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

2 Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0 © 2006 Cisco Systems, Inc.

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

© 2006 Cisco Systems, Inc. Lab Guide 3

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.

© 2006 Cisco Systems, Inc. All rights reserved. CIPT1 v5.0—-5

Visual Objective for Lab 2-1: Performing Platform Administration

POD-x

PC-x

HQ-x

CCM1-x CCM2-x

HQ-x

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

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.


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.


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


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.



Step 2 Visit all pages available under the Settings menu.


Step 3 Answer these questions:

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.



Step 2 Go to Security > Certificate Management > Display Certificate.

Step 3 From Cisco Unified CallManager’s 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 certificate’s authenticity by comparing the received fingerprints against the fingerprints that you wrote down after installation.


Activity Verification You have completed this task when you attain these results:

You viewed the tomcat’s 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.



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.



Step 2 Go to Services > Ping and ping the default gateway of PC-x (10.x.3.101).


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.


Step 1 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.

Note 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.

Step 2 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

Step 3 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).

file search activelog * error

Note You will probably not find any hits when searching for this string on the newly installed Cisco Unified CallManager.

Step 4 Use this utils commands to capture packets sent out by your Cisco Unified CallManager:

utils network capture eth0 src 10.x.1.1 count 20

Note 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.

Activity Verification 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 2-2: Performing General Administration Complete this lab activity to practice what you learned in the related module.

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 for Lab 2-2: Performing General Administration

POD-x

PC-x

HQ-x

CCM1-x CCM2-x

HQ-x

1. Access Cisco Unified CallManager Administration GUI from PC-x

2. Create users with different administrative privileges

From your PC, you will log in to the Cisco Unified CallManager Administration GUI and set up additional administrator accounts.













Cisco Unified CallManager Administration GUI

CCMAdministrator / appuserpass

https://10.x.1.1/ccmadmin




Parameter Value




CCM1-x 10.x.1.1/24

CCM2-x 10.x.1.2/24



The figure shows the menu structure of the Cisco Unified CallManager Administration.



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).


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 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.

Step 3 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 CCMAdministrator’s 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.


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.

User ID: Johnny

Password: password

PIN: 12345

Last name: Jones

Step 4 Click Add New again and add a second user with these attributes.

User ID: Jack

Password: password

PIN: 12345

Last name: Smith

Step 5 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.


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 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).

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 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.



Complete this lab activity to practice what you learned in the related module.

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 for Lab 2-3: Configuring Cisco Unified CallManager 5.0 Basic Settings

POD-x

PC-x

HQ-x

CCM1-x CCM2-x

HQ-x

1. Change hostnames to IP addresses

2. Activate Services

From your PC, you will eliminate the need for Domain Name System (DNS) and activate the minimum services needed to support Cisco IP phones.
















Cisco Unified CallManager Serviceability GUI


https://10.x.1.1/ccmservice




Parameter Value




CCM1-x 10.x.1.1/24

CCM2-x 10.x.1.2/24



The figure shows the menu structure of the Cisco Unified CallManager Serviceability GUI.



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.


Step 1 From PC-x, access Cisco Unified CallManager Administration.

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.


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.




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.

The figure illustrates what you will accomplish in this activity.


Visual Objective for Lab 2-4: Backing Up and Restoring Cisco Unified CallManager 5.0

1. Back up Cisco Unified CallManager cluster to SFTP server on PC-x

2. Restore Cisco Unfied CallManager cluster from SFTP server on PC-x

POD-x

PC-x

HQ-x

CCM1-x CCM2-x

HQ-x




SFTP server software installed on the PC



Use the credentials in the table to log in to the lab applications that require authentication.





PC-x PC with Internet Explorer browser and SFTP server







Cisco Unified CallManager Disaster Recovery System GUI

admin / adminpass https://10.x.1.1/drf

Cisco Unified CallManager Platform Administration CLI

admin / adminpass SSH to 10.x.1.1




Parameter Value




CCM1-x 10.x.1.1/24

CCM2-x 10.x.1.2/24



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.


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: 22

Root path: C:\

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:

Server name: <IP address of your PC>

Note This is the IP address that you wrote down in step 5.

Path name: /

User name: user

Password: pass

Step 11 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.


Change the Configuration 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.

User ID: test

Password: password

PIN: 12345

Last name: test

Step 4 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.


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:

Server name: <IP address of your PC>

Note 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 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 3-1: Configuring Auxiliary or Voice VLANs Complete this lab activity to practice what you learned in the related module.

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 for Lab 3-1: Configuring Auxiliary or Voice VLANs

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

CCM2-x

PC-x

VLAN: x01

VLAN: x03

VLAN: x02

VLAN: x05

Phone1-x Phone2-x

Phone4-x

Phone3-x

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).


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

1 0/1 0/2 0/3 0/4

2 0/1 0/2 0/3 0/4

VLAN Assignment to Pods

Pod Number HQ Voice VLAN HQ Data VLAN BR Voice VLAN

1 102 103 105

2 202 203 205


Task 1: Configuring Auxiliary or Voice VLANs In this task, you will configure auxiliary or voice VLANs on the Cisco IOS Catalyst switch.


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 mode access

switchport voice vlan <HQ Voice VLAN number>

switchport access vlan <HQ Data VLAN number>

spanning-tree portfast

exit

Note Your instructor has already configured the data VLANs. This command is for reference only: vlan <VLAN number>

Step 4 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


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.

Activity Verification

You have completed this task when you attain these results:

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.




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

POD-x

Phone1-x Phone2-x

Phone4-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

Phone3-x

BR-x

CCM2-x

10.x.1.0/24

.1 .2

10.x.2.0/24

10.x.5.0/24

.101.101

.102

DHCP server

DHCP server

.11-.20

.11-.20.11-.20

.11-.20

IP

IP

DN: 3004

DN: Auto


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.



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 excluded-address

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.



DHCP Addressing and Scope Information for Cisco IOS Router at Branch

Cluster Name

Pool Type

Pool Name Network and Mask

Default Router

Option 150

Excluded Addresses, Start and End

Pod1 Voice pod1-br-voice 10.1.5.0 255.255.255.0

10.1.5.102 10.1.1.1 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

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

First Node Subsequent Node Auto-Registration DN Range

Cluster Name

Cisco Unified CallManager Name

IP Address Server Function

Cisco Unified CallManager Name

IP Address Server Function

Start End

Pod1 CCM1-1 10.1.1.1 CCM2-1 10.1.1.2 2001 2002

Pod2 CCM1-2 10.2.1.1 Firs

t

CCM2-2 10.2.1.2 Subs

eque

nt

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.


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


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.


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.


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 Auto-Registration 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.


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:

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)

Step 28 Click Save.


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.


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 Insert an Ethernet cable of the Phone2-x into its port on the switch.

Step 6 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 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 Insert an Ethernet cable of the Phone4-x into its port on the switch. 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.



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.


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 phone’s 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 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. If the IP phone fails to register, reset its network configuration.

Activity Verification 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.


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 phone’s 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 phone’s 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.

Activity Verification 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 3-3: Configuring SIP Endpoints Complete this lab activity to practice what you learned in the related module.

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 for Lab 3-3: Configuring SIP Endpoints

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

CCM2-x

10.x.1.0/24

.1 .2

10.x.2.0/24

10.x.5.0/24

.101.101

.102

3rd-party SIPDN: 2003

SIP

.11-.20

.11-.20.11-.20.11-.20

Phone1-x Phone2-x

Phone4-x

Phone3-x


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.


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.


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 re-register 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.


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.


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 In Cisco Unified CallManager Administration, choose Device > Phone and click Add New.

Step 7 For the Phone Type, choose Third-party SIP Device (Advanced) from the drop-down 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.


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 In Cisco Unified CallManager Administration, choose Device > Phone and click 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.


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 3-4: Using Cisco Unified CallManager BAT Complete this lab activity to practice what you learned in the related module.

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:

Cisco Unified CallManager Release 5.0 first node

Three Cisco IP phones

Cisco Unified CallManager BAT.xlt spreadsheet

MS Excel installed on your PC



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.


Step 1 In Cisco Unified CallManager Administration, choose Bulk Administration > 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.


Step 1 In Cisco Unified CallManager Administration, choose Bulk Administration > 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.


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.


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 In Cisco Unified CallManager Administration, choose Bulk Administration > 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 drop-down 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 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.


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 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.



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:

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”).


Lab 4-1: Configuring Cisco Unified CallManager 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 for Lab 4-1: Configuring Cisco Unified CallManager Trunks

POD-1 POD-2

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

BRANCH-1 BRANCH-2

HQ-1 HQ-2

CCM1-1 CCM2-2

HQ-1 HQ-2

Phone3-1 Phone3-2

BR-1 BR-2

CCM2-1 CCM1-2

10.1.1.0/24 10.2.1.0/24

.1 .1.2 .2.1.101.2.10110.12.0.0/16

IC_TRK

SIP_TRK

IC_TRK

SIP_TRK


Two Cisco Unified CallManager Release 5.0 clusters

IP WAN connectivity between the two clusters


Command List The table describes the command that is used in this activity.

Cisco IOS Commands

Command Description

ping Verifies remote IP address reachablity.


Intercluster Trunk Information

Cluster Name Device Name Description Remote IP Address

First: 10.2.1.1 Pod 1 IC_TRK Intercluster to Pod 2

Subsequent: 10.2.1.2

First: 10.1.1.1 Pod 2 IC_TRK Intercluster to Pod 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.


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)



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.


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.


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.


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 drop-down 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.


The SIP trunk appears in the list when you choose Device > Trunk and then click the Find button in Cisco Unified CallManager Administration.


Lab 4-2: Configuring Voice Gateways Complete this lab activity to practice what you learned in the related module.

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 for Lab 4-2: Configuring Voice Gateways

POD-1 POD-2

BRANCH-1 BRANCH-2

HQ-1 HQ-2

CCM1-1 CCM2-2

HQ-1 HQ-2

BR-1 BR-2

CCM2-1 CCM1-2

10.1.1.0/24 10.2.1.0/24

.1 .1.2 .2

10.1.250.102/32 10.2.250.102/32

.101 .101

PSTN

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2

H-323 gw

H-323 gw

MGCPgw

MGCP gw


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


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

Command Description


configure terminal Enters global configuration mode.

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



mgcp Enables the MGCP application on the gateway.

mgcp call-agent [first-node-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 Configures a POTS dial peer on the gateway.

application mgcpapp Configures the dial peer to use the MGCP application.

port [module-number/slot-number/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.


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


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.


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 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.


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


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.

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

Step 7 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 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.


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 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.


Task 4: Configuring Cisco IOS Voice Gateway to Support MGCP

In this task, you will enable MGCP on a Cisco IOS MGCP gateway.


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 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.

Step 7 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


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.


Lab 4-3: Configuring Basic Dial Plan Elements Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will build basic route plans to place calls over the IP WAN and the PSTN. After completing this activity, you will be able to meet these objectives:

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 for Lab 4-3: Configuring Basic Dial Plan Elements

POD-1 POD-2

PSTNBRANCH-1 BRANCH-2

HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2

Prio 1 Prio 1

Prio 2 Prio 2

8522.2xxx (ICT, SIP)8532.3xxx (ICT, SIP)


DN: 2001

DN: 3004

DN: 2003DN: 2002DN: 2003DN: 2002 DN: 2001

DN: 3004

521-555-2000 to 2199 522-555-2000

to 2199

532-555-3000 to 3199

531-555-3000 to 3199

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2

To PSTN

To PSTN

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.



Two Cisco Unified CallManager clusters with IP phones

MGCP gateway

H.323 gateway

Simulated PSTN

IP WAN



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


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.


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.


Step 1 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt > 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.





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.


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 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt > 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 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt

> 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@HQ-x), 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.


Configure a Route List Step 10 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt

> 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 Configuration window.

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 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt

> 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 24 Click Save. You are notified by a pop-up window that authorization code will not be activated. 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.



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.


Step 1 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt > 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 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt

> Route Pattern.

Step 17 Click Add New to open the Route Pattern Configuration window.


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 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.


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 for Lab 4-4: Configuring Hunt Groups and Call Coverage

POD-1 POD-2


HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2



521-555-2000 to 2199

522-555-2000 to 2199

532-555-3000 to 3199

531-555-3000 to 3199

2ndLG

DN: 3004

1stLG

DN: 2002DN: 2001

2ndLG

DN: 3004

1stLG

DN: 2001 DN: 2002DN: 2001 DN: 2002

1stHL

Hunt Pilot: 2111

1stHL

Hunt Pilot: 2111

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2



Visual Objective for Lab 4-4: Configuring Hunt Groups and Call Coverage (Cont.)

External call Internal call

RNAR: 5 seconds

Try Next Member; Then, Try Next Group in Hunt List

Line Groups

Hunt Pilot

Hunt List

2003

2002

2001

3004

x111

2ndLG

1stLG

1stHL


Two Cisco Unified CallManager clusters

Three Cisco IP phones with LCD display

One third-party SIP phone

IP WAN




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.


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.


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.


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


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.


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 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.


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


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).


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 4-5: Configuring Complex Dial Plan Elements Complete this lab activity to practice what you learned in the related module.

Activity Objective In this activity, you will be able to apply complex route-plan concepts to simple route plans. After completing this activity, you will be able to meet these objectives:

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 for Lab 4-5: Configuring Complex Dial Plan Elements

POD-1 POD-2


HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2

521-555-2000 to 2199

522-555-2000 to 2199

532-555-3000 to 3199

531-555-3000 to 3199

Attendant2XXX=2001

Attendant2XXX=2001

Access Code9.@

Blocked calls to9 1 900 555 XXXX

Enabled calls to9 1 5YZ 555 XXXX

External Mask 521555XXXX531555XXXX

External Mask 522555XXXX532555XXXX

Access Code9.@

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2



Gateway devices (Cisco IOS MGCP [PRI], H.323 [FXO])

Simulated PSTN

IP WAN



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

Configures a POTS dial peer on the gateway.

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.


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


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.


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 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.


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 Configuration window.

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.


Configure PSTN Route Pattern with Access Code Step 1 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt

> 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 6 Check the box Use Calling Party’s 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.


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, 9-1522-555-2001). 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.


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).

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.




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 four-digit 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.


Activity Verification You have successfully completed this task when you attain these results:

If you dial the telephone number with the 900 area code now (for example, 91-900-52y-2001, 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 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt > 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 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 for Lab 4-6: Implementing Calling Privileges and Restrictions

POD-1 POD-2


HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2

Lobby_PT 2001

521-555-2000 to 2199

522-555-2000 to 2199

532-555-3000 to 3199

531-555-3000 to 3199

Lobby_PT 2001

Executive_PT 2003Executive_PT

2003

Employee_PT 3004

Employee_PT 3004

PSTN_PT 9.@

PSTN_PT 9.@

PSTN_PT 9.@

PSTN_PT 9.@

TRK_PT8522.2XXX and

8532.3XXX

TRK_PT8521.2XXX and

8531.3XXX

PLAR

Executive_PT 2002

PLAR

Executive_PT 2002

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2



Cisco IP telephony server cluster


Gateway devices


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

Calling Search Space Name

Description

Contains Partitions

Executive IP phone (2002) MGCP gateway H.323 gateway

Unlimited_Access_CSS Unlimited Calling Privileges

Employee_PT Lobby_PT Executive_PT PSTN_PT TRK_PT

Employee IP phone (2003 and 3004) Intercluster trunk (IC_TRK) SIP Trunks (SIP_TRK)

Company_Only_CSS Intra- and Intercluster Calling Only

Employee_PT Lobby_PT TRK_PT

Lobby IP phone (2001)

Cluster_Only_CSS Intracluster Calling Only

Employee_PT Lobby_PT


Task 1: Configuring Partitions In this task, you will create partitions in Cisco Unified CallManager Administration with partition names and descriptions.


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.


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.



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.


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 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.

Gateways 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 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.

Note 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 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 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.


Step 1 Choose Call Routing > Class of Control > Partition, and click Add New to open the Partition Configuration page.


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 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.

Activity Verification 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:

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 5-1: Configuring Call Admission Control Complete this lab activity to practice what you learned in the related module.

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.

After completing this activity, you will be able to meet these objectives:

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 for Lab 5-1: Configuring Call Admission Control

POD-1 POD-2

BRANCH-1 BRANCH-2

HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2

521-555-2000 to 2199

522-555-2000 to 2199

532-555-3000 to 3199

531-555-3000 to 3199

Max. 128kbps

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2

RSVP

Max. 128kbps

Max. 56kbps Max. 56kbpsSecond Call

Denied !

First CallAdmitted

First CallAdmitted

First CallAdmitted

Second CallDenied !

Second CallDenied !

RSVPRSVP

RSVP

H.323 gkH.323 gk


Two Cisco Unified CallManager clusters with IP phones

Single gatekeeper device per cluster

Two RSVP agents per cluster

IP WAN

Simulated PSTN


Cisco IOS RSVP Commands

Command Description



bandwidth bw Configures available bandwidth on an interface.


Command Description

fair-queue [congestive-discard-threshold [dynamic-queues [reservable-queues]]]

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 group-number

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

Command Description

gatekeeper Enters gatekeeper configuration mode.

zone local name Specifies the zone that is controlled by the gatekeeper.

zone subnet name [ip-address/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.


H.323 Gatekeeper Configuration Parameters

Cluster

Zone Name

Remote Zone Name

H.323 GK IP Address

Remote H.323 GK IP Address

HQ LAN IP Subnet

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


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.


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.


Step 18 Name the new device pool CM_Remote_DP.


Step 19 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

Step 20 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.


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.


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 drop-down 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 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.



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


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


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


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.


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.



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.


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


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 gatekeeper-controlled trunk to an existing route group and remove the nongatekeeper-controlled trunk.


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 gatekeeper-controlled 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.


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 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 5-2: Configuring Cisco Unified SRST Complete this lab activity to practice what you learned in the related module.

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 for Lab 5-2: Configuring SRST

POD-1 POD-2


HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2

DN: 2001

DN: 3004

DN: 2003DN: 2002DN: 2003DN: 2002 DN: 2001

DN: 3004

521-555-2000 to 2199 522-555-2000

to 2199

532-555-3000 to 3199

531-555-3000 to 3199

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2

Backup to reachHQ IP phones

SRST router

Failure

SRST router

Failure




MGCP PRI gateway at HQ site

Gateway device with SRST feature and analog line at Branch site

Simulated PSTN

IP WAN



Cisco IOS SRST Commands

Command Description



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 extension-length 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.



Task 1: Configuring SRST on Cisco IOS Router In this task, you will configure SRST support on the Branch gateway.


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-manager-fallback 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


Note 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.

Step 9 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-manager-fallback. 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.


Step 1 Connect to your first node and, in Cisco Unified CallManager Administration, choose System > SRST.


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.



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.


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.


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 phone’s display).


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.


Lab 6-1: Configuring Media Resources Complete this lab activity to practice what you learned in the related module.

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 for Lab 6-1: Configuring Media Resources

POD-1 POD-2

BRANCH-1 BRANCH-2

HQ-1 HQ-2

CCM1-1 CCM1-2

HQ-1 HQ-2

BR-1 BR-2

PSTN

Phone1-1 Phone2-1

Phone4-1

Phone1-2 Phone2-2

Phone4-2

Phone3-1 Phone3-2

521-555-2000 to 2199

522-555-2000 to 2199

532-555-3000 to 3199

531-555-3000 to 3199

H-323 gw

H-323 gw

Conf. BridgeTranscoder

Conf. BridgeTranscoder

G.711 G.711

G.711 Only G.711 Only

MOH server

Meet-Me 250X

MOH server

Meet-Me 250X

Region: LowBWMax. G.729

Region: LowBWMax. G.729


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)



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.



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


Step 8 From the T1/E1 controller, remove the pri-group by using the command no pri-group 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.


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

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.



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: _________________________________

Step 4 Connect to your first node and, in Cisco Unified CallManager Administration, choose Media Resources > Conference Bridge to open the Find and List Conference Bridges page.


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 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 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


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.


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.



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.


Note The approach that is provided here represents only one of many ways to organize MRGs and MRGLs.

Step 11 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.

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 Meet-Me 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.


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/sw-voice.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.


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.


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 (woman’s voice) is streamed. Hang up.


Lab 6-2: Configuring User Features Complete this lab activity to practice what you learned in the related module.

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 for Lab 6-2: Configuring User Features

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

Phone1-x Phone2-x

Phone4-x

Phone3-x

Pickup: 6686

Park: 266X

Pickup: 6686

BargeLn-2: 2099

Privacy

Pickup: 6685

BargeLn-2: 2099

Privacy


Cisco Unified CallManager





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.


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.


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 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.


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 The CallBack softkey in the On Hook call state enables you to check the status of a call-back request that was previously initiated or to cancel a call-back request that is no longer desired.

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 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.

Cleanup 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.

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.


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.


Step 11 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.

Note 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.


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.


Step 1 Choose System > Service Parameters.

Step 2 From the Server drop-down menu, choose your first node. From the Service drop-down 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


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.



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 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.




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




Visual Objective for Lab 6-3: Configuring Cisco Unified CallManager Extension Mobility

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

Phone1-x Phone2-x

Phone4-x

Phone3-x

mperry logs in

ExtensionMobilityService

DN: 2555



Cisco Unified IP Phone 7940




Task 1: Adding a User In this task, you will add a new user that will be used later for Extension Mobility.


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:

User ID: mperry

Password: cisco

PIN: 12345

Last Name: Perry

First name: Mark

Step 3 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.


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.


Step 9 In Cisco Unified CallManager Administration, choose Device > Device Settings > 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 This URL is case-sensitive, so make sure that you enter it exactly as described.

Step 13 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 behavior—that is, whether you allow the user to log in to more than one device at a time


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.


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 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.


Step 1 In Cisco Unified CallManager Administration, choose Device > Device Settings > Device Profile and click Add New.

Step 2 From the Device Profile Type drop-down menu, choose Cisco 7940 and click Next.


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 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.


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.


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.


Step 1 In Cisco Unified CallManager Administration, choose Device > Phone and click 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.



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 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 for Lab 6-4: Configuring Presence for BLF Functionality

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

Phone1-x Phone2-x

Phone4-xManager

CIOExecutive

Allowsubscription

Blocksubscription



Cisco Unified CallManager Release 5.0 first node

Three Cisco IP Phones (two Cisco Unified IP Phone 7940 phones and one Cisco Unified IP Phone 7961)



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.


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.


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 Presence-enabled 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.


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 Choose Device > Phone and click Find.

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.



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 Choose Device > Phone and click Find.

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 phone’s 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 watcher’s Presence-enabled speed dials is off, and the phone icon displays idle.

When subcribees are in a call, the BLF status on the watcher’s Presence-enabled speed dials is red, and the phone icon associated with the speed dial displays busy.


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.


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

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 6-5: Configuring Additional Features Complete this lab activity to practice what you learned in the related module.

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. After completing this activity, you will be able to meet these objectives:

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 for Lab 6-5: Configuring Additional Features

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

Phone1-x Phone2-x

Phone4-xFront Desk

Room 2Room 1

PC-x

CIPC DN: 2004

HTTP SrvCMC requested:

852y-2001852y-2002

Display Restrictions

Display Unrestricted





IP WAN



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.


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:

User ID: kwong

User Password: cisco

PIN: 12345

Last Name: Wong

First Name: Kim

Step 3 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.


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 Device Pool: Default_CMLocal_DP 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 Wizard—Select 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 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.


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 In Cisco Unified CallManager Administration, choose Device > Device Settings > 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 On your PC, open the Command Prompt with Start > Run, enter cmd, and click OK. At the Command Prompt enter ipconfig.

Step 6 In the Service URL field, enter:

http://<Your PC IP Address>/Sample/sample.asp

Step 7 Click Save.


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.



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.


Create Partitions Step 1 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 group currently assigned to the line (set to None).

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 group currently assigned to the line (set to None).

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 Provide Outside Dial Tone: Unchecked Calling Line ID Presentation and Calling Name Presentation: Restricted Connected Line ID Presentation and Connected Name Presentation: Restricted


Partition: P_CallsFromFrontDeskToRoom CSS: CSS_Room Provide Outside Dial Tone: Unchecked Calling Line ID Presentation and Calling Name Presentation: Allowed Connected Line ID Presentation and Connected Name Presentation: Restricted



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.


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 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 In Cisco Unified CallManager Administration, choose Call Routing > Route/Hunt > 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:

Route Partition: Executive_PT

Gateway/Route List: TRK_RL

Provide Outside Dial Tone: Unchecked

Require Client Matter Code: Checked

Step 6 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.




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.

After completing this activity, you will be able to meet these objectives:

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 for Lab 6-6: Configuring Cisco Unified CallManager Attendant Console

POD-x

BRANCH-x

HQ-x

CCM1-x

HQ-x

BR-x

Phone1-x Phone2-x

Phone4-x

PC-xPilot Point:

2222

Attendant phoneDN: 2002

Attendant Console

Client



PC

Two or more Cisco IP phones




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.


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.


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.


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.


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.


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.


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.


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.


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.


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.


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 Step 4 Write the fingerprints of the tomcat certificate:

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

Note This output varies per Cisco Unified CallManager.

Task 5: Checking Network Connectivity and Setting Up Remote Support Step 3 Write the passphrase for remote support:

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.


The Verification sections within the tasks provide all the verification that is needed.




Lab 3-1: Configuring Auxiliary or Voice VLANs The Verification sections within the tasks provide all the verification that is needed.



Lab 3-3: Configuring SIP Endpoints The Verification sections within the tasks provide all the verification that is needed.

Lab 3-4: Using Cisco Unified CallManager BAT The Verification sections within the tasks provide all the verification that is needed.

Lab 4-1: Configuring Cisco Unified CallManager Trunks The Verification sections within the tasks provide all the verification that is needed.

Lab 4-2: Configuring Voice Gateways The Verification sections within the tasks provide all the verification that is needed.

Lab 4-3: Configuring Basic Dial Plan Elements The Verification sections within the tasks provide all the verification that is needed.

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.


Task 5: Configuring Final Forwarding for Busy and No-Answer Conditions Step 5 What do you expect will be the call-distribution and final forwarding behavior?

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).

Step 6 What do you expect will be the call-distribution and final forwarding behavior?

The call will hunt for 12 seconds and will forward to 2001 because the line is busy.

Lab 4-5: Configuring Complex Dial Plan Elements The Verification sections within the tasks provide all the verification that is needed.

Lab 4-6: Implementing Calling Privileges and Restrictions The Verification sections within the tasks provide all the verification that is needed.

Lab 5-1: Configuring Call Admission Control The Verification sections within the tasks provide all the verification that is needed.

Lab 5-2: Configuring Cisco Unified SRST The Verification sections within the tasks provide all the verification that is needed.

Lab 6-1: Configuring Media Resources The Verification sections within the tasks provide all the verification that is needed.

Lab 6-2: Configuring User Features The Verification sections within the tasks provide all the verification that is needed.



Lab 6-4: Configuring Presence and BLF Functionality The Verification sections within the tasks provide all the verification that is needed.

Lab 6-5: Configuring Additional Features The Verification sections within the tasks provide all the verification that is needed.



Date post:	18-Jul-2016
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CIPT1_5.0_Lab_Guide

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