Communications Manager (CUCM)

CCIE Voice Lab Workbook Volume I Version 3.
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Communications Manager (CUCM)
The following publication, CCIE Voice Lab Workbook Volume I Version 3.0, was developed by Internetwork Expert, Inc. All rights reserved. No part of this publication may be reproduced or distributed in any form or by any means without the prior written permission of Internetwork Expert, Inc. Cisco, Cisco Systems, CCIE, and Cisco Certified Internetwork Expert, are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain countries. All other products and company names are the trademarks, registered trademarks, and service marks of the respective owners. Throughout this manual, Internetwork Expert, Inc. has used its best efforts to distinguish proprietary trademarks from descriptive names by following the capitalization styles used by the manufacturer.
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CCIE Voice Lab Workbook Volume I Version 3.0 Disclaimer
The following publication, CCIE Voice Lab Workbook Volume I Version 3.0, is designed to assist candidates in the preparation for Cisco Systems CCIE Voice Lab Exam. While every effort has been made to ensure that all material is as complete and accurate as possible, the enclosed material is presented on an as is basis. Neither the authors nor Internetwork Expert, Inc. assume any liability or responsibility to any person or entity with respect to loss or damages incurred from the information contained in this workbook. This workbook was developed by Internetwork Expert, Inc. and is an original work of the aforementioned authors. Any similarities between material presented in this workbook and actual CCIE lab material is completely coincidental.
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CCIE Voice Lab Workbook Volume I Version 3.0
Table of Contents
Cisco Unified Communications Manager ............................................1
2.1 CUCM Navigation and GUI Interface ................................................ 1 2.2 CUCM CLI Interface .......................................................................... 1 2.3 Communications Manager Initialization ............................................. 1 2.4 Verifying Database Replication ......................................................... 2 2.5 NTP & Date/Time Groups.................................................................. 2 2.6 CUCM DHCP..................................................................................... 2 2.7 IOS DHCP ......................................................................................... 2 2.8 Phone Auto-Registration ................................................................... 2 2.9 SIP Phone Endpoints ........................................................................ 3 2.10 Basic User Configuration ................................................................. 3 2.11 CUCM OS Users ............................................................................. 3 2.12 Softkey Templates ........................................................................... 3 2.13 Phone Button Templates ................................................................. 4 2.14 Common Phone Profile ................................................................... 4 2.15 Directory Number/Line Configuration .............................................. 4 2.16 Partitions.......................................................................................... 5 2.17 Calling Search Spaces .................................................................... 5 2.18 Device Pools.................................................................................... 5 2.19 Gateway - MGCP T1 PRI ................................................................ 5 2.20 Gateway - MGCP T1 CAS ............................................................... 6 2.21 Gateway H.323 ............................................................................. 6 2.22 SIP Trunk......................................................................................... 6 2.23 Media Resources Conference Bridge........................................... 6 2.24 Media Resources Transcoder ...................................................... 7 2.25 Music On Hold ................................................................................. 7 2.26 CUCM Applications - IPMA ............................................................. 7
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Cisco Unified Communications Manager Solutions ............................8

2.1 CUCM Navigation and GUI Interface ................................................ 8 2.2 CUCM CLI Interface ........................................................................ 12 2.3 Communications Manager Initialization ........................................... 14 2.4 Verifying Database Replication ....................................................... 25 2.5 NTP & Date/Time Groups................................................................ 28 2.6 CUCM DHCP................................................................................... 32 2.7 IOS DHCP ....................................................................................... 36 2.8 Phone Auto-Registration ................................................................. 38 2.9 SIP Phone Endpoints ...................................................................... 42 2.10 Basic User Configuration ............................................................... 47 2.11 CUCM OS Users ........................................................................... 51 2.12 Softkey Templates ......................................................................... 52 2.13 Phone Button Templates ............................................................... 56 2.14 Common Phone Profile ................................................................. 59 2.15 Directory Number/Line Configuration ............................................ 61 2.16 Partitions........................................................................................ 65 2.17 Calling Search Spaces .................................................................. 68 2.18 Device Pools.................................................................................. 71 2.19 Gateway - MGCP T1 PRI .............................................................. 73 2.20 Gateway - MGCP T1 CAS ............................................................. 86 2.21 Gateway H.323 ........................................................................... 91 2.22 SIP Trunk....................................................................................... 98 2.23 Media Resources Conference Bridge....................................... 107 2.24 Media Resources Transcoder .................................................. 109 2.25 Music On Hold ............................................................................. 111 2.26 CUCM Applications - IPMA ......................................................... 114
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Cisco Unified Communications Manager

Note
Many lab scenarios require previous labs to be properly configured. Therefore, it is recommended to save your config, and ensure you load the config before attempting further labs.
2.1

CUCM Navigation and GUI Interface

Become familiar with the CUCM version 7 GUI interface Login to the system using the following credentials through webrowser: o Username: admin o Password: cciecisco (lowercase) Explore the menu system and learn the locations of configuration pages Use the navigation links to move between subsystems Use the find function to search within configuration pages Use the related links navigation menu
2.2

CUCM CLI Interface

Become familiar with the CUCM version 7 CLI interface Login to the system using the following credentials through SSH: o Username: admin o Password: cciecisco (lowercase) Explore the command line interface Learn the help system for command reference Learn useful show commands
2.3

Communications Manager Initialization

Change Server Hostnames to IP Addresses Change Phone reference URLs to reflect the hostname change Activate both CUCM systems Add the CUCM-SUB to the CUCM Group Ensure auto registration is enabled
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2.4

Verifying Database Replication

Verify the database replication between the Publisher and Subscriber Servers Use the GUI reporting system to verify the replication User the CLI show commands to verify the replication Check configuration pages on the Subscriber server
2.5

NTP & Date/Time Groups

Configure R1 as an NTP server Set the timezone on R1 to PST -8 Hours from UTC Ensure the most reliable source is used for NTP on R1 Configure the Publisher for NTP reference Configure a Date/Time Group for the HQ Branch (Timezone PST) Configure a Date/Time Group for the BR1 Branch (Timezone CST) Ensure group uses a 12 Hour Time Format
2.6

CUCM DHCP
Configure the Publisher as a DHCP server Configure a DHCP subnet for the HQ Site (Subnet: 177.1.11.1) Configure a DHCP subnet for the BR1 Site (Subnet: 177.1.11.1) Configure the respective switchports for the phones at both sites
2.7

IOS DHCP
Configure R1 as an IOS DHCP server Allocate addresses for both the HQ and BR1 Phones using the correct subnet (HQ: 177.1.11.1, BR1: 177.1.11.1) Ensure the correct options are specified to support the TFTP server for SIP and SCCP phones
2.8

Phone Auto-Registration
Auto-register phones from the HQ site Auto-register phones from the BR1 site Assign names and numbers to the phones as shown below: o HQ HQ IP Ph1 Number 1001 o HQ HQ IP Ph2 Number 1002 o BR1 HQ IP Ph1 Number 2001
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2.9

SIP Phone Endpoints

Migrate HQ IP Ph2 from SCCP to SIP firmware Assign names and numbers to the phones as shown below: o BR1 BR1 IP Ph2 Number 2002 Create a new phone template to support SIP phones Change the auto-registration enterprise parameter to support SIP phones
2.10 Basic User Configuration

Create users with the following parameters: o Password: cisco o Pin: 12345 o hquser1: HQ IP Ph1 o hquser2: HQ IP Ph2 o br1user1: BR1 IP Ph1 Assign the user to the respective phone Assign a primary extension to each user using the primary DN of the associated phone Assign both HQ users to the Standard CCM Admin Users group Assign the BR1 user to the Standard CCM End Users group
2.11 CUCM OS Users

Change the GUI admin username to: administrator Change the GUI admin password to: ciscocisco
2.12 Softkey Templates

Create a new softkey template for SCCP phones using the following settings: o Include all defaults o Add the Quality Reporting Tool to the On Hook State o Add the Callback button to the Ring Out State Assign the new softkey template to HQ IP Ph1
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2.13 Phone Button Templates

Create a new Phone Button template for 7960 SCCP phones using the following settings: o 3 Line Buttons o 3 Speed Dial Buttons Assign the new button template to HQ IP Ph1, and BR1 IP Ph1
2.14 Common Phone Profile

Create a new Common Phone Profile using the following settings: o The DND option should reject calls o Implement a Flash alert for incoming DND Calls o Disable User access to change phone background settings o Set a phone unlock password of 1111 Assign the new profile to HQ IP Ph2
2.15 Directory Number/Line Configuration

Create a new line (1010) to be assigned to HQ IP Ph2 and apply the following settings: o Unanswered calls should be forwarded to 2001 o Set the busy trigger to 4 calls, and no answer timer to 10 seconds o External Phone Number mask: 775201XXXX o New calls should only ring once Add a description, alerting name, line text label, and external phone number mask to each line on all registered phones Ensure unanswered calls to line 2 of HQ IP Ph2 are forwarded to 2001 External Phone Number Masks should be in the following pattern: o HQ: 775201XXXX o BR1: 312301XXXX
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2.16 Partitions
Create the following partitions and apply them to the following device lines: o Partition: Internal Device: HQ IP Ph1 Line 1, BR1 IP Ph 1 Line 1, HQ IP Ph2 Line 2 o Partition: Restricted Device: HQ IP Ph2 Line 1 Apply the Partition Internal as the auto registration partition
2.17 Calling Search Spaces

Create three calling search spaces containing the partitions shown below: o CSS_HQ_STANDARD Partition: Internal o CSS_HQ_PREMIUM Partition: Internal, Site2Site, Restricted o CSS_BR1 Partition: Site2Site Create the new partition Site2Site and assign to new line 2 of HQ IP Ph2 and BR1 Ph1 Line 2 Assign CSS_HQ_Standard to HQ Ph1, CSS_HQ_PREMIUM to HQ Ph2, and CSS_BR1 to BR1 Ph1
2.18 Device Pools

Create the following device pools and assign the phones at the respective sites: o DP_HQ o DP_BR1 Rename the Default pool to create DP_HQ Assign CSS_HQ_STANDARD for auto-registration under the DP_HQ and CSS_BR1 for DP_BR1
2.19 Gateway - MGCP T1 PRI

Configure R1s T1 PSTN connection as an MGCP gateway Ensure Layer 3 is bound to the CUCM Use only the first 3 channels of the T1 Ensure all remaining channels are busied out on the CUCM system Set channel selection to Top Down Add the gateway to the CUCM system INE.com 5
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2.20 Gateway - MGCP T1 CAS

Configure R1s T1 PSTN connection as an MGCP T1 CAS gateway Use only the first 3 channels of the T1 Use E&M Wink-Start signaling type Use Top Down Port Selection Set Port 1 to Inbound, Port 2 to Outbound, and Port 3 to Bothways Add the gateway to the CUCM system
2.21 Gateway H.323

Configure R2s T1 PSTN connection as a H.323 gateway Configure R2 for outgoing PSTN and incoming call routing Use only the first 3 channels of the T1 Incoming calls should first be routed to the Subscriber, then the Publisher Ensure the fallback timer is set to the minimum possible value Add the gateway to the CUCM system
2.22 SIP Trunk

Configure the CUCM to use R2 as a SIP Gateway Configure R2 for outgoing PSTN and incoming call routing Use only the first 3 channels of the T1 Incoming calls should first be routed to the Subscriber, then the Publisher Use the most reliable interface of R2 to source SIP signaling messages and media traffic Ensure SIP timers are set for minimal delay between call agents
2.23 Media Resources Conference Bridge

Configure R1s DSP Resources as a Conference Bridge Add the Conference Bridge to the CUCM server SCCP traffic should be sourced from the Voice Vlan Ensure only one conference session is allowed at any given time Ensure SCCP packets are marked as IP Precedence 3
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2.24 Media Resources Transcoder

Configure R2s DSP Resources as a Transcoder Add the Transcoder to the CUCM server as R2_XCODER SCCP traffic should be sourced from the Voice Vlan Ensure only 3 sessions are allowed at any given time Ensure SCCP packets are marked as IP Precedence 3
2.25 Music On Hold

Configure the Publisher and Subscriber to stream Music on Hold Music should be sent using the G.711 format inside the HQ region and using G.729 when streaming to BR1 devices Enable Multicast media delivery to the HQ and Br1 sites
2.26 CUCM Applications - IPMA

Configure the IPMA service in proxy mode to intercept calls to HQ Phone 1s primary line Use the following table as reference for user names and extension numbers to complete this task Phone HQ IP Ph1 HQ IP Ph2 Extension 1001 1002 1011 Comment IPMA Manager Line Assistant Primary Line Assistant Proxy Line
HQ Phone 1 is assigned to a manager and HQ Phone 2 to the managers assistant
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Cisco Unified Communications Manager Solutions

2.1

CUCM Navigation and GUI Interface

Become familiar with the CUCM version 7 GUI interface Login to the system using the following credentials: o Username: admin o Password: cciecisco (lowercase) Explore the menu system and learn the locations of configuration pages Use the navigation links to move between subsystems Use the find function to search within configuration pages Use the related links navigation menu
Configuration Login to the CUCM system through a web browser. (Internet Explorer is the only browser officially supported by Cisco, however Firefox will work) (Complete access details for the servers can be located in the Rack Rental Users Guide). The login page is shown below:
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The landing or home page of the CUCM system is shown below. Note the headings listed at the top of the page: System / Call Routing / Media Resources / Voice Mail / Device / Application / User Management / Bulk Administration / Help Each heading is a separate menu structure through which configuration pages are accessed. Become familiar with each section, as time management and quick GUI navigation are key aspects to success in the CCIE lab.
The image below is from the top right corner of the CUCM web interface. This menu is always accesible from each page in the interface. The menu is used for navigation between each subsystem within the CUCM interface. Each subsystem and its description are listed below the image:
Cisco Unified CM Administration: Primary configuration location. Used to access configuration pages for the Communications Manager.
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Cisco Unified Serviceability: The Serviceability subsystem includes Service activation, Control Center, trace settings, and various database mangement tools.
Cisco Unified OS Administration: The OS Administration subsystem includes systems for viewing server parameters and tools for low level server administration such as network parameters.
Disaster Recovery System: The Disaster Recovery subsystem is used to create backups and restore the system from them. (This is similar to the BARS system in CCM ver 4 and below)
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Cisco Unified Reporting: The Unified Reporting subsystem handles all reporting for the server including reporting on the Communications Manager.
Explanation The web based gui interface for CUCM ver 7 is designed to be easier to navigate than previous versions, and navigation is supported through the use of related links and aggregation of common systems. Exploring each menu and subsystem to learn the locations of configuration pages and options will greatly enhance the speed configurations can be made.
Verification To verify the task above, simply ensure that you can reach each page, and become familiar with the navigation and menu structure.
Further Reading: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/confi guration/guide/cmebasic.html
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2.2

CUCM CLI Interface

Become familiar with the CUCM version 7 CLI interface Login to the system using the following credentials through SSH: o Username: admin o Password: cciecisco (lowercase) Explore the command line interface Learn the help system for command reference Learn useful show commands Explore the utils commands Restart the CUCM Pub server
Configuration
computer_prompt$ ssh admin@177.1.10.10 admin@177.1.10.10's password: Last login: Fri May 29 16:28:35 2009 Welcome to the Platform Command Line Interface WARNING, VMware Virtual Environment Detected! VMware is NOT a supported platform!
(Note: This warning will not affect your lab session)

admin:? delete* file* help quit run* set* show* unset* utils* admin:utils system restart Do you really want to restart ? Enter (yes/no)? yes Appliance is being Restarted ... Warning: Restart could take up to 5 minutes. Shutting down Service Manager will take some time.. \ Service Manager shutting down services... Please Wait DONE!!!! Broadcast message from root (Thu Jun 11 14:26:15 2009): The system is going down for reboot NOW! Waiting . Operation succeeded restart now.
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Explanation CUCM version 7 is a linux based server platform as are Unity Connections and Unified Presence (CUPS). While the CLI is accessible, its command structure is limited to basic system administration tasks, and cannot be used for administration of the actual Communications Manager system. Access is only through SSH, as telnet is not supported. The same username and password used for the GUI interface are used to access the CLI interface. Once logged in through SSH, a prompt will appear denoting that you are an admin user. admin: From this prompt, typing a question mark ? will load the text based help system similar to the question mark in the IOS CLI on routers and switches. The commands available are shown above in the configuration section. Each command has a submenu which can also be referenced for help using the ?. The primary use of the CLI interface is low level system administration. The servers network configuration can modified here, various parameters can be set, configuration details and database status can be verified, and the system can be controlled through the utils commands. Note the messages displayed after the utils system restart command is issued. The time required for the system to reboot can be significant, however the command line interface is the easiest location to do so.
Verification To verify this section, simply ensure that you are able to complete all parts of this task: logging in, using CLI based commands and using the help structure.
Further Reading: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cli_ref/7_1_2 /cli_ref_712.html
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2.3

Communications Manager Initialization

Change Server Hostnames to IP Addresses Ensure auto registration is enabled Activate both CUCM systems Add the CUCM-SUB to the CUCM Group Change Phone reference URLs to reflect the hostname change
Configuration Step 1: From the CM Administration page, navigate to the Server configuration page: System > Server
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Next click on the CUCM7-PUB server link:
Change the Host Name to the IP address of the Publisher Server (177.1.10.10). Next add a description with the Host Name:
Ensure you click the save button which is located at both the top and bottom of the configuration pages. A warning window will appear, which must be accepted:
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Repeat this step for the Subscriber Server, replacing the Host Name with IP Address (177.1.10.20). Next, confirm both servers through the find page:
Step 2: Navigate to System > Cisco Unified CM:
Click the find button to display all CM systems:
Click the CUCM7-PUB link:
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Change the CM name to CM_177.1.10.10. Next define the starting and ending directory number whch will later be used for auto-registration. Uncheck the Autoregistration Disabled box. This will enable Auto-registraton for the Publisher server:
Finally, confirm the name change on the Find and List page:
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CCIE Voice Lab Workbook Volume I Version 3.0 Step 3:
Use the subsystem navigation menu and navigate to the Serviceability System: (Note that you will be required to login to the system, using the same credentials to access the CM Administation system)
Navigate to Tools > Service Activation:
Next, select the Publisher Server:
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From the service activation page, activate all necessary services: Notice that the services are grouped into categories, unlike the CCM 4.X servers. Activate each service in the screenshots below that show a checked box: (Note that not all services will be activated, as they are not required for the lab environment. These include the Messaging Interface, Webdialer Service, CTL Provider, and the Certificate Authority Proxy Function.)
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Finally, ensure you click save, and accept the prompt window that will appear:
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To verify that the services have been activated, navigate to Tools > Control Center Feature Services, and also Control Center Network Services. From these windows, you should see a service status of Started and a Activation Status of Activated next to each service you activated on the previous window:
Repeat the process for the Subscriber server, activating the same services except for the TFTP service, as the Publisher will act as the only TFTP server.
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Navigate back to the CM Administration subsystem. Next, navigate to System > Cisco Unified CM Group. Click find, and click the link for the group named Default. Change the name of the group to CUCM-GROUP, and add the Subscriber server to the group using the arrow below the selection window:
Step 5: Navigate to System > Enterprise Parameters. Under the Phone URL Parameters section, change all hostnames in the URLs to the IP addresses of the publisher server
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CCIE Voice Lab Workbook Volume I Version 3.0 Explanation
In order to implement any configuration, the Communications Manager servers must be enabled, and the basic configuration must be applied. This process includes service activation, and changing basic parameters such as hostnames, adding all applicable servers to the CUCM group, and enabling options such as Auto Registation. Although Auto Registration is not required for functionality, it is recommended for the lab environment, as the time savings it offers is extremely valuable. The process outlined in this task is in a specific order, as certain tasks, such as service activation must take place before the Subscriber server can be added to the CUCM group, and will be recognized as a CallManager by the Publisher. Verification Verification for this task is included in the configuration steps above. Ensure each step is completed, and displayed on the configuration page. Finally, Navigate to the OS Reporting subsystem, and click System Reports > Unified CM Cluster Overview. Generate a new report, and ensure the report is successfully generated, the Publisher server is recognized, and both servers are recognized in the group.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2servr.html Copyright 2009 Internetwork Expert 24 INE.com
2.4

Verifying Database Replication

Verify the database replication between the Publisher and Subscriber Servers Use the GUI reporting system to verify the replication User the CLI show commands to verify the replication Check configuration pages on the Subscriber server
Configuration Step 1: Navigate to the OS Reporting subsystem. Click System Reports > Unified CM Database Status. Generate a new report, and ensure the database replication is working correctly:
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Ensure the replication count is the same on both servers (Note that the replication count can be any value, as long as both servers match)
Step 2: SSH to the Publisher server, and perform the following steps:
admin:utils dbreplication status *********************************************************************** ***************************** This command reads and writes database information from all machines and will take quite some time...please be patient. *********************************************************************** ***************************** -------------------- utils dbreplication status -------------------Output is in file cm/trace/dbl/sdi/ReplicationStatus.2009_06_16_10_47_17.out Please use "file view activelog cm/trace/dbl/sdi/ReplicationStatus.2009_06_16_10_47_17.out " command to see the output admin:file view activelog cm/trace/dbl/sdi/ReplicationStatus.2009_06_16_10_47_17.out SERVER ID STATE STATUS QUEUE CONNECTION CHANGED ----------------------------------------------------------------------g_cucm7_pub_ccm7_0_1_11000_2 2 Active Local 0 g_cucm7_sub_ccm7_0_1_11000_2 3 Active Connected 0 Jun 11 No Errors or Mismatches found. Replication status is good on all available servers.
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Login to the Publisher server and ensure previous configuration applied on the Publisher server is present. This can be verified through any of the configuration pages, such as the CUCM group page. Explanation Checking the database replication between servers is a critical task before beginning your lab. Although it is rare that there is an issue, the consequences of not checking could be a failed CCIE lab. Each method can be used for checking the replication status, and is simply a matter of preference. A simple verification through the configuration pages should also be completed, to ensure configuration applied to the Publisher is written to the Subscriber server. If any status does not show a good replication status, a reboot will often fix the problem. If you find that the replication is not working in the CCIE lab, immediately notify the proctor. Verification The verificaton procedure for this task is included in the above configuration section. Simply ensure that all output shows a good replication status.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/7_1_2/report/cur ptg.html
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2.5

NTP & Date/Time Groups

Configure R1 as an NTP server Set the timezone on R1 to PST -8 Hours from UTC Ensure the most reliable source is used for NTP on R1 Configure the Publisher for NTP reference Configure a Date/Time Group for the HQ Branch (Timezone PST) Configure a Date/Time Group for the BR1 Branch (Timezone CST) Ensure group uses a 12 Hour Time Format
Configuration Step 1:
R1: ! ! Configure the timezone ! clock timezone PST -8 ! ! Enable the NTP server, and source the most reliable interface ! ntp master 5 ntp source loopback 0 ! VORack01R1#clock set 09:00:00 June 16 2009 Jun 16 09:00:00.000: %SYS-6-CLOCKUPDATE: System clock has been updated from 11:16:04 UTC Tue Jun 16 2009 to 09:00:00 UTC Tue Jun 16 2009, configured from console by console.
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On the publisher server, navigate to System > NTP Phone Reference. Set the IP address to the Loopback 0 interface IP of R1 (177.1.254.1)
Step 3: Navigate to System > Date/Time Group. Click the CMLocal link, and set the parameters shown in the screenshot below. Ensure you add the NTP Reference.
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After clicking save, add a new Date/Time Group for BR1. Fill in the following parameters:
Explanation Date/Time Groups are used to display times on phones in addition to logging purposes. The CUCM servers do not act as an NTP server themselves. Therefore, you must configure an IOS NTP server. Once the router is configured as an NTP server the CUCM server will sync its clock to this NTP reference. Ensure that you set the correct timezone for each site. Verification Verification can be completed on R1 for the NTP server and its configuration. Verification of time for each site can be accomplished in a later lab, once phones are registered to the CUCM system.
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R1: VORack01R1#show clock .09:01:04.519 PST Tue Jun 16 2009 VORack01R1#show ntp status Clock is synchronized, stratum 5, reference is 127.127.7.1 nominal freq is 250.0000 Hz, actual freq is 250.0000 Hz, precision is 2**18 reference time is CDE1E20C.21C8C02A (09:01:08.131 PST Tue Jun 16 2009) clock offset is 0.0000 msec, root delay is 0.00 msec root dispersion is 0.02 msec, peer dispersion is 0.02 msec
Further Reading:
http://www.cisco.com/en/US/docs/ios/12_1/configfun/configuration/guide/fcd303. html#wp1001170 - IOS NTP Configuration http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2ntpsv.html - NTP Reference http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2dtgrp.html - Date/Time Group
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2.6

CUCM DHCP
Configure the Publisher as a DHCP server Configure a DHCP subnet for the HQ Site (Subnet: 177.1.11.1) Configure a DHCP subnet for the BR1 Site (Subnet: 177.2.11.1) Only allocate addresses in the .100-.110 range Configure the respective switchports for the phones at both sites
Configuration Step 1: On the Publisher server, navigate to System > DHCP > DHCP Server. Configure the parameters as shown below:
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Navigate to System > DHCP > DHCP Subnet. Add a new DHCP Subnet for the HQ Site. Configure the parameters as shown below:
Repeat the procedure of adding a new DHCP subnet for BR1. Ensure you change the IPs to reflect the addressing at BR1. Note: CUCM version 7 and above is subject to a bug related to the Cisco security agent. If your DHCP server does not seem to be working, login to the CLI, and issue the command: utils csa disable. Then type yes to reset. Copyright 2009 Internetwork Expert 33 INE.com

SW1: ! interface range fastethernet 0/2 3 switchport mode access switchport access vlan 12 switchport voice vlan 11 spanning-tree portfast ! R1: ! interface fa 0/0.11 ip helper-address 177.1.10.10 R2: ! VORack01R2(vlan)#vlan 11 VLAN 11 created: VORack01R2(vlan)#vlan 12 VLAN 12 created: VORack01R2(vlan)#apply APPLY completed. VORack01R2(vlan)#exit APPLY completed. Exiting.... ! interface fastethernet 0/1/0 switchport trunk encapsulation dot1q switchport mode trunk switchport access vlan 12 switchport voice vlan 11 spanning-tree portfast ! interface vlan 11 ip helper-address 177.1.10.10 !
Explanation The configuration of DHCP on the CUCM server is greatly simplified in Version 7 compared to version 4. The configuration is completed through the CUCM interface, rather than through the OS. Note that in order for the DHCP server to work, the DHCP Monitor Server must be enabled, which was completed in the Initialization lab. Ensure you configure both option 66 and option 150. Both DHCP IP Options specify a TFTP server address, however option 66 is required for SIP phones and option 150 is required for SCCP phones. Once the configuration on the CUCM server is complete, the switchports for each phone must be configured, and the ip helper-address must be specified on the voice vlan the phones are connected to. Copyright 2009 Internetwork Expert 34 INE.com
CCIE Voice Lab Workbook Volume I Version 3.0 Verification
To verify the DHCP configuration, user the show cdp neighbor detail command on each switchport connected to a phone:
VORack01R2#show cdp neighbors fa 0/1/0 detail ------------------------Device ID: SEP00164661326D Entry address(es): 177.2.11.100 Platform: Cisco IP Phone 7960, Capabilities: Host Interface: FastEthernet0/1/0, Port ID (outgoing port): Port 1 Holdtime : 179 sec Version : P00308000800 advertisement version: 2 Duplex: full Power drawn: 6.300 Watts ! VORack01SW1#show cdp neighbors fa 0/2 detail ------------------------Device ID: SIP000E83286110 Entry address(es): IP address: 177.1.11.101 Platform: Cisco IP Phone 7960, Capabilities: Host Phone Interface: FastEthernet0/2, Port ID (outgoing port): Port 1 Holdtime : 179 sec Version : P0S3-08-8-00 <snip> ! VORack01SW1#show cdp neighbors fa 0/3 detail ------------------------Device ID: SEP000785356951 Entry address(es): IP address: 177.1.11.102 Platform: Cisco IP Phone 7960, Capabilities: Host Phone Interface: FastEthernet0/3, Port ID (outgoing port): Port 1 Holdtime : 125 sec Version : P00308000900 <snip>
Further Reading: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2dhsrv.html - DHCP Server http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2dhsub.html - DHCP Subnet Copyright 2009 Internetwork Expert 35 INE.com
2.7

IOS DHCP
Configure R1 and R2 as IOS DHCP servers Allocate addresses for both the HQ and BR1 Phones using the correct subnet (HQ: 177.1.11.1, BR1: 177.1.11.1) Only allocate addresses in the .50-.60 range Ensure the correct options are specified to support the TFTP server for SIP and SCCP phones
Configuration **Note**: Before applying this configuration, ensure the switchports have been shutdown, and you have removed the CUCM DHCP server.
R1: ! ! Exclude the proper range of addresses ! ip dhcp excluded-address 177.1.11.1 177.1.11.49 ip dhcp excluded-address 177.1.11.61 177.1.11.255 ! ! Configure the actual DHCP pool, defining the TFTP server options ! ip dhcp pool VOICE network 177.1.11.0 255.255.255.0 default-router 177.1.11.1 option 66 ip 177.1.10.10 option 150 ip 177.1.10.10 ! interface fa 0/0.11 ip helper-address 177.1.10.10 ! SW1: ! interface range fastethernet 0/2 3 switchport mode access switchport access vlan 12 switchport voice vlan 11 spanning-tree portfast ! R2: ! ip dhcp excluded-address 177.2.11.1 177.2.11.49 ip dhcp excluded-address 177.2.11.61 177.2.11.255 ! ! ip dhcp pool VOICE network 177.2.11.0 255.255.255.0 default-router 177.2.11.1 option 66 ip 177.1.10.10 option 150 ip 177.1.10.10 !
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! interface fastethernet 0/1/0 switchport mode access switchport access vlan 12 switchport voice vlan 11 spanning-tree portfast ! VORack01R2(vlan)#vlan 11 VLAN 11 created: VORack01R2(vlan)#vlan 12 VLAN 12 created: VORack01R2(vlan)#apply APPLY completed. VORack01R2(vlan)#exit APPLY completed. Exiting.... ! !
Explanation Configuring an IOS DHCP server is quicker than a CUCM based server. In general the parameters are similar, with exception to the configuration of excluded addresses rather than just allocated addresses. Ensure the switchports are configured properly, and the VLANs have been added to the VLAN database on R2. Verification Repeat the verification procedure from the CUCM DHCP lab, but ensure the phones have received the new addresses. In addition, you can use IOS show commands to ensure the DHCP server is distributing the addresses:
VORack01R1#show ip dhcp binding Bindings from all pools not associated with VRF: IP address Client-ID/ Lease expiration Type Hardware address/ User name 177.1.11.100 0100.1646.8070.81 Jun 16 2009 11:20 AM Automatic 177.1.11.101 0100.1646.6132.73 Jun 16 2009 11:17 AM Automatic
Further Reading:
http://www.cisco.com/en/US/docs/ios/12_2/ip/configuration/guide/1cfdhcp.html Copyright 2009 Internetwork Expert 37 INE.com
2.8

Phone Auto-Registration
Auto-register phones from the HQ site Auto-register phones from the BR1 site Assign names and numbers to the phones as shown below: o HQ HQ IP Ph1 Number 1001 o HQ HQ IP Ph2 Number 1002 o BR1 BR1 IP Ph1 Number 2001
Configuration Step 1: Disable the switchport 0/1/0 of R2 to ensure that the HQ phones register first. Once step 2 has been completed, ensure you enable the port on R2.
R2: ! interface fa 0/1/0 shut !
Step 2: Navigate to System > Cisco Unified CM. Ensure that the Auto-registration disabled check box is not checked. Next, navigate to System > Cisco Unified CM Group and ensure the auto-registration checkbox is checked:
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Once you have confirmed that Auto-registration is enabled, the phones should register with the Publisher server. To confirm that the HQ phones have registered, navigate to Device > Phone: and click find to display all phones.
Once both HQ phones have registered, remember to enable the port on R2. You should now see three phones registered.
Step 3: Discover the MAC addresses of each phone from SW1 and R2. Once you have each mac address you will be able to configure the proper phone parameters.
SW1: VORack01SW1#show cdp neighbor fa 0/2 detail | include Device ID Device ID: SEP000E83286110 VORack01SW1#show cdp nei fa 0/3 de | inc Device ID Device ID: SEP000785356951 R2: VORack01R2#show cdp nei fa 0/1/0 detail | include Device ID Device ID: SEP00164661326D
Step 3: On the publisher server, click each phone link and configure the respective parameters for each phone referencing the mac addresses found above. (HQ Ph1: sw1 fa 0/2, HQ Ph2: sw1 fa 0/3, BR1 Ph1 R2 fa 0/1/0)
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First configure the name, and ensure you click save:
Next, click the Line [1] link, under Association Information:
Enter the number for the phone and click save:
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Auto-registration is an extremely important time saving measure to be used in the CCIE lab, however basic phone registration may be completed for you. While the process may be completed, knowing the process can be important for troubleshooting. Basic phone parameters such as name and number are easy to assign, but time consuming, which is the reason for defining a number range in the auto registration parameters defined on the CM configuration. In order for the number assignment to work on the correct phones, use the method of shutting down switchports and bringing them up one by one. With this method, you can enable the port for HQ Ph1 and it will be assigned 1001, then proceed with each further phone. This eliminates the need to find each phone by mac address. Verification Simple verification can be completed by viewing the summary page of the phone list, and also by looking at the phone, or clicking the link to access the phones web page. In addition, you can use the Route Plan Report to verify each DN and its device association. Navigate to Call Routing > Route Plan Report:
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2autor.html
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2.9

SIP Phone Endpoints

Migrate HQ IP Ph2 from SCCP to SIP firmware Assign names and numbers to the phones as shown below: o BR1 BR1 IP Ph2 Number 2002 Create a new phone template to support SIP phones Change the auto-registration enterprise parameter to support SIP phones
Configuration Step 1: Create a new Phone Template: Navigate to Bulk Administration > Phones > Phone Template. Click Add New, then select 7960 as the Phone Type:
Next, select SIP as the Device Protocol:
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Adjust the following parameters shown in the screenshot. Note that any highlighted area indicates a required parameter. All other fields can be left to their default settings. Once all of the fields have been set, ensure you click save:
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Step 2: Navigate to Bulk Adminsitration > Phones > Migrate Phones > SCCP to SIP. From this page, you must perform a search function to limit the results to the phone you want to convert. Once you have located the phone, click next:
Select the Phone Template Name, ensure the Reset Phones option is checked, and check the Run Immediately option. Then click submit:
Navigate to Device > Phone, and click find to display all phones. HQ IP Ph2 will now be displayed, however the state will show rejected:
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In order to correct the issue of the phone being rejected, the auto-registration Enterprise Parameters must be modified to support SIP registration. Navigate to System > Enterprise Parameters. Change the Auto Registration Phone Protocol to SIP. Click save, then reset for the changes to take place:
Once all devices have reset, navigate back to Device > Phone. Confirm that all three phones are registered, including HQ IP Ph2 as a SIP phone:
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While both SCCP and SIP endpoints are supported by the CUCM system, only one protocol type can be set for auto-registration. Manual registration supports both phones types. Note that if auto-registration is set to one type, all phones that first attempt to auto-register will download the firmware for that protocol type, and migrate to that protocol, which is the reason for the phones first registering as SCCP. Using the migration tool you can convert SCCP to SIP, but not the reverse. If a phone needs to be coverted back from SIP to SCCP, you must change the enterprise parameter to SCCP, delete the phone, and wait for auto-registration to complete the firmware update process. Verification The verification process for this task is contained within the steps in the task. Simply ensure that all aspects of the steps above are completed, and use the outputs shown to verify the phone registration with SIP. Once a phone registers as a SIP phone, note that the screen display will appear quite differently from that of a SCCP phone.
Further Reading:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note0918 6a0080094584.shtml#siptosccpccm5 - SCCP to SIP Migration http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2autor.html - Auto-registration Enterprise Parameters
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2.10 Basic User Configuration

Create users with the following parameters: o Password: cisco o Pin: 12345 o hquser1: HQ IP Ph1 o hquser2: HQ IP Ph2 o br1user1: BR1 IP Ph1 Assign the user to the respective phone Assign a primary extension to each user using the primary DN of the associated phone Assign both HQ users to the Standard CCM Admin Users group Assign the BR1 user to the Standard CCM End Users group
Configuration Step 1: Navigate to User Management > End User. Click Add New. Fill in the fields using the information in the task list as shown below for each user. All other fields can be left blank. Click save once the fields have been filled:
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Next, click the Device Association button:
Click the Find button and select the matching phone. Then click the Save Selected/Changes button.
Use the Related Links navigation menu to go back to the user configuration page:
Scroll down the page, and locate the Directory Number Associations drop down menu. Select the appropriate DN to associate with the user. Ensure you click save before leaving the page.
Step 2: Repeat these steps for the remaining users/phones.
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Navigate to User Management > User Group. (Note that you can add a user to a group from the user configuration page as well.) Click the find button to locate all pre-configured groups. Then click the appropriate group name to add a user:
Click the Add End Users to Group button:
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A new pop up window will appear in which you can chose the users to add. Once you have selected the users, click Add Selected and close the window.
Repeat this step for the End User group for br1user1. Explanation End Users and their association to phone endpoints are used by the CUCM system for multiple purposes. With their username and pin users can login to their phone, allowing access to further services, applications and various options, or permissions such as bypassing certain calling restrictions. User accounts are also needed for features such as IPMA which will be covered in a later lab. User groups are primarily used in large installations, however you can set certain permissions for groups, and apply them to entire groups, rather than having to set parameters for individual users. Therefore, groups can be a great time saving measure. Verification To verify this task, first check to see that all user accounts have been created on the end user page, and that the proper users are associated to the correct groups.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 8user.html Copyright 2009 Internetwork Expert 50 INE.com
2.11 CUCM OS Users

Change the GUI admin username to: administrator Change the GUI admin password to: ciscocisco
Configuration
CUCM CLI Interface: admin:utils reset_ui_administrator_name -------------------- utils reset_ui_administrator_name ---------------Reset user interface administrator user name New administrator user name: Administrator user name has been successfully updated to administrator
admin:utils reset_ui_administrator_password New password: Confirm new password: -------------------- utils reset_ui_administrator_password ----------Reset user interface password admin 06/17/2009 14:56:48.033 : Update SUCCEEDED
Explanation While resetting the admin username and password are not typically recommended, a lab task may require it, or you may have to due to a troubleshooting scenario. Knowing the syntax to do so will save time. Verification To verify that the username and password have been changed, login with the new credentials through the GUI interface of the CUCM server.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cli_ref/7_1_2/cli_ref_712 .html#wp42401
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2.12 Softkey Templates

Create a new softkey template using the following settings: o Include all defaults o Add the Quality Reporting Tool to the On Hook State o Add the Callback button to the Ring Out State Assign the new softkey template to HQ IP Ph1
Configuration Step 1: Navigate to Device > Device Settings > Softkey Template. Click Find to list all pre-configured templates. Click on the Standard User template and then click copy on the template page:
Once the template has copied, type in a new name, and click save:
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Next, use the Related Links Navigation Menu to access the Configure Softkey Layout page.
The first Call State that will be displayed is On Hook. Select the Quality Report Tool and move it to the Selected Softkeys pane using the middle movement arrow:
Next, change the state to Ring Out and select the Call Back option. Finally, click Save. Step 2: Navigate to Device > Phone and select HQ IP Ph1. On the configuration page, assign the new softkey template (Standard User New) Click Save, then Reset
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Softkey Templates control the softkeys or buttons below the screen of the phone. The CUCM system has pre-configured templates that are assigned by default, which you can copy and modify to meet custom needs or requirements. Once you have copied a pre-configured template, you must choose which softkeys will be assigned to which phone states. The two softkeys used in this task are QRT, the Quality Reporting Tool which allows users to submit call data for troubleshooting to a system administrator, and Call Back which will alert the phone user when an extension they tried to reach that was previously busy is now available. To learn about all of the available softkeys and their use, click the Cisco E-Learning link in the further reading section below. Each phone state and its description are shown below: Phone States: On Hook: This indicates the phone is currently not on an active call. (Sometimes referred to as the idle state) Connected: Indicates the phone is currently connected to an active call. On Hold: The phone is currently On Hold on a call. Ring In: The phone has received an incoming call and is currently ringing Off Hook: The phone is currently in an in use state, typically indicating that digits will be dialed to complete a call. Connected Transfer: The phone is performing a consultation call to initiate a transfer. Digits After First: The off hook state after the first digit has been dialed (such as 9 for secondary dialtone) Connected Conference: The phone is performing a consultation call to initiate a conference. Ring Out: The phone has initiated a call, and the remote destination is currently ringing. Off Hook With Feature: The Off Hook state for Connected Conference and Connected Transfer calls. Remote In Use: If a phone shares a DN with another phone, this indicates that the line is in use by the other phone. Connected No Feature: Whisper intercom call in progress. Copyright 2009 Internetwork Expert 54 INE.com
Verification To verify this task, check your configuration and ensure the correct template has been applied to the phone. HQ IP Ph1 should now display the new softkeys:
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 6skey.html - Softkey Templates http://www.cisco.com/E-Learning/bulk/public/celc/7960_Tutorial/index.html Softkey Definitions
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2.13 Phone Button Templates

Create a new Phone Button template for 7960 SCCP phones using the following settings: o 3 Line Buttons o 3 Speed Dial Buttons Assign the new button template to HQ IP Ph1, and BR1 IP Ph1
Configuration Step 1: Navigate to Device > Device Settings > Phone Button Template. Click the Find button and locate the Standard 7960 SCCP link. Click the link and copy the template. Next, name the new template: 7960 SCCP 3+3
Set the first three buttons to be used for Line and the next three for Speed Dial:
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Navigate to Device > Phone. Click the link for HQ IP Ph1 and apply the new button template, ensuring to click save:
Repeat this step for BR1 IP Ph1.
Explanation Phone button templates control the buttons located on the right side of the Cisco IP Phone. Each button on an SCCP phone can be set to either a Line or Speed Dial button. SIP phones however can only have buttons set to Line. On the 7960 button page, you will notice up to 34 buttons. 6 buttons are built into the phone itself, and the remaining button slots are reserved for add on modules. By default, the first two buttons are assigned to lines, and the remaining four are assigned to speed dials.
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CCIE Voice Lab Workbook Volume I Version 3.0 Verification
To verify this task, click reset on the phone configuration page, and ensure the Association Information pane now shows the correct button settings:
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmsys/a0 8ipph.html#wp1106111
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2.14 Common Phone Profile

Create a new Common Phone Profile using the following settings: o The DND option should reject calls o Implement a Flash alert for incoming DND Calls o Disable User access to change phone background settings o Set a phone unlock password of 1111 Assign the new profile to HQ IP Ph2
Configuration Step 1: Navigate to Device > Device Settings > Common Phone Profile. Select the pre-configured profile, and click the copy button. Next, change the settings as shown below and click save:
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Navigate to Device > Phone and select HQ IP Ph 2. Apply the new profile to the phone, click save, and then reset:
Explanation Common Phone Profiles have limited options, but can be useful for creating a template to apply to phones to modify any of these options. Any options configured at the device level will override these settings.
Verification First, ensure the correct profile is applied on the phone settings page once the phone has reset. To test the actual settings applied in the profile, you must first add the DND key to the On Hook state of a softkey template and apply it to the phone. Reference the previus softkey lab for detailed instructions. Once the template has been applied, reset the phone and press the DND key on HQ Ph2, then call from Br1 Ph1 and, ensure that the call is rejected, but a visual flash indicates the call attempt.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 6cpprf.html Copyright 2009 Internetwork Expert 60 INE.com
2.15 Directory Number/Line Configuration

Create a new line (1010) to be assigned to HQ IP Ph2 and apply the following settings: o Unanswered calls should be forwarded to 2001 o Set the busy trigger to 4 calls, and no answer timer to 10 seconds o External Phone Number mask: 775201XXXX o New calls should only ring once Add a description, alerting name, line text label, and external phone number mask to each line on all registered phones Ensure unanswered calls to line 2 of HQ IP Ph2 are forwarded to 2001 External Phone Number Masks should be in the following pattern: o HQ: 775201XXXX o BR1: 312301XXXX
Configuration Step 1: Create a new line for HQ IP Ph2 (1010). Navigate to Device > Phone, and for HQ IP Ph2 click the Add a New DN link on the phone configuration page:
Configure the appropriate settings for the new DN:
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Step 2: Set the relevant settings for the DN/Line configuration: Notice that you can set call forward to either voicemail using the checkbox, or a configured destination as shown below:
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The display settings for the DN/Line are configured below. Notice that the phone number mask uses all uppercase Xs. This is required for the configuraton page to recognize the digits. All other settings are left to their defaults:
Finally, set the Busy trigger to 4 calls, and click the save button:
Step 3: Repeat the applicable steps to configure the appropriate settings for the lines on each of the remaning phones. Copyright 2009 Internetwork Expert 63 INE.com
Each DN/Line can be modified separately from the phones device level settings and other DN/Lines. This means that any configuration, such as a call forward destination will apply to that line only, and not as a device level global setting. Any configuration to a DN/Line will override settings applied to the device configuration page. This will affect later configuration such as partitions and calling search spaces. The Display or Alerting name will only be shown for internal calls. Calls to the PSTN will not recognize this information, and the data will therefore not be transferred, but any calls within the CUCM system will carry this information. With this capability, a call between phones registered to the CUCM system will display a name instead of just a phone number. The external phone number mask is what the system uses to determine the fully qualified E.164 number, and build internal routing mechanisms. The E.164 number is the standard numbering plan used in the PSTN, and therefore is required for external calls. You can either configure the full number in the field, including the 4 digits assigned to the line, known as the extension, or use XXXX to indicate 4 variable digits. In the details above, the mask for the HQ site is denoted as 775201XXXX, this indicates that all calls to 775-201-Any four digits will be considered as local to the CUCM system. For internal call routing, the system simply uses the 4 digit extension, and ignores the phone number mask. Settings such as busy out timers and ring length timers are typically set in either seconds or a quantity of calls/rings. Each field will indicate which variable value is required. Verification To verify this task, first check to see that all of the settings are correct on the configuration page for the DN. Once the phone has reset after you click save, ensure the display on the phone is correct. Notice the label next to line 2. Finally, call HQ Ph2 from Ph1 to ensure the call forward and ring settings are correct.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 3dn.html Copyright 2009 Internetwork Expert 64 INE.com
2.16 Partitions
Create the following partitions and apply them to the following device lines: o Partition: Internal Device: HQ IP Ph1 Line 1, BR1 IP Ph 1 Line 1, HQ IP Ph2 Line 2 o Partition: Restricted Device: HQ IP Ph2 Line 1 Apply the Partition Internal as the auto registration partition
Configuration Step 1: Navigate to Call Routing > Class of Control > Partitions. Click Add New. Next, simply type the names of the two partitions being added:
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Step 2: Navigate to Device > Phone. Select HQ IP Ph 1. Set the Partition on the DN configuration page to Internal, then click save:
Step 3: Repeat the step above to configure the appropriate partition for each line defined in the task details. Step 4: Navigate to System > Cisco Unified CM. Click the link to configure the Publisher server. Set the partition that will automatically be applied to phones auto-registering with the CUCM system:
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Partitions are logical groupings used to restrict access to lines. Any DN within a partition can reach any other DN within the same partition. If one DN is in partition Internal, by default, it will not be able to reach a DN in partition Restricted. The ability to reach other partitions is accomplished with the use of Calling Search Spaces which will be explained further in the next lab. As you noticed when adding the partitions, the submenu they are located in is labeled Class Of Control. Essentially Class of Control is the ability to set restrictions on calls. Partitions are one of the most basic control mechanisms, and limit the locations within the sytem a certain DN can call or be called from. Partition based restrictions are used for configurations such as IPMA, and limiting certain phones from calling others such as limiting a lobby phone from calling a managers line. As a note, Partitions and Calling Search Spaces can be named anthing you chose, but should be something easy to remember. One easy method of differentiation is to prefix naming within the CUCM system. As an example for a partition, add PT_ before the name = PT_INTERNAL, PT_RESTRICTED, etc Verification Once the partitions have been applied to each line, ensure that you can place calls from each line within the same parition to each other line within the same parition. Next, place a call from one to partition to another, the call should not go through, as the system will automatically block it due to the class of control.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 3parti.html
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2.17 Calling Search Spaces

Create three calling search spaces containing the partitions shown below: o CSS_HQ_STANDARD Partition: Internal o CSS_HQ_PREMIUM Partition: Internal, Site2Site, Restricted o CSS_BR1 Partition: Site2Site Create the new partition Site2Site and assign to new line 2 of HQ IP Ph2 and BR1 Ph1 Line 2 Assign CSS_HQ_Standard to HQ Ph1, CSS_HQ_PREMIUM to HQ Ph2, and CSS_BR1 to BR1 Ph1
Configuration Step 1: Repeat the steps from the previous lab to add a new partition (Site2Site). Assign the partition to HQ Ph2 Line 2 and Br1 Ph1 Line 2. Step 2: Navigate to Call Routing > Class of Control > Calling Search Space. Click Add New, and create the new CSS. Use the arrows to add the correct partition:
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Repeat the step above to create the next two CSSs. After clicking Save, you can use the Copy button to easily create a new CSS. For each additional CSS, ensure that the correct partitions are added. Once you have completed the configuration, the list of CSSs should match that shown below:
Step 3: Navigate to Device > Phone. Select the first phone to configure and add the Calling Search Space as a device level setting on the phone configuration page:
Repeat this step to assign the proper CSS to each phone.
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Calling Search Spaces work directly with Partitions as a Class of Control method within the CUCM system. CSSs are used to group partitions and define access between different partitions. As explained in the Partitions lab, only DNs within the same partition can contact each other. Calling Search Spaces allow a list of partitions to be configured that define which partitions a device can reach. If a CSS with three partitions is assigned to a phone, that phone can then reach each of those three partitions, even if the line on that phone is configured in a single partition. CSSs can be configured at both the device level, and for individual DNs. Any configuration of the DN will override device level configuration. This means that if a CSS is assigned to the device that only allows access to partitions X+Y, but a CSS is assigned to a DN on the phone that allows access to X+Y+Z, a call on that phone line will be able to reach X,Y, or Z. In addition to simply allowing access to partitions, CSSs also allow you to sort the order of the partitions assigned to a specific CSS. The CUCM routing process will hunt through the partitions in the order they are listed. Use the arrows on the right side of the Selected Partitions windows to arrange the order:
Verification To verify the task, place calls from each phone and ensure that you can reach the phones and lines in the appropriate partitions. Each partition listed in the Calling Search Space should be able to be reached by any phone with that CSS assigned.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 3csspc.html
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2.18 Device Pools

Create the following device pools and assign the phones at the respective sites: o DP_HQ o DP_BR1 Rename the Default pool to create DP_HQ Assign CSS_HQ_STANDARD for auto-registration under the DP_HQ and CSS_BR1 for DP_BR1
Configuration Step 1: Navigate to System > Device Pool. Click the link to access the pre-configured Default device pool. Rename the pool and configure the settings shown below:
Step 2: After clicking save, use the Add New button to create a new device pool for BR1. Configure the applicable settings and click save. Ensure you change the CSS for Auto-registration, and also change the Date/Time Group to the group configured for BR1.
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Assign the BR1 device pool to BR1 IP Ph1:
(Note: You will not have to assign the other device pool to the HQ devices, as the default device pool will always apply during Auto-registration, and the phones are therefore already configured)
Explanation Device Pools are used as templates to set parameters for groups of devices such as phones and gateways on the CUCM system. A device pool has many parameters which can be configured in one location and then applied to multiple devices. The pool eliminates the need for each setting to be configured per device. Once the device is put into a device pool, it will inherit all configuration settings for that device pool. (Additional parameters will be used in further labs, but for this lab, all other settings can be left to their defaults) Verification To verify the task, ensure that the settings have all been applied correctly, and that each phone has the correct device pool set. You can also check a setting such as the phone time to ensure the phone inherited the Date/Time Group setting from the device pool configuration.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 2devpl.html
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2.19 Gateway - MGCP T1 PRI

Configure R1s T1 PSTN connection as an MGCP gateway Ensure Layer 3 is bound to the CUCM Use only the first 3 channels of the T1 Ensure all remaining channels are busied out on the CUCM system Set channel selection to Top Down Add the gateway to the CUCM system
Configuration Step 1: Locate the port. The location will be used to configure the T1 and gateway.
VORack01R1#show diag 3725 Backplane EEPROM: <snip> Slot 0: C3725 Mother board 2FE(TX)-3W Port adapter, 7 ports Port adapter is analyzed Port adapter insertion time unknown EEPROM contents at hardware discovery: PCB Serial Number : JAD06490P9P Processor type : 61 Top Assy. Part Number : 800-16147-01 Board Revision : G0 Fab Part Number : 28-4226-06 Deviation Number : 65535-65535 Manufacturing Test Data : FF FF FF FF FF FF FF FF RMA Number : 255-255-255-255 RMA Test History : FF RMA History : FF Chassis Serial Number : JMX0651L1US Chassis MAC Address : 000b.5ffd.d300 MAC Address block size : 48 Field Diagnostics Data : FF FF FF FF FF FF FF FF Hardware Revision : 0.1 Number of Slots : 2 Product (FRU) Number : C3725-2FE <snip> WIC Slot 0: T1 (2 port) Multi-Flex Trunk WAN daughter card Hardware revision 1.0 Board revision B0 Serial number 31504287 Part number 800-04477-01 FRU Part Number VWIC-2MFT-T1= Test history 0x0 RMA number 00-00-00 Connector type PCI <snip>
From the output above, the location is determined to be 0/0 (Slot 0 of the router, Slot 0 of the motherboard) Copyright 2009 Internetwork Expert 73 INE.com

R1:
! ! Define the isdn switch-type (In this case NI2) ! isdn switch-type primary-ni ! ! Enable clock participate ! network-clock-participate wic 0 ! ! Configure the controller, and ensure the PRI protocol is MGCP ! controller t1 0/0 framing esf linecode b8zs pri-group timeslots 1-3 service mgcp ! ! ! <Notice the console messages displayed below. The T1 controller ! Interface has come up, in addition to a new Serial interface that has ! been automatically created by the configuration (Serial 0/0:23)> ! Jun 21 14:34:26.812: %CONTROLLER-5-UPDOWN: Controller T1 0/0, changed state to up VORack01R1(config-controller)# Jun 21 14:34:28.812: %LINK-3-UPDOWN: Interface Serial0/0:23, changed state to up ! ! Bind the Layer 3 packets to the CUCM system ! interface serial 0/0:23 isdn bind-l3 ccm-manager ! ! Configure the MGCP settings, ensuring the call agent and redundant ! host are both configured ! mgcp mgcp call agent 177.1.10.20 mgcp bind control source loopback 0 mgcp bind media source loopback 0 mgcp dtmf voip codec all mode out-of-band ! ccm-manager redundant-host 177.1.10.10 ccm-manager mgcp ccm-manager switchback immediate !
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On the CUCM Publisher server, navigate to Device > Gateway. Click the Add New button, and follow through the steps to configure the gateway:
After selecting the type (3725), select the protocol (MGCP)
Configure the gateway. Use the full hostname of R1 for the domain name. Select the NM-4VWIC-MBRD as the module in Slot 0, and NI2 for the switchtype:
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Step 4: Click save, then select the subunit where the card is located. This location was determined earlier (0/0):
Click save once again, then click the link for the T1 Card:
Chose the gateway protocol to be used, in this case PRI:
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Configure the applicable settings for the gateway. The End-Point name and description are automatically configured, and will be used later for configuration of the channel status. All settings required must be configured as shown below:
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Continue with further configuration of the gateway. The PRI Protocol Type is automatically set, however you must change the Channel Selection Order. Next, uncheck the Inhibit restarts at PRI initialization, and check the Enable status poll option boxes.
Also check the following three options, which will enable IE (information element) delivery and redirection:
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Click the save button, and copy the End-Point Name.
Step 5: Navigate to System > Service Parameters. Chose the Publisher server, and the Call Manager Service:
Once the page has loaded, click the Advanced button, which will display all of the advanced parameters:
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Locate the Change B-Channel Maintenance Status 1 Field under the Clusterwide Parameters, and enter the endpoint name of the gateway followed by = 0001 1111 1111 1111 1111 1111. The field should appear like this: S0/SU0/DS1-0@VORack01R1 = 0001 1111 1111 1111 1111 1111
The digit string entered above is used to busy out the remaining channels on the PRI. As you can see, the first three channels are denoted as open with the use of 0 and the remaing 21 channels are denoted as busy out with the use of 1. This can be verified in the verification section of the lab using the router IOS show commands. Ensure you click save after entering the data in the field. Explanation Voice gateways are used to communicate with the PSTN by transmitting voice packets from a call processing system such as CUCM across a connection such as a T1 PRI. In this case, the gateway is controlled by the CUCM system using MGCP (Media Gateway Control Protocol). The configuration first begins on the IOS platform, as you enable the T1 controller, define the channels to be used, and the router creates both a voice-port and serial interface. Within the IOS configuration, certain parameters must be set for the PRI and MGCP to work properly. The ISDN switch-type must first be configured, which specifies what type of switch-type is supported by the remote end. In this case, a North American standard of NI2 is used for the HQ location. As a note, the switch-type can be configured as either a global or interface specific setting. The next parameter which must be set is network-clock-participate for the wic card. This parameter must be set in order to ensure correct processing and sequenced delivery of voice packets. The reason for this command necessity is related to the routers default behavior, which is to use free running clocking on serial interfaces which does not affect data packets. This essentially means that the backplane of the router will keep a running clock, but it in no way affects the data processing timeline for the packets arriving on a serial interface. By enabling network-clock-participate, the router backplane will use the same time structure or clocking from the WIC card for its data processing and will deliver the voice packets to the DSP resources in proper sequence. Copyright 2009 Internetwork Expert 80 INE.com
After the global parameters have been set, the controller specific configuration must be applied to instruct the router to use the T1 controller as a serial voice interface (in this case PRI) rather than a serial data connection. The framing and linecode are both default settings, but should be configured to ensure the proper types have been set. Next, the pri-group command enables the PRI voice interface, and the channels to be used are set. Finally, the controller must be enabled for MGCP control, using the service mgcp option. As shown in the configuration above, a serial interface is automatically created by the router, in addition to a voice-port. Both the serial interface and voice-port use the same location identifier 0/0 followed by the T1 addressing of :23 (serial 0/0:23, voice-port 0/0:23) Under the serial interface, the isdn bind-l3 ccmmanager command must be issued, which binds the layer 3 ISDN Q.931 (call control and signaling) on the interface to the CUCM system through backhaul. Finally, the MGCP specific parameters are configured. As in other configurations, the loopback interface is used to source the packets from the MGCP process, as the loopback interface is the most reliable. Ensure that you configure both the primary call agent and the redundant-host in case of CUCM fail over. Notice that the SUB address is used for the primary call agent. This configuration is a best practice, as the Subscriber server should be used for all call processing whenever available. Next is the CUCM configuration to enable and configure a voice gateway that the CUCM system can use for PSTN communication. Similar to the IOS configuration, the MGCP parameters must be defined. Some parameters are already defined, while others must be set or changed. The Channel Selection Order is set to Top Down, which defines the order the system hunts for available channels. The logic here is reversed from a numerical perspective, as the channels are considered to be top at 1 and bottom at 24, therefore Top Down selection will select channel 1 first. The inhibit restarts at PRI initialization box is unchecked to allow the system to monitor the D channel state. If the box is left checked, the system always assumes the channels are available, which can lead to dropped calls if the PRI is not in fact ready or in service. The enable status poll box must be checked to set the channel busy out string in the service parameters, as only 3 channels are being used. The significant digits field is used to define the digits the CUCM system recognizes for internal calls, and truncates incoming numbers to. As all internal configurd DNs are 4 digits, the same value is set for the significant digits. As an example, if a call for HQ Ph1 comes in from the PSTN, the called number is 7752011001. Once the significant digits field is set, the gateway will automatically strip the leading digits, and keep 1001, the significant digits.
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Finally, the Display IE options are set, which control whether or not Information Elements will be displayed for the call. The IEs contain the call identification parameters such as caller ID and ANI. Once the gateway has been configured, the last step is modifying the service parameters to busy out the unused channels of the PRI. Verification Verification of the task will be completed using both the CLI and the CUCM GUI. First, begin by verifying the configuration and status of the T1 controller and ISDN connection to the PSTN:
VORack01R1#show isdn status Global ISDN Switchtype = primary-ni %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply ISDN Serial0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) <snip>
From the output above, we can see that the switchtype is correct, layer 3 is backhauled to the CUCM, and that the Layer 2 status is MULTIPLE_FRAME_ESTABLISHED.
VORack01R1#show isdn service PRI Channel Statistics: %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply ISDN Se0/0:23, Channel [1-24] Configured Isdn Interface (dsl) 0 Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend) Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 State : 0 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend) Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 State : 0 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2
The show isdn service command, as shown above displays the status of each PRI channel. The first three channels are available, which the remaining 21 are busy. Copyright 2009 Internetwork Expert 82 INE.com
You can also verify the automatic creation and status of the voice-port and serialinterface:
VORack01R1#show voice port 0/0:23 DS0 Group 0/0:23 - 0/0:23 Type of VoicePort is XCC Operation State is DORMANT Administrative State is UP No Interface Down Failure <snip> DS0 channel specific status info: PORT 0/0:23 0/0:23 0/0:23 CH 01 02 03 SIG-TYPE xcc-voice xcc-voice xcc-voice OPER none none none IN STATUS none none none Method NVRAM NVRAM NVRAM NVRAM NVRAM OUT STATUS none none none Status up up up up up TIP RING
VORack01R1#show ip interface brief Interface IP-Address OK? FastEthernet0/0 unassigned YES FastEthernet0/0.10 177.1.10.1 YES FastEthernet0/0.11 177.1.11.1 YES FastEthernet0/0.12 177.1.12.1 YES <snip> Serial0/0:23 unassigned YES <snip>
Protocol up up up up up
Next, verify the mgcp call manager configuration:

VORack01R1#show ccm-manager MGCP Domain Name: VORack01R1 Priority Status Host ============================================================ Primary Registered 177.1.10.20 First Backup Backup Ready 177.1.10.10 Second Backup None Current active Call Manager: <snip> 177.1.10.20
Backhaul Link info: Link Protocol: TCP Remote Port Number: 2428 Remote IP Address: 177.1.10.20 Current Link State: OPEN Statistics: <snip> PRI Ports being backhauled: Slot 0, port 0
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CCIE Voice Lab Workbook Volume I Version 3.0 Troubleshooting
T1 PRIs may experience issues in establishing a successful connection. The most common issue is a layer two status of TEI_ASSIGNED as shown below:
VORack01R1#show isdn status Global ISDN Switchtype = primary-ni ISDN Serial0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x80000007 Number of L2 Discards = 0, L2 Session ID = 4 Total Allocated ISDN CCBs = 0
Without a status of MULTIPLE_FRAME_ESTABLISHED, the connection will not work properly. The following steps can be used to fix this issue: ! First, shut down the controller and source interface of the MGCP packets
VORack01R1(config)#controller t1 0/0 VORack01R1(config-controller)#shut VORack01R1(config-controller)# Jun 22 20:29:44.542: %CONTROLLER-5-UPDOWN: Controller T1 0/0, changed state to administratively down VORack01R1(config-controller)# Jun 22 20:29:44.550: %ISDN-6-LAYER2DOWN: Layer 2 for Interface Se0/0:23, TEI 0 changed to down Jun 22 20:29:46.542: %LINK-3-UPDOWN: Interface Serial0/0:23, changed state to down VORack01R1(config-controller)#interface loopback 0 VORack01R1(config-if)#shut Jun 22 20:29:54.550: %LINK-5-CHANGED: Interface Loopback0, changed state to administratively down Jun 22 20:29:55.550: %LINEPROTO-5-UPDOWN: Line protocol on Interface Loopback0, changed state to down
! Remove the MGCP process

VORack01R1(config-if)#no mgcp WARNING: no mgcp: Teardown MGCP application may take a while to clean up resources Jun 22 20:29:55.806: %MGCP_APP-6-MGCP_SHUTDOWN_COMPLETE: MGCP Shutdown has completed
! Remove the layer 3 backhaul configuration

VORack01R1(config)#interface serial 0/0:23 VORack01R1(config-if)#no isdn bind-l3 ccm-manager
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! Now re-enable each interface, the MGCP process, and backhaul configuration
VORack01R1(config-if)#controller t1 0/0 VORack01R1(config-controller)#no shut Jun 22 20:30:38.426: %CONTROLLER-5-UPDOWN: Controller T1 0/0, changed state to up VORack01R1(config)# Jun 22 20:30:38.430: %ISDN-6-LAYER2UP: Layer 2 for Interface Se0/0:23, TEI 0 changed to up Jun 22 20:30:40.426: %LINK-3-UPDOWN: Interface Serial0/0:23, changed state to up VORack01R1(config)#interface loopback 0 VORack01R1(config-if)#no shut VORack01R1(config-if)#mgcp VORack01R1(config)# Jun 22 20:30:47.694: %LINK-3-UPDOWN: Interface Loopback0, changed state to up Jun 22 20:30:48.694: %LINEPROTO-5-UPDOWN: Line protocol on Interface Loopback0, changed state to up VORack01R1(config)#interface serial 0/0:23 VORack01R1(config-if)#isdn bind-l3 ccm-manager
! Now check the status of the interface, which has come up properly:
VORack01R1(config-if)#do show isdn status Global ISDN Switchtype = primary-ni %Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply ISDN Serial0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni L2 Protocol = Q.921 0x0000 L3 Protocol(s) = CCM MANAGER 0x0003 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: <snip>
These steps can be completed quickly to fix T1 PRI layer 2 problems. The other option would be a reload of the router, which will accomplish the same thing, as the configuration is reapplied as the router boots.
Further Reading:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note091 86a008017787b.shtml - IOS MGCP Configuration http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 6gtway.html#wp1311169 - Adding an MGCP Gateway in CUCM http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a008 01509d7.shtml - T1 Troubleshooting Copyright 2009 Internetwork Expert 85 INE.com
2.20 Gateway - MGCP T1 CAS

Configure R1s T1 PSTN connection as an MGCP T1 CAS gateway Use only the first 3 channels of the T1 Use E&M Wink-Start signaling type Use Top Down Port Selection Set Port 1 to Inbound, Port 2 to Outbound, and Port 3 to Bothways Add the gateway to the CUCM system
Configuration Step 1: Locate the port. The location will be used to configure the T1 and gateway.
VORack01R1#show diag 3725 Backplane EEPROM: <snip> Slot 0: C3725 Mother board 2FE(TX)-3W Port adapter, 7 ports Port adapter is analyzed Port adapter insertion time unknown EEPROM contents at hardware discovery: PCB Serial Number : JAD06490P9P Processor type : 61 Top Assy. Part Number : 800-16147-01 Board Revision : G0 Fab Part Number : 28-4226-06 Deviation Number : 65535-65535 Manufacturing Test Data : FF FF FF FF FF FF FF FF RMA Number : 255-255-255-255 RMA Test History : FF RMA History : FF Chassis Serial Number : JMX0651L1US Chassis MAC Address : 000b.5ffd.d300 MAC Address block size : 48 Field Diagnostics Data : FF FF FF FF FF FF FF FF Hardware Revision : 0.1 Number of Slots : 2 Product (FRU) Number : C3725-2FE <snip> WIC Slot 0: T1 (2 port) Multi-Flex Trunk WAN daughter card Hardware revision 1.0 Board revision B0 Serial number 31504287 Part number 800-04477-01 FRU Part Number VWIC-2MFT-T1= Test history 0x0 RMA number 00-00-00 Connector type PCI <snip>
From the output above, the location is determined to be 0/0 (Slot 0 of the router, Slot 0 of the motherboard) Copyright 2009 Internetwork Expert 86 INE.com

R1:
! ! Define the isdn switch-type (In this case NI2) ! isdn switch-type primary-ni ! ! Enable clock participate ! network-clock-participate wic 0 ! ! Configure the controller, ensuring to enable the signal type ! controller T1 0/0 framing esf linecode b8zs ds0-group 0 timeslots 1-3 type e&m-wink-start ! ! Bind the Layer 3 packets to the CUCM system ! interface serial 0/0:23 isdn bind-l3 ccm-manager ! ! Configure the MGCP settings, ensuring the call agent and redundant ! host are both configured ! mgcp mgcp call agent 177.1.10.20 mgcp bind control source loopback 0 mgcp bind media source loopback 0 mgcp dtmf voip codec all mode out-of-band ! ccm-manager redundant-host 177.1.10.10 ccm-manager mgcp ccm-manager switchback immediate !
Step 3: On the CUCM Publisher server, navigate to Device > Gateway. Click the Add New button, and follow the steps from the MGCP lab above to select the Chassis type (3725), Mainboard and Subunit cards. Next, select the Protocol Type:
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Set the Device Pool, and Port Selection Order:
Click save, then click the Add a New Port Link:
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Set the Port Type to EANDM:
Set the port parameters, ensuring you select the proper Port Direction as required by the task. Then click Save.
Repeat the process to add two additional ports, until all three ports are configured, setting the direction on each port.
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T1 CAS or (Channel Associated Signaling) also referred to as Robbed Bit Signaling differs from CCS (Common Channel Signaling) used in PRIs in the method it uses to transport signaling information. As the name implies, the signaling is carried on the same channels as the voice or data payload. Bits are robbed from the frames on each timeslot (The 6th, 12th, 18th, and 24th frames) This configuration is not commonly used in the CCIE lab, however knowing how to setup a gateway with this configuration is recommended, as it is mentioned in the CCIE blueprint. Verification To verify the configuration of the gateway. First ensure that the gateway is registered in the CUCM gui. Next, use the CLI of R1 to verify that the gateway created a voice-port (0/0:0) and that port is operational.
VORack01R1#show voice port 0/0:0 recEive and transMit Slot is 0, Sub-unit is 0, Port is 0 Type of VoicePort is E&M Operation State is UP Administrative State is UP <snip>
Further Reading:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a008 00e2560.shtml
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2.21 Gateway H.323

Configure R2s T1 PSTN connection as a H.323 gateway Configure R2 for outgoing PSTN and incoming call routing Use only the first 3 channels of the T1 Incoming calls should first be routed to the Subscriber, then the Publisher Ensure the fallback timer is set to the minimum possible value Add the gateway to the CUCM system
R2: ! ! Configure the switchtype and clocking ! isdn switchtype primary-ni network-clock-participate wic 0 ! controller t1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3 ! interface serial 0/0/0:23 isdn outgoing display-ie ! ! Configure the h323 gateway, binding to the loopback interface ! interface loopback 0 h323-gateway voip interface h323-gateway voip bind srcaddr 177.1.254.2 ! ! Create a voice class, using the minimal timeout value ! voice class h323 1 h225 timeout tcp establish 1 ! ! Create the incoming dial-peers ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! !
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! ! Configure two dial-peers. One to the SUB and one to the PUB ! dial-peer voice 3120 voip destination-pattern 312301....$ session target ipv4:177.1.10.20 voice-class h323 1 ! dial-peer voice 3121 voip destination-pattern 312301....$ session target ipv4:177.1.10.10 voice-class h323 1 ! ! Configure the outgoing dial-peers ! dial-peer voice 911 pots description == 911 == destination-pattern 911 port 0/0/0:23 forward-digits 3 ! dial-peer voice 9911 pots description == 9911 == destination-pattern 9911 port 0/0/0:23 forward-digits 3 ! dial-peer voice 900 pots description == Local Calls == destination-pattern 9[2-9]......$ port 0/0/0:23 ! dial-peer voice 901 pots description == Long Distance == destination-pattern 91[2-9]..[2-9]......$ port 0/0/0:23 prefix 1 ! dial-peer voice 902 pots description == International == destination-pattern 9011T port 0/0/0:23 prefix 011
Step 2: On the CUCM Publisher server, navigate to Device > Gateway. Click the Add New button, and follow through the steps to configure the gateway:
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Enter the configuration settings for the gateway: Use the Loopback IP from R2 as the Device Name. Also define the Device Pool.
Set the significant digits to 4, and the Calling Search Space to CSS_BR1. Check the Redirecting Number IE Delivery Inbound box.
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Finally, check the Display IE Delivery and Redirecting Number IE Delivery Outbound checkboxes. Then slick save.
Explanation An H.323 gateway performs the same basic functions as the MGCP gateway configured in the previous lab. The primary difference between the operation and configuration is related to the actual gateway control. While MGCP gateways are controlled by the CUCM system, H.323 gateways control themselves. Because the gateway control is handled by the IOS, the creation of dial-peers is necessary for call routing decisions. In the configuration steps, dial-peers are created for both inbound and outbound calls. Notice the first dial-peer configured with direct-inward-dial and incoming called-number . The reason for this configuration is related to incoming calls and the routers default behaviour of initiating dial-tone for a new call. By creating this dial-peer, the router will match all calls coming in from the T1 connection using the . and ensure they are processed as incoming DID calls, instead of issuing dial-tone to the call. Inbound dial-peers are configured next that direct calls to the CUCM system. Two dial-peers are created, first pointing to the Subscriber server, and then the Publisher. The destination-pattern is the same, but the session target is different. The gateway will hunt the dial-peer tags numerically, preferring the lower number, which points to the subscriber server. If the SUB is not available, the gateway will then redirect the call to the second dial-peer (PUB). Copyright 2009 Internetwork Expert 94 INE.com
The first outgoing dial-peers created are for calls to 911 (Emergency Services) Two dial-peers are created to ensure calls to 911 are processed correctly. If a user dials the standard access code of 9 then 911, or simply dials 911, both calls will be completed successfully. This configuration is to assist users during emergency periods, and is often a requirement of call routing tasks. Finally, the remaining outgoing dial-peers are configured. Notice the prefix statement, which will ensure that the necessary digits before the [ ] are included in the called number, as the router will automatically strip these digits during the digit collection process. In addition to the dial-peer configuration, a voice class was created and applied to each dial-peer pointing to the CUCM cluster. This voice class is configured to ensure proper redirection of the call if the subscriber server is not available. The H225 timeout controls the length of time a call will be attempted to the specific dial-peer. By default, the H225 timeout is set to 40 seconds, which will result in a failed call, as the ISDN Q.931 call pickup timer is 10 seconds. This configuration will therefore ensure that if the first dial-peer is not available, the call will immediately transfer to the next dial-peer. The CUCM configuration is less complex that an ISDN PRI gateway, as the gateway is not controlled by the CUCM system. Only simple parameters must be set for proper operation. Note that H.323 gateways will not appear as registered with the CUCM system. Verification Check the T1 PRI Status
VORack01R2#show isdn status **** No Global ISDN Switchtype currently defined **** ISDN Serial0/0/0:23 interface dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED <snip>
Check the operation of the H.323 Gateway

VORack01R2#show gateway H.323 ITU-T Version: 4.0 H.323 service is up H323 Stack Version: 0.1
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CCIE Voice Lab Workbook Volume I Version 3.0 Check the configuration of the dial-peers
VORack01R2#show dial-peer voice summary dial-peer hunt 0 AD PRE PASS OUT TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT 1 pots up up 0 down 3120 voip up up 312301....$ 0 syst ipv4:177.1.10.20 3121 voip up up 312301....$ 0 syst ipv4:177.1.10.10 911 pots up up 911 0 up 0/0/0:23 9911 pots up up 9911 0 up 0/0/0:23 900 pots up up 9[2-9]......$ 0 up 0/0/0:23 901 pots up up 1 91[2-9]..[2-9]......$ 0 up 0/0/0:23 902 pots up up 011 9011T 0 up 0/0/0:23
The output above shows the summary details and status of the configured dialpeers. Each dial-peer listed is identified by the tag number defined in the dialpeer configuration (dial-peer voice 911 pots). The type, status, prefixed digits, destination-pattern and port are all easily viewed from this output. To verify the operation of the gateway, place test calls from the PSTN phone to BR1 Ph1 (ex 13123012001) (Test with the csim start 13123032001 command on the PSTN router) Verify the call is procedding correctly with the following commands:
VORack01R2#debug isdn q931 Jun 30 09:59:48.792: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0087 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Called Party Number i = 0xA1, '3123012001' Plan:ISDN, Type:National Jun 30 09:59:48.792: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x8087 callID = 0x000D switch = primary-ni interface = User Jun 30 09:59:48.832: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8087 Channel ID i = 0xA98381 Exclusive, Channel 1 VORack01R2# Jun 30 09:59:49.416: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x8087 VORack01R2# Jun 30 10:00:09.428: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x0087 Cause i = 0x8290 - Normal call clearing Jun 30 10:00:09.432: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x8087 Jun 30 10:00:09.440: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0087
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The output of the ISDN q931 debug shows that the call was completed successfully. The Alerting stage indicates that the call is ringing, and the Normal Call Clearing indicates that the call completed without an error. The next debug command used is the debug dial-peer voice summary. Note that this command has large amounts of output, which is mostly ommited below. The statement to look for in the output is Matched. In this debug, both dial-peers are matched, with the dial-peer pointing to the subscriber (3120) matched first.
VORack01R2#debug voice dialpeer all voip dialpeer all debugging is on Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=3123012001 Result=SUCCESS(0) <snip> List of Matched Outgoing Dial-peer(s): 1: Dial-peer Tag=3120 2: Dial-peer Tag=3121
Troubleshooting The most common issue that will lead to call failure on an H.323 gateway is an interface in a shut down state, or an incorrectly configured dial-peer. Using the debugs shown above you will be able to more easily diagnose the problem. If no dial-peer match is found, then check your dial-peer configuration. If the call fails, and following output is displayed using a debug isdn q931, check the interface status and port status:
VORack01R2#debug isdn q931 debug isdn q931 is ON. Jun 30 09:58:05.456: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref <snip> Exclusive, Channel 1 Called Party Number i = 0xA1, '3123012001' Plan:ISDN, Type:National <snip> Jun 30 09:58:05.460: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8086 Cause i = 0x80AF - Resource unavailable, unspecified VORack01R2#show run | section voice-port voice-port 0/0/0:23 shutdown
The voice-port is in a shutdown state. Once the port is brought back up, using the no shut command, test the call again, and the call will proceed correctly.
Further Reading:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_ example09186a0080094852.shtml Copyright 2009 Internetwork Expert 97 INE.com
2.22 SIP Trunk

Configure the CUCM to use R2 as a SIP Gateway Configure R2 for outgoing PSTN and incoming call routing Use only the first 3 channels of the T1 Incoming calls should first be routed to the Subscriber, then the Publisher Use the most reliable interface of R2 to source SIP signaling messages and media traffic Ensure SIP timers are set for minimal delay between call agents
Configuration **Note: Remove the H.323 gateway configuration on R2 and the CUCM system from the previous task before proceeding with SIP configuration. The dial-peers can be left in place, but modified to remove the H.323 class.
Step 1:
R2: ! ! Configure the switchtype and clocking ! isdn switchtype primary-ni network-clock-participate wic 0 ! ! Configure the T1 ! controller t1 0/0/0 framing esf linecode b8zs pri-group timeslots 1-3 ! ! Enable IE delivery for the T1 ! interface serial 0/0/0:23 isdn outgoing display-ie ! ! Bind signaling and media to the most reliable interface ! voice service voip sip bind all source-interface loopback 0 ! ! Set the SIP INVITE timer to the lowest possible setting ! sip-ua timer trying 100 !
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! ! Configure the incoming dial-peer ! dial-peer voice 1 pots incoming called-number . direct-inward-dial ! ! Configure two dial-peers. One to the SUB and one to the PUB ! Add the session protocol statement ! dial-peer voice 3120 voip destination-pattern 312301....$ session target ipv4:177.1.10.20 session protocol sipv2 ! dial-peer voice 3121 voip destination-pattern 312301....$ session target ipv4:177.1.10.10 session protocol sipv2 ! ! Configure the outgoing dial-peers ! dial-peer voice 911 pots description == 911 == destination-pattern 911 port 0/0/0:23 forward-digits 3 ! dial-peer voice 9911 pots description == 9911 == destination-pattern 9911 port 0/0/0:23 forward-digits 3 ! dial-peer voice 900 pots description == Local Calls == destination-pattern 9[2-9]......$ port 0/0/0:23 ! dial-peer voice 901 pots description == Long Distance == destination-pattern 91[2-9]..[2-9]......$ port 0/0/0:23 prefix 1 ! dial-peer voice 902 pots description == International == destination-pattern 9011T port 0/0/0:23 prefix 011 !
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On the CUCM Publisher server, navigate to Device > Trunk. Click the Add New button, and follow through the steps to configure the gateway:
Set the Device Name and Device Pool, leave all other settings to their defaults.
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Set the Significant Digits filed to 4. Enable Inbound/Outbound Header delivery.
Set the destination address to the loopback IP of R2. Select the following SIP settings, then click Save, and Reset.
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SIP is an open standards, text based call control protocol. SIP implementation is common for open source call solutions, but is often used in Cisco CUCM deployments for site to site communication with the use of SIP Trunks. The configuration is similar to an H.323 gateway in IOS, however the configuration in CUCM is for a trunk, rather than a gateway. One particular advantage in using SIP is the clear text messages, which are easily readable and can be used for efficient verification and troubleshooting of calls. The message types and explanations are expained in more detail in the verification section. SIP messages use Status Codes that are similar to those used for HTTP. The following table shows each code, the Class Of Response and an explanation: Class Of Response Status Code 100 180 Informational/Provisional 181 182 183 200 Success 202 300 301 Redirection 302 305 380 Client-Error 400 401 402 403 404 405 406 407 408 409 410 411 413 414 415 Copyright 2009 Internetwork Expert 102 Explanation Trying Ringing Call Being Forwarded Call Queued Session Progress OK Accepted Multiple Choices Moved Permanently Moved Temporarily Use Proxy Alternative Service Bad Request Unauthorized Payment Required Forbidden Not Found Method Not Allowed Not Acceptable Proxy Authentication Required Request Timeout Conflict Gone Length Required Request Entity Too Large Request URI Too Long Unsupported Media Type INE.com
Communications Manager (CUCM) Unsupported URI Scheme Bad Extension Explanation Extension Required Interval Too Brief Temporarily Unavailable Call/Transaction Does Not Exist Loop Detected Too Many Hops Address Incomplete Ambiguous Busy Here Request Terminated Not Acceptable Here Request Pending Undecipherable Server Internal Error Not Implemented Bad Gateway Service Unavailable Server Time Out Version Not Supported Message Too Large Busy Everywhere Declined Does Not Exist Anywhere Not Acceptable
416 Class Of Response 420 Status Code 421 423 480 481 Client-Error 482 483 484 485 486 487 488 491 493 500 501 502 503 504 505 513 600 603 604 605 Verification
Server-Error
Global Failure
Use the same verification procedure from the H.323 gateway lab to check the voice interface, dial-peer and perform a test call. SIP specific verification is shown below: First confirm that the SIP service is running, and the trying timer is set properly:
VORack01R2#show sip service SIP Service is up VORack01R2#show sip-ua timers SIP UA Timer Values (millisecs unless noted) trying 100, expires 180000, connect 500, disconnect 500
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prack 500, rel1xx 500, notify 500, update 500 refer 500, register 500, info 500, options 500, hold 2880 minutes tcp/udp aging 5 minutes tls aging 60 minutes
The output from the debug ccsip messages command is shown below. This debug displays the SIP call flow and cleartext messages used by SIP for call control. The first message sent is the INVITE, which initiates the call. The calling party or UA (User Agent) address is shown in a format similar to an email address of phone-number@domain:port. Next, the From field indicates the origination point of the message which is R2s loopback address (which was configured as the source of signaling messages). The To field then shows the next-hop destination of the phone located at the Subscriber CUCM. The next three highlighted messages show that the message is using the standard Session Description Protocol, the message is being transported VIA R2, and the call sequence number is 101.
VORack01R2#debug ccsip messages Jul 1 10:23:11.494: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:3123012001@177.1.10.20:5060 SIP/2.0 Date: Wed, 01 Jul 2009 10:23:11 GMT Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER From: <sip:177.1.254.2>;tag=514019B4-2397 Allow-Events: telephone-event Supported: 100rel,timer,resource-priority,replaces,sdp-anat <snip> Content-Length: 235 User-Agent: Cisco-SIPGateway/IOS-12.x To: <sip:3123012001@177.1.10.20> Contact: <sip:177.1.254.2:5060> Expires: 180 Content-Disposition: session;handling=required Content-Type: application/sdp Call-ID: 3559723-656011DE-802A8042-7BB8DE91@177.1.254.2 Via: SIP/2.0/UDP 177.1.254.2:5060;branch=z9hG4bK421D8 CSeq: 101 INVITE Max-Forwards: 70
The next section in the debug displays the SDP message contents, which are used for endpoint capabilities exchange, and call negotiation features.
v=0 o=CiscoSystemsSIP-GW-UserAgent 5148 5832 IN IP4 177.1.254.2 s=SIP Call c=IN IP4 177.1.254.2 t=0 0 m=audio 19076 RTP/AVP 18 19 c=IN IP4 177.1.254.2 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:19 CN/8000 a=ptime:20
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The next message denotes the call is Trying (Status Code 100). This indicates that the gateway has processed the call.
Received: SIP/2.0 100 Trying <snip>
The call is now Ringing (Status Code 180)

Jul 1 11:29:33.501: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Date: Wed, 01 Jul 2009 18:29:08 GMT <snip>
Next, an OK response (Status Code 200) indicating the call has been answered.
Jul 1 11:29:47.621: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Date: Wed, 01 Jul 2009 18:29:08 GMT <snip>
The call is now Active, and the relevant Call Setup Information is shown. In this example, the test was performed using the csim start command from the PSTN, therefore the Calling Number is not included. The origination (R2) and destination (SUB) addresses for the Signaling traffic are included in the message:
The Call Setup Information Call Control Block (CCB) : State of The Call : TCP Sockets Used : Calling Number : Called Number : Source IP Address (Sig ): Destn SIP Req Addr:Port : Destn SIP Resp Addr:Port : Destination Name : is: 0x4A0012A0 STATE_ACTIVE NO 3123012001 177.1.254.2 177.1.10.20:5060 177.1.10.20:5060 177.1.10.20
The Media traffic information is now displayed. Notice the Destination IP is different from the Signaling traffic. The actual media destination is the phone (10.0.11.27 in this example)
Number of Media Streams: 1 Media Stream : Negotiated Codec : Negotiated Codec Bytes : Nego. Codec payload : Negotiated Dtmf-relay : Dtmf-relay Payload : Source IP Address (Media): 1 g729r8 20 18 (tx), 18 (rx) 0 0 (tx), 0 (rx) 177.1.254.2
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Source IP Port (Media): Destn IP Address (Media): Destn IP Port (Media): Orig Destn IP Address:Port 19076 10.0.11.27 5004 (Media): [ - ]:0
An ACK message is the final response to the INVITE showing success

ACK sip:3123012001@177.1.10.20:5060 SIP/2.0 <snip>
The call is now hung up, showing a BYE message, and call state of DEAD
Received: BYE sip:177.1.254.2:5060 SIP/2.0 <snip> The Call Setup Information is: Call Control Block (CCB) : 0x4A0012A0 State of The Call : STATE_DEAD <snip>
The Status Code of 200 is displayed indicating OK. This is similar to Normal Call Clearing in an isdn Q.931 status.
Disconnect Cause (SIP) : 200
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmcfg/b0 6trunk.html#wp1127633 - CUCM Trunk Configuration http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cgbasic_cfg_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp12649 66 - IOS SIP Configuration http://docwiki.cisco.com/wiki/Cisco_IOS_Voice_Troubleshooting_and_Monitoring _--_Cisco_SIP_Gateway_Troubleshooting - Troubleshooting IOS SIP Gateway
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2.23 Media Resources Conference Bridge

Configure R1s DSP Resources as a Conference Bridge Add the Conference Bridge to the CUCM server SCCP traffic should be sourced from the Voice Vlan Ensure only one conference session is allowed at any given time Ensure SCCP packets are marked as IP Precedence 3
R1: voice-card 0 no dspfarm dsp service dspfarm ! sccp local Vlan 11 sccp ccm 177.1.10.10 identifier 1 sccp ccm 177.1.10.20 identifier 2 sccp ip precedence 3 sccp ! sccp ccm group 1 associate ccm 1 priority 2 associate ccm 2 priority 1 associate profile 1 register R1_CONF ! dspfarm profile 1 conference maximum-sessions 1 associate application SCCP no shut !
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Step 2: Navigate to Media Resource > Conference Bridge, then click Add a New Conference Bridge link. Set the Conference Bridge Type Cisco IOS Enhanced Conference Bridge, Name R1_CONF, Device Pool DP_HQ, the Location field value to HUB_NONE, and Security Mode to Non Secure:
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/b0 4cnbrg.html
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2.24 Media Resources Transcoder

Configure R2s DSP Resources as a Transcoder Add the Transcoder to the CUCM server as R2_XCODER SCCP traffic should be sourced from the Voice Vlan Ensure only 3 sessions are allowed at any given time Ensure SCCP packets are marked as IP Precedence 3
R2: voice-card 0 no dspfarm dsp service dspfarm ! sccp local Vlan 11 sccp ccm 177.1.10.10 identifier 1 version 7 sccp ccm 177.1.10.20 identifier 2 version 7 sccp ip precedence 3 sccp ! sccp ccm group 1 associate ccm 1 priority 2 associate ccm 2 priority 1 associate profile 1 register R2_XCODER ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729r8 codec g729ar8 maximum sessions 3 associate application SCCP no shut !
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Step 2: Navigate to Media Resource > Transcoder. Click the Add a New Transcoder link. Set the Transcoder Type to Cisco IOS Enhanced Media Termination Point, set the Device Name to R2_XCODER and the Device Pool field to DP_BR1
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmcfg/b0 4trans.html
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2.25 Music On Hold

Configure the Publisher and Subscriber to stream Music on Hold Music should be sent using the G.711 format inside the HQ region and using G.729 when streaming to BR1 devices Enable Multicast media delivery to the HQ and Br1 sites
Configuration Step 1: Add a new device pool named DP_MOH: copy it from DP_HQ. Ensure the Region field is set to REG_HQ for this device pool. Step 2: Enable the use of the G.729 codec for Music on Hold. Navigate to System > Service Parameters and select 177.1.10.10 as the Server and IP Voice Media Streaming Application as the Service. Set Supported MoH Codecs to G711mulaw and G729AnnexA (**Note that you must use Control + Click to select both Codecs)
Step 3: Navigate to Media Resource > Music on Hold Server and then click the Find button. For each of the MoH Servers listed, set the Device Pool to DP_MOH and Location to HUB_NONE. Check the Enable Multicast Audio Sources on this MOH Server field. Set the Base Multicast IP Address to 239.1.1.1 and port to 16384. Choose Increment Multicast on IP Address, and then click the Update button:
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Repeat the same operations for the Subscriber MOH server, changing the IP to 239.1.1.2. Step 4: In the upper right hand navigation section of the configuration page, click the "Configure Audio Sources" link.
Next, click MOH Audio Source 1. Check the Allow Multicasting field, and then click the Save button:
Next, click the Related Link for the Music On Hold Server Configuration. Set the Maximum Hops to 4 so that the MOH multicast stream will reach Br1:
Step 5: Enable multicast routing at the HQ and Br1 sites:

R1: ip multicast-routing
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! interface FastEthernet 0/0.11 ip pim dense-mode ! interface FastEthernet 0/0.10 ip pim dense-mode ! interface Serial 0/0/1:0.1 ip pim dense-mode
R2: ip multicast-routing ! interface Vlan 11 ip pim dense-mode ! interface Serial 0/0/1:0.1 ip pim dense-mode ! interface Loopback0 ip pim dense-mode
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fs moh.html#wpxref59089
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2.26 CUCM Applications - IPMA

Configure the IPMA service in proxy mode to intercept calls to HQ Phone 1s primary line Use the following table as reference for user names and extension numbers to complete this task Phone HQ IP Ph1 HQ IP Ph2 Extension 1001 1002 1011 Comment IPMA Manager Line Assistant Primary Line Assistant Proxy Line
HQ Phone 1 is assigned to a manager and HQ Phone 2 to the managers assistant
Configuration Step 1: Navigate to Call Routing > Intercom > Intercom Route Partition. Add a new partition named PT_INTERCOM. Click Save. Step 2: Navigate to Device > Device Settings > Phone Button Template. Select the template corresponding to the phone model such as 7960, 61, 62, 65, etc... Click the Copy button. Name the new template "IMPA Template 3+3". Set the second button to Intercom and the third button to be a line appearance, instead of a Speed Dial:
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Step 3: Navigate to Device > Phone section and locate HQ IP Phone 1 using the Find function. Change the Softkey Template field to Standard Manager, and the "Button Template" to IMPA Template 3+3:
Step 4: Choose Line 1 of the phone and change the Partition field to PT_MANAGER then click the Save button. Next, Choose Line 2 and set the Intercom DN number to *1001, choose the partition created for the Intercom line, as well as the CSS. Finally, set the speed dial to *1002 which will point to the assistants phone:
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Step 4: Under HQ IP Ph1 device settings click the Subscribe/Unsubscribe Services link in the related links navigation menu. Under Select a Service menu choose IPMA, click the Continue button then the Subscribe button, and Save. Step 5: Navigate to Device > Phone and locate HQ IP Phone 2 using the Find button. Under the device settings ensure the Phone Button Template is set to "IMPA Template 3+3. Change the Softkey Template to Standard IPMA Assistant. Step 6: Choose Line 2 and set the Intercom DN number to *1002, choose the partition created for the Intercom line, as well as the CSS. Finally, set the speed dial to *1001 which will point to the managers phone. On the phone settings page, choose the Line 2 Add a DN link, set the Directory Number to 1011, and the Calling Search Space to CSS_MGR. Change Display (Internal Called ID) field to IPMA Proxy Line then click the Add button.
Perform the following steps on the Publisher CallManager to create new users: Step 1: Navigate to User Management > End User. Click Add New, and set up the following user: hqmanager as the First Name,Last Name and User ID. Set the User Password to cisco and PIN to 12345. Check the Allow Control Of Device From CTI checkbox then click the Save button. Copyright 2009 Internetwork Expert 116 INE.com
Step 2: Click the Device Association link on the User page. Locate HQ IP Phone 1, check the device on the page and click the Save button to associate the device with the user. Step 3: Create a new user for the assistant phone. Use hqast as the First Name, Last Name and User ID. Set the User Password to cisco and the PIN field to 12345. Check the Allow Control Of Device From CTI checkbox then click the Save button. Step 4: Click the Device Association link on the User page. Locate HQ IP Phone 2, check the device on the page and click the Save button to associate device with the user.
Perform the following steps to configure the IPMA application user settings: Step 1: Locate the hqmanager user. Under the Related Links menu select Manager Confguration. Uncheck the Automatic Configuration box. Set Intercom Line value to *1001 and move line 1001 from Available Lines to Selected Lines. Set the Assistant, and click save:
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Step 2: Click on the hqast user on the hqmgr page and click View Details. Uncheck the Automatic Configuration box, and configure the assistant user with the relevant line settings configured earlier:
Step 3: To install the IPMA applicaton, navigate to Application > Plugins. Click the download icon next to the Cisco Unified CallManager Assistant Console. Once the application has downloaded, complete the installation process. After the application has installed, configure it for the IPMA server IP address (177.1.10.10), and hqast username. Authenticate using the password value you configured previously. Once the application has launched and autheticated, you can test the IPMA configuration, by placing a call from another phone to 1001, HQ Ph2 should ring.
Further Reading:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsi pma.html
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CCIE Voice Lab Workbook Volume I Version 3.