco Unified CallManager Disaster Recovery System from PC-x.
Step 7
Go to Backup > Configure Features.
Step 8
At the Configure Cluster Features page under Select Features, check both features
(CCM and CDR_CAR) and click Save.
Step 9
Go to Backup > Storage Location.
Step 10
At the Storage Location page under Select Destination, select Network Directory
with these attributes:
Note
Step 11
24
Server name:
<IP address of your PC>
This is the IP address that you wrote down in step 5.
Path name:
User name:
user
Password:
pass
Click Save.
Step 12
Go to Backup > Manual Backup.
Step 13
At the Manual Backup page under Features selected for Backup, click the Start
Backup button.
Step 14
The Backup Status page will appear and indicate the progress of the backup. The
window refreshes automatically every few seconds. Wait until the status that is
displayed on top of the window is SUCCESS: Backup Completed.
Step 15
At the Backup Status page under Backup details, you will see the Tar Filename.
Write the Tar Filename of the backup file. You will need it later for the restore.
Step 16
At your PC, verify that one xml file and several tar files have been copied to C:\.
They should all start with the Tar Filename that you wrote down in the previous
step.
The manual backup is successful as indicated by these outcomes:
The backup completes 100 percent, and the result status indicates successful
completion.
Backup files are stored on your PC in the C:\ directory.
Task 2: Restoring the System Using the Cisco Disaster

Recovery System GUI
In this task, you will restore Cisco Unified CallManager Release 5.0 from the previously
created system backup.
Activity Procedure
Change the Configuration
Step 1
Step 2
Go to User Management > End User and click Add New.
Step 3
Configure a user with the attributes that follow, and save the newly created account
by clicking the Save button at the bottom of the page or the Save symbol at the top
of the End User Configuration window.
Step 4
User ID:
test
Password: password
PIN:
Last name: test
12345
Make sure that the SFTP server is still running on PC-x. If it is not, follow the first
steps of the previous task to restart it.
Lab Guide
25
Note
Do not start multiple instances of the SFTP server application on PC-x.
Restore the Backed Up Configuration

Step 5
From PC-x, access the Cisco Unified CallManager Disaster Recovery System.
Step 6
Go to Restore > Restore Wizard.
Step 7
At the Step 1 Restore window, select Network Directory with these attributes:
Note
Server name:
<IP address of your PC>
This is the IP address that you wrote down in step 5 of the previous task.
Path name:
User name:
user
Password:
pass
Step 8
Click Next.
Step 9
At the Step 2 Restore window, select the backup file that corresponds with the Tar
Filename that you wrote down in step 15 of the previous task. Then click Next.
Step 10
At the Step 3 Restore window, select both features (CCM and CDR_CAR) to be
restored. Then click Next.
Step 11
At the Step 4 Restore window, select check all features on all servers and click
Restore.
Step 12
The Restore Status page will appear and indicate the progress of the restore. The
window refreshes automatically every few seconds. Wait until the status that is
displayed on top of the window is SUCCESS: Restore Completed.
Restart the Servers

Step 13
From PC-x, use an SSH client (PuTTY, which is stored in the CIPT1 directory on
the desktop of PC-x) to access the Cisco Unified CallManager Platform
Administration CLI of CCM1-x.
Step 14
Enter the command utils system shutdown. You will be asked if you really want to
shut down the server. Confirm by entering yes.
Step 15
Restart the second server of your cluster using the same procedure, but this time
accessing the Cisco Unified CallManager Platform Administration CLI on server
CCM2-x. Wait until the servers are back up again. This will take about 5 to 10
minutes.
Step 16
Access Cisco Unified CallManager Administration. Go to User Management >

End User and click Find.
Verify the Restore

Step 17
26
Verify that user test that you added at the beginning of this task (after the backup) is
not shown any more because the system has been restored from the backup.
The restore is successful as indicated by these outcomes:
The backup completes 100 percent, and the result status indicates successful
completion.
The changes you made after the backup (adding username test) are gone.
Lab Guide
27

Activity Objective
In this activity, you will configure auxiliary or voice VLANs to distinguish between voice and
data traffic. After completing this activity, you will be able to meet this objective:
Configure auxiliary or voice VLANs on the provided Cisco Catalyst switch
Visual Objective

Configuring Auxiliary or Voice VLANs
Phone1-x
Phone2-x
Phone3-x
VLAN: x03
CCM1-x CCM2-x
HQ-x
VLAN: x01
BRANCH-x
VLAN: x02
PC-x
HQ-x
BR-x
Phone4-x
VLAN: x05
POD-x
CIPT1 v5.0-12
At the moment, the Cisco Catalyst switch is preconfigured with data VLANs only, making it
possible for the PC to access the Cisco Unified CallManager server. In this lab exercise, ports
on the Cisco Catalyst switch will be reconfigured to offer either voice and data VLANs through
the same physical FastEthernet port (HQ) or voice VLAN only (BR).
Required Resources
28
Cisco IP phones
Cisco IOS Catalyst switch
Cisco Unified CallManager cluster
Command List
The tables describe the commands that are used in this activity.
Cisco IOS Catalyst Series Switch Commands
Command
Description
enable
Enables you to enter privileged EXEC mode or any other

security level that is set by a system administrator.
interface fastethernet
Begins interface configuration for the Catalyst 3500 Series

Switch.
switchport mode access
Configures the switchport to be an access (nontrunking) port
switchport voice vlan

voice_vlan_ID
Configures the Cisco IP phone to transmit voice traffic in 802.1Q

frames, tagged with the voice VLAN ID and a Layer 2 CoS value
(the default is 5). Valid VLAN IDs are from 1 to 4094. The switch
puts the 802.1Q voice traffic into the voice VLAN.
switchport access vlan

data_VLAN_ID
Configures the interface as a static access port with the access

VLAN ID (262 in this example); the range is 1 to 4094.
spanning-tree portfast
Causes a port to enter the spanning-tree forwarding state

immediately and bypass the listening and learning states.
Job Aids
This job aid is available to help you complete the lab activity.
Port Assignment on LAN Switch
Pod
Phone1-x
Phone2-x
Phone3-x
Phone4-x
0/1
0/2
0/3
0/4
0/1
0/2
0/3
0/4
VLAN Assignment to Pods

Pod Number
HQ Voice VLAN
HQ Data VLAN
BR Voice VLAN
102
103
105
202
203
205
Lab Guide
29
Task 1: Configuring Auxiliary or Voice VLANs

In this task, you will configure auxiliary or voice VLANs on the Cisco IOS Catalyst switch.
Activity Procedure
Step 1
Refer to the Port Assignment on LAN Switch table in the Job Aids section and
obtain the Cisco Catalyst switch ports that apply to IP phone in your pod. Record the
port numbers in the space that follows:
Port number of Phone1-x: ________________________
Step 2
Connect to the Cisco Catalyst switch and enter the configuration mode. You can
reach the switch from your PC using Telnet to the IP address 3.x.0.2 (where x is
your pod number).
Step 3
Configure voice and data VLANs on the switch ports connected to Phone1-x and
Phone2-x. The configuration commands for the Cisco Catalyst series switches
follow. For the VLAN numbers, refer to the VLAN Assignment to Pods table in
the Job Aids section. Enter these commands to configure HQ voice and HQ data
VLANs on a Cisco Catalyst Series switch running Cisco IOS:
interface fastethernet<module/port>
switchport voice vlan <HQ Voice VLAN number>
switchport access vlan <HQ Data VLAN number>
exit
Note
Step 4
Your instructor has already configured the data VLANs. This command is for reference only:
vlan <VLAN number>
Configure the switch port that is connected to Phone3-x for access mode with the
HQ voice VLAN. This is necessary because the third-party SIP phone does not work
with voice (auxiliary) VLANs. For the VLAN number, refer to the VLAN
Assignment to Pods table in the Job Aids section.
switchport access vlan <HQ Voice VLAN number>
exit
30
Step 5
Configure BR voice VLAN on the switch port that is connected to Phone4-x. For the
VLAN number, refer to the VLAN Assignment to Pods table in the Job Aids
section.
switchport voice vlan <BR Voice VLAN number>
end
Step 6
Save the switch configuration with copy running-config startup-config.
Disconnect your PC and connect IP phones in your cluster to their respective ports on
Cisco Catalyst switch and verify that they obtain correct voice VLAN numbers. Use one of
these Cisco IP phone model-specific sequences to display the VLAN information:
Cisco Unified IP Phone 7940: Press the Settings button, and then press 3 to view
the Network Configuration menu. Press 19 to see the Operational VLAN ID that
shows the voice VLAN that the IP phone has obtained from the switch.
Cisco Unified IP Phone 7961: Press the Settings button, and then press 2 to view
the Network Configuration menu. Press 20 to see the Operational VLAN ID that
shows the voice VLAN that the IP phone has obtained from the switch.
Disconnect all IP phones from the switch again and connect your PC to the port where it
was connected before.
Lab Guide
31

to Support Cisco IP Phones
Activity Objective
In this activity, you will add and register Cisco IP phones to an IP telephony cluster. After
completing this activity, you will be able to meet these objectives:
Configure a DHCP scope on the first node or default gateway router to provide IP
addresses and other options to network devices
Configure system parameters, including the Cisco Unified CallManager server, Cisco
Unified CallManager group, and device pools, to prepare the Cisco Unified CallManager
cluster to autoregister telephones
Add IP phones to the Cisco Unified CallManager database using auto-registration
Manually configure an IP phone and directory number
Visual Objective

Unified CallManager to Support Cisco IP Phones
Phone2-x
Phone1-x
Phone3-x
DN: Auto
.11-.20
.11-.20
10.x.2.0/24
.11-.20
IP
CCM1-x CCM2-x
DHCP
server
.1
HQ-x
.2
10.x.1.0/24
BRANCH-x
.101
.101
HQ-x
DHCP
server
BR-x
.102
IP
Phone4-x
DN: 3004
.11-.20
10.x.5.0/24
POD-x
32
CIPT1 v5.0-13
In this activity, you will configure DHCP on the first node and the BR-x router to provide an IP
address, default gateway IP address, and TFTP server IP address to the IP phones. The IP
phones then contact their TFTP server (first node) to obtain their load and a prioritized list of
Cisco Unified CallManager servers. The IP phones will then register with the Cisco Unified
CallManager servers and obtain a directory number (if not manually assigned) and other
configuration information (except the third-party SIP phone, which is registered later). You will
then be able to place calls to other IP phones within your cluster.
Required Resources
Cisco IP phones (such as Cisco Unified IP Phone 7910, 7940, or 7960 models)
Third-party SIP phone (such as Linksys SPA942)
Cisco Catalyst switch (Cisco Catalyst 3560 recommended)
DHCP servers (first node server for HQ IP phones and BR-x router for BR IP phone)
Command List
The table describes the commands that are used in this activity.
Cisco Router IOS Commands
Command
Description
enable
Enables you to enter privileged EXEC mode or any other security

level set by a system administrator.
configure terminal
or
config t
Enters global configuration mode.
ip dhcp excludedaddress
Specifies the IP addresses that the DHCP server should not assign
to DHCP clients.
ip dhcp pool
Creates a name for the DHCP server address pool and places you
in DHCP pool configuration mode (identified by the config-dhcp#
prompt).
network
Specifies the subnet network number and mask of the DHCP

address pool.
default-router
Specifies the IP address of the default router for a DHCP client.

One IP address is required, although you can specify up to eight
addresses in one command line.
option 150 ip-address
Provides the TFTP server address in the option field 150 of DHCP.
telnet ip-address
Enables you to remotely access systems.
copy running-config
startup-config
Saves the router configuration to NVRAM.
Lab Guide
33
Job Aids
DHCP Addressing and Scope Information for Cisco IOS Router at Branch
Cluster
Name
Pool
Type
Pool Name
Pod1
Voice
pod1-br-voice
Network and
Mask
Default
Router
Option
150
10.1.5.0
255.255.255.0
10.1.5.102
10.1.1.1
Excluded
Addresses, Start
and End
10.1.5.1 to
10.1.5.10
and
10.1.5.21 to
10.1.5.254
Pod2
Voice
pod2-br-voice
10.2.5.0
255.255.255.0
10.2.5.102
10.2.1.1
10.2.5.1 to
10.2.5.10
and
10.2.5.21 to
10.2.5.254
DHCP Server Information for Cisco Unified CallManager Server

Cluster
Name
Primary
TFTP Server
(Option 150)
ARP Cache
Timeout (sec)
IP Address
Lease Time (sec)
Renewal (T1)
Time (sec)
Rebinding (T2)
Time (sec)
Pod1
10.1.1.1
300
36000
36000
36000
Pod2
10.2.1.1
300
36000
36000
36000
DHCP Addressing and Subnet Information for Cisco Unified CallManager Server
34
Cluster
Name
DHCP
Server
Subnet IP
Address
Primary
Start IP
Address
End IP
Address
Primary
Router IP
Address
Subnet Mask
Primary TFTP
Server
(Option 150)
Pod1
CCM1-1
(10.1.1.1)
10.1.2.0
10.1.2.11
10.1.2.20
10.1.2.101
255.255.255.0
10.1.1.1
Pod2
CCM1-2
(10.2.1.1)
10.2.2.0
10.2.2.11
10.2.2.20
10.2.2.101
255.255.255.0
10.2.1.1
Cisco Unified CallManager Configuration
Cisco
Unified
CallManager
Name
IP Address
Pod1
CCM1-1
10.1.1.1
Pod2
CCM1-2
10.2.1.1
Subsequent Node
Server
Function
Cisco
Unified
CallManager
Name
IP Address
CCM2-1
10.1.1.2
CCM2-2
10.2.1.2
AutoRegistration
DN Range
Server
Function
Start
End
Subs
equen
t
First Node
First
Cluster
Name
2001
2002
2001
2002
Task 1: Configuring the DHCP Service

You will configure the DHCP service on the default router at Branch and the first node server
at HQ to provide the IP address and other scope options to IP phones.
Activity Procedure
DHCP on Branch Router
You will configure an IP DHCP pool on the Branch default router.
Step 1
Refer to the DHCP Addressing and Scope Information for Cisco IOS Router at
Branch table in the Job Aids section of this lab activity. Refer to the Default
Router column.
Write the IP address of the Branch default router: __________________________
Step 2
Establish a Telnet session to the Branch default router and log in.
Step 3
Configure the exclusion of IP addresses from the DHCP pool. Refer to the
Excluded Addresses, Start and End column of the DHCP Addressing and Scope
Information for Cisco IOS Router at Branch table. Enter this command for both
excluded addresses of the voice pool:
ip dhcp excluded-address <start IP address> <end IP address>
Step 4
Refer to the DHCP Addressing and Scope Information for Cisco IOS Router at
Branch table. Configure an IP DHCP pool for voice by entering these commands:
ip dhcp pool <voice pool name>
network <voice subnet address> <subnet mask>
default-router <voice default router>
option 150 ip <first node IP address>
end
Step 5
Save the router configuration. Press Ctrl-Z, and then enter this command:
copy running-config startup-config
Lab Guide
35
DHCP on the First Node

You will configure the DHCP service for the HQ on the first node.
Step 6
Establish the connection to the Cisco Unified CallManager Administration tool of

the first node in your cluster and log in.
Step 7
Change to the Cisco Unified CallManager Serviceability tool by choosing it from

the Navigation drop-down menu, and then click the Go button.
Step 8
Choose Tools > Service Activation and, on the new page shown, choose your first
node from the Server drop-down menu.
Step 9
On the new page shown, activate the Cisco DHCP Monitor Service.
Step 10
Change back to the Cisco Unified CallManager Administration tool by choosing it

from the Navigation drop-down menu, and then click the Go button.
Step 11
Choose System > DHCP > DHCP Server from the menu and, on the new page
shown, click the Add New button.
Step 12
Choose your first node server from the Host Server drop-down menu.
Step 13
Refer to the DHCP Server Information for Cisco Unified CallManager Server
table in the Job Aids section and enter all parameters mentioned in the table. Keep
other fields that are not mentioned in the table empty.
Step 14
When done, click Save.
Step 15
Choose System > DHCP > DHCP Subnet from the menu and, on the new page
shown, click Add New button.
Step 16
Refer to the DHCP Addressing and Subnet Information for Cisco Unified
CallManager Server table in the Job Aids section and enter all parameters
mentioned in the table. Keep other fields that are not mentioned in the table empty.
Step 17
When done, click Save.
36
You can verify that all your IP phones have obtained the IP addresses from the configured
range. Use one of these IP phone model-specific procedures to display the IP address on the
IP phone:
Cisco Unified IP Phone 7940: Press the Settings button, select 3 (Network
Configuration), and then select 6 (IP Address).
Cisco Unified IP Phone 7961: Press the Settings button, select 2 (Network
Configuration), and then select 6 (IP Address).
Linksys SPA942: Press the Settings button (paper), select 9 (Network), and then
select 2 (Current IP).
Task 2: Configuring System Configuration Parameters

In this task, you will configure the Cisco Unified CallManager server in your cluster to prepare
it before connecting devices.
Activity Procedure
Cisco Unified CallManager Server Configuration
Step 1
In Cisco Unified CallManager Administration, choose System > Server to open the
Find and List Servers window.
Step 2
Click Find.
Step 3
Select the first node from the list shown. Change the name of the server to the IP
address of that server by deleting the name that appears in the Host Name/IP
Address field. This will eliminate the requirement of DNS in the cluster.
Step 4
Enter the former Host Name into the Description field and click Save.
Step 5
Click the Go button next to the Related Links: Back to Find/List drop-down menu.
Step 6
For the second node in your cluster, repeat the last 3 steps to replace its host name
with the IP address and also enter the server name into Description field.
Cisco Unified CallManager Configuration

You will now select the first node server in the cluster, edit the Cisco Unified CallManager
name, and fill in the description with the function of the server from the Cisco Unified
CallManager Configuration table.
Step 7
Choose System > Cisco Unified CallManager and click the Find button.
Step 8
In the list shown, choose the first node server.
Step 9
Do not change the name of the first node. Enter the function of the server in the
cluster into the Description field (example, First node).
Step 10
Enter the appropriate starting and ending directory number from the Cisco Unified
CallManager Configuration table. The system will automatically uncheck the AutoRegistration Disabled on This Cisco Unified CallManager check box.
Step 11
Click Save.
Step 12
Click the Go button next to the Back to Find/List at the upper-right side of the
window.
Step 13
Choose the name of the subsequent node server.
Step 14
Do not change the name of the subsequent node. Enter the function of the server in
the cluster in the Description field.
Step 15
Enter the starting and ending directory number. (Refer to the Cisco Unified
CallManager Configuration table in the Job Aids section of this lab activity.)
Step 16
Click Save.
Lab Guide
37
Cisco Unified CallManager Groups

Step 17
Choose System > Cisco Unified CallManager Group and click the Find button.
Step 18
Choose the name Default from the list displayed.
Step 19
Rename the Default group to the new name CMG.
Step 20
In the Cisco CallManager Group Members area, the first node (its IP address) should
already be in the Selected Cisco CallManagers pane.
Highlight the subsequent server in the Available Cisco CallManagers pane. Use the
arrow between the two boxes to move the subsequent server to the Selected Cisco
CallManagers pane.
Step 21
Use the Up arrow and Down arrow to place the first Cisco Unified CallManager at
the top of the list, making it the primary Cisco Unified CallManager for the group, if
this is not done automatically.
Step 22
Click Save. You might be notified about the reset of devices. Click OK if a pop-up
window is shown.
Step 23
Verify that the CMG Cisco Unified CallManager group is shown next to the Cisco
CallManager Group at the top of the page.
Configure Device Pools

In this section, you will create device pools that are named to reflect their Cisco Unified
CallManager group, date and time group, and region. A device pool is named
BA_Default_CMLocal_DP if it uses the BA_CMG Cisco Unified CallManager group, the
default region, and the CMLocal date and time group.
Step 24
Choose System > Device Pool and click the Find button.
Step 25
Click Default in the list shown.
Step 26
Change the Device Pool Name field from Default to Default_CMLocal_DP.
Step 27
Use the drop-down lists to ensure that these characteristics are selected:
Step 28
Cisco Unified CallManager Group: CMG-x
Date/Time Group: CMLocal (use the Cisco Unified CallManager internal

clock)
Region: Default (default)
Softkey Template: Standard User (default)
SRST Reference: Disable (default)
Click Save.
38
In Cisco Unified CallManager Administration, choose System > Device Pool and the list
automatically displayed should show the new device pool name.
Task 3: Adding IP Phones Using Auto-Registration

In a previous task, you configured auto-registration of the Cisco Unified CallManager group
and assigned a directory number range that this group can provide when the telephone
autoregisters. In this task, you will plug the remaining Cisco IP phones into the network, and
the IP phones will autoregister, getting the directory number from the Cisco Unified
CallManager Configuration window and then registering to the appropriate Cisco Unified
CallManager group that was assigned by the device pool on the Default Devices Configuration
window.
Activity Procedure
Step 1
Determine the correct switch ports for your pod and record the port numbers here:
Switch port number for Phone1-x: ________________________
Step 2
Insert an Ethernet cable of the Phone1-x into its port on the switch.
Step 3
Observe as the IP phone cycles through the registration process. When the IP phones
have successfully registered, it will display the date, time, and directory number.
Step 4
On the registered IP phone, press the Settings key, and then press the 3 button on
your keypad to view the Network Configuration in the telephone. Press 21 to view
the Cisco Unified CallManager settings. (The numbers can differ based on the Cisco
IP phone model used.)
The IP address of the Cisco Unified CallManager that is designated as the primary
Cisco Unified CallManager is displayed first, with the word Active to the right.
The IP address of the backup (secondary) Cisco Unified CallManager is displayed
next, with the word Standby to the right.
Step 5
Step 6
have successfully registered, it will display the date, time, and directory number.
Step 7
In Cisco Unified CallManager Administration, choose Device > Phone and click
Find without changing any parameters. A list of all registered IP phones is
displayed.
The device pool and IP address of the Cisco Unified CallManager to which the IP
phone is currently registered is displayed. The IP address of the Cisco Unified
CallManager is listed under the Status column. Note that the Description field
shows Auto with the directory number to indicate that a phone is autoregistered.
Step 8
Observe as the IP phone cycles through the registration process. This IP phone
should fail obtaining the directory number and should be unable to autoregister.
Lab Guide
39
The two Cisco IP phones have directory numbers, and you are able to call from one
registered IP phone to another IP phone within the cluster.
The third IP phone in sequence, attempting to register, has failed to autoregister.
Task 4: Manually Adding and Configuring an IP Phone

In this task, you will manually add a Cisco IP phone (Phone4-x) and assign a directory number.
The other telephones remain configured through auto-registration.
Activity Procedure
40
Step 1
In Cisco Unified CallManager Administration, choose Device > Phone and click the
Add New button.
Step 2
On the new page shown, from the Phone Type drop-down menu, choose the actual
type of the Phone4-x you use (for instance 7940). Click Next.
Step 3
On the new page shown, confirm that the SCCP device protocol is selected and click
the Next button.
Step 4
Obtain the MAC address from your Phone4-x. On the Phone4-x, click the Settings
button, then select the Network Configuration, and find the IP phones MAC
address under the row number 3. Another way to obtain the MAC address is to look
on the bar code that is located on the bottom back of the Cisco IP phone. Record the
MAC address here:
____________________________________________________________________
Step 5
Enter the Phone4-x MAC address into the MAC Address field.
Step 6
Enter 3004 in the Description field.
Step 7
Choose Default_CMLocal_DP from the Device Pool drop-down list.
Step 8
From the Phone Button Template drop-down list, choose Standard 7940 SCCP.
Step 9
Scroll the page down to locate the Protocol Specific Information section. From the
SCCP Phone Security Profile drop-down menu, choose Non Secure SCCP Profile
auth by null String.
Step 10
Click Save. (You will configure additional parameters later in the course.) On the
pop-up window shown, notifying about the reset, click OK.
Step 11
On the new page, click the Line [1] Add a new DN link to continue to configure
line 1 extension. The Directory Number Configuration page appears.
Step 12
Enter 3004 in the Directory Number field.
Step 13
Click Save.
Step 14
On the new page, click the Reset button to reset the IP phone.
Step 15
have successfully registered, it will display the date, time, and directory number. If
the IP phone fails to register, reset its network configuration.
You have completed this activity when you attain this result:
The Phone4-x obtains the directory number that you specified.
Post-Task 5: Manually Configuring All Cisco IP Phones

In this task, you will reconfigure your HQ and BR Cisco IP phones for predictable directory
number s. After this lab, all your Cisco IP phones will always keep the same directory numbers,
and autoconfiguration will be disabled.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Device > Phone and click the
Find button.
Step 2
Select the IP phone that is supposed to be the Phone1-x, regardless of the directory
number shown in the Description column.
If you are not sure which in the list is your Phone1-x, look at the IP phones display
to see the directory number that it shows. Then click the IP phone displaying that
directory number in the Description column.
Step 3
On the IP phones configuration page, change the Description field to 2001, and
click Save and OK on the pop-up shown.
Step 4
Click the Line[1] link, change the directory number to 2001, and then click Save.
Step 5
Reset the IP phone.
Step 6
Repeat the last four steps for the Phone2-x with these parameters:
Description: 2002
DN (Line[1]): 2002
Step 7
Disable Auto-Registration. Choose System > Cisco Unified CallManager, and then
click Find.
Step 8
From the list, select your first node and, when its configuration page opens, check
the check box next to the Auto-registration Disabled on this Cisco Unified
CallManager.
Step 9
Click Save and click OK to accept the message.
You have completed this activity when you attain this result:
Your Cisco IP phones should display directory numbers in this predictable order:
Phone1-x has 2001, Phone2-x has 2002, and Phone4-x has 3004.
Lab Guide
41

Activity Objective
In this activity, you will use the Cisco Unified CallManager BAT SCCP to SIP migration tool
to convert an SCCP phone to SIP, add a third-party SIP phone, and configure local dial rules on
a Cisco Unified SIP Phone 7961 so that the user does not have to press the dial key after
entering digits. After completing this activity, you will be able to meet these objectives:
Use the Cisco Unified CallManager BAT Migration Tool to convert an SCCP phone to SIP
Add a third-party SIP phone
Create and apply SIP dial rules to an existing Cisco Unified SIP Phone 7961
Visual Objective

Configuring SIP Endpoints
Phone2-x
Phone1-x
Phone3-x
3rd-party SIP
DN: 2003
SIP
.11-.20
.11-.20
CCM1-x CCM2-x
10.x.2.0/24
.11-.20
.101
.1
HQ-x
.101
.2
10.x.1.0/24
BRANCH-x
HQ-x
BR-x
.102
Phone4-x
.11-.20
10.x.5.0/24
POD-x
CIPT1 v5.0-14
Required Resources
42
Cisco Unified CallManager server
Two Cisco Unified SCCP Phone 7940 or 7960 phones that remain SCCP and one Cisco
Unified SCCP Phone 7961 that gets converted to SIP
One third-party SIP phone (Linksys SPA942 used for this lab)
Command List
There are no commands used in this activity.
Job Aids
There are no job aids for this activity.
Task 1: Using the Cisco Unified CallManager BAT Migration

Tool to Convert SCCP Phones to SIP
In this task, you use the Cisco Unified CallManager BAT Migration Tool to convert one SCCP
phone to SIP. The Cisco Unified CallManager BAT SCCP to SIP Migration Tool can converts
one phone, many, or all SCCP phones at a time. To convert all SCCP phones to SIP, navigate
to the Migrate Phones - SCCP to SIP Configuration page, click Find, and proceed with the
migration. To selectively choose which phones to migrate, enter search criteria from the
Migrate Phones - SCCP to SIP Configuration page.
In this task, you are going to use the Cisco Unified CallManager BAT migration tool to migrate
the Phone1-x, which is in use, using SCCP, to SIP.
Activity Procedure
Step 1
Make sure that all three of your Cisco IP phones are registered to their Cisco Unified
CallManager first node as SCCP phones. In Cisco Unified CallManager, choose
Device > Phones, and then click Find. You should see three IP phones with the
status Registered with.
Step 2
For this task to complete, Bulk Administration service has to be started. In the Cisco
Unified CallManager Serviceability tool, choose Tools > Service Activation and,
on your first node server, activate the service named Cisco Bulk Provisioning
Service.
Step 3
In Cisco Unified CallManager Administration, choose Bulk Administration >

Phones > Phone Template. Click Add New.
Step 4
For the Phone Type, select 7961 and click Next.
Step 5
For the protocol, select the SIP and click Next.
Step 6
Create a new phone template for 7961 with these characteristics:
Template Name: SCCP-SIP-7961
Description: SCCP-SIP-7961
Device Pool: Default_CMLocal_DP
Phone Button Template: Standard 7961 SIP
SIP Phone Security Profile: Standard SCCP Profile for Auto Registration
SIP Profile: Standard SIP Profile
Set all other required settings to their defaults if it is not done automatically.
Lab Guide
43
Step 7
Choose Bulk Administration > Phones > Migrate Phones > SCCP to SIP.
Step 8
Find the Phone1-x by its Directory Number equals to (is exactly) 2001 to produce a
Search Results list containing only this IP phone. If you see more IP phones than
Phone1-x in the list, do not proceed to the next step.
Step 9
Click Next.
Step 10
Select SCCP-SIP-7961 for the Phone Template Name.
Step 11
Make sure that the Reset phones radio button is selected.
Step 12
Click the Run Immediately radio button and leave the offered Job Description
name.
Step 13
Click Submit.
Step 14
The IP phone will restart, obtain a SIP phone load and upgrade the firmware, and reregister with Cisco Unified CallManager. This process takes several minutes (it
could take about 10-15 minutes), while the IP phone shows the firmware upgrade
progress on its display. Wait until the Phone1-x registers to Cisco Unified
CallManager again.
Caution
Known Issue: There is a problem with the reset from within the Cisco Unified CallManager
BAT migration tool. This feature was added in 5.0(2) release. The phones (Cisco Unified IP
Phone 7960 or 7961 models) never register. Try it, and if it does not work, delete the phone
from the database. Repeat the previously described migration process, but this time choose
not to reset/restart the phone. Instead, go to Bulk Administration > Phones >
Reset/Restart Phone > Query, find the IP phone by its directory number (2001), reset the
phone from this page, and run the job immediately. The reset from the Query page works on
Cisco Unified IP Phone 7961 phones but not 7960. In 5.0(1), it worked on both.
The Phone1-x registers to its Cisco Unified CallManager and runs the SIP device protocol.
Verify this from the Find and List Phones page and on the phone itself, using the Settings
button, then Model Information, and scroll to the Load File entry. (On different phone
types, this syntax might vary.)
You can make calls to all other IP phones in your pod.
Task 2: Adding a Third-Party SIP Phone

In this task, you will add the Linksys SPA942 third-party SIP phone.
Activity Procedure
Gather Information on the Phone
Step 1
Plug the third-party SIP phone into its Cisco Catalyst switch port, if it has not
already been done.
44
Step 2
Verify that the third-party SIP phone has obtained IP address using DHCP. Write the
IP address of the phone here:
Step 3
The Linksys SPA942 supports HTTP access. Start a web browser on your PC and
navigate to the IP address of the third-party SIP phone.
Step 4
When the page opens, record the MAC address of the phone (located on the Info
tab) on the space that follows:
____________________________________________________________________
Configure Cisco Unified CallManager to Support the Third-Party Phone

Step 5
In Cisco Unified CallManager Administration, create an end user with these
parameters:
User ID: dburke
Password: cisco
PIN: 12345
Last Name: Burke
Step 6
Add New.
Step 7
For the Phone Type, choose Third-party SIP Device (Advanced) from the dropdown menu. The Linksys SPA942 is a multiline phone (four lines), so you must
choose Advanced for licensing and configuration purposes. Click Next.
Step 8
On the Phone Configuration page for the Third-party SIP Device, add a phone with
these parameters:
MAC address of the phone obtained earlier from the web browser.
Description: 2003 third-party SIP phone
Phone Button Template: Third-party SIP Device (Advanced)
Common Phone Profile: Standard Common Phone Profile
SIP Phone Security Profile: Standard SIP Profile for Auto Registration
SIP Profile: Standard SIP Profile
Digest User: dburke
Line [1]: 2003
Configure the Third-Party Phone to Register with Cisco Unified CallManager

Step 9
Return to the web browser page that displays the third-party SIP phone IP address
and click the Admin Login link in the upper-right page corner.
Step 10
Click the advanced link in the upper-right page corner.
Step 11
Click the Ext 1 tab from the menu bar.
Lab Guide
45
Step 12
Configure these parameters on extension 1:
Proxy under Proxy and Registration:<First node IP address 10.x.1.1>
Auth ID: dburke
User ID: 2003
User Auth ID: yes
Step 13
Click the Submit All Changes button at the bottom of the current page.
Step 14
Wait until the third-party SIP phone obtains the directory number, and then place
test calls to other IP phones in your group.
Step 15
Find. You should see the third-party SIP phone listed, with its status showing
Registered. Try to restart the phone from Cisco Unified CallManager
Administration.
The third-party SIP phone has registered to Cisco Unified CallManager with extension
2003.
Directory number 2003 is displayed on all phone buttons.
You can place calls to other phones in your pod from the third-party SIP phone and receive
calls from IP phones in your pod.
The third-party SIP phone is listed on the Find and List Phone as a SIP phone. However, it
cannot be reset from Cisco Unified CallManager Administration. You must manually
reboot the phone from the phone itself or from the web interface when you submit a
change.
Task 3: Configuring SIP Dial Rules

In this task, you will configure SIP dial rules for the Cisco Unified IP Phone 7961 SIP phone so
that the user does not have to wait for the interdigit timeout or push the Dial soft key before the
call is placed.
Activity Procedure
46
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Dial Rules >
SIP Dial Rules.
Step 2
Add a new dial rule for with these characteristics:
SIP Dial Pattern :7940_7960_OTHER
Name: 7940-61_4-digit_DR
Description: 4 digit internal
Pattern Description: 4 digit internal (then click Add Pattern)
Choose Pattern from the Dial Parameter drop-down menu.
Value: Enter a pattern value that will enable your phone to match any four-digit
directory number in your pod (2XXX).
Timeout: 0 (dial as soon as a local dial rule match is found)
Step 3
Assign the new SIP Dial Rule to the Cisco Unified IP Phone 7961 SIP phone
(Phone1-x) from the Phone Configuration page.
Step 4
Save the change and reset the Cisco Unified IP Phone 7961.
Step 5
Test dialing behavior on the Cisco Unified IP Phone 7961 SIP phone (Phone1-x) to
other IP phones in your pod. The Cisco Unified IP Phone 7961 SIP phone should
extend the call immediately as soon as the four-digit directory number is entered.
Step 6
Test dialing to an invalid destination (5555, for example). If you do not press the
Dial softkey, the phone still waits for the interdigit timer to expire before returning a
reorder tone because no local dial rule has been defined for phone numbers in the
5 range.
Step 7
Test dialing to a valid but unassigned number (2555, for example). The phone now
returns a reorder tone as soon as digits are entered because a local dial rule is
matched.
When you dial a valid or unassigned four-digit directory number in your pod from the
Cisco Unified IP Phone 7961 SIP phone, the call is extended or a reorder tone is returned as
soon as the four digits are dialed, without pressing the Dial softkey.
All other digits, valid or invalid, still require that you press the Dial softkey or wait for the
interdigit timer to expire.
Lab Guide
47

Activity Objective
In this activity, you will use Cisco Unified CallManager BAT to bulk-add Cisco IP phones to
an IP telephony network. After completing this activity, you will be able to meet these
objectives:
Create an IP phone template to use with Cisco Unified CallManager BAT
Create a CSV file using the Cisco Unified CallManager BAT spreadsheet
Validate the IP phone template and CSV file to check for errors
Insert the IP phones and users into the Cisco Unified CallManager database
Visual Objective
There is no figure illustrating what you will accomplish in this activity.
Required Resources
These are the resources and equipment required to complete this activity:
Three Cisco IP phones
Cisco Unified CallManager BAT.xlt spreadsheet
MS Excel installed on your PC
Command List
Job Aids
Task 1: Creating an IP Phone Template to Use with Cisco

Unified CallManager BAT
In this task, you will create a Cisco IP phone template to define common attributes.
Activity Procedure
48
Step 1

Phones > Phone Template and click Add New.
Step 2
Select Cisco 7960 for Phone Type and click Next.
Step 3
Leave SCCP as device protocol and click Next.
Step 4
Create the new phone template with these parameters:
Template Name: BAT-7960
Description: 7960 template
Phone Button Template: Standard 7960 SCCP
SCCP Phone Security Profile: Standard SCCP Profile for Auto Registration
Leave all other settings at their default values and click Save.
Step 5
Click the Line[1] link in the left column, then enter 2100 for Line Template Name,
and click Save.
Step 6
Highlight the BAT-7960 in the Associated Devices pane and click Edit Device.
Step 7
Click the Line[2] link in the left column, then enter 2101 for Line Template Name,
and click Save.
Task 2: Creating the CSV File Using the Cisco Unified

CallManager BAT Spreadsheet
In this task, you will use the Cisco Unified CallManager BAT spreadsheet to add lines and
directory numbers, and you will export the Excel spreadsheet to a CSV file for use with Cisco
Unified CallManager BAT.
Activity Procedure
Step 1

Upload/Download Files and the click Find.
Step 2
Check the check box next to the bat.xlt file and click the Download Selected
button.
Step 3
On the new dialog window, click Save, and then navigate to the C:\ directory on
your local PC.
Step 4
Change the filename in the File name field to bat7960.xlt and click Save. Wait until
the file is downloaded to your local PC.
Step 5
Open the file on your local PC using MS Excel.
Step 6
On the Phones spreadsheet tab, click the Phones radio button, and then click the
Create File Format button.
Step 7
On the new dialog window, highlight the Directory Number in the Line Fields
pane, and then click the right arrow button to move it to the Selected Line Fields
pane.
Step 8
Click Create button, and then click Yes on the pop-up window to overwrite the
existing file.
Step 9
In the Maximum Number of Phone Lines field, enter 2 to create two lines for new IP
phones. Then click any other portion of the spreadsheet to confirm the modification
and click OK on the new pop-up window.
Lab Guide
49
Step 10
Check the check box next to the Dummy MAC Address field.
Step 11
On the Phones spreadsheet, enter five new IP phones with the parameters that
follow, and leave the MAC Address column empty.
Description
Directory Number 1
Directory Number 2
IPPhone6
2006
3006
IPPhone7
2007
3007
IPPhone8
2008
3008
IPPhone9
2009
3009
IPPhone10
2010
3010
Step 12
Scroll right and click the Export to BAT Format button.
Step 13
Save the new file as C:\Phones7960. You should be notified that the file was
successfully exported. Click OK.
Step 14
Exit MS Excel without saving the changes to the .xlt file.
Task 3: Validating the IP Phone Template and CSV File

In this task, you will enable Cisco Unified CallManager BAT to run a validation routing to
check the CSV file that you created in Task 2, and the IP phone template that you created in
Task 1, to ensure that all fields are populated.
Activity Procedure
50
Step 1
In Cisco Unified CallManager Serviceability, choose Tools > Service Activation

and make sure that the Cisco Bulk Provisioning Service is activated on your first
node. If not, activate it.
Step 2

Upload/Download Files and click Add New.
Step 3
Click Browse and, on the new dialog window, navigate to the C:\Phone7960.txt file
highlight it, and then click Open. This will populate the File field on the File Upload
Configuration page.
Step 4
Choose Phones from the Select Transaction Type drop-down menu.
Step 5
Choose Insert Phones Specific Details from the Select Transaction Type dropdown menu and click Save.
Step 6
Choose Bulk Administration > Phones > Validate phones.
Step 7
From the File Name drop-down menu, choose the CSV file Phones7960.txt
Step 8
From the Phone Template Name drop-down menu, choose the BAT-7960 template
created earlier.
Step 9
Click Submit.
Step 10
Choose Bulk Administration > Job Scheduler and click Find.
Step 11
In the job list shown, click the Job Id link that has Validate Specific Phones in the
Description column.
Step 12
The job results should display the validation status without errors.
The validation status is Validate Completed, and when you click the Log File Name link,
the Result Summary message reads Validate for 5 Phones passed. Validate for 0 Phones
failed.
Cleanup
Close the log file window and click the Delete button to remove the validation job from the list
of jobs.
Task 4: Inserting Phones and Users into the Cisco Unified

CallManager Database
In this task, you will complete the last step in the Cisco Unified CallManager BAT process by
inserting the IP phones into the Cisco Unified CallManager database.
Activity Procedure
Step 1
Choose Bulk Administration > Phones > Insert Phones.
Step 2
Choose Phones7960.txt from the File Name drop-down menu.
Step 3
Choose BAT-7960 from the Phone Template Name drop-down menu.
Step 4
Check the check box next to Create Dummy MAC Address.
Step 5
Click the Run Immediately radio button.
Step 6
Click Submit.
Step 7
Choose Bulk Administration > Job Scheduler and click Find.
Step 8
If the Status column of the phone insertion job does not show Completed, click the
Find button again. Repeat this step until you see the status Completed.
Step 9
Click the phone insertion Job Id link and make sure that the job has completed
successfully. Alternatively, you can review the log file.
Step 10
Click Delete button to remove the phone insertion job from the list of jobs and click
OK on the pop-up window.
Lab Guide
51
The phones insertion job has completed with success.
When you choose Device > Phone and click Find, you see in the list the five new IP
phones that you bulk-added. Notice that the telephones added with Cisco Unified
CallManager BAT all start with BAT followed by a dummy MAC address in the Device
Name field. Click one of the device names and view the results. You can customize any of
these fields on a per-telephone basis.
Cleanup
To prepare for future labs, complete these steps:
52
Step 1
Delete the Cisco Unified CallManager BAT telephones that were added using Cisco
Unified CallManager BAT. Choose Device > Phone and click Find.
Step 2
Check the check box next to each bulk-added IP phone (their names start with
BAT) and click Delete Selected. Click OK on the pop-up window.
Step 3
Make sure that the only IP phones that remain are the four active IP phones in your
cluster (those starting with SEP).

Trunks
Activity Objective
In this activity, you will add and configure an intercluster trunk to enable calling to a remote
cluster over the IP WAN. (Calling will not be fully enabled until you add a route pattern to the
remote cluster in the next lab activities.) After the intercluster trunk is set, you will add and
configure a SIP trunk between the same clusters. After completing this activity, you will be
able to meet these objectives:
Configure an intercluster trunk to a remote cluster in Cisco Unified CallManager

Administration
Configure a SIP trunk to a remote cluster in Cisco Unified CallManager Administration
Visual Objective
Visual Objective for Lab 4-1: Configuring

Cisco Unified CallManager Trunks
Phone2-1
Phone1-1
Phone1-2
Phone3-1
Phone2-2
Phone3-2
IC_TRK
SIP_TRK
CCM1-1 CCM2-1
CCM1-2 CCM2-2
10.12.0.0/16
.1
HQ-1
.2.101
.1.101
.2
10.1.1.0/24
.2
.1
HQ-2
HQ-1
10.2.1.0/24
HQ-2
IC_TRK
BRANCH-1
SIP_TRK
BR-1
BR-2
Phone4-1
BRANCH-2
Phone4-2
POD-1
POD-2
CIPT1 v5.0-15
Required Resources
Two Cisco Unified CallManager Release 5.0 clusters
IP WAN connectivity between the two clusters
Lab Guide
53
Command List
The table describes the command that is used in this activity.
Cisco IOS Commands
Command
Description
ping
Verifies remote IP address reachablity.
Job Aids
Intercluster Trunk Information
Cluster Name
Device Name
Description
Remote IP Address
Pod 1
IC_TRK
Intercluster to Pod 2
First: 10.2.1.1
Subsequent: 10.2.1.2
Pod 2
IC_TRK
Intercluster to Pod 1
First: 10.1.1.1
Subsequent: 10.1.1.2
SIP Trunk Information

Cluster Name
Device Name
Description
Destination Address
Pod 1
SIP_TRK
SIP trunk to Pod 2
10.2.1.1
Pod 2
SIP_TRK
SIP trunk to Pod 1
10.1.1.1
Pre-Lab Task 1: Verifying IP Connectivity Between Pods

In this task, you will verify the IP connectivity between the pods across IP WAN. This IP
connectivity will be used for the intercluster and SIP trunks over IP WAN between the pods.
Activity Procedure
Step 1
Connect to the cluster gateway HQ-x (where x is your pod number), log in, and enter
the privileged mode.
Step 2
Use the extended ping command on the gateway to verify if the remote cluster first
node can be reached over the IP WAN. Use these parameters for ping (leave other
parameters at their defaults):
Remote IP address: 10.y.1.1 (where y is the remote pod number)
Source IP address: 10.x.1.101 (where x is your pod number)
54
You could successfully reach the remote cluster first node with ping.
Task 1: Configuring Intercluster Trunks

In this task, you will use the IP address of the Cisco Unified CallManager nodes in the remote
cluster to add an intercluster trunk in Cisco Unified CallManager Administration. Pod 1 cluster
will connect to Pod 2 cluster over IP WAN.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Device > Trunk.
Step 2
Click Add New button.
Step 3
Choose Inter-Cluster Trunk (Non-Gatekeeper Controlled) from the Trunk Type

drop-down list. Click Next.
Step 4
Enter the device name for the intercluster trunk and a description from the
Intercluster Trunk Information table in the Job Aids section of this lab activity.
Step 5
Choose Default_CMLocal_DP for the device pool.
Step 6
Choose Allowed for the Calling Line ID Presentation (Outbound calls). Cisco
Unified CallManager uses calling line ID presentation as a supplementary service to
control the display of the calling party number on the called party IP phone screen.
Choosing Allowed displays the calling number information.
Step 7
Enter the IP address for the remote cluster first node in the Server 1 IP Address/Host
Name field. Refer to the Intercluster Trunk Information table in this lab activity to
find the correct IP address.
Step 8
Enter the IP address of the remote cluster subsequent node in the Server 2 IP
Address/Host Name field. Refer to the Intercluster Trunk Information table in this
lab activity to find the correct IP address.
Step 9
Click Save, and then click OK on the pop-up window.
Step 10
Click the Reset button, and then click the Reset button on the new dialog box. Click
the Close button on the new dialog box.
The intercluster trunk appears as a trunk when you choose Device > Trunk and then click
the Find button in Cisco Unified CallManager Administration.
Lab Guide
55
Task 2: Configuring SIP Trunks

In this task, you will add a SIP trunk in Cisco Unified CallManager Administration to the Cisco
Unified CallManager first node in the remote cluster.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Device > Trunk.
Step 2
Click the Add New button.
Step 3
Choose SIP Trunk from the Trunk Type drop-down list. Click Next.
Step 4
Enter the device name for the intercluster trunk and a description from the SIP
Trunk Information table in the Job Aids section of this lab activity.
Step 5
Choose Default_CMLocal_DP for the device pool.
Step 6
Choose Allowed for the Calling Line ID Presentation (Outbound calls section).
Cisco Unified CallManager uses calling line ID presentation as a supplementary
service to control the display of the calling party number on the called party IP
phone screen. Choosing Allowed displays the calling number information.
Step 7
Enter the IP address for the remote cluster first node in the Destination Address field
of the SIP Information section. Refer to the SIP Trunk Information table in this lab
activity to find the correct IP address.
Step 8
Make sure that Destination Address is an SRV box is not selected.
Step 9
Choose Non Secure SIP Trunk Profile from the SIP Trunk Security Profile dropdown menu.
Step 10
Choose Standard SIP Profile from the SIP Profile drop-down menu.
Step 11
Click Save, and then click OK on the pop-up window.
Step 12
Click the Reset button, and then click the Reset button on the new dialog box. Click
the Close button on the new dialog box.
56
The SIP trunk appears in the list when you choose Device > Trunk and then click the Find
button in Cisco Unified CallManager Administration.

Activity Objective
In this activity, you will configure a Cisco IOS H.323 gateway that uses the FXO port on the
Cisco router, then a Cisco IOS MGCP gateway that uses the digital PRI line on the Cisco router
to reach the emulated PSTN and the remote cluster through the PSTN. (Calling will be enabled
after you add a route pattern to the remote cluster in the next lab activity.) After completing this
activity, you will be able to meet these objectives:
Configure an H.323 voice gateway on Cisco Unified CallManager Release 5.0
Configure an IOS H.323 voice gateway
Configure an MGCP voice gateway on Cisco Unified CallManager Release 5.0
Configure an IOS MGCP voice gateway
Visual Objective

Configuring Voice Gateways
Phone2-1
Phone1-1
CCM1-1 CCM2-1
.1
HQ-1
Phone1-2
Phone3-1
.101
HQ-2
HQ-1
Phone3-2
CCM1-2 CCM2-2
MGCP
gw
MGCP
gw
.101
.2
10.1.1.0/24
Phone2-2
.2
.1
10.2.1.0/24
HQ-2
PSTN
BRANCH-1
BR-1
10.1.250.102/32
BR-2
H-323
gw
H-323
gw
Phone4-1
BRANCH-2
10.2.250.102/32
Phone4-2
POD-1
POD-2
CIPT1 v5.0-16
Required Resources
Cisco Unified CallManager Release 5.0
Cisco IOS MGCP gateway (for example Cisco 2600, 2800, 3700, and 3800 Series routers)
with T1/E1 PRI connection to PSTN
Lab Guide
57
H.323 gateway (for example Cisco 2600, 2800, 3700, or 3800 Series routers) with FXO
connection to PSTN
Simulated PSTN
Command List
The tables describe the commands that are used in this activity.
Cisco IOS H.323 Gateway Commands
58
Command
Description
enable

configure terminal
voice class h323 tag
Creates an H.323 voice class that is used to configure a TCP

timeout duration.
h225 timeout tcp

establish seconds
Configures the H.225 TCP timeout duration in seconds.
h323-gateway voip
interface
Enables H.323 gateway functions on the interface.
h323-gateway voip bind

srcaddr [ipaddress]
Sets the source IP address to be used for all signaling and

media terminated on this H.323 gateway.
dial-peer voice tag

pots
Configures a POTS dial peer on the gateway.
destination-pattern
digits
Configures the destination pattern for the configured dial peer.
preference number
Identifies the priority order of the dial peers.
port [x/x/x]
Specifies the voice port that is used with the configured dial peer.
session target
ipv4:ip-address
Specifies the IP address of the Cisco Unified CallManager to

send the call to when the destination pattern is matched.
dial-peer voice tag

voip
Configures a VoIP dial peer on the gateway.
show running-config
Displays the router or switch (if using Cisco IOS software)

current configuration
show dial-peer voice

summary
Display the list of all configured dial peers.
direct-inward-dial
Enables the direct inward dialing (DID) call treatment for an

incoming called number.
no digit-strip
Prevents stripping digits from the called number when a call is

progressed along a POTS dial peer.
Cisco IOS MGCP Gateway Commands

Command
Description
enable

configure terminal
mgcp
Enables the MGCP application on the gateway.
mgcp call-agent [firstnode-ip-address]
Identifies the first node Cisco Unified CallManager for the

MGCP gateway.
mgcp dtmf-relay codec all

mode out-of-band
Relays DTMF digits through the IP network.
direct-inward-dial
Enables the DID call treatment for an incoming called

number.
controller e1
Enters the controller configuration mode.
ccm-manager redundant-host
[backup-ccm-ip-address]
Identifies the secondary or tertiary Cisco Unified

CallManager for the MGCP gateway.
ccm-manager mgcp
Indicates that the Cisco Unified CallManager is using the

MGCP application.
isdn switch-type
Configures the type of ISDN switch connected to the

MGCP gateway PRI line.
isdn bind-l3 ccm-manager
Backhauls the ISDN Q.931 channel to the Cisco Unified

CallManager.
dial-peer voice tag pots
application mgcpapp
Configures the dial peer to use the MGCP application.
port [module-number/slotnumber/port-number]
Specifies the voice port that is used with the configured dial
peer.
show mgcp endpoint
Displays a list of all MGCP endpoints on an MGCP

gateway.
show ccm-manager hosts
Displays the registration status of the MGCP gateway on

its Cisco Unified CallManagers.
Job Aids
MGCP Gateway Information
Cisco IOS MGCP Gateway Configuration
Cluster Name
Call Agent
Redundant Host
Pod 1
10.1.1.1
10.1.1.2
Pod 2
10.2.1.1
10.2.1.2
Lab Guide
59
Cisco Unified CallManager Administration Configuration

Parameter
Data
Domain Name
HQ-x (where x is your port number)
Description
MGCP gateway
Cisco Unified CallManager Group
CMG
Module in Slot 1
NM-HDV2-2PORT-E1 (depends on actual hardware used)
Subunit 1
VWIC-2MFT-E1 (depends on actual hardware used)
Port
1/0
Channel Selection Order
Top Down
H.323 Gateway Information

Cisco IOS H.323 Gateway Configuration
Session Targets
Cluster Name
Preference 1
Preference 1
Pod 1
10.1.1.1
10.1.1.2
Pod 2
10.2.1.1
10.2.1.2
Cisco Unified CallManager Administration, H.323 Gateway Configuration

Parameter
Data
Device Name
10.x.250.102 (where x is your pod number) - IP address of H.323 gateway
Description
H.323 gateway
Device Pool
Default_CMLocal_DP
Calling Party
Presentation
Allowed
Task 1: Configuring Cisco Unified CallManager Release 5.0 for

H.323 Support
In this task, you will configure Cisco Unified CallManager to use the H.323 protocol toward
the H.323 gateway BR-x in your cluster.
Activity Procedure
Step 1
Write the IP address of your H.323 gateway BR-x (where x is your cluster number).
The loopback 0 interface on the gateway should be used for all H.323 functions.
Refer to the H.323 Gateway Information table to obtain the correct IP address.
H.323 gateway IP address: ________________________________________
Step 2
60
In Cisco Unified CallManager Administration, choose Device > Gateway and click
the Add New button.
Step 3
From the Gateway Type drop-down list, choose H.323 Gateway, and then click
Next.
Step 4
In the Device Name field, enter the H.323 gateway IP address recorded earlier.
Step 5
Enter H.323 gateway for the description.
Step 6
Choose Default_CMLocal_DP for the Device Pool.
Step 7
Choose Allowed for the Calling Party Presentation. (The system default is to send
calling line ID information, so by choosing Allowed, you are simply making the
parameter explicit.)
Step 8
Click Save and, on the pop-up window shown, click OK.
Step 9
Reset the gateway.
The H.323 gateway appears in the list when you choose Device > Gateway and click Find.
Task 2: Configuring Cisco IOS Voice Gateway to Support H.323

In this task, you will configure the Cisco IOS router to become the H.323 gateway in your
cluster.
Activity Procedure
Step 1
Connect to your BR-x router and log in. This router will be your H.323 gateway.
Step 2
Configure the H.225 TCP timeout interval by using these commands:

voice class h323 1
h225 timeout tcp establish 5
exit
Step 3
Enable the H.323 gateway functionality (signaling and media) on the router
loopback 0 interface. Enter the commands in this sequence:
interface loopback 0
h323-gateway voip interface
h323-gateway voip bind srcaddr <loopback0 IP address>
exit
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Step 4
Configure the VoIP dial peer, which allows the calling from the H.323 gateway
(actually from the PSTN) to IP phones in your cluster using the first node. The
H.323 gateway should forward to the first node all calls with these dialed numbers,
without any manipulation of digits (x is your pod number and N is any number
placeholder):
Calling to the HQ IP phones in your cluster 52x-555-2NNN
Calling to the BR IP phones in your cluster 53x-555-3NNN
Enter this command sequence:

dial-peer voice y000 voip (y is the IP host address of the
first node)
destination-pattern 5[2,3]x555[2,3]... (x is your pod number)
session target ipv4:<First node IP address>
preference y (y is the IP host address of the first node)
voice-class h323 1
exit
Step 5
Configure another VoIP dial peer with lower preference, based on the previous step,
which allows the calling between the H.323 gateway (actually PSTN) and IP phones
in your cluster using the subsequent node as a backup. When entering the Cisco IOS
commands according to the previous step, be aware of the different IP host address
that the subsequent node uses.
Step 6
Configure the POTS dial peers on the H.323 gateway that allow for calling of these
numbers and number formats in the PSTN through the gateway FXO port. For
outbound PSTN call, no digits should be stripped.
Step 7
Emergency 911
Services [2-8]11
Local numbers [2-9]NNNNNN (N is any number placeholder)
National [2-9]NN[2-9]NNNNNN (N is any number placeholder)
Long distance numbers 1[2-9]NN[2-9]NNNNNN (N is any number

placeholder)
International numbers 011<one or more digits>
Toll-free numbers 1[800,866,877,888]NNNNNNN
Premium numbers 1[900,976]NNNNNNN
Enter these commands to shut down and then enable the voice ports that you have
just configured:
voice-port module/slot/port (! gateway FXO to PSTN)
shutdown
no shutdown
end
Step 8
62
Save your configuration made on the H.323 gateway.
Using the show dial-peer voice summary command on the H.323 gateway, you can see all
requested VOIP and POTS dial peers.
You are not able to dial through the H.323 gateway until you configure a route pattern to
the H.323 gateway in the next lab activities.
Task 3: Configuring Cisco Unified CallManager Release 5.0 for

MGCP Support
In this task, you will configure Cisco Unified CallManager to use the MGCP protocol toward
the MGCP gateway HQ-x in your cluster.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Device > Gateway and, on

the new page shown, click the Add New button.
Step 2
From the Gateway Type drop-down menu, choose the gateway platform (for
example Cisco 26XX) that is used for Cisco IOS MGCP gateway HQ-x (x is your
cluster number). Then click Next.
Step 3
Enter the MGCP gateway name (HQ-x, where x is your pod number) for the Domain
Name (the name is case-sensitive). Enter MGCP gateway in the Description field.
Step 4
Choose CMG for the Cisco Unified CallManager Group.
Caution
These steps are platform dependent. Cross-check with your instructor to find out the actual
hardware that is used for the MGCP gateway. This lab guide is based on Cisco 2600 series
router platform with E1 PRI 1/0.
Step 5
In the Module in Slot 1 drop-down menu, choose NM-HDV2-2PORT-E1 for the

type of the network module (network module with two E1 ports). Then click Save.
Step 6
From the subunit 0, select VWIC-2MFT-E1. Click Save. When the pop-up
window appears, click OK.
Step 7
Click the port icon 1/0 (far left endpoint with the question mark) to open the
Gateway Configuration page.
Step 8
Step 9
Choose Top Down for the Channel Selection Order.
Step 10
Click Save and, when the pop-up window appears, click OK.
The MGCP gateway appears in the list when you choose Device > Gateway and click
Find.
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Task 4: Configuring Cisco IOS Voice Gateway to Support

MGCP
In this task, you will enable MGCP on a Cisco IOS MGCP gateway.
Activity Procedure
Step 1
Connect to your MGCP gateway HQ-x.
Step 2
To configure the primary Call Agent information on the MGCP gateway, enter this
sequence of commands in the global configuration mode:
mgcp
mgcp call-agent <First node IP address> service-type mgcp
version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
Step 3
Continue with the enabling the MGCP gateway to support Cisco Unified
CallManager. Enter this command sequence:
ccm-manager mgcp
Step 4
Configure Cisco Unified CallManager redundancy in the MGCP gateway. Enter this
command in configuration mode:
ccm-manager redundant-host <Subsequent node IP address>
Step 5
Configure the ISDN switch type to Primary NET5. Enter this command in the global
configuration mode:
isdn switch-type primary-net5
Step 6
Configure E1 PRI controller 1/0, connecting your cluster to the simulated PSTN
network, to use MGCP service for all time slots. Enter this sequence of commands:
controller e1 1/0
pri-group timeslots 1-31 service mgcp
no shutdown
exit
Note
Step 7
At this point, the message voice_port is not initialized is shown. This does not mean any
error. The voice port will initialize when the MGCP gateway configuration is completed.
Configure the channel backhauling. Bind the ISDN Q.931 layer (layer 3) to the
Cisco Unified CallManager. Enter this sequence of commands:
interface Serial1/0:15
isdn bind-l3 ccm-manager
exit
64
Step 8
Configure the POTS dial peer associated to the E1 PRI port, so that it starts using the
MGCP application. On this dial peer, also enable the DID call treatment for an
incoming called number. Enter this sequence of commands in the configuration
mode.
dial-peer voice 10 pots
application mgcpapp
port 1/0:15
direct-inward-dial
end
Step 9
Save your configuration made on the MGCP gateway.
You can verify that your MGCP gateway has successfully registered to the Cisco Unified
CallManager by using the command show ccm-manager hosts (Status shows Registered)
and using the show mgcp endpoint command to check that the controlled E1 PRI endpoint
ports are up.
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Activity Objective
In this activity, you will build basic route plans to place calls over the IP WAN and the PSTN.
Create a route pattern to the H.323 gateway to enable calls to be placed to the simulated
PSTN using an analog line and the remote cluster over the simulated PSTN
Create a route pattern that points to the digital gateway (IOS MGCP) to enable dialing to
the simulated PSTN and the remote cluster over the simulated PSTN
Configure a route plan that includes a route group, route list, and route pattern that enables
calls to the PSTN and remote cluster over the digital and analog line
Add intercluster and SIP trunks to a route group, add that route group to a route list, and
build a route pattern that is directed at the route list to allow calls to another cluster via the
IP WAN
Visual Objective

Configuring Basic Dial Plan Elements
Phone1-1
Phone2-1
DN: 2001
DN: 2002
Phone3-1
DN: 2003
8522.2xxx (ICT, SIP)

Phone1-2
Phone2-2
DN: 2001
DN: 2002
521-555-2000
to 2199
CCM1-2
Prio 1
Prio 1
To
PSTN
Pr
io
BRANCH-1
531-555-3000
to 3199
DN: 2003
522-555-2000
to 2199
CCM1-1
HQ-1
Phone3-2
HQ-2
HQ-1
To
PSTN
PSTN
BR-1
BR-2
io
Pr
BRANCH-2
Phone4-1
Phone4-2
DN: 3004
DN: 3004
POD-1
HQ-2
532-555-3000
to 3199
POD-2
CIPT1 v5.0-17
MGCP and H.323 gateways were already configured. In this activity, you create a route plan
that allows for calling between the Cisco Unified CallManager clusters and also implements
redundancy for PSTN calls for the case when one of gateways is not operational.
66
Required Resources
Two Cisco Unified CallManager clusters with IP phones
MGCP gateway
H.323 gateway
Simulated PSTN
IP WAN
Command List
Job Aids
Trunk Route Patterns
Cluster Name
Remote HQ Pattern
Remote Branch Pattern
Route List
Pod 1
8522.2XXX
8532.3XXX
TRK_RL
Pod 2
8521.2XXX
8531.3XXX
TRK_RL
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Task 1: Configuring a Route Pattern to the H.323 Gateway

This is the simplest route plan to configure. You are going to create a route pattern for
emergency services through the H.323 gateway that is already added in the Cisco Unified
CallManager database.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt

> Route Pattern, and then click Add New to open the Route Pattern Configuration
window.
Step 2
In the Route Pattern field, enter the 911 route pattern.
Step 3
Choose the H.323 gateway IP address from the Gateway/Route List drop-down
menu (either 10.x.250.102, where x is your pod number).
Step 4
Uncheck the Provide Outside Dial Tone check box.
Step 5
Click Save. You are notified by a pop-up window that the authorization code will
not be activated. Click OK.
Step 6
Then you are notified by another pop-up window that changes will reset the
gateway. Click OK.
From a Cisco IP phone in your pod, you can successfully reach the emergency number
(part of the simulated PSTN).
Task 2: Configuring a Route Pattern to the IOS MGCP Gateway

In this task, you will reconfigure the emergency route pattern 911 to point to the digital
gateway (IOS MGCP) that is already added in the Cisco Unified CallManager database, instead
of the H.323 gateway configured in the previous task.
Activity Procedure
68
Step 1

> Route Pattern, and then click Find.
Step 2
In the Search Results list, click the 911 route pattern to open the Route Pattern
Configuration window.
Step 3
In the Gateway/Route List drop-down menu, choose the IOS MGCP gateway,
instead of H.323 gateway. The name should look similar to this: S0/DS1-0@HQ-x
(where x is your pod number).
Step 4
Verify that the Provide Outside Dial Tone check box is unchecked.
Step 5
Click Save. You are notified by a pop-up window that the authorization code will
not be activated. Click OK.
Step 6
gateway. Click OK.
Task 3: Configuring a Route Plan that Includes a Route Group,

Route List, and Route Pattern Using the H.323 and MGCP
Gateways
In this task, you will configure routing to the PSTN through both gateways, where the MGCP
gateway will be used with higher priority than the H.323 gateway. This functionality will be
achieved by configuring route groups, route lists, and route patterns in Cisco Unified
CallManager Administration.
Activity Procedure
Pre-Lab Task: Remove the Route Pattern Pointing to the MGCP Gateway
You cannot have both a route pattern pointing directly to a gateway and a route group that
points to that gateway, so you will remove the route pattern that is pointing to the MGCP
gateway.
Step 1

> Route Pattern and click Find.
Step 2
Check the check box next to the route pattern 911 and click Delete Selected.
Add a New Route Group

Step 3
> Route Group. Click Add New to open the Route Group Configuration window.
Step 4
In the Route Group Name field, enter MGCP_ H.323_RG.
Step 5
Choose Top Down for the Distribution Algorithm.
Step 6
From the Available Devices section, highlight the MGCP gateway (S1/DS1-0@HQx), and then click Add to Route Group. The MGCP gateway will appear in the
Selected Devices section.
Step 7
In the Available Devices section, choose the H.323 gateway (IP address
10.x.250.102, where x is your pod number) and click Add to Route Group.
Step 8
Make sure that the MGCP gateway is at the top, having the highest priority. If not,
use the up and down arrows to move it to the top of the list.
Step 9
Click Save.
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69
Configure a Route List

Step 10
> Route List. Click Add New to open the Route List Configuration window.
Step 11
In the name field, enter MGCP_ H.323_RL.
Step 12
Enter PSTN via MGCP and H323 for the Description.
Step 13
Choose CMG for the Cisco Unified CallManager Group.
Step 14
Click Save.
Step 15
When the new page appears, click Add Route Group to open the Route List Detail
Step 16
In the Route Group drop-down menu, choose MGCP_ H.323_RG-[NON-QSIG].
Step 17
Leave the remaining Route List Details Configuration window parameters as they
are. This window is used for transformation information.
Step 18
Click Save and click OK on a pop-up window.
Step 19
Click Reset and Reset again on the dialog window. Then click Close.
Add a Route Pattern

Step 20

> Route Pattern. Click Add New to open the Route Pattern Configuration window.
Step 21
In the Route Pattern field, enter the route pattern 911 for emergency services.
Step 22
Choose MGCP_ H.323_RL from the Gateway/Route List drop-down list.
Step 23
Step 24
Click Save. You are notified by a pop-up window that authorization code will not be
activated. Click OK.
Step 25
gateway. Click OK.
Step 26
Repeat the previous steps to enter the route pattern for international calling and point
this route pattern to the MGCP_H.323_RL. The international numbers are variable
size, and they start with the 011 prefix followed by any digits (011). Disable the
providing of outside dial tone.
70
From a Cisco IP phone in your pod, you can successfully reach international numbers, for
instance 011 44 1632 4960 (part of the simulated PSTN). When dialing, you must either
wait for the interdigit timeout to expire or, at the end of the number dialed, press the
number terminator (#).
Task 4: Configuring a Route Plan of Route Group, Route List,

and Route Pattern Using the Intercluster Trunks
In this task, you will add intercluster and SIP trunks to a route group, then add that route group
to a route list, and finally, build a route pattern that is directed at the route list to allow calls to
another cluster (pod) via IP WAN.
Activity Procedure
Step 1

> Route Group, and then click Add New to open the Route Group Configuration
window.
Step 2
Enter TRK_RG in the Route Group Name field.
Step 3
From the Available devices section, highlight the intercluster trunk IC_TRK to the
remote-cluster Cisco Unified CallManager nodes, and then click Add to Route
Group.
Step 4
Highlight the SIP trunk SIP_TRK to the remote-cluster Cisco Unified CallManager
nodes and click Add to Route Group.
Step 5
Make sure that the IC_TRK is at the top of the list. If not, move it to the top using
the up arrow.
Step 6
Click Save.
Configure a Route List

Step 7
Choose Call Routing > Route/Hunt > Route List.
Step 8
Click Add New to open the Route List Configuration window.
Step 9
Enter TRK_RL for the route list name, and enter Trunks to remote cluster for the
description.
Step 10
Choose CMG for the Cisco Unified CallManager group.
Step 11
Click Save.
Step 12
Click Add Route Group to open the Route List Details Configuration window.
Step 13
Choose TRK_RG-[NON-QSIG] for the route group and click Save and OK.
Step 14
Leave the remaining Route Details Configuration window parameters as they are.
This window is used for transformation information.
Step 15
Click Save and reset the route list.
Add a Route Pattern

Step 16

> Route Pattern.
Step 17
Click Add New to open the Route Pattern Configuration window.
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71
Step 18
In the Route Pattern field, enter the route pattern to reach the remote cluster HQ IP
phones. The pattern consists of a prefix followed by the directory number as seen in
the Remote HQ pattern column of the Trunk Route Patterns table in the Job
Aids section.
Step 19
Choose TRK_RL from the Gateway/Route List drop-down menu.
Step 20
Step 21
From the Discard Digits drop-down menu, choose PreDot. This discards the prefix
digits seen before the dot and sends only the directory number to the remote cluster.
Step 22
Click Save, then OK and OK again.
Step 23
Add a route pattern to reach the remote cluster Branch IP phone, based on the
pattern information in the Remote Branch Pattern column of the Trunk Route
Patterns table in the Job Aids section. Discard the prefix digits (852y and 853y,
where y is the remote pod number) PreDot and keep only the directory number.
72
From an IP phone in your pod, you can dial an IP phone in another pod.
Lab 4-4: Configuring Hunt Groups and Call

Coverage
Activity Objective
In this task, you will configure hunt groups that consist of line groups, hunt lists, and a hunt
pilot number with internal and external forwarding support specifications for busy, no-answer,
and no-call-coverage conditions. First, you will create two line groups and add members to the
line groups. Next, you will create a hunt list and add the line groups. Next, you will create a
hunt pilot number and associate it with the hunt list. After completing this activity, you will be
able to meet these objectives:
Create a line group and add members
Create a hunt list and add line groups
Create a hunt pilot number and associate it with a hunt list
Test the call distribution
Configure final forwarding on the hunt pilot number for busy and no-answer conditions
Visual Objective

Hunt Groups and Call Coverage
Phone1-1
Phone2-1
DN: 2001
DN: 2002
Phone3-1

1stLG
Phone1-2
Phone2-2
DN: 2001
DN: 2002
1stLG
Phone3-2
522-555-2000
to 2199
521-555-2000
to 2199
CCM1-2
CCM1-1
HQ-2
HQ-1
HQ-1
1stHL
Hunt Pilot: 2111
BRANCH-1
BR-1
HQ-2
1stHL
PSTN
Hunt Pilot: 2111
BR-2
531-555-3000
to 3199
532-555-3000
to 3199
Phone4-1
Phone4-2
DN: 3004
DN: 3004
2ndLG
2ndLG
POD-1
BRANCH-2
POD-2
CIPT1 v5.0-18
Lab Guide
73

Hunt Groups and Call Coverage (Cont.)
Internal call
External call
Hunt Pilot
x111
Hunt List
1stHL
Line Groups
1stLG
2002
2001
RNAR: 5 seconds
Try Next Member; Then,
Try Next Group in Hunt List
2003
2ndLG
3004
CIPT1 v5.0-19
Required Resources
Two Cisco Unified CallManager clusters
Three Cisco IP phones with LCD display
One third-party SIP phone
IP WAN
Command List
Job Aids
74
Task 1: Creating a Line Group and Adding Members

In this task, you will create two line groups and add members to the new line groups. You will
create a 1stLG line group and add directory numbers 2001 and 2002 to it and a 2ndLG line
group with directory number 3004 assigned to it.
Activity Procedure
Step 1
Choose Call Routing > Route/Hunt > Line Group.
Step 2
Click Add New.
Step 3
Enter 1stLG in the Line Group Name field.
Step 4
For now, leave the distribution algorithm at the default (longest idle time). Leave the
hunt options for busy, no answer, and not available at their default values (Try next
member, then, try next group in Hunt List).
Step 5
Change the RNA Reversion (RNAR) Timeout from its default value of 10 seconds
to 5 seconds. This value might not be appropriate in a call center environment,
however, in a classroom or test environment, a shorter timeout enables you to more
quickly validate call-distribution behavior.
Add Members to the Line Group

Step 6
Add the 2001 and 2002 directory numbers to the line group. In the Available
DN/Route Partition pane, choose 2001 and click Add to Line Group to move it to
the Selected DN/Route Partition pane. Repeat this step to add 2002 to this line
group.
Step 7
The order of the directory numbers in the Selected DN/Route Partition pane
determines the order in which the directory numbers are accessed in this line group.
Change the order of the line group members so that 2002 is the first member and
2001 is the second member. To change the order, click a directory number and use
the up and down arrows to the right of the pane to change the order of directory
numbers, or choose Reverse Order of Selected DNs.
Step 8
Click Save to add the new directory numbers to the line group.
Step 9
Repeat steps 1 through 8 to create a second line group named 2ndLG with the
third directory number in your pod, 3004, as its only member.
Two line groups have been added to Cisco Unified CallManager with their respective
members. You can see these two line groups if you choose Call Routing > Route/Hunt >
Line Group and click Find.
Lab Guide
75
Task 2: Creating a Hunt List and Adding Line Groups

In this task, you will create a hunt list called 1stHL and add the 1stLG and 2ndLG line groups
to it.
Activity Procedure
Step 1
Choose Call Routing > Route/Hunt > Hunt List.
Step 2
Click Add New.
Step 3
In the Hunt List Name field, enter the name 1stHL.
Step 4
From the drop-down list, choose CMG for Cisco Unified CallManager group.
Step 5
To add this hunt list, click Save. The Hunt List Configuration window displays the
newly added hunt list.
Step 6
Add both line groups to the new hunt list. To add a line group, click Add Line
Group. The Hunt List Detail Configuration window appears.
Step 7
From the Line Group drop-down list, choose the 1stLG line group to add to the hunt
list.
Step 8
To add the line group, click Save. Then click OK on the pop-up window.
The line group name appears in the Hunt List Configuration window in the Selected
Groups pane.
Step 9
Click Add Line Group and repeat last two steps to add the remaining line group,
2ndLG, to the hunt list.
Step 10
When you finishing adding line groups to the hunt list, click Save on the Hunt List
Configuration window. Then click OK on the pop-up that reminds you to reset the
hunt list.
Step 11
Click Reset to reset the hunt list. When the dialog window appears, click Reset and
click Close.
Note
Cisco Unified CallManager accesses line groups in the order in which they appear in the
hunt list. You can change the access order of line groups, if needed, by choosing a line
group from the Selected Groups list and clicking the up or down arrow on the right side of
the pane to move the line group up or down in the list.
76
One hunt list that contains two line groups has been added to Cisco Unified CallManager.
You can verify that if you choose Call Routing > Route/Hunt > Hunt List and then click
Find.
Task 3: Creating a Hunt Pilot Number and Associating It with a

Hunt List
In this task, you will create a hunt pilot number, 2111, and assign it to the 1stHL hunt list.
Activity Procedure
Complete these steps
Step 1
Choose Call Routing > Route/Hunt > Hunt Pilot.
Step 2
Click Add New.
Step 3
In the Hunt Pilot number field, enter 2111.
Step 4
Assign the hunt pilot to the 1stHL hunt list using the Hunt List field drop-down
menu.
Step 5
Click Save.
You created a hunt pilot number 2111 and assigned it to the hunt list. You can verify that if
you choose Call Routing > Route/Hunt > Hunt Pilot and then click Find.
Task 4: Testing Call Distribution

In this task, you will test and validate the call-distribution behavior to ensure that it performs as
you expect.
Activity Procedure
Step 1
From the third-party SIP phone with the directory number 2003, call the hunt pilot
number 2111 and observe the call distribution behavior. The call should first ring on
2002. Answer the call when it rings on 2002.
Directory number 2001 should now be the member with the longest idle time and
should be the telephone that rings first when a call comes in to the hunt pilot
number.
Step 2
Call 2111 again and validate this hypothesis, but this time, do not answer the call.
Observe the call hunt to 2001 after 2002 does not answer for 5 seconds (RNAR
timeout). When 2001 does not answer, the call will then hunt to 3004 in the second
line group (2ndLG) because the hunt option specified is Try next member, then,
try next group in Hunt List.
Step 3
To test call-distribution behavior for busy conditions in a classroom or lab

environment, from the Directory Number Configuration window, set the maximum
number of calls for the line to 1 and the busy trigger to 1 for each directory number.
This action disables call waiting, allowing a directory number to receive only one
call at a time. Any additional calls receive a busy signal. (To reach the Directory
Number Configuration window, choose Device > Phone, click Find, click the
device name in the list, click the line, and scroll down to Multiple Call/Call Waiting
Lab Guide
77
Settings. Change the value of the Maximum Number of Calls and Busy Trigger
fields to 1. Click Save.)
Step 4
Remove the handsets on 2001 and 2002, making both telephones appear to be busy.
What do you expect will happen when you call the hunt pilot? Test your hypothesis.
Step 5
Hang up telephones 2001 and 2002. Spend a few moments experimenting with other
line group distribution algorithms (Circular, Broadcast, or Top-Down) and other
hunt options (Stop Hunting; Skip Remaining Members, and Go Directly to Next
Group; and Try Next Member, but Do Not Go to the Next Group).
Step 6
For the next task, set the call-distribution algorithm on the first line group to Top
Down and hunt options to Try next member, then, try next group in Hunt List.
Step 7
Refer to step 3 and change the Maximum Number of Calls and Busy Trigger fields
back to their default values (4 and 2, respectively).
Calls to the hunt pilot or to a specific directory number in the hunt group will hunt and
achieve call distribution according to the behavior outlined for each lab task.
Task 5: Configuring Final Forwarding for Busy and No-Answer

Conditions
In this task, you will configure final forwarding on the hunt pilot number so that a call
forwarded to the hunt pilot that is not answered (because of no-answer or busy conditions) is
forwarded to a destination directory number (presumably, in a production network, to a
designated number in the hunt group or an autoattendant that is always available as backup to
the hunt group).
Activity Procedure
78
Step 1
Choose Call Routing > Route/Hunt > Hunt Pilot, and then click Find.
Step 2
Choose 2111 to go to the Hunt Pilot Configuration window.
Step 3
In the Hunt Forward Settings section, configure these final forwarding settings:
Forward Hunt No Answer Destination: 2001
Forward Hunt Busy Destination: 2001
Maximum Hunt Timer: 12 seconds
Step 4
Click Save.
Step 5
Call the hunt pilot number 2111 and do not answer the call. What do you expect will
be the call-distribution and final forwarding behavior? Test your hypothesis and
write it in the space provided here.
Step 6
Go off hook on all three Cisco IP phones. From the third-party SIP phone, call the
hunt pilot number 2111. What do you expect will be the call-distribution and final
forwarding behavior? Test your hypothesis and write it in the space provided here.
Step 7
Hang up all three Cisco IP phones.
Calls to the hunt pilot receive final forwarding to a directory number where the call will be
answered by an individual, autoattendant, or voice mail (in a production network).
Cleanup
To prepare for future labs, remove the hunt groups, line groups, and hunt pilot numbers. The
order in which you delete these items must be the reverse of the order in which you configured
them.
Step 1
Delete the hunt pilot number (choose Call Routing > Route/Hunt > Hunt Pilot).
Step 2
Delete the hunt list (choose Call Routing > Route/Hunt > Hunt List).
Step 3
Delete the line groups (choose Call Routing > Route/Hunt > Line Group).
Step 4
On the Directory Number Configuration page for the directory number 2001,
uncheck all checked boxes in the Call Forward and Call Pickup Settings sections.
Lab Guide
79

Activity Objective
In this activity, you will be able to apply complex route-plan concepts to simple route plans.
Configure translation patterns to handle all directory numbers within the DID range of the
cluster
Configure an external telephone number mask on Cisco IP phones and on the MGCP FXS
port, and apply calling transformation settings to change the calling party directory number
Modify route patterns by adding an access code and discard digits instructions
Configure a route pattern that has a route filter to block 900 numbers
Visual Objective

Configuring Complex Dial Plan Elements
Phone1-1
Phone2-1
Phone1-2
Phone3-1
Attendant
2XXX=2001
Phone2-2
Phone3-2
Attendant
2XXX=2001
521-555-2000
to 2199
522-555-2000
to 2199
Blocked calls to
9 1 900 555 XXXX
CCM1-1
CCM1-2
Access Code
9.@
Access Code
9.@
Enabled calls to
9 1 5YZ 555 XXXX
HQ-1
HQ-1
BRANCH-1
External Mask
521555XXXX
531555XXXX
PSTN
BR-1
HQ-2
External Mask
522555XXXX
532555XXXX
BR-2
Phone4-1
HQ-2
BRANCH-2
Phone4-2
532-555-3000
to 3199
531-555-3000
to 3199
POD-1
POD-2
CIPT1 v5.0-20
Required Resources
80
Gateway devices (Cisco IOS MGCP [PRI], H.323 [FXO])
Simulated PSTN
IP WAN
Command List
Command
Description
enable
Enables you to enter privileged EXEC mode, or any other security

level that is set by a system administrator.
dial-peer voice tag

pots
destination-pattern
digits
Configures the destination pattern for the configured dial peer.
no digit-strip
When a called number is received and matched to a POTS dial

peer, the default router behavior is to strip matched digits and
forward the remaining digits to the voice interface. The no form of
this command disables digit stripping.
port [x/x/x]
Specifies the voice port that is used with the configured dial peer.
Job Aids
Transformation Masks
Cluster Name
IP Phones
External Phone Number Mask
Pod 1 HQ
Phone1-1, Phone2-1, Phone3-1
521555XXXX
Pod 1 BR
Phone4-1
531555XXXX
Pod 2 HQ
Phone1-2, Phone2-2, Phone3-2
522555XXXX
Pod 2 BR
Phone4-2
532555XXXX
Lab Guide
81
Task 1: Configuring a Translation Pattern

To ensure that a company does not miss dialed calls, in this task, you will configure a
translation pattern to send all calls with the given DID range to a known directory number,
typically an attendant.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Translation

Pattern.
Step 2
Click Add New to open the Translation Pattern Configuration window.
Step 3
Enter 2XXX for the translation pattern.
Step 4
Step 5
Enter 2001 for the Called Party (not calling party) Transform Mask.
Step 6
Click Save.
From an IP phone in your cluster, other than directory number 2001, you can call an
unassigned directory number in your cluster (starting with 2); it should ring the 2001
directory number that you configured as the attendant.
Check with another cluster to see whether it has configured its translation pattern, and then
call an unassigned directory number in that cluster; it should ring the directory number
2001 in the remote cluster.
Task 2: Configuring Transformation Masks

In this task, you will configure an external phone number mask on Cisco IP phones and apply
calling transformation settings to change the calling party directory number.
Activity Procedure
82
Step 1
In Cisco Unified CallManager Administration, choose Device > Phone, and click
Find to list the IP phones in the cluster.
Step 2
Choose the Phone1-x (where x is your pod number) from the list to access the Phone
Step 3
From the Phone Configuration window for the selected IP phone, click Line [1] to
open the Directory Number Configuration window.
Step 4
Scroll down to the Line 1 on Device SEP section and, in the External Phone
Number Mask field, enter the mask that applies for the selected IP phone, as shown
in the Transformation Masks table in the Job Aids section.
Step 5
Click Save.
Step 6
Repeat Steps 1 though 5 for the other IP phones in your cluster.
Your IP phones have updated their displays to show the external phone number in the
upper-right corner.
Task 3: Configuring Route Patterns to Use Access Codes and

Discard Digits Instructions
In this task, you will configure the route patterns to use an outside line access code and apply
the DDI to send the correct expected digits to the next call-processing agent. You will also
configure translation patterns that translate full PSTN numbers into the directory numbers for
inbound call processing.
Activity Procedure
Configure PSTN Route Pattern with Access Code
Step 1
> Route Pattern and click Add New.
Step 2
In the Route Pattern field, enter these three characters: 9.@. Here, 9 is the access
code and @ covers all NANP numbers.
Step 3
From the Number Plan drop-down menu, choose NANP.
Step 4
From the Gateway/Route List drop-down menu, choose MGCP_H.323_RL.
Step 5
Step 6
Check the box Use Calling Partys External Phone Number Mask. This will
present the mask configured earlier when calling through the PSTN.
Step 7
Choose Allowed from the Calling Line ID Presentation drop-down menu.
Step 8
From the Discard Digits drop-down menu, choose PreDot to remove the access
code that is separated with . (dot) from the remaining PSTN digits.
Step 9
Click Save. Then click OK on the pop-up window and one more OK on the window
notifying you about the automated resetting.
Configure Translation Patterns for Incoming Calls

Step 10
In Cisco Unified CallManager Administration, choose Call Routing > Translation
Pattern and click Add New. This translation pattern should translate called numbers
for calls coming from the other pod to your HQ IP phones via the PSTN, from full
PSTN to the directory number only.
Step 11
In the Translation Pattern field, enter 52y5552XXX (where y is your pod number).
Step 12
Make sure that the Provide Outside Dial Tone box is not checked.
Lab Guide
83
Step 13
In the Called Party Transform Mask field, enter 2XXX.
Step 14
Click Save.
Step 15
Repeat the last five steps to enter the translation pattern for a call coming from the
other pod to your BR IP phone (Phone4-x) via the PSTN. That would be the
translation pattern translating from 53y5553XXX (where y is your pod number) to
3XXX.
Ask the other pod if their configuration is completed. From an IP phone in your cluster, dial
the complete PSTN number (11 long-distance digits) of an IP phone in the remote cluster
preceded by the access code 9, to route a call via the PSTN (for example, 9-1522-5552001). At the receiving IP phone, you should see the external phone number mask applied
to the directory number of the calling IP phone (this does not work over a PSTN analog
line).
Verify also that the External Phone Number Mask is not applied to internal calls by placing
test calls between IP phones within your cluster. In this case, you should see the original
directory number of the calling IP phone, not the masked one.
Task 4: Configuring Route Filters to Apply to Route Patterns

In this task, you will block dialing to particular area codes using a route filter that is applied to
a route pattern.
Activity Procedure
Create an Additional Route Group and Route List
You are going to create an additional route group and route list to reach the IP phones of the
remote pod through the PSTN network (using your MGCP gateway only).
84
Step 1
Choose Call Routing > Route/Hunt > Route Group and click Add New to open
the Route Group Configuration page.
Step 2
In the Route Group Name field, enter MGCP_RG.
Step 3
From the devices in the Available Devices pane, select S1/D1-0@HQ-x and click
the Add to Route Group button.
Step 4
Click the Save button.
Step 5
Choose Call Routing > Route/Hunt > Route List and click the Add New button to
open the Route List Configuration page.
Step 6
In the Route List Name field, enter MGCP_RL and, in the Description field, enter
PSTN Route.
Step 7
Choose CMG from the Cisco CallManager Group drop-down menu. Click Save.
Step 8
Click the Add Route Group button to open the Route List Detail Configuration
page.
Step 9
Choose MGCP_RG[NON-QSIG] from the Route Group drop-down menu.
Step 10
Click Save and, when the pop-up window is shown, click OK.
Step 11
Reset the new MGCP_RL.
Create the Route Filter

You are going to create the Route Filer to match the 900 calls.
Step 12
Choose Call Routing > Route Filter, and click Add New to access the Route Filter
Configuration page.
Step 13
Click Next to confirm the NANP.
Step 14
Enter Match 900 Calls for the Route Filter Name.
Step 15
For the AREA-CODE, choose == from its drop-down menu and then, in the new
entry field shown, enter 900. Click Save.
Configure Route Patterns

You are going to modify the existing route pattern and create a new route pattern to reach the
remote cluster IP phones. The existing route pattern will use the new MGCP_RL, while the
new one will block the 900 calls using the existing TRK_RL.
Step 16
Choose Call Routing > Route/Hunt > Route Pattern, and then click Find.
Step 17
From the list shown, click the existing 9.@ route pattern to open its configuration
window.
Step 18
Connect the route pattern to the MGCP_RL instead of the route list it currently uses
(MGCP_H.323_RL).
Step 19
Make sure that the check box Provide Outside Dial Tone is not checked.
Step 20
Click Save and click OK on two pop-up windows.
Step 21
Click Add New to add a new pattern.
Step 22
Enter 9.@ in the Route Pattern field.
Step 23
Enter 900 Calls for the Description.
Step 24
Choose NANP for the Numbering Plan.
Step 25
From the Route Filter drop-down menu, choose the Match 900 Calls filter.
Step 26
Choose TRK_RL from the Gateway/Route List drop-down menu.
Step 27
Uncheck the check box next to Provide Outside Dial Tone.
Step 28
Enter the XXXX in the Called Party Transform Mask field.
Step 29
Lab Guide
85
When calling between local and remote cluster IP phones, the call is primarily routed either
through the WAN (TRK_RL), or through the PSTN (MGCP_RL) according to the area
code used. Follow this verification procedure:
Step 30
From an HQ IP phone within your cluster, dial an NANP-based telephone number

with some area code different than 900 (for example, 91-52y-555-2001, where y is
the number of the remote pod).
Step 31
You should be connected to the remote cluster IP phone via the PSTN (your MGCP
gateway). This can be verified at the remote IP phone display, where you should see
the calling phone number transformed to the entire PSTN telephone number.
Step 32
Dial a telephone number with the 900 area code (for example, 91-900-555-2001)
from an IP phone within your cluster.
Step 33
You should be connected to the remote cluster IP phone via the Intercluster Trunk.
This can be verified at the remote IP phone display, where you should see the fourdigit directory number of the calling IP phone.
Block 900 Calls

So far, the 900 calls have been routed using the intercluster trunks to the remote cluster. In this
section, you are going to configure the blocking of the 900 calls.
Step 34
Choose Call Routing > Route/Hunt > Route Pattern.
Step 35
From the list shown, click 9.@, which has the route filter applied.
Step 36
Change the Route Option to select Block this pattern instead of the current Route
this pattern radio button.
Step 37
From the drop-down menu next to Block this pattern, choose Call Rejected.
Step 38
You have successfully completed this task when you attain these results:
86
If you dial the telephone number with the 900 area code now (for example, 91-900-52y2001, where y is the number of remote pod), your call should be rejected by the local Cisco
Unified CallManager. You should hear the reorder dial tone.
However, the call to the same remote directory number with an area code different than 900
(routed via PSTN), should still get through.
Cleanup
Complete these tasks to prepare for the next lab activities:
Step 1

> Route Pattern and click Find. Delete the route pattern 9.@ that has the route
filter applied.
Step 2
Click the only route pattern 9.@ that is left, and connect it to the original
MGCP_H.323_RL. Click Save and click OK twice.
Lab Guide
87
Lab 4-6: Implementing Calling Privileges and

Restrictions
Activity Objective
In this activity, you will configure partitions, assign them to calling search spaces, and then
assign the partitions and calling search spaces to directory numbers, route patterns, and
gateways to enable the calling privileges. After completing this activity, you will be able to
meet these objectives:
Configure partitions in Cisco Unified CallManager Administration
Configure calling search spaces in Cisco Unified CallManager Administration
Assign partitions and calling search spaces to configure a telephony CoS
Assign partitions and calling search spaces to route patterns and gateways
Using calling search spaces, partitions, and a translation pattern, configure a telephone to
use the PLAR feature
Visual Objective
Visual Objective for Lab 4-6: Implementing

Calling Privileges and Restrictions
Phone2-1
Phone1-1
Lobby_PT
2001
Phone1-2
Phone3-1
Executive_PT
2002
Executive_PT
2003
Lobby_PT
2001
Phone2-2
Executive_PT
2002
PLAR
Executive_PT
2003
PLAR
521-555-2000
to 2199
CCM1-1
HQ-1
522-555-2000
to 2199
CCM1-2
TRK_PT
8521.2XXX and
8531.3XXX
TRK_PT
8522.2XXX and
8532.3XXX
HQ-1
Phone3-2
PSTN_PT
9.@
PSTN_PT
9.@
HQ-2
PSTN_PT
9.@
BR-2
HQ-2
PSTN
BRANCH-1
BR-1
PSTN_PT
9.@
Phone4-1
531-555-3000
to 3199
Phone4-2
Employee_PT
3004
Employee_PT
3004
POD-1
88
BRANCH-2
532-555-3000
to 3199
POD-2
CIPT1 v5.0-21
Required Resources
Cisco IP telephony server cluster
Gateway devices
Command List
Job Aids
These job aids are available to help you complete the lab activities.
Partitions
Partition Name
Description
Assigned to Phones
Employee_PT
Regular Employees
2003
3004
Lobby_PT
Lobby Directory Number
2001
Executive_PT
Executive Employees
2002
Partition Name
Description
Assigned to Route Patterns
PSTN_PT
PSTN Route Patterns
9.@
TRK_PT
Trunk Route Patterns
852y.2XXX (y=remote pod number)

853y.3XXX (y=remote pod number)
Calling Search Spaces

Devices
Executive IP phone
(2002)
Calling Search Space

Name
Unlimited_Access_CSS
Description
Contains Partitions
Unlimited Calling
Privileges
Employee_PT
MGCP gateway
Executive_PT
PSTN_PT
H.323 gateway
Employee IP phone
(2003 and 3004)
Lobby_PT
TRK_PT
Company_Only_CSS
Intra- and Intercluster

Calling Only
Intercluster trunk
(IC_TRK)
Employee_PT
Lobby_PT
TRK_PT
SIP Trunks
(SIP_TRK)
Lobby IP phone
(2001)
Cluster_Only_CSS
Intracluster Calling
Only
Employee_PT
Lobby_PT
Lab Guide
89
Task 1: Configuring Partitions

In this task, you will create partitions in Cisco Unified CallManager Administration with
partition names and descriptions.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Class of

Control > Partition, and click Add New to open the Partition Configuration
window.
Step 2
Using the partition configuration data from the Partitions table (in the Job Aids
section of this lab exercise), enter all partition names and their descriptions using
this format:
<partitionName> , <description>
<partitionName> , <description>
...
Step 3
Click Save.
When you choose Call Routing > Class of Control > Partition and then click Find, you
are able to see all partitions that you added.
Task 2: Configuring Calling Search Spaces

In this task, you will create calling search spaces.
Activity Procedure
90
Step 1
Choose Call Routing > Class of Control > Calling Search Space, and click Add
New to open the Calling Search Space Configuration window.
Step 2
Using the Calling Search Spaces table (in the Job Aids section of this lab activity),
enter the first calling search space name and description.
Step 3
Using the information in the Calling Search Spaces table column Contains
Partitions, choose from the Available Partitions pane the appropriate partitions to
add to the first calling search space. (Highlight a partition, and use the down arrow
below the Available Partitions pane to move the highlighted partition to the Selected
Partitions pane. Use the Shift key to select multiple contiguous entries and the Ctrl
key to select multiple noncontiguous entries.)
Step 4
Use the Up and Down arrow keys to place the partitions in the same order as seen in
the Calling Search Spaces table.
Step 5
Click Save.
Step 6
Click the Add New button to open the Calling Search Space Configuration window.
Step 7
Repeat the last six steps to create the remaining two calling search spaces that are
listed in the Calling Search Spaces table.
When you choose Call Routing > Class of Control > Calling Search Space and click
Find, you are able to see all calling search spaces that you have added.
Task 3: Assigning Partitions and Calling Search Spaces to

Configure a Telephony CoS for IP Phones
In this task, you will assign partitions and then assign calling search spaces to your IP phones
(directory numbers), to apply telephony CoS within your IP telephony cluster.
Step 1
Make sure that all your IP phones are registered to your first node.
Step 2
In Cisco Unified CallManager Administration, choose Device > Phone.
Step 3
Set the search parameter Find phone where to Directory Number, and then click
Find.
Step 4
Choose the IP phone with the directory number 2001 to open the Phone
Configuration page.
Step 5
Click Line [1] - 2001 from the left page column to access the Directory Number
Configuration page.
Step 6
Choose Lobby_PT for the Route Partition, and then click Save.
Step 7
In the Associated Device pane, highlight the phone device and click the Edit Device
button.
Step 8
Choose Cluster_Only_CSS for the Calling Search Space and click Save. When the
pop-up window appears, click OK.
Step 9
Click Reset, then click Reset again, and click OK on the pop-up window shown.
Step 10
Click the Go button next to the Back to Find/List link, in the upper-right corner of
the screen, to list the IP phones that are registered in your cluster by directory
numbers.
Step 11
Repeat this task for the other three IP phones in your cluster. Refer to the tables
Partitions and Calling Search Spaces in the Job Aids section to locate the
partition and the calling search space configuration parameters applicable to these
remaining IP phones.
Lab Guide
91
The Executive IP phone (2002) is able to call both the Employee (2003 or 3004) and Lobby
(2001) IP phones.
The Lobby (2001) and Employee (2003 or 3004) IP phones can call between each other but
cannot call to the Executive IP phone (2002).
Task 4: Assigning Partitions and Calling Search Spaces to

Route Patterns and Gateways
In this task, you will assign partitions and calling search spaces to route patterns, translation
patterns, gateways, and trunks.
Activity Procedure
Route Patterns
Step 1
Choose Call Routing > Route/Hunt > Route Pattern. Click Find to list all Route
Patterns if none are listed.
Step 2
Click the PSTN Route Pattern 9.@ to access its Route Pattern Configuration page.
Step 3
Refer to the Partition Name column from the Partitions table in the Job Aids
section, and find the partition that corresponds to the selected PSTN route pattern.
Choose this corresponding partition from the Route Partition drop-down menu.
Step 4
Step 5
Click the Go button next to the Back to Find/List Route, and repeat the last three
steps for the other two trunk route patterns from the Partitions table.
Step 6
Choose Device > Gateway, and click Find to list all the gateways.
Step 7
Choose your H.323 gateway from the list (10.x.250.102, where x is your pod
number) to access its Gateway Configuration page.
Step 8
Scroll down the page to see the Call Routing Information Inbound Calls section
and, from the Calling Search Spaces drop-down menu, based on the parameters in
the Calling Search Space table of the Job Aids section, choose the CSS that
corresponds to the H.323 gateway.
Step 9
Step 10
Click Go next to the Back to Find/List and, from the list shown, click your MGCP
gateway to open its Gateway Configuration page.
Step 11
Click the icon representing the 1/0 Endpoint.
Gateways
Note
92
Because each endpoint on an MGCP gateway is considered to be a separate gateway, to

configure the calling search space, you must select the endpoint.
Step 12
Scroll down the page to see the Call Routing Information Inbound Calls section
and, from the Calling Search Spaces drop-down menu, based on the parameters in
the Calling Search Space table of the Job Aids section, choose the CSS that
corresponds to that MGCP gateway.
Step 13
Step 14
Choose Device > Gateway and click Find. Then click the Select All button to select
all gateways in the list.
Step 15
Click Reset Selected, on the new pop-up window click Reset, and then click OK.
This action resets all the gateways at once.
Intercluster Trunk
Step 16
Choose Device > Trunk and click Find.
Step 17
From the list, select the IC_TRK intercluster trunk listed to open its Trunk
Configuration page.
Step 18
Scroll down the page to see the Call Routing Information section and, from the
Calling Search Spaces drop-down menu, based on the parameters in the Calling
Search Space table, choose the CSS that corresponds to the IC_TRK intercluster
trunk.
Step 19
Step 20
Reset the trunk to make the changes take effect.
Step 21
Repeat steps 34-38 for the SIP trunk SIP_TRK.
From the Employee IP phone (2003 or 3004), you are able to call within the cluster (except
for the Executive IP phone) and use the Intercluster and SIP trunk to call the remote cluster
IP phones; however, you are not able to call the PSTN numbers.
From the Executive IP phone (2002), you are able to call anywhere; however, other clusters
cannot call the Executive IP phone because the gateway calling search space
(Cluster_Only_CSS) does not have the executive partition listed in the selected partitions.
From the Lobby IP phone (2001), you are able to call only the Employee IP phone (2003 or
3004).
Task 5: Configuring PLAR

In this task, you will configure the PLAR feature using partitions, calling search spaces, and a
special use of a translation pattern.
Activity Procedure
Step 1
Choose Call Routing > Class of Control > Partition, and click Add New to open
the Partition Configuration page.
Lab Guide
93
Step 2
Create a partition with the name Hotline_PT and description PLAR Translation
Pattern.
<< partitionName >>, << description >>
Step 3
Click Save.
Step 4
Choose Call Routing > Class of Control > Calling Search Space, and click Add
New to open the Calling Search Space Configuration page.
Step 5
Enter the new calling search space Hotline_CSS in the Calling Search Space Name
field. Add the Hotline_PT partition to the Selected Partitions pane.
Step 6
Click Save.
Step 7
Choose Call Routing > Translation Pattern, and click Add a New to access the
Translation Pattern Configuration page.
Step 8
Leave the Translation Pattern field blank.
Step 9
Choose Hotline_PT for the Partition, and choose Unlimited_Access_CSS for the
Calling Search Space.
Step 10
Step 11
Enter 2001 for the Called Party Transform Mask (not the Calling Party). This is the
PLAR destination number.
Step 12
Click Save.
Step 13
Choose Device > Phone and choose the Directory Number from the Find Phone
where drop-down menu. Click Find to list all directory numbers in the cluster.
Step 14
Choose the Phone2-x with 2002 directory number (Executive IP phone). Assign it
the new Hotline_CSS, and then click Save and OK on the pop-up window.
Step 15
Reset the 2002 Executive IP phone.
You have successfully completed this task when you attain this result:
Click the New Call softkey on the Executive IP phone (2002), and it should automatically
ring the 2001 extension. If it does not work, make sure that you entered the correct PLAR
destination number in the Called Party Transform Mask (not the Calling Party Transform
Mask).
Note
The blank Translation Pattern is equal to no digits, and the Called Party Transformation
Mask specifies the directory number or Route Pattern to be automatically dialed.
Cleanup
To prepare the equipment for the next lab, complete these steps:
94
Step 1
Assign all IP phone lines (directory numbers) to the Executive_PT partition.
Step 2
Assign all IP phones to the Unlimited Access_CSS calling search space.
Step 3
Assign all gateways to the Unlimited Access_CSS calling search space.
Step 4
Assign all trunks to the Unlimited Access_CSS calling search space.
Step 5
Assign all translation patterns to the Executive_PT partition and to the Unlimited
Access_CSS calling search space.
Step 6
Assign all route patterns to the Executive_PT partition.
Step 7
Delete the PLAR translation pattern.
Step 8
Reset all IP phones, gateways, and trunks.
All IP phones and gateways are able to dial all directory numbers and route patterns.
Lab Guide
95

Activity Objective
In this activity, you will configure call admission control using locations and regions in Cisco
Unified CallManager Administration. You will create a remote location and region called
Branch that has 40 kbps of available bandwidth and uses the G.729 codec. You will attempt to
establish two parallel calls to the Branch region to discover that only a single call will be
admitted.
Then you will configure gatekeeper-controlled intercluster trunks in Cisco Unified
CallManager Administration. You will also configure call admission control on the Cisco IOS
gatekeeper so that any calls that would cause the zone bandwidth to exceed the 128-kbps
maximum will be rejected. Finally, you will configure call admission control using RSVP in
Cisco Unified CallManager and Cisco IOS router to allow 56 kbps maximum bandwidth
between HQ and Branch sites.
96
Configure locations and regions in Cisco Unified CallManager Administration to provide

locations-based call admission control in a centralized call-processing model
Configure zone, bandwidth, and Cisco Unified CallManager information settings on the
gatekeeper
Add the gatekeeper in Cisco Unified CallManager Administration
Configure RSVP on the Cisco Unified CallManager
Configure the Cisco IOS gateway for RSVP
Visual Objective

Configuring Call Admission Control
Phone1-1
Phone2-1
Phone1-2
Phone3-1
Phone2-2
521-555-2000
to 2199
Phone3-2
522-555-2000
to 2199
CCM1-2
CCM1-1
HQ-1
Max. 128kbps
Max. 128kbps
H.323 gk
HQ-1
Second Call
Denied !
Second Call
Denied !
Max. 56kbps
First Call
Admitted
BRANCH-1
BR-1
H.323 gk
First Call
Admitted
RSVP
RSVP
HQ-2
RSVP
HQ-2
Max. 56kbps
First Call
Admitted
Second Call
Denied !
BRANCH-2
RSVP
BR-2
Phone4-1
Phone4-2
532-555-3000
to 3199
531-555-3000
to 3199
POD-1
POD-2
CIPT1 v5.0-22
Required Resources
Two Cisco Unified CallManager clusters with IP phones
Single gatekeeper device per cluster
Two RSVP agents per cluster
IP WAN
Simulated PSTN
Command List
Cisco IOS RSVP Commands
Command
Description
enable
Enables you to enter privileged EXEC mode, or any other

configure terminal
bandwidth bw
Configures available bandwidth on an interface.
Lab Guide
97
Command
Description
fair-queue
[congestive-discardthreshold [dynamicqueues [reservablequeues]]]
Enables weighted fair queuing (WFQ) for an interface.
max-reserved-bandwidth
bw-percentage
Adjusts the amount of bandwidth that can be reserved on an

interface.
ip rsvp bandwidth
[interface-kbps]
[single-flow-kbps]
Enables RSVP for IP on an interface.
sccp local interface
Selects the local interface that SCCP applications use to register

with Cisco Unified CallManager.
sccp ccm ccm-address

identifier number
version 5.0.1
Adds a Cisco Unified CallManager server to the list of available

servers to which the Cisco voice gateway can register.
sccp
Enables SCCP and brings it up administratively.
sccp ccm group groupnumber
Enters SCCP Cisco Unified CallManager configuration mode.
bind interface
interface
Binds an interface to the Cisco Unified CallManager group.
associate ccm
identifier priority
priority
Adds a Cisco Unified CallManager server to the Cisco Unified

CallManager group and establishes its priority within the group.
associate profile
profile-id register
name
Associates a DSP farm profile to the Cisco Unified CallManager

group.
dspfarm profile
profile-id mtp
Enters DSP farm profile configuration mode to define a profile for

DSP farm services.
codec pass-through
Enables codec pass-through mode for the DSP farm profile.
rsvp
Enables RSVP support in the DSP farm profile.
maximum sessions
{hardware | software}
number
Specifies the maximum number of sessions that are supported by

the profile.
associate application
SCCP
Associates the SCCP protocol to the DSP farm profile.
no shutdown
Enables the profile, allocates DSP farm resources, and

associates the application.
Cisco IOS Gatekeeper Commands
98
Command
Description
gatekeeper
Enters gatekeeper configuration mode.
zone local name
Specifies the zone that is controlled by the gatekeeper.
zone subnet name [ipaddress/bit-mask]

enable
Configures the gatekeeper to accept discovery and registration

messages that are sent by endpoints in designated subnets.
Command
Description
no zone subnet name

[ip-address/bit-mask]
enable
Configures the gatekeeper to deny discovery and registration

messages that are sent by endpoints in configured subnets.
zone prefix name

[digits]
Configures the prefix or digits that are associated with a zone.
gw-type-prefix 1#*
default-technology
Configures the default technology.
bandwidth total zone

name [kbps]
Configures the total bandwidth that is allowed in the specified

zone.
bandwidth session zone

name [kbps]
Configures the total bandwidth that is allowed during a session

(call) in a zone.
no shutdown
Enables the gatekeeper.
show gatekeeper
endpoints
Display the status of all registered endpoints for a gatekeeper,
show gatekeeper calls
Displays the status of each ongoing call of which a gatekeeper is

aware.
Job Aids
H.323 Gatekeeper Configuration Parameters
Cluster
Zone Name
Remote Zone
Name
Pod 1
hq-1.pod1.com
hq-2.pod2.com
10.1.1.101
10.2.1.101
10.1.1.0
Pod 2
hq-2.pod2.com
hq-1.pod1.com
10.2.1.101
10.1.1.101
10.2.1.0
H.323 GK IP
Address
Remote H.323
GK IP Address
HQ LAN IP
Subnet
Lab Guide
99
Task 1: Configuring Locations and Regions

In this task, you will configure locations and regions in Cisco Unified CallManager
Administration to provide location-based call admission control in a centralized call-processing
model.
Activity Procedure
Add a Region and Define the Codec Used
Step 1
In Cisco Unified CallManager Administration, choose System > Location to open
the Find and List Location Configuration window.
Step 2
Click Add New to open the Location Configuration window.
Step 3
Enter Branch as the location name
Step 4
For the Audio Bandwidth, select the other radio button next to kbps field, and enter
40 kbps for the available audio bandwidth.
Step 5
Click Save.
Step 6
Repeat the last four steps to add a new location named HQ with the available
bandwidth 40 kbps.
Step 7
Choose System > Region to open the Find and List Regions window.
Step 8
Click Add New to open the Region Configuration window.
Step 9
In the Region Name field, enter Branch and click Save.
Step 10
Repeat the last two steps to add a new region named HQ.
Step 11
Change the default codec between the Branch and HQ regions to G.729. Click
Branch in the Regions pane and, from the Audio Codec drop-down menu, choose
G.729.
Step 12
Click Save.
Step 13
Change the Audio Codec for HQ (within this region) to G.711. Click HQ in the
Regions pane and, from the Audio Codec drop-down menu, choose G.711. Click
Save.
Step 14
Choose System > Region and click Find. Select the Branch region from the list.
Step 15
Change the Audio Codec for Branch (within this region) to G.711. Click Branch in
the Regions pane and, from the Audio Codec drop-down menu, choose G.711. Click
Save.
Create a New Device Pool for the Remote Region

Step 16
Choose System > Device Pool.
100
Step 17
Click Add New.
Step 18
Name the new device pool CM_Remote_DP.
Step 19
Step 20
Assign these parameters to the new device pool:
Cisco Unified CallManager Group: CMG
Date/Time Group: CMLocal
Region: Branch
Softkey Template: Standard User
SRST Reference: Disable
Click Save to save the configuration in the database.
Modify the Device Pool for the HQ Region

Step 21
Choose System > Device Pool.
Step 22
Click Find and, from the list, select Default_CMLocal_DP.
Step 23
Change the Region to HQ.
Step 24
Click Save to save the configuration in the database.
Assign the New Device Pool to the Remote IP Phone

Step 25
Choose Device > Phone, and click Find to list all IP phones.
Step 26
Select the IP phone with directory number 3004 (Phone4-x, where x is your pod
number) to open its Phone Configuration window. You will configure this IP phone
as a Branch IP phone.
Step 27
Choose CM_Remote_DP for the device pool and Branch as the location for the IP
phone.
Step 28
Click Save and OK on the pop-up window.
Step 29
Reset all IP phones. Choose Device > Phone and click Find. Click the Select All
button and then the Reset Selected button.
Step 30
On the new dialog window, click Reset and Close.
Place a call from the Phone3-x to the Phone 4-x (3004) and answer the call. While the call
is active, press the ? button twice on the Phone4-x to verify that the call has been placed
using the G.729 codec. Leave the call active and continue to the next step.
Attempt to place another call from the Phone1-x to the Phone4-x (3004). The Phone1-x
displays the message Not Enough Bandwidth because there is no bandwidth available
between HQ and Branch locations.
Release the active call.
Lab Guide
101
Task 2: Configuring RSVP on Cisco Unified CallManager

In this task, you will configure the RSVP policy between locations and configure the RSVP
agent as a Media Termination Point (MTP) in Cisco Unified CallManager. Finally, you will
configure Media Resource Groups (MRGs) and Media Resource Group Lists (MRGLs) to
apply the RSVP to device pools and hence individual IP phones.
Activity Procedure
Reconfigure Locations
Step 1
In Cisco Unified CallManager Administration, choose System > Location and click
Find.
Step 2
From the list, select the HQ and, when the Location Configuration page opens,
change the Audio Bandwidth to Unlimited.
Step 3
For the HQ location, configure the RSVP Setting toward the Branch location to
Mandatory. Click Branch in the Location pane and, from the RSVP Setting dropdown menu, choose Mandatory. Cisco Unified CallManager will now use RSVP
for call admission control to this location instead of locations-based call admission
control.
Step 4
Click Save.
Step 5
Choose System > Location and click Find.
Step 6
From the list, select the Branch and, when the Location Configuration page opens,
change the Audio Bandwidth to Unlimited.
Step 7
For the Branch location, verify that the RSVP Setting toward the HQ location is set
to Mandatory.
Step 8
Click Save.
Step 9
Click the Resync Bandwidth button to reset all call admission control bandwidth
usage.
Create MTPs
102
Step 10
Add a new MTP, using MTP_HQ as the name. Choose Media Resources >
Media Termination Point and click Add New.
Step 11
Make sure that the Media Termination Point Type is set to Cisco IOS Enhanced
Software Media Termination Point.
Step 12
Enter MTP_HQ as a Media Termination Point Name and RSVP HQ MTP for the
Description.
Step 13
Select the Default_CMLocal_DP for the device pool and click Save.
Step 14
Repeat the last four steps to add MTP with these parameters:
MTP Type: Cisco IOS Enhanced Software Media Termination Point
MTP Name: MTP_BR
Description: RSVP Branch MTP
Device Pool: CM_Remote_DP
Create MRGs and MRGLs

Step 15
Add a new MRG named HQ_RSVP_MRG to Cisco Unified CallManager, and
add the MTP named MTP_HQ as the only member of that group. Choose Media
Resources > Media Resource Group and click Add New.
Step 16
Enter HQ_RSVP_MRG for the Name and, from the Available Media Resources
pane, select MTP_HQ and move it to the Selected Media Resources pane.
Step 17
Click Save.
Step 18
Repeat the last three steps to add a new MRG named Branch_RSVP_MRG and add
the MTP named MTP_BR as the only member of that group.
Step 19
Add a new MRGL named HQ_MRGL to Cisco Unified CallManager, and add the
MRG named HQ_RSVP_MRG as the only member of that list. Choose Media
Resources > Media Resource Group List and click Add New.
Step 20
Enter HQ_MRGL for the name, and move HQ_RSVP_MRG from the Available
MRGs pane to the Selected MRGs pane.
Step 21
Click Save.
Step 22
Repeat the last three steps to add a new MRGL named Branch_MRGL and add the
MRG named Branch_RSVP_MRG as the only member of that list.
Step 23
Choose System > Device Pool and click Find.
Step 24
From the list, choose Default_CMLocal_DP and, when its configuration page
opens, assign it with HQ_MRGL. Click Save.
Step 25
Reset the device pool.
Step 26
Repeat the last three steps to assign Branch_MRGL to the CM_Remote_DP.
Attempt for a call between Branch and HQ locations. No call can be placed between
Branch and HQ locations.
Task 3: Configuring RVSP on Cisco IOS Gateways

In this task, you configure RSVP in the Cisco IOS gateways in the HQ and Branch locations.
Activity Procedure
Lab Guide
103
Configure RSVP Agent on Router at HQ Location

Step 1
Connect to your HQ-x router (where x is your pod number).
Step 2
Enable RSVP on major serial interface toward the Branch site to allow for a single
G.729 call (40 kbps required initially, but use 56 kbps). Enter these commands:
interface serial0/0/1
bandwidth 128
fair-queue 64 256 3
max-reserved-bandwidth 80
ip rsvp bandwidth 56
exit
Step 3
Configure the SCCP for RSVP support on the HQ-x router. Enter these commands:
sccp local fastethernet0/0.x02 (x is your pod number)
sccp ccm <First node IP Address> identifier 1 version 5.0.1
sccp
Step 4
Configure SCCP Cisco Unified CallManager group on HQ-x router. Enter these
commands:
sccp ccm group 1
bind interface fastethernet0/0.x02 (x is your pod number)
associate ccm 1 priority 1
associate profile 1 register MTP_HQ
exit
Step 5
Configure and active the MTP functions on the DSPFARM of the HQ-x router.
Enter these commands:
dspfarm profile 1 mtp
codec pass-through
rsvp
maximum sessions software 2
associate application SCCP
no shutdown
end
Configure RSVP Agent on Router at Branch Location

Step 6
Connect to your BR-x router (where x is your pod number).
Step 7
Enable RSVP on major serial interface toward the HQ site, to allow for a single
G.729 call (40 kbps required initially, but use 56 kbps). Enter these commands:
interface serial0/0/1
bandwidth 128
fair-queue 64 256 3
max-reserved-bandwidth 80
ip rsvp bandwidth 56
exit
104
Step 8
Configure the SCCP for RSVP support on the BR-x router. Enter these commands:
sccp local fastethernet0/0.x05 (x is your pod number)
sccp ccm <First node IP Address> identifier 1 version 5.0.1
sccp
Step 9
Configure SCCP Cisco Unified CallManager group on HQ-x router. Enter these
commands:
sccp ccm group 1
bind interface fastethernet0/0.x05 (x is your pod number)
associate ccm 1 priority 1
associate profile 1 register MTP_BR
exit
Step 10
Configure and active the MTP functions on the DSPFARM of the BR-x router.
Enter these commands:
dspfarm profile 1 mtp
codec pass-through
rsvp
maximum sessions software 2
associate application SCCP
no shutdown
end
Verify that the DSPFARM MTP is operational. Use the command show dspfarm profile 1
on HQ-x, and verify that Admin State is UP and Operation State is Active.
Place a call from Phone1-x to Phone4-x (3004). This call should succeed. While holding
the call active, display on HQ-x, using the command show ip rsvp reservation, the
installed RSVP reservation.
Attempt for a second call from Phone2-x to Phone4-x (3004). The second call is blocked
because Cisco Unified CallManager is configured with RSVP MANDATORY reservation,
and only 56K of voice bandwidth is allocated across the HQ to Branch WAN link.
Hang up all active calls.
Cleanup
Step 1
Choose Device > Phone. Click Find and assign all IP phones back to the location
Hub_None.
Step 2
Choose Device > Phone. Click Find and, from the list, choose IP phone with
directory number 3004 (Phone4-x).
Step 3
Assign the Phone4-x back to the Default_CMLocal_DP device pool.
Step 4
Click Save.
Lab Guide
105
Step 5
Choose System > Device Pool. Click Find and, from the list, choose
Default_CMLocal_DP.
Step 6
Disassociate the device pool from the Media Resource Group List HQ_MRGL
(choose <None>).
Step 7
Change the Region back to Default.
Step 8
Click Save.
Step 9
Verify that you can place two parallel calls between HQ and Branch sites.
Task 4: Configuring Gatekeeper

In this task, you will configure zone, bandwidth, and Cisco Unified CallManager information
settings on the gatekeeper and add the gatekeeper in Cisco Unified CallManager
Administration.
Activity Procedure
Step 1
Connect to your H.323 gatekeeper HQ-x (where x is your pod number) and enter the
router configuration mode.
Step 2
Configure the H.323 gatekeeper on the router. Refer to the H.323 Gatekeeper
Configuration Parameters table in the Job Aids section for all the information to
supply and enter this command sequence:
gatekeeper
zone local <Your Zone Name> company.com <Your H.323 GK IP
Address>
zone remote <Remote Zone Name> company.com <Remote H.323 GK IP
Address>
zone prefix <Remote Zone Name> *
gw-type-prefix 1#* default-technology
bandwidth total zone <Your Zone Name> 128
bandwidth session zone <Your Zone Name> 128
no shutdown
end
106
You can see your H.323 gatekeeper state UP when using the show gatekeeper status
command.
Task 5: Configuring the Gatekeeper and Intercluster Trunk in

Cisco Unified CallManager Administration
In this task, you will add the gatekeeper device in Cisco Unified CallManager Administration
and configure the intercluster gatekeeper-controlled trunk. You will also add the gatekeepercontrolled trunk to an existing route group and remove the nongatekeeper-controlled trunk.
Activity Procedure
Add the Gatekeeper in Cisco Unified CallManager Administration
Step 1
Connect to your first node and, in Cisco Unified CallManager Administration,
choose Device > Gatekeeper. Click Add New.
Step 2
In the Host Name/IP Address field, enter the IP address of your H.323 Gatekeeper
(10.x.1.101).
Step 3
For the Description, enter Gatekeeper.
Step 4
Leave the Registration Request Time to Live and Registration Retry Timeout fields
at their default values. Ensure that Enable Device box is checked.
Step 5
Click Save.
Step 6
Click Reset, then Reset, and finally Close for the dialog window.
Add an Intercluster Gatekeeper-Controlled Trunk

Step 7
Choose Device > Trunk. Click Add New.
Step 8
Choose Inter-Cluster Trunk (Gatekeeper Controlled) for the Trunk type.
Step 9
The Device Protocol field will automatically populate with the Inter-Cluster
Trunk. Click Next.
Step 10
In the Device Name field, enter the name of the new intercluster gatekeepercontrolled trunk ICT_GK.
Step 11
In the Description field, enter the Gatekeeper Controlled ICT.
Step 12
For Device Pool, choose Default_CMLocal_DP.
Step 13
For Calling Line ID Presentation, choose Allowed.
Step 14
Scroll the current page down to see the Gatekeeper Information section and choose
your H.323 Gatekeeper IP address from the Gatekeeper Name drop-down menu.
Step 15
Choose the Gateway from the Terminal Type drop-down menu.
Note
Terminal would not be inappropriate because it would be an H.323 client.
Step 16
Enter 1# for the Technology Prefix.
Step 17
In the Zone field, enter your zone name from the H.323 Gatekeeper Configuration
Parameters table in the Job Aids section. This entry is case-sensitive and must be
typed exactly as it is shown in the table.
Lab Guide
107
Step 18
Click Save and, when the pop-up window appears, click OK
Step 19
Reset the new trunk by clicking Reset, then Reset and Close.
You can see your first node registered on the H.323 gatekeeper as the H.323 endpoint,
when using the command show gatekeeper endpoints on the H.323 gatekeeper. The first
node should be identified with the ICT_GK_1 H.323 ID.
Add the New Intercluster Gatekeeper-Controlled Trunk to an Existing Route Group

Step 20
Choose Call Routing > Route/Hunt > Route Group. Click Find to list all the route
groups if none are visible.
Step 21
Click the TRK_RG to open its Route Group Configuration page.
Step 22
In the Selected Devices pane, highlight the old intercluster nongatekeeper-controlled

trunk (IC_TRK) and SIP trunk (SIP_TRK) and move it to the Removed Devices
pane using the down arrow between them.
Step 23
In the Available Devices pane, highlight the new intercluster gatekeeper-controlled

trunk ICT_GK and move it to the Selected Devices pane by clicking the Add to
Route Group button.
Step 24
Click Save.
Caution
108
Coordinate here with your colleagues in the remote cluster, so that this verification is
performed by a single cluster at a time. Also, to verify, the other cluster has to be finished
with the configuration.
Place the first call from one of your IP phones to any IP phone in the remote cluster using
the former intercluster SIP trunk prefix (for example, to call from Phone2-1 to Phone2-2,
dial 85222002). The first call should be successful. Hold on the call. On your H.323
gatekeeper, display the active call along with the bandwidth it uses (for the current G.711 it
would be by default 128 kbps) by using the command show gatekeeper calls.
Enable the RAS debugging on your H.323 gatekeeper by entering the command debug ras,
and attempt for a second call initiated from another IP phone within your cluster toward a
remote cluster IP phone. The second call should fail because the call admission control on
your H.323 gatekeeper has denied it because of lack of intercluster bandwidth. Your H.323
gatekeeper debug output should show the Admission Reject message (ARJ) sent to the
Cisco Unified CallManager.
When done, disable all debugging on your H.323 gatekeeper and hang up to release the
active call. Let the other pod perform the same test.
Cleanup
Step 1
Choose Call Routing > Route/Hunt > Route Group. Click Find and, from the list,
choose TRK_RG.
Step 2
Remove the ICT_GK from the Selected Device pane.
Step 3
From the Available Device pane, move the IC_TRK into the Selected Device pane.
Step 4
From the Available Device pane, move the SIP_TRK into the Selected Device pane.
Step 5
Click Save.
Verify that you can place two parallel intercluster calls via IP WAN and that the H.323
gatekeeper is no longer used for call admission control.
Lab Guide
109

Activity Objective
In this activity, you will configure Cisco Unified SRST to provide call-processing redundancy
during WAN failure. After completing this activity, you will be able to meet these objectives:
Configure SRST on the remote location gateway
Configure the device pool for SRST and assign the device pool to a phone located in a
remote location
Isolate the remote location IP phone from the Cisco Unified CallManager
Visual Objective

Configuring SRST
Phone1-1
Phone2-1
DN: 2001
DN: 2002
Phone3-1
DN: 2003
Phone1-2
Phone2-2
DN: 2001
DN: 2002
Phone3-2
DN: 2003
521-555-2000
to 2199
522-555-2000
to 2199
CCM1-1
CCM1-2
HQ-2
HQ-1
HQ-1
Failure
BRANCH-1
PSTN
Backup to reach
HQ IP phones
BR-1
SRST
router
531-555-3000
to 3199
HQ-2
Failure
BR-2
BRANCH-2
SRST
router
Phone4-1
Phone4-2
DN: 3004
DN: 3004
POD-1
532-555-3000
to 3199
POD-2
CIPT1 v5.0-23
Required Resources
110
MGCP PRI gateway at HQ site
Gateway device with SRST feature and analog line at Branch site
Simulated PSTN
IP WAN
Command List
Cisco IOS SRST Commands
Command
Description
enable
Enables you to enter privileged EXEC mode, or any other

configure terminal
call-manager-fallback
Enables SRST feature support and enters Cisco Unified

CallManager fallback mode.
ip source-address
<router ip address>
<port>
Enables the router to receive messages from the Cisco IP

phones through the specified IP addresses and ports. The default
port number is 2000.
max-ephones <number IP
Phones>
Configures the maximum number of Cisco IP phones that can be

supported by the router. The default is 0. The maximum number
is platform dependent.
max-dn <number DNs>
Sets the maximum number of directory numbers or virtual voice

ports that can be supported by the router. The default is 0. The
maximum number is platform dependent.
show ephone
Displays the Cisco IP phones that have registered to the SRST

router.
destination-pattern
pattern
Configures the destination pattern under the dial peer.
dial-peer voice tag

{pots | voip}
Enters the dial-peer configuration mode.
dialplan-pattern tag
pattern extensionlength number
Creates a global prefix that can be used to expand the

abbreviated extension numbers into fully qualified E.164
numbers, under call-manager-fallback mode.
prefix prefix
Configures the prefix to prepend on outbound calls under the dial

peer.
keepalive seconds
Configures the time interval between sending keepalive

messages by the router to the Cisco IP phones when SRST is
enabled, under call-manager-fallback mode.
Job Aids
Lab Guide
111
Task 1: Configuring SRST on Cisco IOS Router

In this task, you will configure SRST support on the Branch gateway.
Activity Procedure
Step 1
Make sure that your Branch IP phone (Phone4-x, where x is your pod number) is
registered to the Cisco Unified CallManager in addition to at least one of your HQ
IP phones.
Step 2
Connect to your Branch router BR-x (where x is your pod number). This will be
your SRST gateway.
Step 3
On the SRST gateway, configure the dial peer for outbound calls, for the case when
the Cisco Unified CallManager is unreachable. Branch site employees should be
able to call to the cluster HQ site via PSTN using the abbreviated dialing, still using
the four-digit directory number only. The SRST router must expand the directory
number into a full PSTN number (for instance, the directory number 2001 should be
expanded into 152x5552001, where x is your pod number).
dial-peer voice 2 pots
destination-pattern 2...
port module/slot/port (FXO to PSTN)
prefix 152x5552 (where x is your pod number)
exit
Step 4
Enter the command call-manager-fallback. This transfers you to the call-managerfallback configuration mode.
Step 5
Enter the command that follows to enable the SRST gateway to receive messages
from the Cisco IP phones through the specified IP addresses using the default port
2000. The default router IP address is 10.x.5.102 (where x is your pod number).
ip source-address <Default Router IP Address>
Step 6
On the SRST gateway, configure the keepalive timer to maintain the communication
to BR site IP phone every 20 seconds.
Step 7
On the SRST gateway, create the global PSTN prefix for inbound call processing,
which is used to expand the abbreviated four-digit extensions into fully qualified
long-distance PSTN numbers (for instance, 3xxx into 53x5553NNN, where x is your
pod number, and N is any digit):
dialplan-pattern 1 53x5553... extension-length 4 (x is your
pod number)
Step 8
Enter the command that follows to configure the maximum number of Cisco IP
phones that can be supported by the router. You will enter 1 because you have only
one IP phone at the Branch location.
max-ephones 1
112
Note
Step 9
Although the Cisco routers can support more than one IP phone, it is important to consider
that for each IP phone that is configured, the router reserves resources for SRST that
cannot be used for other functions, such as routing.
Enter this command to set the maximum number of directory numbers or virtual
voice ports that can be supported by the router:
max-dn 1
Step 10
Type end or Ctrl-Z to exit the configuration mode.
You can verify the SRST configuration by using the command show call-managerfallback. More detailed verification will be performed in a later task.
Task 2: Configuring SRST on Cisco Unified CallManager

In this task, you will configure SRST support on the Cisco Unified CallManager.
Activity Procedure
Step 1

choose System > SRST.
Step 2
Click Add New.
Step 3
In the SRST Reference Name field, type the name of your branch router BR-x
(where x is your pod number).
Step 4
In the IP Address field, enter 10.x.5.102 (where x is your pod number).
Step 5
Click Save.
Step 6
Choose System > Device Pool and click Find to list available Device Pools.
Step 7
Click Device Pool CM_Remote_DP.
Step 8
In the SRST Reference field, from its drop-down menu, choose the SRST reference
configured earlier with the name of your branch router BR-x.
Step 9
Click Save.
Step 10
You can see that the Branch IP phone has obtained the SRST reference. This can be
checked on the IP phone by pressing the Settings button and then selecting Network
Settings. The SRST reference should be visible as CallManager 2 SRST with the state
Standby.
Lab Guide
113
Task 3: Isolating Branch from the Cisco Unified CallManager

In this task, you will purposely isolate the Branch IP phone from its Cisco Unified
CallManager, to test the SRST operation.
Activity Procedure
Step 1
Connect to your Branch router BR-x and shut down the interface serial0/0/1. This
will imitate a failure in the WAN, and the Cisco Unified CallManager will become
unreachable to the Branch IP phone.
Step 2
Branch IP phone (Phone4-x) has lost the connectivity to the Cisco Unified
CallManager and therefore it fails to register to it. Wait for about 2 to 3 minutes until
the Branch IP phone displays the message CM Fallback Service Operating.
Step 3
On your SRST gateway, enter the command show ephone to verify that the Branch
IP phone (Phone4-x) has successfully registered to it. On the Branch IP phone, now
in the fallback mode, press Settings, and then select Network Settings to observe
that the active Cisco Unified CallManager is now the SRST router (seen here as the
Cisco Unified CallManager x SRST with the state Active).
Step 4
While the Cisco Unified CallManager is unreachable, place a call from the Branch
IP phone4-x to the HQ IP phone (stay within your cluster). Call setup will take a bit
longer than usual, but the call should succeed. Disconnect the call after the test.
Step 5
Repair the imitated WAN failure by reactivating the serial0/0/1 interface and wait
for about 2 to 3 minutes for the Branch IP phone to resume the normal operation
using its Cisco Unified CallManager (the message CM Fallback Service Operating
disappears from the IP phones display).
114
You could call from Branch to HQ using abbreviated directory number when the Cisco
Unified CallManager was unreachable and the Branch IP phone was in SRST fallback
mode.

Activity Objective
In this activity, you will configure media resources (MTP, CONF, XCODE, and MOH) for
devices to use. After completing this activity, you will be able to meet these objectives:
Configure hardware media resources in the Cisco Unified CallManager database
Configure Meet-Me directory numbers to support Meet-Me conferences
Configure MRGs and MRGLs and assign the MRGL to devices
Configure and add new audio source files used by the MOH servers, and assign the new
audio source files to devices
Visual Objective

Configuring Media Resources
Phone1-1
Phone2-1
Phone1-2
Phone3-1
Phone2-2
Region: LowBW
Max. G.729
Region: LowBW
Max. G.729
521-555-2000
to 2199
522-555-2000
to 2199
CCM1-2
CCM1-1
MOH server
HQ-1
Phone3-2
HQ-1
Conf. Bridge
Transcoder
Conf. Bridge
Transcoder
MOH server
HQ-2
Meet-Me 250X
Meet-Me 250X
G.711 Only
HQ-2
G.711 Only
PSTN
BRANCH-2
BRANCH-1
BR-1
H-323
gw
G.711
H-323
gw
BR-2
G.711
Phone4-1
Phone4-2
532-555-3000
to 3199
531-555-3000
to 3199
POD-1
POD-2
CIPT1 v5.0-24
Required Resources
Two Cisco Unified CallManager clusters with three Cisco IP phones each
Cisco IOS voice gateway with hardware conference capability
Cisco IOS H.323 gateway connected to the simulated PSTN
Audio files (.wav format)
Lab Guide
115
Command List
H.323 Gateway IOS Commands
Command
Description
codec g711alaw
Configures the audio codec on a voice dial peer. Only this

codec becomes allowed on that voice dial peer.
show dial-peer voice
Displays configuration information and call statistics for a

voice dial peer.
DSPFARM IOS Commands

Command
Description
voice-card <card number>
Enters voice-card configuration mode to configure its

resources.
dsp services dspfarm
Configures DSP farm services for a voice card.
Sccp
Enables the Skinny Client Control Protocol (SCCP) and its

related applications (transcoding and conferencing).
sccp local <interface>
Select the local interface that the SCCP applications should

use to register with Cisco Unified CallManager.
sccp ccm <CCM ipaddress>

priority
Adds Cisco Unified CallManager server to the list of

available servers and sets its priority.
dspfarm transcoder maximum

sessions
Specifies the maximum number of transcoding sessions to

be supported by a DSP farm.
dspfarm confbridge maximum

sessions
Specifies the maximum number of concurrent conference

sessions for which DSP farm resources should be allocated.
dspfarm
Enables DSP farm services.
Job Aids
116
Pre-Lab Task 1: Activating Services and Default Media

Resource Availability
This section activates the MOH and tests the default media resources that are available when
the IP Voice Media Streaming Application service is started on Cisco Unified CallManager.
Step 1
In the Cisco Unified CallManager Serviceability, choose Tools > Service

Activation.
Step 2
Choose the first node from the Server drop-down menu.
Step 3
Activate the Cisco IP Media Streaming App service.
Step 4
Make sure that all your IP phones are registered to your first node.
Step 5
Set up a call from your Phone1-x to your Phone2-x in your cluster.
Step 6
On the Phone1-x, press the Hold softkey. After about 5 seconds, the Phone2-x will
start generating beeps (or playing music) to signal that the call was put on hold.
Step 7
On the Phone1-x, press the Resume softkey to reconnect the call. Hang up the call.
Step 8
From any of your IP phones, call an invalid (nonexisting) directory number. You
should hear this annunciator announcement: Your call cannot be completed as
dialed. Please consult your directory and call again or ask your operator for
assistance. This is a recording.
Step 9
Hang up.
Pre-Lab Task 2: Deactivating MGCP Gateway and Release E1

Resources
This section deactivates the MGCP gateway and releases its E1 resources. This is needed for
two reasons. The first reason is to cause the PSTN to route inbound calls to your HQ via H.323
gateway. This will be used to test the media transcoding. The second reason is to make the E1
DSP resources available for transcoding services. The MGCP gateway functionality is not
required in your cluster any more.
Step 1
Connect to your MGCP gateway HQ-x and enter the configuration mode.
Step 2
Shut down the T1/E1 controller.
Step 3
Shut down the voice-port associated with the T1/E1 controller.
Step 4
Deactivate the MGCP by entering the no mgcp command.
Step 5
Deactivate the communication to call-agent by entering the no ccm-manager mgcp

command.
Step 6
Remove the only dial peer using the MGCP application.
Step 7
Remove the PRI backhauling from the D-channel serial interface by entering this
sequence of commands (this is E1 example):
interface serial module/port:15
no isdn bind-l3 ccm-manager
exit
Lab Guide
117
Step 8
From the T1/E1 controller, remove the pri-group by using the command no prigroup timeslot 1-31 (this is E1 example).
Step 9
Try to call to your HQ IP phones from the other cluster. Because T1/E1 PRI is
deactivated, the call is routed through your H.323 gateway. The call setup takes
longer, but it should get through.
Note
This functionality is not common in standard PSTN networks, and it was implemented only
to fit this lab environment.
Task 1: Creating the Requirement for the Transcoding Services

Currently, your H.323 gateway uses G.729 codec (the default). The IP phones also support this
codec, and therefore the connectivity was established. All devices have used the same region
(Default). This region allows G.711 as the maximum codec (therefore also G.729).
You will configure the H.323 gateway to use only the G.711 codec, instead of G.729. All IP
phones will still be able to connect to the PSTN through H.323 gateway because they support
G.711 in addition to G.729, and the only region being used allows G.711 as the maximum
codec.
To create the need for transcoding services, you will configure your HQ IP phones to be in a
different region (LowBW) with a maximum codec of G.729. It denies these IP phones from
using G.711 when connecting to the H.323 gateway (PSTN);therefore, set the requirement for
transcoding services between G.729 and G.711.
Activity Procedure
Step 1
Connect to your H.323 gateway BR-x.
Step 2
Change the codec used for all VoIP calls to G.711 by entering this sequence of
commands (for fist node and subsequent node dial peers):
dial-peer voice <tag>
codec g711alaw
end
118
Step 3
Verify that this change has not impacted the reachability between your IP phones
and PSTN. From any of your HQ IP phones, place a call to the Branch IP Phone4-x
via PSTN. Dial 9-153x5553004
Step 4
In Cisco Unified CallManager Administration, choose System > Region, and click
Add New to open the Region Configuration page.
Step 5
For the Region Name, enter LowBW, and then click Next.
Step 6
Select the LowBW region in the Regions pane and choose G.729 from the Audio
Codec drop-down menu. Click Save.
Step 7
Select the Default region in the Regions pane, and choose G.729 from the Audio
Codec drop-down menu. Click Save.
Step 8
Choose System > Device Pool, and click Find to list all existing Device Pools.
Step 9
Click the Copy icon in the right-most column of the only Default_CMLocal_DP
device pool. This opens the Device Pool Configuration page for that new copy.
Step 10
Change the Device Pool Name to LowBW_DP, and change the Region from the
Default to LowBW. Click Save.
Step 11
Choose Device > Phone, and click Find to list all available phones.
Step 12
In the list shown, click your first HQ IP phone (Phone1-x) to open its Phone
Configuration page.
Step 13
Change the Device Pool to LowBW_DP and click Save. When the pop-up window
appears, click OK, you will reset the phone later.
Step 14
Similarly, assign your second and third HQ IP phones (Phone2-x and Phone3-x) to
the LowBW_DP device pool.
Step 15
Make sure that your Branch IP phone (Phone4-x) is in the Default_CMLocal_DP

device pool. If not, assign it there.
Step 16
Click Go next to the Back to Find/List to see the list of available IP phones and
check if the Device Pool column reflects the changes. Only your Phone4-x should
remain in the device pool Default_CMLocal_DP.
Step 17
Click Select All, and then click the Reset Selected button. Click Reset on the new
pop-up window, and then click Close. The third-party SIP phone Phone3-x must be
reset manually.
When both pods are finished, try to call to either Phone1-x or Phone2-x from the other pod
via PSTN, and vice versa. The called phone will ring, but when you try to answer that call,
the codec negotiation for the media fails. You will hear the fast busy tone. You can also
verify this by entering the command show dial-peer voice <tag> | incl Last on your
H.323 gateway BR-x. The command output shows you the Last Disconnect Cause-Code
(2F) and the text no resource (47).
Note
When the call is originated by the other pod via PSTN, the call is set up, but you will hear a
busy tone. When the call is originated from the other side (your own IP phone), the call is not
even set up, and you hear the busy instantly. The signaling FXS/FXO circuitry commits the
call prematurely, while the call setup is not fully completed. This is the standard behavior of
analog lines.
Try also calling between any of your IP phones. This should be successful because the IP
phones use the G.729 codec, whereas the H.323 gateway, where the PSTN is attached, is
being forced to the G.711 codec only.
Task 2: Configuring Hardware Media Resources

In this task, you will configure hardware media resources in the Cisco Unified CallManager
database.
Lab Guide
119
Activity Procedure
Prepare Router DSP Resources
Step 1
Connect to your HQ-x. This will be your transcoder router. Enter the configuration
mode.
Step 2
Make the router DSP resources available for the transcoder. Enter this sequence of
commands:
voice-card 1
dsp services dspfarm
exit
sccp local FastEthernet0/0.x01 (where x is your pod number)
sccp
sccp ccm <Your First Node IP Address> priority 1
dspfarm transcoder maximum sessions 3
dspfarm confbridge maximum sessions 3
dspfarm
end
Configure Cisco Unified CallManager to Use the DSP Resources

Step 3
Discover the current MAC address on the FastEthernet0/0 interface of your
transcoder router and record it on the line that follows the example. Enter the
command show interface fastethernet0/0 to see the output similar to this example.
Make note of the MAC address shown on your output.
FastEthernet0/0 is up, line protocol is up
Hardware is AmdFE, address is 000f.34d9.0d00 (bia
000f.34d9.0d00)
(output truncated)
Your transcoder router MAC address: _________________________________
120
Step 4

choose Media Resources > Conference Bridge to open the Find and List
Conference Bridges page.
Step 5
Click Add New.
Step 6
From the Conference Bridge Type drop-down menu, choose Cisco IOS Conference
Bridge.
Step 7
For the Conference Bridge Name, enter the name CFB<MAC >, where <MAC> is
the MAC address of your Conference Bridge (transcoder) router taken in the earlier
step, without the dots. The name, for instance, based on the output in the earlier step,
looks like CFB000F34D90D00.
Step 8
For the Description, enter the Conference Bridge.
Step 9
Step 10
Choose Hub_None for Location.
Step 11
Click Save.
Step 12
Click Reset, then Reset on the dialog window, and Close.
Configure Transcoding Services

Step 13
Choose Media Resources > Transcoder, and then click Add New to open the
Transcoder Configuration page.
Step 14
From the Transcoder Type drop-down menu, choose Cisco IOS Media
Termination Point. This transcoder type supports Cisco IOS routers used for
transcoding services.
Step 15
For the Description, enter the IOS Transcoder.
Step 16
For the Device Name, enter the name MTP<MAC >, where <MAC> is the MAC
address of your transcoding router taken earlier. The name would look like
MTP000F34D90D00, for instance.
Step 17
Step 18
Click Save.
Step 19
Reset the Transcoder by clicking Reset, then Reset on the dialog window, and
Close.
Connect to your Bridge/Transcoder router HQ-x and enter the command show sccp. Verify
that both the Transcoding Oper State and the Conferencing Oper State are ACTIVE
and that the TCP Link Status is CONNECTED, as visible on this example output. If they
are not up, enter the configuration mode and reactivate the SCCP by entering the command
no sccp. Wait a few seconds and activate it again by entering the sccp command.
SCCP Admin State: UP
Gateway IP Address: 10.1.1.101
Switchover Method: IMMEDIATE, Switchback Method: GUARD_TIMER
Switchback Guard Timer: 1200 sec, IP Precedence: 5
Max Supported MTP sessions: 0
User Masked Codec list: None
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.1.1.1, Port Number: 2000
TCP Link Status: CONNECTED
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.1.1.1, Port Number: 2000
TCP Link Status: CONNECTED
Call Manager: 10.1.1.1, Port Number: 2000
Priority: 1, Version: 3.1 or Higher
Lab Guide
121
Task 3: Configuring Meet-Me Directory Numbers

In this task, you will configure a Meet-Me directory number range so that you can use the
Meet-Me Conference feature.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Meet-Me

Number/Pattern.
Step 2
Click Add New to open the Meet-Me Number/Pattern Configuration page.
Step 3
In the Directory Number or Pattern field, enter 250X.
Step 4
In the Description field, enter Meet-me DN.
Step 5
Select Executive_PT for the Partition.
Step 6
Click Save.
From any of your Cisco IP phones, you can initiate a Meet-Me conference by taking the IP
phone off hook (click the NewCall softkey), then pressing the softkey more, and afterward
pressing the softkey MeetMe and dialing the number from the Meet-Me conference range
2500-2509. The IP phone should indicate the called party To Conference.
After the Meet-Me conference is established, from another IP phone within your cluster,
simply dial the Meet-Me directory number (the same you used for the Meet-Me initiation)
to join the Meet-Me conference from another IP phone.
Caution
Do not press the MeetMe softkey on the second IP phone because this key is used by the
conference controller only to initiate the Meet-Me conference.
Add to the Meet-Me conference a third IP phone.
Disconnect all conferenced IP phones after the test.
Task 4: Configuring Media Resource Groups and Media

Resource Group Lists
In this task, you will configure MRGs and MRGLs and apply the MRGLs to devices to allow
them access to media resources.
Activity Procedure
122
Add and Configure Media Resource Groups

Step 1
In Cisco Unified CallManager Administration, choose Media Resource > Media
Resource Group, and then click Add New to open the Media Resource Group
Configuration page.
Step 2
In the Media Resource Group Name field, enter Software_MRG.
Step 3
For the Description, enter Software type media resources.
Step 4
From the Available Media Resources pane, highlight all software resources. These
are the resources starting with MTP and CFB<_digit>, but not the CFB<MAC>.
These are the hardware resources. Also exclude the MOH and the ANN
(annunciator). Move the highlighted resources into the Selected Media Resources
pane by clicking the down arrow between the panes.
Step 5
Click Save.
Step 6
By repeating the previous five steps, add the MRGs from the table.
Media Resource Groups

MRG Name
Description
Selected Media Resources
Hardware_MRG
Hardware resources
CFB<MAC>
MTP<MAC>
MOH_MRG
MOH resources
MOH_<digit>
CFB_MRG
Conference bridges
CFB_<digit>
CFB<MAC>
XCODE_MRG
XCODERS (MTP and XCODE)
MTP_<digit>
MTP<MAC>
Note
An annunciator is considered a media device. It can also be included in MRGs if you want to
control which annunciator resource is selected for use by phones and gateways.
Click Go next to the Back to Find/List, then click Find, and verify if you can see all five
MRGs entered.
Add and Configure Media Resource Group Lists

Step 7
Choose Media Resource > Media Resource Group List. Click Add New to open
the Media Resource Group List Configuration page.
Step 8
In the Name field, enter All_MRGL.
Step 9
Move all the MRGs added in the previous section of this task from the Available
Media Resource Groups pane to the Selected Media Resource Groups pane, but get
them prioritized in this order: Hardware_MRG, Software_MRG, CFB_MRG,
MOH_MRG, and XCODE_MRG.
Step 10
Click Save.
Lab Guide
123
Note
Step 11
The approach that is provided here represents only one of many ways to organize MRGs
and MRGLs.
Repeat the previous four steps to add the two MRGLs from the table and associate
them with the MRGs.
Caution
Remember that the software and hardware MRGs both contain conference bridge
resources.
Media Resource Group Lists

MRGL Name
MRGs to Assign
NOCFB_MRGL
All MRGs added in this task that do not contain conference

bridges.
Hint: The Hardware_MRG and Software_MRGs contain
conference bridges.
MOH_MRGL
MOH_MRG
Click Go next to the Back to Find/List, then click Find, and verify if you can see all three
added MRGLs in the list.
Apply Media Resource Group Lists to Devices

Step 12
Choose System > Device Pool, and then click Find to list the available device
pools.
124
Step 13
Choose Default_CMLocal_DP from the list shown to open the Device Pool
Configuration page.
Step 14
From the Media Resource Group List drop-down menu, choose All_MRGL.
Step 15
Click Save.
Step 16
Click Reset, then click Reset on the dialog window, and click Close.
Step 17
Choose Device > Phone, and then click Find to list all available IP phones.
Step 18
Select your first HQ IP Phone1-x to open its Phone Configuration page.
Step 19
From the Media Resource Group List drop-down menu, choose NOCFB_MRGL,
and then click Save and OK on the pop-up window.
Step 20
Reset your first IP Phone1-x.
Step 21
Click Go next to the Back to Find/List. In the list shown, select your second HQ IP
Phone2-x and assign it similarly to the All_MRGL Media Resource Group List. Do
not forget to reset the IP Phone2-x when the page loading is completed.
From your first IP Phone1-x (assigned to the NOCFB_MRGL), try to establish a Meet-Me
conference call. The IP phone should not be able to establish the conference and should
display No Conference Bridge.
From your second IP Phone2-x (assigned to the All_MRGL), successfully establish a MeetMe conference with the other IP phones in your cluster (including the Phone1-x). Hang up
the conference.
Task 5: Configuring Audio Sources for MOH

In this task, you will upload sample audio files to the Cisco Unified CallManager MOH. Then
you will select these files to be used by a device as MOH audio sources.
Activity Procedure
Cleanup Before MOH
Step 1
Assign all your IP phones back to the original device pool Default_CMLocal_DP
and, for all your IP phones, select All_MRGL for the Media Resource Group List.
Step 2
Reset all your IP phones for the changes to take effect. The third-party SIP phone
must be reset manually.
Add MOH Audio Sources

Step 3
Choose Media Resource > MOH File Management.
Step 4
The new page opens, with the list, where only the single SampleAudioSource file
appears.
Step 5
Most standard .wav and MP3 files serve as valid input audio source files. Your
instructor has provided two .wav audio files for the MOH, which are stored on your
PC on the desktop in the CIPT1 folder. Click the Upload File button and, in the new
dialog window, click the Browse button. Then navigate to the first file, select the
file, and click Open.
Step 6
Click Upload. Wait until the file gets uploaded. (The larger file,
WelcomeHome.wav, can take about 5 minutes.). Then click Close. The file appears
in the list of audio sources.
Note
Registered Cisco.com users with software download privileges can access the full set of
music and prompts for Cisco Unified CallManager MOH at the Voice Software index page
under Cisco Unified CallManager 3.3 at http://www.cisco.com/kobayashi/sw-center/swvoice.shtml.
Step 7
Upload also the second .wav audio file using the same procedure.
Step 8
Choose Media Resources > Music On Hold Audio Source and click Add New.
Step 9
From the MOH Audio Stream Number drop-down menu, choose 2.
Lab Guide
125
Step 10
From the MOH Audio Source File drop-down menu, choose the audio file
Thank_you_for_waiting.
Step 11
Check the Play continuously (repeat) check box. Click Save.
Step 12
Repeat the last four steps to add a new audio source from the second source file
(WelcomeHome) with the MOH Audio Stream Number 3.
Step 13
Click Go next to the Music On Hold Server Configuration, then click Find, and
reset the MOH server.
Assign MOH to Device Pool

Step 14
Choose System > Device Pool, and then click Find.
Step 15
From the list shown, select the device pool Default_CMLocal_DP.
Step 16
From the Network Hold MOH Audio Source drop-down menu, choose the audio
source 2 - Thank_you_for_waiting.
Step 17
From the User Hold Audio Source drop-down menu, choose the audio source
3 - WelcomeHome.
Step 18
Click Save.
Step 19
Click Reset, and click Reset on the dialog window, and then click Close.
126
Place a call within your cluster and put the call on hold by pressing Hold softkey (user
hold). WelcomeHome (piano music) is streamed. Hang up.
Place a call within your cluster and transfer the call by pressing Transfer softkey (network
hold). Thank_you_for_waiting (womans voice) is streamed. Hang up.

Activity Objective
In this activity, you will configure and use the Call Park, Call Pickup, Cisco Call Back, Barge,
and Privacy features. After completing this activity, you will be able to meet these objectives:
Configure a Call Park number, park calls on one phone, and pick them up on another
Configure a Call Pickup group and assign the Call Pickup group to telephones
Configure a Cisco Call Back softkey template and assign it to a device pool within your
cluster
Configure the Barge feature to barge into a call on a shared line
Assign a Privacy button to a line and invoke the Privacy feature to prevent other users of a
shared line from viewing call information or barging into the call
Visual Objective

Configuring User Features
Phone2-x
Phone1-x
Barge
Ln-2: 2099
Privacy
Phone3-x
Pickup: 6686
Pickup: 6685
CCM1-x
HQ-x
Park: 266X
BRANCH-x
HQ-x
BR-x
Phone4-x
Barge
Ln-2: 2099
Privacy
Pickup: 6686
POD-x
CIPT1 v5.0-25
Required Resources
Lab Guide
127
Command List
Job Aids
Task 1: Configuring Call Park

In this task, you will configure a Call Park number and then use the Call Park feature within
your cluster.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Call Park.
Step 2
Click Add New to open the Call Park Configuration page.
Step 3
Enter 266X for the Call Park Number/Range field. In the Description field, enter
Call Park Range.
Step 4
Select Executive_PT for Partition.
Step 5
Choose CM_<your first node> to indicate the Cisco Unified CallManager to which
the Call Park is assigned. Click Save.
Establish a call between any pair of your Cisco IP phones and, on one of them, park the call
by pressing the Park softkey. The IP phone from which the call was parked goes on hook,
while the other displays the message To Park Number: 266x. Take note of the Call Park
number.
Take your third IP phone off hook and pick up the parked call by dialing the Call Park
number.
Disconnect the call.
Task 2: Configuring Call Pickup

In this task, you will configure a Call Pickup Group and assign it to the IP phones in your
cluster, then use the Call Pickup feature within your cluster.
Activity Procedure
128
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Call Pickup
Group.
Step 2
Click Add New to open the Call Pickup Group Configuration page.
Step 3
Enter Pickup-6685 for the Call Pickup Group Name.
Step 4
Enter 6685 as the Call Pickup Group Number and, in the Description field, type Call
Pickup Group 6685.
Step 5
Select Executive_PT for Partition.
Step 6
Click Save.
Step 7
Add another Call Pickup Group:
Call Pickup Group Name: Pickup-6686
Call Pickup Number: 6686
Description: Call Pickup Group 6686
Partition: Executive_PT
Assign Call Pickup to a Directory Number

Step 8
Choose Device > Phone and click Find.
Step 9
Select your first IP Phone1-x from the list to open its Phone Configuration page.
Step 10
In the left page column, click Line [1], and then scroll down to see the Call
Forward and Pickup Settings section.
Step 11
Choose 6685 from the Call Pickup Group drop-down menu.
Step 12
Click Save.
Step 13
Reset the IP phone.
Step 14
Choose Device > Phone, click Find, and repeat the procedure for your Phone2-x
and Phone4-x, but this time, assign them to the Call Pickup Group 6686.
Remember to reset all three modified IP phones at the end.
From your Phone1-x, call your Phone4-x , but do not answer the call. From your Phone2-x
(which should be in the 6686 Call Pickup Group), go off hook, press the more softkey, and
then press the PickUp softkey. Press the Answer softkey to pick up the call. The call
should be rerouted. Hang up the call.
From one of the two IP phones that have been assigned to Call Pickup Group 6686 (either
Phone2-x or Phone4-x), call the Phone1-x (assigned to 6685). Do not answer the call. From
the idle IP phone (which should be in the 6686 Call Pickup Group), go off hook, press the
more softkey, and press the GPickup softkey. Enter 6685, and then press the Answer
softkey. The call should be rerouted. Hang up the call.
Note
If the IP phone that you use to try to pick up a call is not in any Call Pickup Group, you will
receive an error message on the IP phone, and the call will not be rerouted.
Lab Guide
129
Task 3: Configuring Cisco Call Back

In this task, you will configure a Cisco Call Back softkey template and assign it to a device
pool within your cluster. The Cisco Call Back feature allows you to receive call-back
notification on your Cisco IP phone when a called-party line becomes available.
Activity Procedure
Complete these steps.
Step 1
In Cisco Unified CallManager Administration, choose Device > Device Settings >
Softkey Template to open the Find and List Softkey Templates window. Click
Find.
Step 2
Click the Copy icon next to Standard User template to create a copy of this
template.
Step 3
Rename the template copy to Standard CallBack User and change the Description
field to CallBack Softkey Template. Click Save.
Step 4
Click Go next to the Configure Softkey Layout in the upper-right corner of the
window.
Step 5
On Hook should be highlighted in the drop-down menu. Select Call Back

(CallBack) in the Unselected Softkeys pane and use the right arrow to move it to the
Selected Softkeys pane. Move Call Back (CallBack) to the top of the list. Click
Save.
Note
Step 6
Choose Ring Out from the Select a call state drop-down menu.
Step 7
Select Call Back (CallBack) in the Unselected Softkeys pane and click the right
arrow to move it to the Selected Softkeys pane. Move it to the top of the list. Click
Save.
Note
130
The CallBack softkey in the On Hook call state enables you to check the status of a callback request that was previously initiated or to cancel a call-back request that is no longer
desired.
The CallBack softkey in the Ring Out state is used during a call-back request when a call is
in the ringing or busy state or when the call has been forwarded to voice mail.
Step 8
Choose System > Device Pool, and then click the Find button to bring up the list of
device pools.
Step 9
Choose Default_CMLocal_DP from the list of device pools.
Step 10
In the Softkey Template field, choose Standard CallBack User, and then click
Save.
Step 11
Reset the device pool for the change to take effect on your IP phones.
Temporarily, change Line [1] on the Phone1-x to accept maximally one call. Open its IP
phone directory number configuration page and change both the Maximum Number of
Calls field and the Busy Trigger field to 1. Click Save and reset the IP phone.
Take the Phone1-x off hook. From other Cisco IP phone, call the busy Phone1-x, and then
press the CallBack softkey. Hang up the busy IP phone. Cisco Unified CallManager should
notify you that the line is available and offer you the option of dialing the number.
Call any registered directory number in your cluster. While it is ringing, press the CallBack
softkey. On the IP phone that you called, go off hook, then on hook again. Cisco Unified
CallManager should notify you that the line is available and offer you the option to dial the
number.
Return the field Maximum Number of Calls to 4 and the field Busy Trigger to 2 on
Phone1-x again. Click Save and reset the IP phone.
Cleanup
Task 4: Configuring Barge with Built-In Conference Bridge

In this task, you will configure and use the Barge feature. Barge allows a user to be added to a
connected call that is made to or from another device on a shared line. To use Barge, you must
first configure a shared line appearance and enable the built-in Barge cluster-wide parameter.
Activity Procedure
Create a Shared Line
Step 1
Choose Device > Phone, and click Find to show the list of available IP phones.
Step 2
Select your second IP Phone1-x.
Step 3
Click Line 2 Add a new DN.
Step 4
In the Directory Number field, enter 2099, and choose Executive_PT for the Route
Partition.
Step 5
Click Save. Reset the Phone1-x.
Step 6
Choose Device > Phone, click Find, and choose your IP Phone4-x.
Step 7
Click the Modify Button Items button and click OK on the pop-up window.
Step 8
On the new page, move Add a new SD from the Associated Items to the Unassigned
Associated Items pane, using the right arrow between the panes.
Step 9
Move Line [2] Add a new DN from the Unassigned Associated Items to the
Associated Items pane, using the left arrow between the panes.
Step 10
Click Save and Close.
Lab Guide
131
Step 11
Note
Assign the Phone4-x Line [2] the same directory number 2099 as to your Phone1-x.
Select Executive_PT for the Route Partition. Notice that the Associated Devices
pane lists the two IP phones instead of one. This indicates the shared line.
Depending on the IP phone model used for the Phone4-x, the modification of button layout
may not be needed to show the second line.
Step 12
Reset the Phone4-x.
Step 13
Take the Phone1-x off hook and press the Line 2 button. On the Phone4-x (which
shares the line), notice that the IP phone Line 2 icon also goes off hook. Hang up
the line.
Configure Barge
Step 14
Choose System > Service Parameters.
Step 15
From the Server drop-down, choose your first node.
Step 16
From the Service drop-down, choose Cisco CallManager and wait until the page
refreshes.
Step 17
Scroll down to locate the Clusterwide Parameters (Device - Phone) section. Change
the setting of the Builtin Bridge* Enable to On, and click OK when the pop-up
window appears.
Step 18
Scroll down to locate the Clusterwide Parameters (Feature - General) section.

Ensure that the Party Entrance Tone* parameter is set to True. This parameter will
enable or disable a tone on the telephone of the Barge target when another user is
barging into the call.
Step 19
Click Save. Wait until the page refreshes.
Note
Both the Standard User and Standard Feature softkey templates contain the Barge softkey.
Because the Standard CallBack User softkey template that you are using is based on the
Standard User, it also contains the Barge softkey.
132
From your Phone2-x (IP phone without the shared line appearance), dial 2099. Answer the
call on one of the ringing IP phones.
On the other Cisco IP phone with the shared line appearance that has not answered the call,
go off hook (New Call softkey) and press the Line 2 button (shared line). Watch for the
Barge softkey to appear at the bottom of IP phone display and press it to join the
conversation. The IP phone displays the To Barge message.
On the IP phone with the shared line appearance, but not the one showing To Barge,
hang up the call. The Barger (the Barging IP phone) and the original caller will still be
connected.
Hang up both connected IP phones.
Task 5: Configuring Privacy

In this task, you will assign a Privacy button to a line so that you can invoke Privacy to prevent
other users of a shared line from viewing call information or using Barge to join the call.
Activity Procedure
Step 1
Choose System > Service Parameters.
Step 2
From the Server drop-down menu, choose your first node. From the Service dropdown menu, choose the Cisco CallManager service.
Step 3
Scroll the page down to locate the Clusterwide Parameters (Device - Phone) section.
Ensure that the Privacy Setting* is True.
Create a New Phone Button Template

Step 4
Choose Device > Device Settings > Phone Button Template to open the Phone
Button Template window. Click Find to list all button templates.
Step 5
To create a copy of the Standard 7961 SIP phone button template, click the Copy
icon next to the Standard 7961 SIP phone button template.
Step 6
Rename the template to Privacy 4-1 7961 SIP and click Save.
Step 7
Change the line and speed-dial settings as follows:
Button 3: Line
Button 4: Line
Button 5: Privacy
Button 6: Speed Dial
Step 8
Click Save.
Step 9
Choose Device > Phone, and then click Find.
Step 10
Select the Phone1-x (configured earlier with the shared line).
Step 11
Choose Privacy 4-1 7961 SIP from the Phone Button Template drop-down menu.
Step 12
Click Save and reset the IP phone.
Step 13
Choose Device > Device Settings > Phone Button Template to open the Phone
Button Template window. Click Find to list all button templates.
Step 14
To create a copy of the Standard 7960 SCCP phone button template, click the Copy
icon next to the Standard 7960 SCCP phone button template.
Step 15
Rename the template to Privacy 4-1 7960 SCCP and click Save.
Step 16
Change the line and speed-dial settings as follows:
Button 3: Line
Button 4: Line
Button 5: Privacy
Button 6: Speed Dial

Lab Guide
133
Step 17
Click Save.
Step 18
Choose Device > Phone, and then click Find.
Step 19
Select the Phone4-x (the other IP phone with the shared line).
Step 20
Choose Privacy 4-1 7960 SCCP from the Phone Button Template drop-down menu.
Step 21
IP phones with Privacy enabled indicate the Privacy feature at their displays.
From your Phone2-x (the one without the shared line appearance), dial 2099. Answer the
call on one of the ringing IP phones. On the other IP phone with the shared line appearance,
go off hook and attempt to join the call using Barge. You should not be successful. With
Privacy enabled (a black circle in the middle of the Privacy icon), neither the Barge softkey
nor the calling name and number are displayed on the telephone of the partner after the call
is answered.
While the call is still active, toggle the Privacy softkey on the IP phone with an active call.
This action disables Privacy on the line. Notice that the other IP phone with the shared line
now shows an off-hook icon. You should now be able to join the call from that other IP
phone using Barge.
Cleanup
134
Step 1
To the Phone1-x, assign back the Standard 7961 SIP Phone Button Template.
Reset the Phone1-x.
Step 2
To the Phone3-x, assign back the Standard 7960 User Phone Button Template.
Reset the Phone3-x.

Extension Mobility
Activity Objective
In this activity, you will configure and use the Cisco Unified CallManager Extension Mobility
feature. You create the device profiles for the Cisco Unified CallManager Extension Mobility
service in Cisco Unified CallManager and associate the user device profile with the username
mperry. Then, you will log in as user mperry to an IP phone that has subscribed to the Cisco
Unified CallManager Extension Mobility service, and the IP phone will assume the user device
profile. After completing this activity, you will be able to meet these objectives:
Add a user to Cisco Unified CallManager
Add the Cisco Unified CallManager Extension Mobility Service in Cisco Unified
CallManager
Set the login service parameters in the Cisco Unified CallManager Service Parameters
Configuration
Add a device profile default for the Cisco Unified IP Phone 7940
Create a device profile for a user, with attributes such as name and description
Associate a user device profile to the user for Cisco Unified CallManager Extension
Mobility
Subscribe to the Cisco Unified CallManager Extension Mobility feature
Lab Guide
135
Visual Objective

Unified CallManager Extension Mobility
Phone1-x
Phone2-x
Phone3-x
mperry logs in
DN: 2555
CCM1-x
Extension
Mobility
Service
HQ-x
HQ-x
BRANCH-x
BR-x
Phone4-x
POD-x
CIPT1 v5.0-26
Required Resources
Cisco Unified IP Phone 7940
Command List
Job Aids
136
Task 1: Adding a User

In this task, you will add a new user that will be used later for Extension Mobility.
Activity Procedure
Step 1
In Cisco Unified CallManager Serviceability, choose User management > End

User, and then click Add New.
Step 2
Add a user with these parameters:
Step 3
User ID: mperry
Password: cisco
PIN: 12345
Last Name: Perry
First name: Mark
Click Save
The new user appears in the list when you choose User management > End User and
click Find.
Task 2: Adding the Cisco Unified CallManager Extension

Mobility Service
In this task, you will configure the Cisco Unified CallManager Extension Mobility service.
Activity Procedure
Step 1
In Cisco Unified CallManager Serviceability, choose Tools > Service Activation.
Step 2
Choose your first node server.
Step 3
Check the check box next to the Cisco Extension Mobility service.
Step 4
Check the check box next to the Cisco CallManager Cisco IP Phone Services
service.
Step 5
Click Save, and click OK on the pop-up window.
Step 6
In Cisco Unified CallManager Administration, choose System > Enterprise

Parameters.
Step 7
Scroll down to the Phone URL Parameters section and, in this section, replace all
server names (CCM1-1) with the IP address (10.x.1.1, where x is your pod number),
leaving the rest of URL string as is.
Step 8
Click Save.
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137
Step 9
Phone Services and click Add New.
Step 10
The Cisco IP Phone Service Configuration window opens. In the Service Name
field, enter Extension Mobility for the name of the service. This name is displayed
on the IP phone when the user presses the Services button.
Step 11
In the Service Description field, enter Extension Mobility.
Step 12
In the Service URL field, enter this URL with IP address of the Cisco Unified
CallManager first node server:
http://10.x.1.1:8080/emapp/EMAppServlet?device=#DEVICENAME# (where x
is your pod number)
Note
Step 13
This URL is case-sensitive, so make sure that you enter it exactly as described.
Click Save.
You can verify that the Cisco Unified CallManager Extension Mobility service is listed in
the Cisco IP Phone Services Configuration window when you choose Device > Device
Settings > Phone Services and then click Find.
Task 3: Setting the Cisco Unified CallManager Extension

Mobility Service Parameters
In this task, you will set the service parameters in the Cisco Extension Mobility Parameters
configuration window to accomplish these objectives:
Define a system-wide maximum login time
Define the multilogin behaviorthat is, whether you allow the user to log in to more than
one device at a time
Activity Procedure
138
Step 1
In Cisco Unified CallManager Administration, choose System > Service

Parameters.
Step 2
From the Server drop-down menu, choose your first node server.
Step 3
From the Service drop-down menu, choose the Cisco Extension Mobility service.
Step 4
At the Enforce Maximum Login Time, choose True to specify a system-wide

maximum time for logins. After the specified time, the system automatically logs
out the device.
Step 5
As the Maximum Login Time, specify 2 minutes. (Enter 2 minutes as 0:02.)
Note
In a production deployment, you would normally set a maximum login time that corresponds
to a normal workday, such as 9 or 10 hours.
Step 6
In the Multiple Login Behavior field, choose Auto Logout. The automatic logout
behavior specifies that after a user logs in to a second device, Cisco Unified
CallManager automatically logs the user out of the first device.
Step 7
Click Save.
The Cisco Extension Mobility Service Parameters page refreshes with the entered values.
Task 4: Creating the Device Profile Default for the Cisco Unified
IP Phone 7940
In this task, you will add a device profile for the Cisco Unified IP Phone 7940.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, go to Device > Device Settings >

Default Device Profile.
Step 2
From the Device Profile Type drop-down menu, choose Cisco 7940 and click Next.
Step 3
Leave the User Hold Audio Source field as None. If you do not choose an audio
source, Cisco Unified CallManager uses the audio source that is defined in the
device pool or, if the device pool does not specify an audio source ID, the system
default.
Step 4
From the User Locale drop-down menu, choose English United States.
Step 5
Step 6
From the Softkey Template drop-down list, choose Standard Feature.
Step 7
Click Save.
Task 5: Creating the Device Profile for a User

The user device profile contains attributes such as name, description, phone template, add-on
modules, directory numbers, subscribed services, and speed-dial information. In this task, you
will add a default device profile for a new user of Cisco Unified CallManager
Extension Mobility.
Activity Procedure
Step 1
Device Profile and click Add New.
Step 2
From the Device Profile Type drop-down menu, choose Cisco 7940 and click Next.
Lab Guide
139
Step 3
In the User Device Profile Name field, enter rperry_EM. (You can use any word or
phrase that describes this particular user device profile.)
Step 4
In the Phone Button Template field, choose Standard 7960.
Step 5
Enter the Device Profile Name mperry-em.
Step 6
Leave the User Hold Audio Source field as None.
Step 7
From the User Locale drop-down menu, choose English United States.
Step 8
Step 9
From the Softkey Template drop-down list, choose Standard Feature.
Step 10
From the Login User Id drop-down menu, choose the user mperry.
Step 11
Click Save.
Step 12
When new page appears, click the Line [1] link in the left column of the page.
Step 13
In the Directory Number field, enter 2555.
Step 14
Choose Executive_PT from the Route Partition drop-down menu.
Step 15
Click Save.
The Directory Number Configuration window shows an association between the profile
mperry-em and the directory number 2555.
Task 6: Associating a User Device Profile with a User

In this task, you will associate a user device profile with the user for Cisco Unified
CallManager Extension Mobility.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose User Management > End

User and click Find.
Step 2
From the list of users shown, click mperry to open the User Configuration window.
Step 3
Under Extension Mobility, highlight the mperry-em profile in the Available

Profiles pane and move it to the Controlled Profiles pane by clicking the down arrow
between the panes.
Step 4
Click Save.
140
The mperry-em profile is assigned to the user mperry.
Task 7: Subscribing Cisco IP Phones to Cisco Unified

CallManager Extension Mobility
In this task, you will subscribe to the Cisco Unified CallManager Extension Mobility feature.
Activity Procedure
Step 1
Find.
Step 2
Choose your Phone2-x (where x is your pod number) as the IP phone that you want
to configure.
Step 3
When the Phone Configuration window opens, choose Subscribe/Unsubscribe

Services from the Links drop-down menu in the upper-right corner.
Step 4
Click the Go button next to the Related Links drop-down menu.
Step 5
The Subscribe Cisco IP Phone Services dialog window opens. Choose Extension
Mobility from the Select a Service drop-down menu.
Step 6
Click Next.
Step 7
Click Subscribe, and then click Save.
Step 8
Close the dialog window.
Step 9
Scroll down to the bottom of the Cisco Unified CallManager Phone Configuration
window. Check the Enable Extension Mobility check box.
Step 10
From the Log Out Profile drop-down menu, choose Use Current Device Settings.
Step 11
On the Cisco IP phone where Cisco Unified CallManager Extension Mobility is enabled
(Phone2-x), press the Services button. Highlight Extension Mobility and press Select.
Log in as mperry, and the user profile after new IP phone registration appears (extension
2555). Recall that the PIN is 12345.
After 2 minutes, the IP phone automatically logs out and resumes the original profile and
directory number (2002).
Lab Guide
141
Lab 6-4: Configuring Presence and BLF

Functionality
Activity Objective
In this activity, you will configure a Presence-enabled speed dial on the Cisco Unified IP Phone
7961 to watch the device status of the other two phones in your pod. You will then assign each
phone directory number to different Presence groups, and allow one of the phones to see the
status of the other phone and block it from seeing the status of the other phone. Recall that
Presence groups apply to Presence-enabled call lists and not to the Presence-enabled speed
dials. This means that regardless of the Presence group subscription permissions, the Cisco
Unified IP Phone 7961 with BLF speed dial configured will always see the status of the other
phones. After completing this activity, you will be able to meet these objectives:
Configure Presence service parameters and enterprise parameters
Customize phone button templates for the Presence-enabled speed-dial buttons
Configure Presence-enabled speed-dial buttons for the phone
Configure Presence groups
Visual Objective

Presence for BLF Functionality
Phone1-x
Phone2-x
Executive
CIO
Block
subscription
CCM1-x
HQ-x
HQ-x
BRANCH-x
BR-x
Allow
subscription
Phone4-x
Manager
POD-x
142
CIPT1 v5.0-27
Required Resources
Three Cisco IP Phones (two Cisco Unified IP Phone 7940 phones and one Cisco Unified IP
Phone 7961)
Command List
Job Aids
Task 1: Enabling Presence Service Parameters and Enterprise

Parameters
Note
The order of the steps in this lab is different from the order presented in the Student Guide
and in the Cisco Unified CallManager documentation so that the learner can first enable
Presence on phones and then see the impact of adding Presence Groups.
In this task, you enable the Presence service parameters and enterprise parameters.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose System > Enterprise

Parameters and enable the BLF for Call Lists enterprise parameter.
Step 2
Click Save.
Step 3
Choose System > Service Parameters and select the Cisco CallManager service.
Step 4
Scroll down to find the Default Inter-Presence Group Subscription service parameter
and select Allow Subscription.
Step 5
Click Save.
The services required to support Presence-enabled call lists and inter-group Presence
subscriptions are enabled.
Lab Guide
143
Task 2: Creating Presence Phone Button Templates and

Applying Them to Phones
In this task, you will customize the standard phone button templates to include Presenceenabled speed-dial buttons and apply the templates to the Cisco Unified IP Phone 7961
(Phone1-x) and Cisco Unified IP Phone 7940 (Phone2-x and Phone4-x). The speed-dial BLF
buttons will be configured so that 2001 can call 2002 and 3004 using the BLF speed-dial button
and can see the status of either of these phones when they are on hook or off hook.
Activity Procedure
Create a Phone Button Template and Apply It to a Cisco Unified IP Phone 7961
Step 1
In Cisco Unified CallManager, choose Device > Device Settings > Phone Button
Template and click Find.
Step 2
Click the copy icon next to the Standard 7961 SIP template.
Step 3
Change the name of the template to BLF 7961 SIP and click Save.
Step 4
Create two Speed Dial BLF buttons at button positions 3 and 4.
Step 5
Click Save.
Step 6
Step 7
From the list, select the Phone1-x and assign it with the new phone button template
BLF 7961 SIP.
Step 8
Scroll the Phone Configuration page down and find the SUBSCRIBE Calling Search
Space drop-down menu. Choose Unlimited_Access_CSS.
Step 9
Click Save.
Step 10
Click 3 Add a new BLF SD in the left column of the Phone Configuration page.
Step 11
A new dialog window opens. From the Directory Number drop-down menu
applicable for the top Busy Lamp Field (number 1), choose 2002. Enter BLF 2002
in the corresponding Label field.
Step 12
Repeat the previous step for the second Busy Lamp Field (number 2). Choose
directory number 3004 and, for the Label, enter BLF 3004.
Step 13
Click Save, and close the dialog window by clicking Close button at the bottom of
the window.
Caution
Do not close the dialog window by clicking the [x] window control because the changes will
not be transferred to the Phone Configuration page.
Step 14
144
Reset the IP phone.
Create a Phone Button Template and Apply it to Cisco Unified IP Phone 7940
Phones
Step 15
In Cisco Unified CallManager, choose Device > Device Settings > Phone Button
Template and click Find.
Step 16
Click the copy icon next to the Standard 7940 SCCP template.
Step 17
Change the name of the template to BLF 7940 SCCP and click Save.
Step 18
Create one Speed Dial BLF button at the button position 2.
Step 19
Click Save.
Step 20
Step 21
From the list, select the Phone2-x and assign it the new phone button template BLF
7940 SCCP.
Step 22
Scroll the Phone Configuration page down and find the SUBSCRIBE Calling Search
Space drop-down menu. Choose Unlimited_Access_CSS.
Step 23
Click Save.
Step 24
Click 2 Add a new BLF SD in the left column of the Phone Configuration page.
Step 25
A new dialog window opens. From the Directory Number drop-down menu
applicable for the top Busy Lamp Field (number 1), choose 2001. Enter BLF 2001
in the corresponding Label field.
Step 26
Click Save and close the dialog window by clicking the Close button at the bottom
of the window.
Caution
Do not close the dialog window by clicking [x] window control, because the changes will not
be transferred to the Phone Configuration page.
Step 27
Repeat Steps 20 through 26 to apply the BLF 7940 SCCP phone button template to
Phone4-x and configure the phones second button position to monitor the status of
the directory number 2001.
Step 28
Reset the IP phones Phone2-x and Phone4-x.
The Presence-enabled speed dials with the correct labels appear on your IP phones.
When you press the Presence-enabled speed dials, they dial the correct destinations.
When subcribees are idle, the LED on the watchers Presence-enabled speed dials is off,
and the phone icon displays idle.
When subcribees are in a call, the BLF status on the watchers Presence-enabled speed
dials is red, and the phone icon associated with the speed dial displays busy.
Lab Guide
145
Task 3: Enabling Presence Groups

Currently, all phones are in the Standard Presence group. In this task, you create three Presence
groups: Executive Presence group (EXEC_PG), CIO Presence Group (CIO_PG), and Manager
Presence group (MGR_PG), such that the EXEC_PG can subscribe to the MGR_PG but not the
CIO_PG.
Then, you assign all phones at the device (phone) level to the EXEC_PG. At the line (directory
number) level, you assign directory number 2001 to the EXEC_PG, directory number 2002 to
the CIO_PG, and directory number 3004 to the MGR_PG. This configuration disallows the
Executive phone from subscribing to (watching) the CIO line. The Executive will not be able to
see the status of the CIO directory number when the Executive presses the Directories button to
retrieve the call history list. The Executive will still be able to see the CIO directory number in
call history list, but cannot see the CIO directory number status (busy or available).
Because the default inter-group Presence subscription is set to allow from Task 1, you must
explicitly disallow the Exec_PG to CIO_PG subscription. By default, the Exec_PG can see the
MGR_PG Presence-enabled call history list.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose System > Presence Group

and click Add New.
Step 2
Create three presence groups: Exec_PG, CIO_PG, and MGR_PG, with the
bidirectional relationships that follow. Click Save for each Presence Group to be
added, then edit presence groups and build the relationships.
Presence Group Relationships
146
Group
Group
Relationship
EXEC_PG
CIO_PG
Disallow subscription
EXEC_PG
MGR_PG
Allow subscription
Step 3
Assign device Phone1-x to the Exec_PG (Presence Group drop-down menu) and
save the configuration. Assign line 2001 (Presence Group drop-down menu) to the
Exec_PG, save the configuration, and reset the phone.
Step 4
Assign device Phone2-x to the Exec_PG and save the configuration. Assign line
2002 to the CIO_PG, save the configuration, and reset the phone.
Step 5
Assign device Phone4-x to the Exec_PG and save the configuration. Assign line
3004 to the MGR_PG, save the configuration, and reset the phone.
Step 6
Place some calls from line 2001 to the other two IP phones and from the other two
IP phones to line 2001.
Step 7
From IP phone 2001, observe the call history list (Directories button) while you
complete the remaining steps.
Step 8
Go off hook on line 3004 (MGR_PG) and, from the Executive IP phone (2001),
observe the Presence status of line 3004. It should change to busy.
Step 9
Place line 3004 (MGR_PG) back on hook and, from the Executive IP phone (2001),
observe the Presence status of line 3004. It should change to available.
Step 10
Go off hook on line 2002 (CIO_PG) and, from the Executive IP phone (2001),
observe the Presence status of line 2002. It should not change because the Executive
IP phone is not allowed to subscribe (watch) the CIO line.
The phone assigned to the Exec_PG (Phone1-x with directory number 2001) can see the
status of the MRG_PG line (directory number 3004) but cannot see the CIO_PG line
(directory number 2002).
The Presence-enabled speed-dial button on Phone1-x Executive phone can still monitor the
status of the CIO phone (directory number 2002). This is because the Presence-group
feature applies to call history lists but not Presence-enabled speed-dial buttons because the
latter are under administrative control.
Lab Guide
147

Activity Objective
In this activity, you will install and configure Cisco IP Communicator on your PC to place and
receive calls using your PC. Then you will enable IP phones to access sample IP phone
services. In this activity you will also configure and use Call Display Restrictions and CMC.
Prepare Cisco Unified CallManager for deploying Cisco IP Communicator, and install
Cisco IP Communicator on a client PC to place and receive calls using a PC
Use a sample services file to add Cisco IP phone services to the Cisco Unified CallManager
database for end users
Enable Call Display Restrictions so that call display information is restricted when guest
rooms call one another and presented to both the guest room and the lobby phone when
either calls the other
Add CMCs in Cisco Unified CallManager Administration, assign them to route patterns,
and then use CMCs to reach certain dialed numbers
Visual Objective

Configuring Additional Features
Phone1-x
Phone2-x
Room 1
Room 2
Display
Restrictions
PC-x
HTTP Srv
CMC requested:
852y-2001
852y-2002
CIPC
CCM1-x DN: 2004
HQ-x
HQ-x
BRANCH-x
BR-x
Display
Unrestricted
Phone4-x
Front Desk
POD-x
148
CIPT1 v5.0-28
Required Resources
IP WAN
Command List
Job Aids
Task 1: Configuring Cisco IP Communicator

In this task, you will add a user that will be associated with Cisco IP Communicator, run the
Cisco IP Communicator Administration Tool to enable automatic updates, and install the latest
Cisco IP Communicator phone load for devices that register to Cisco Unified CallManager.
Activity Procedure
Add the User
Step 1
In Cisco Unified CallManager Administration, choose User Management > End

User and click Add New.
Step 2
Add a new user with this information:
Step 3
User ID: kwong
User Password: cisco
PIN: 12345
Last Name: Wong
First Name: Kim
Click Save.
Prepare and Configure Cisco Unified CallManager for Cisco IP Communicator

Step 4
You need to obtain the MAC address of the NIC of your PC, where Cisco IP
Communicator is to be installed. Write the MAC address of your PC here:
____________________________
Note
Follow these instructions to obtain the MAC address. From your PC, get a command prompt
(choose Start > Run, enter cmd in the Open field, and click OK). From the command
prompt, enter ipconfig /all. The Physical Address field under the Ethernet adapter Local
Area Connection heading is the MAC address. When you have finished, close the command
prompt window.
Lab Guide
149
Step 5
In Cisco Unified CallManager Administration, choose Device > Phone. Click Add
New.
Step 6
Choose Cisco IP Communicator from the Phone Type drop-down list. Click Next.
The Phone Configuration window appears.
Step 7
Fill in these details in the Phone Configuration window:
MAC Address: <MAC of the PC>
Description: Kwong CIPC
Phone Button Template: Default IP Communicator Template
Calling Search Space: Unlimited_Access_CSS
Owner User ID: kwong
SCCP Phone Security Profile: Standard SCCP Profile for Auto Registration
Step 8
Click Save and OK.
Step 9
Click Line [1] and enter 2004 as the Directory Number.
Step 10
Choose Executive_PT for Route Partition.
Step 11
Click Save.
Install Cisco IP Communicator

Step 12
Make sure you have administrator privileges on your PC. If not, log in as
administrator.
Step 13
Your instructor has placed Cisco IP Communicator on the desktop of your PC in a

folder called CIPT1. Navigate to this folder and double-click the file named
CiscoIPCommunicatorSetup.exe.
Step 14
Click Next when the Cisco IP Communicator InstallShield Wizard appears.
Step 15
Accept the license agreement, click Next, and follow the on-screen prompts to
complete the installation.
Step 16
When the InstallShield Wizard Completed window appears, check the Launch
Cisco IP Communicator check box and click Finish. The Audio Tuning Wizard
screen appears.
Step 17
Read the Audio Tuning Wizard window and click Next to open the Audio Tuning
WizardSelect Audio Devices window.
Step 18
Choose the appropriate devices from the drop-down lists. If no additional audio
devices have been installed, you can accept the default values. Click Next.
Note
150
If multiple audio devices are installed, such as a USB handset device, then choose the
desired devices from the drop-down lists.
Step 19
Follow the on-screen instructions to adjust the listening volume, and then click
Next. The Adjust the Microphone Volume window appears.
Step 20
Follow the on-screen instructions to adjust the microphone volume, and then click
Next. The last window appears.
Step 21
Click Finish. The Cisco IP Communicator displays a pop-up message that it is not
able to reach TFTP server. Click OK.
Step 22
The Preferences window appears on your PC desktop. Click the Use these TFTP
servers radio button and, in the TFTP Server 1 field, enter the IP address of your
first node (10.x.1.1).
Step 23
Click OK.
Step 24
Watch as your Cisco IP Communicator cycles through the registration process.
The Cisco IP Communicator launch window appears on your PC desktop with the correct
directory number displayed.
You can make and receive calls using your PC.
Task 2: Configuring IP Phone Services

In this task, you will use a sample services file to add Cisco IP phone services to the Cisco
Unified CallManager database.
Activity Procedure
Step 1
Your instructor has installed a simple HTTP server on your PC. On the desktop in
the CIPT1\Sample IP Phone Service\Sample folder, you will find the file
sample.asp, which contains only a single line that will be displayed on your IP
phones at the end of this task.
Step 2
Phone Services and click Add New.
Step 3
Enter Sample Service in the Service Name field.
Step 4
Enter Sample Service in the Service Description field.
Step 5
Obtain the IP address of your PC and record it here:

______________________________
Note
Step 6
On your PC, open the Command Prompt with Start > Run, enter cmd, and click OK. At the
Command Prompt enter ipconfig.
In the Service URL field, enter:

http://<Your PC IP Address>/Sample/sample.asp
Step 7
Click Save.
Lab Guide
151
Subscribe to Cisco IP Phone Services

Step 8
In Cisco Unified CallManager Administration, choose Device > Phone click Find.
Step 9
Click the Phone2-x to display its Phone Configuration window.
Step 10
From the Related Links drop-down menu in the upper-right corner, choose
Subscribe/Unsubscribe Services, and click the Go button next to it.
Step 11
When a new dialog window opens, choose Sample Service from the Select a
Service drop-down menu, and then click Next.
Step 12
Click Subscribe, and then click Save.
Step 13
Close the dialog window.
Step 14
Reset the IP phone.
On the Phone2-x, on which you configured the sample services, press the Services button.
Press the Select softkey and choose the Sample service.
The IP phone displays Welcome to CIPT1 class!
Task 3: Enabling Call Display Restrictions

In this task, you will configure display restrictions with requirements typical of the hospitality
industry. You will configure directory numbers 2001 and 2002 as guest room telephones and
3004 as the front desk telephone. The call information display is restricted when guest rooms
call one another. The call information is presented to both the guest room and the front-desk
telephone when either calls the other.
Activity Procedure
Create Partitions
Step 1
152
The real partitions will eventually be assigned to the guest room and front-desk
telephone lines. The translation partitions will be assigned to the translation
patterns. The procedure for adding them in Cisco Unified CallManager
Administration is identical, so you can add all of them at the same time. Add these
partitions:
Insert the real partition P_Room.
Insert the real partition P_FrontDesk.
Insert the translation partition P_CallsFromRoomToRoom.
Insert the translation partition P_CallsFromRoomToFrontDesk.
Insert the translation partition P_CallsFromFrontDeskToRoom.
Create Calling Search Spaces

Step 2
Add these calling search spaces in Cisco Unified CallManager Administration and
assign the partition in parentheses to the calling search space.
Insert the calling search space CSS_Room (P_Room).
Insert the calling search space CSS_FrontDesk (P_FrontDesk).
Insert the calling search space CSS_FromRoom

(P_CallsFromRoomToFrontDesk and P_CallsFromRoomToRoom).
Insert the calling search space CSS_FromFrontDesk

(P_CallsFromFrontDeskToRoom).
Assign Partitions and Calling Search Spaces to Devices and Lines

Step 3
Configure Phone1-x with directory number 2001 (Room 1) with the device calling
search space CSS_FromRoom and the partition P_Room. Unassign the Call Pickup
group currently assigned to the line (set to None).
Step 4
Configure Phone2-x with directory number 2002 (Room 2) with the device calling
search space CSS_FromRoom and the partition P_Room. Unassign the Call Pickup
Step 5
Configure Phone4-x with directory number 3004 (Front Desk) with the device
calling search space CSS_FromFrontDesk, and check the Ignore Presentation
Indicators check box and the partition P_FrontDesk. Unassign the Call Pickup
Step 6
Reset all three IP phones.
Create Translation Patterns

Step 7
In Cisco Unified CallManager Administration, configure these translation patterns:
1. Insert a translation pattern 3XXX with these parameters:
Partition: P_CallsFromRoomToFrontDesk
CSS: CSS_FrontDesk
Provide Outside Dial Tone: Unchecked
Calling Line ID Presentation and Calling Name Presentation: Restricted
Connected Line ID Presentation and Connected Name Presentation: Allowed
Partition: P_CallsFromRoomToRoom
CSS: CSS_Room
Calling Line ID Presentation and Calling Name Presentation: Restricted
Connected Line ID Presentation and Connected Name Presentation: Restricted
Partition: P_CallsFromFrontDeskToRoom
CSS: CSS_Room
Calling Line ID Presentation and Calling Name Presentation: Allowed
Connected Line ID Presentation and Connected Name Presentation: Restricted
Lab Guide
153
Place a call from Room 1 to Room 2 (2001 to 2002) and vice versa. The call
information display is restricted, and the IP phone displays From Private or
From Anonymous (depending on the IP phone model).
Place a call from the front desk (3004) to Room 1 and then to Room 2 (2001 and
2002, respectively). The call information should be displayed to both parties
involved in the call. This is because the front desk IP phone is set to allow the
Calling Line ID Presentation and Calling Name Presentation options and to
enable the Ignore Presentation Indicators (Internal Calls Only) option. As a
result, the Connected Line ID Presentation and Connected Name Presentation
option restrictions that were set for the room IP phones are ignored.
Place a call from Room 1 (2001) to the front desk (3004). The call information
should be displayed to both parties involved in the call, even though the Calling
Line ID Presentation and Calling Name Presentation options are restricted for
the room telephones, because the Ignore Presentation Indicators (Internal Calls
Only) option is enabled for the front desk IP phone.
Cleanup
To prepare for the future labs, complete these steps:
Step 1
Delete all 2XXX and 3XXX translation patterns.
Step 2
Assign all IP phone lines (directory numbers) to the Executive_PT partition.
Step 3
Assign all IP phones to the Unlimited Access_CSS calling search space.
Step 4
Reset all IP phones.
Step 5
Ensure that you can call within and between pods.
Task 4: Configuring Client Matter Codes

In this task, you will add client matter codes in Cisco Unified CallManager Administration,
assign client matter codes to route patterns, and then use the codes to reach certain dialed
numbers.
Note
Remember, you can optionally use Cisco Unified CallManager BAT to add client matter
codes in a bulk transaction.
Activity Procedure
Add Client Matter Codes in Cisco Unified CallManager Administration
Step 1
In Cisco Unified CallManager Administration, choose Call Routing > Client
Matter Codes and click Add New.
Step 2
154
Referring to the table, add a code in the Client Matter Code field and add its
associated client name in the Description field. Click Save when finished. (You will
use the client number later in the lab.)
Client Matter Codes

Code
Client Name
Client Number
1111
ABC Company
852y.2001 (where y is your

partner pod number)
2222
XYZ Corporation
852y.2002 (where y is your

partner pod number)
Step 3
Make sure that you have added both client matter codes (Call Routing > Client
Matter Codes, then Find).
Enable Client Matter Codes for Route Patterns

Use the Client Number column from the table in step 2 to add two new route patterns to reach
two directory numbers in your partner pod, connected via intercluster trunk.
Step 4

> Route Pattern and click Add New.
Step 5
Configure the first route pattern from the Client Number column of the Client
Matter Codes table (852y.2001) with these parameters:
Step 6
Route Partition: Executive_PT
Gateway/Route List: TRK_RL
Require Client Matter Code: Checked
Configure the second route pattern from the Client Number column of the Client
Matter Codes table (852y.2002) with the same parameters as in the previous step.
Call the first directory number in your partner pod by dialing 852y.2001 (where y is your
partner pod number), and enter the client matter code when prompted. Remember to enter #
after the code to bypass the interdigit timeout and extend the call.
Call the second directory number in your partner pod by dialing 852y.2002, and enter the
client matter code when prompted. Remember to enter # after the code to bypass the
interdigit timeout and extend the call.
Call the third directory number in your partner pod by dialing 852y.2003. The call should
extend without requiring you to enter a client matter code.
Call the first directory number in your partner pod by dialing 852y.2001, and enter an
invalid code (5432, for example). You should receive a reorder tone.
Cleanup
To prepare for the future labs, complete this step:
Step 1
Delete the 852y.2001 and 852y.2002 route patterns.
Lab Guide
155

Attendant Console
Activity Objective
In this activity, you will add an attendant console user, jjones, configure 2222 as the pilot point
number to reach the hunt group, add jjones and extension 2002 to the hunt group, create the
attendant console user account, and associate extension 2002 and the pilot 2222 with the
attendant console user. Then you will install and configure the Cisco Unified CallManager
Attendant Console client application on your PC, and log in to the application as user jjones.
You will then make and receive calls using the Cisco Unified CallManager Attendant Console
application.
156
Add an attendant console user through the Cisco Unified CallManager Attendant Console
User Configuration window so that the user can log in to an attendant console
Configure the pilot point to receive and direct calls and hunting algorithm to distribute calls
Associate the pilot point with the hunt group and add members to the hunt group
Create the attendant console user and associate the attendant phones and the pilot points
with the user
Associate the attendant console phones and pilot points with the ac user
Install and configure the Cisco Unified CallManager Attendant Console client
Visual Objective

Unified CallManager Attendant Console
Phone1-x
Phone2-x
Attendant phone
DN: 2002
PC-x
CCM1-x
Attendant
Console
Client
Pilot Point:
2222
HQ-x
HQ-x
BRANCH-x
BR-x
Phone4-x
POD-x
CIPT1 v5.0-29
Required Resources
PC
Two or more Cisco IP phones
Command List
Job Aids
Lab Guide
157
Task 1: Adding a Cisco Unified CallManager Attendant Console

User
In this task, you will add an attendant console user. You must add users through the Cisco
Unified CallManager Attendant Console User Configuration window in Cisco Unified
CallManager Administration before the user can log in to an attendant console.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Application > Cisco Unified

CM Attendant Console > Cisco Unified CM Attendant Console User and click
Add New.
Step 2
In the User ID field, enter jjones.
Step 3
In the Password field, enter cisco.
Step 4
In the Confirm field, enter cisco.
Step 5
Click Save to add the new user.
When you choose Application > Cisco Unified CM Attendant Console > Cisco Unified
CM Attendant Console User and click Find, the attendant console user that you created is
displayed.
Task 2: Configuring the Pilot Point

In this task, you will configure the Cisco Unified CallManager Attendant Console server pilot
point in preparation for installing and configuring the Cisco Unified CallManager Attendant
Console client.
Activity Procedure
158
Step 1
In Cisco Unified CallManager Administration, choose Application > Cisco Unified

CM Attendant Console > Pilot Point and click Add New.
Step 2
In the Pilot Name field, enter Pilot1.
Step 3
From the Device Pool drop-down menu, choose Default_CMLocal_DP.
Step 4
From the Location drop-down menu, choose Hub_None.
Step 5
From the Media Resource Group drop-down menu, choose All_MRGL.
Step 6
Click Save.
Step 7
Click the Line[1] link to define a pilot number.
Step 8
In the Directory Number field, enter 2222.
Step 9
Choose Executive_PT for the Route Partition.
Step 10
Choose Unlimited_Access_CSS for the Calling Search Space.
Step 11
Click Save.
When you choose Application > Cisco Unified CM Attendant Console > Pilot Point and
click Find, the new pilot point is listed.
Task 3: Configuring the Hunt Group

You will configure the Cisco Unified CallManager Attendant Console server hunt group in
preparation for installing and configuring the Cisco Unified CallManager Attendant Console
client.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose Service > Cisco Unified CM

Attendant Console > Pilot Point and click Find.
Step 2
Choose Pilot1 from the list.
Step 3
Click Add Member. A new dialog window opens.
Step 4
Click the User Member radio button.
Step 5
From the User Name drop-down menu, choose jjones.
Step 6
From the Line Number drop-down menu, choose 1 (because you have only one line
configured on the IP phone).
Step 7
Click Save, and then click Close. The user jjones appears in the Available Members
pane.
Step 8
Click Add Member.
Step 9
From the Directory Number drop-down menu, choose 2002 in Executive_PT.
Step 10
Check the Always Route Member check box so that the voice-mail number can
receive multiple calls at the same time.
Step 11
Click Save, and then click Close. The directory number 2002 appears in the
Available Members pane.
The Available Members pane under Hunt List Member Information lists one user and one
directory number.
Lab Guide
159
Task 4: Creating the ac User Account

You must configure one user with the user ID ac in Cisco Unified CallManager
Administration and associate the attendant IP phones and the pilot points with this user. If you
do not configure this user, the attendant console cannot interact with Cisco CTIManager, and
the attendant cannot receive calls.
Activity Procedure
Step 1
In Cisco Unified CallManager Administration, choose User Management >

Application User and click Add New.
Step 2
In the First Name and Last Name fields, enter ac.
Step 3
In the User ID field, enter ac.
Step 4
In the User Password field, enter 12345.
Step 5
In the Confirm Password field, enter 12345.
Step 6
Click Save.
Step 7
Choose User Management > User Group.
Step 8
In the list, click Standard CTI Allow Call Park Monitoring. A new User Group
Configuration page opens.
Step 9
Click the Add Application User to Group button and, on the new dialog window,
check the check box ac, and then click Add Selected. The selected user appears in
the Users in Group list.
Step 10
Click the Go button next to the Back to Find/List.
Step 11
In the list, click Standard CTI Enabled group and, by repeating last two steps,
assign the ac application user to this group.
When you choose User Management > Application User and click Find, the ac user
appears in the list.
Task 5: Associating Devices and Pilot Points with the ac User

Before the attendant uses the attendant console, you must associate the attendant console IP
phones and pilot points with the ac user.
Activity Procedure
160
Step 1
Get the MAC address of the Phone2-x (2002) and record it here:
Step 2
Choose User Management > Application User and click Find.
Step 3
Select the ac user from the list.
Step 4
In the Application User Configuration window for the ac user, move Pilot1 from
the Available Devices pane to the Controlled Devices pane.
Step 5
Similarly, move the identifier (MAC) of the Phone2-x (2002) from the Available
Devices pane to the Controlled Devices pane. This extension should be the primary
extension.
Step 6
Click Save.
Pilot1 and the identifier of Phone2-x appear in the Controlled Devices pane.
Task 6: Activating Services

In this task, you will activate the necessary services for Cisco Unified CallManager Attendant
Console.
Activity Procedure
Step 1
In the Cisco Unified CallManager Serviceability, choose Tools > Service

Activation and select your first node server.
Step 2
Activate the Cisco CTIManager service and Cisco CallManager Attendant

Console Server service.
Both services report activation status Activated.
Task 7: Installing and Configuring Cisco Unified CallManager

Attendant Console Client
In this task, you will install and configure the Cisco Unified CallManager Attendant Console
client.
Activity Procedure
Install the Cisco Unified CallManager Attendant Console Plug-In
Step 1
Make sure that you have logged in to your PC with administrator privileges.
Step 2
In Cisco Unified CallManager Administration, choose Applications > Plugins and

click Find.
Step 3
Click the Download link next to the Cisco CallManager Attendant Console to
download the client to your PC.
Lab Guide
161
Step 4
On the new dialog window, navigate to the C:\Temp directory, where the client
(CiscoAttendantConsoleClient.exe) is going to be downloaded, and click Save.
Step 5
Wait until the file is completely downloaded.
Step 6
On your PC, navigate to the CiscoAttendantConsoleClient.exe file and start it. The
installation wizard opens.
Step 7
Follow the on-screen instructions to install the client on your PC.
Configure Cisco Unified CallManager Attendant Console Settings

Step 8
Start the Cisco Unified CallManager Attendant Console client by choosing Start >
Programs > Cisco CallManager Attendant Console, or double-click the Cisco
Unified CallManager Attendant Console icon on the desktop.
Step 9
In the Basic Settings tab, enter your first node IP address in the Attendant Server
Host Name or IP Address field.
Step 10
Enter 2002 in the Directory Number of Your Phone field.
Step 11
Click Save. If a pop-up window appears, saying that multiple devices match, from
the pop-up window drop-down menu, choose the device starting with SEP and
click OK.
Step 12
At this point, you see the Connecting to the Server message. Wait until the Log In
dialog window opens.
Step 13
Enter jjones as the user ID and cisco as the password. Click Log In. The Cisco
Unified CallManager Attendant Console application appears shortly.
Step 14
From the client menu, choose File > Go Online.
Step 15
From another IP phone in your cluster, place a call to the IP phone associated with
the attendant console (2002) and answer the call from the attendant console.
Answer, transfer (normal transfer and consult transfer), hold, resume, hang up, and
make calls from the attendant console to telephones in your cluster.
162
The Cisco Unified CallManager Attendant Console application can be started, and calls can
be made and answered from the attendant console.
Answer Key
The correct answers and expected solutions for the activities that are described in this guide
appear here.
Lab 2-1 Answer Key: Performing Platform Administration

When you complete this activity, your solution will be similar to this, with differences that are
specific to your device or workgroup.
Task 1: Checking Platform Status

Q1)
Show > Cluster
Q2)
Show > Network
Q3)
Show > System
Task 2: Viewing and Updating Platform Settings

Q1)
Settings > IP > Ethernet
Q2)
No.
Task 3: Viewing Security Certificates

Write the fingerprints of the tomcat certificate:
Step 4
Note
SHA1:
A4 7D 15 29 44 4A 7C 84 B3 C9 1B CC C1 83 4D 64 C3 11 DD 82
MD5:
11 83 A3 C7 7C 19 22 4B 73 A1 11 23 55 A5 CA B1
This output varies per Cisco Unified CallManager.
Task 5: Checking Network Connectivity and Setting Up Remote Support

Write the passphrase for remote support:
Step 3
MB8GQTSHP0
Note
This output varies per remote support account.
Lab 2-2 Answer Key: Performing General Administration

The Verification sections within the tasks provide all the verification that is needed.
Lab 2-3: Configuring Cisco Unified CallManager Release 5.0

Basic Settings
Lab Guide
163
Lab 2-4: Backing Up and Restoring Cisco Unified CallManager

Release 5.0

Lab 3-2: Configuring Cisco Unified CallManager to Support

Cisco IP Phones


Lab 4-1: Configuring Cisco Unified CallManager Trunks



Lab 4-4: Configuring Hunt Groups and Call Coverage

When you complete this activity, your solution will be similar to this, with differences that are
specific to your device or workgroup.
Task 4: Testing Call Distribution

Step 4
What do you expect will happen when you call the hunt pilot?
The call should hunt directly to directory number 3004 because the hunt option for
ring, no-answer, and not-available conditions is Try Next Member; Then, Try Next
Group in Hunt List. Hunting exhausts because no final forwarding is configured.
164
Task 5: Configuring Final Forwarding for Busy and No-Answer Conditions

What do you expect will be the call-distribution and final forwarding behavior?
Step 5
The call will forward to 2001 because 2002 and 3004 are not answered. (In a
production network, the final forwarding number would probably be a voice-mail or
autoattendant number).
What do you expect will be the call-distribution and final forwarding behavior?
Step 6
The call will hunt for 12 seconds and will forward to 2001 because the line is busy.

Lab 4-6: Implementing Calling Privileges and Restrictions





Lab 6-3: Configuring Cisco Unified CallManager Extension

Mobility
Lab 6-4: Configuring Presence and BLF Functionality


Lab 6-6: Configuring Cisco Unified CallManager Attendant

Console
Lab Guide
165

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Lab 2-1: Performing Platform Administration

Lab 2-2: Performing General Administration

Lab 3-1: Configuring Auxiliary or Voice VLANs

Lab 3-2: Configuring Cisco Unified CallManager to Support Cisco IP Phones

Lab 3-3: Configuring SIP Endpoints

Lab 3-4: Using Cisco Unified CallManager BAT

Lab 4-1: Configuring Cisco Unified CallManager Trunks

Lab 4-2: Configuring Voice Gateways

Lab 4-3: Configuring Basic Dial Plan Elements

Lab 4-4: Configuring Hunt Groups and Call Coverage

Lab 4-5: Configuring Complex Dial Plan Elements

Lab 4-6: Implementing Calling Privileges and Restrictions

Lab 5-1: Configuring Call Admission Control

Lab 5-2: Configuring Cisco Unified SRST

Lab 6-1: Configuring Media Resources

Lab 6-2: Configuring User Features

Lab 6-3: Configuring Cisco Unified CallManager Extension Mobility

Lab 6-4: Configuring Presence and BLF Functionality

Lab 6-5: Configuring Additional Features

Lab 6-6: Configuring Cisco Unified CallManager Attendant Console

Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0

2006 Cisco Systems, Inc.

Lab 2-1: Performing Platform Administration

Check the status of various platform components

View IP, NTP, and SMTP settings

View security certificates

Use the CLI to perform basic platform functions

Visual Objective for Lab 2-1:

Check platform status

View platform settings

View security certificates

Upload TFTP files

Check network connectivity

Use the CLI

2006 Cisco Systems, Inc.

First node running Cisco Unified CallManager Release 5.0

PC running Windows Explorer

SSH client software on PC (for example, PuTTY)

Lab Devices and Their Roles

Device Role in the Activity

Cisco Unified CallManager Release 5.0 first node

Cisco Unified CallManager Release 5.0 subsequent node

PC with Internet Explorer browser

Cisco IPT Platform Administration

Cisco Unified CallManager

Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0

2006 Cisco Systems, Inc.

Displays the basic platform status information:

Memory and disk usage

show tech ccm_service

Displays information on all Cisco Unified CallManager

show version active

Displays the software version on the active partition.

show network eth0

Displays information about the first Ethernet interface.

file list tftp *

Shows a list of all TFTP files.

file view install

Shows the content of the install.log installation log file.

file search activelog *

Searches for the string error in all active log files.