Cipt150sg Vol.1

CIPT1
Implementing Cisco Unified CallManager Part 1

Volume 1
Version 5.0
Student Guide
Text Part Number: 97-2428-01
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Students, this letter describes important course evaluation access information!
Welcome to Cisco Systems Learning. Through the Cisco Learning Partner Program, Cisco Systems is committed to bringing you the highest-quality training in the industry. Cisco learning products are designed to advance your professional goals and give you the expertise you need to build and maintain strategic networks. Cisco relies on customer feedback to guide business decisions; therefore, your valuable input will help shape future Cisco course curricula, products, and training offerings. We would appreciate a few minutes of your time to complete a brief Cisco online course evaluation of your instructor and the course materials in this student kit. On the final day of class, your instructor will provide you with a URL directing you to a short post-course evaluation. If there is no Internet access in the classroom, please complete the evaluation within the next 48 hours or as soon as you can access the web. On behalf of Cisco, thank you for choosing Cisco Learning Partners for your Internet technology training. Sincerely, Cisco Systems Learning
Table of Contents
Volume 1 Course Introduction
Overview Learner Skills and Knowledge Course Goal and Objectives Course Flow Additional References Cisco Glossary of Terms Your Training Curriculum
1
1 1 3 4 5 5 6
Get Started with Cisco Unified CallManager

Overview Module Objectives
1-1
1-1 1-1
Introducing Cisco Unified CallManager Release 5.0

Overview Objectives Cisco Unified Communications Solution Cisco Unified Communications Network Cisco Unified Communications Components Cisco Unified CallManager Functions Example: Basic IP Telephony Call Cisco Unified CallManager Release 5.0 as an Appliance Platform Access and Support Database Operation Database Clustering Cisco Unified CallManager Clustering Cluster Communication Dynamic Firewall (iptables) Cisco Unified CallManager Operating System, Database, and Supporting Applications Cisco Unified CallManager Servers Licensing Model Licensing Compliance Device License Units License File Request Process License File Specifics Example License File Obtaining Additional Licenses Migrating from Windows to Appliance Licensing Functional Diagram Calculating License Units Generating a License Unit Report Uploading a License File Summary References
1-3
1-3 1-3 1-4 1-5 1-6 1-7 1-8 1-9 1-11 1-12 1-12 1-13 1-14 1-15 1-16 1-17 1-18 1-19 1-20 1-21 1-22 1-23 1-24 1-25 1-26 1-28 1-29 1-30 1-34 1-35
Evaluating Cisco Unified CallManager Release 5.0 Deployment Options

Overview Objectives Call-Processing Deployment Models Single-Site Deployment Single-Site Deployment: Design Guidelines Multisite WAN with Centralized Call Processing Multisite WAN with Centralized Call Processing: Design Guidelines Multisite WAN with Distributed Call Processing Multisite Distributed Call Processing: Design Guidelines Clustering over the IP WAN Clustering over the IP WAN: Design Guidelines
1-37
1-37 1-37 1-38 1-39 1-40 1-42 1-44 1-45 1-47 1-48 1-49
Cisco Unified CallManager Release 5.0 Database Redundancy 2:1 Redundancy Design Summary
1-50 1-51 1-52
Installing Cisco Unified CallManager Release 5.0

Overview Objectives Installation Framework Installation Methods Installation Options System Requirements NTP Strongly Recommended Hardware Requirements Reasons for Hard Drive Space Expansion Dual Partitions Enable Version Switching Preinstallation Tasks Important Notes Configuration Data Navigating Within the Installation Wizard Hardware Configuration Performing a Basic Installation: First Node Basic Installation (Platform Configuration) Basic Installation (Product Installation) Basic Installation (Timezone) Basic Installation (Auto Negotiation) Basic Installation (DHCP) Basic Installation (DNS) Basic Installation (Account Login) Basic Installation (First Node) Basic Installation (NTP External Source) Basic Installation (NTP Internal Source) Basic Installation (Database Access Security) Basic Installation (SMTP Host) Basic Installation (Platform Configuration Confirmation) Basic Installation (Application User) Basic Installation (Installation Complete) Logon Window for Platform Administration Interface Access and Log On Performing a Basic Installation: Subsequent Nodes Basic Installation (Access to First Node) Basic Installation (NTP) Performing an Upgrade During Installation Upgrade During Installation (Retrieval Mechanism) Upgrade During Installation (Remote Source - DHCP) Upgrade During Installation (Remote Patch Access Information) Upgrade During Installation (Remote Patch Selection) Upgrade During Installation (Local Source) Upgrade During Installation (Patch Location) Examining the Installation Log Files Perform Postinstallation Tasks Summary Module Summary Module Self-Check Module Self-Check Answer Key
1-53
1-53 1-53 1-54 1-55 1-56 1-57 1-58 1-59 1-60 1-61 1-62 1-63 1-64 1-67 1-68 1-69 1-70 1-71 1-72 1-73 1-74 1-75 1-76 1-77 1-78 1-79 1-80 1-81 1-82 1-83 1-84 1-85 1-86 1-87 1-88 1-89 1-90 1-91 1-92 1-93 1-94 1-95 1-96 1-97 1-98 1-99 1-100 1-101 1-105
ii
Implementing Cisco Unified CallManager Part I (CIPT1) v5.0
2006 Cisco Systems, Inc.
Administration of Cisco Unified CallManager Release 5.0

2-1
2-1 2-1
Administering the Cisco IP Telephony Platform

Overview Objectives Cisco IP Telephony Platform Administration Overview Accessing the GUI GUIMain Page Expanded Menus Reference Checking Platform Status Showing Hardware, Software, and System Status Viewing and Updating Platform Settings Changing IP Settings Changing NTP, SMTP, and Time Settings Switching Versions and Restarting the Server Restarting and Switching Software Versions Managing Security Certificates and IPsec Display Certificate Display Cisco Unified CallManager Certificate Upgrading Software or Adding Options Backing Out of an Upgrade Installing or Upgrading Software Options Uploading TFTP Files Checking Network Connectivity and Setting Up Remote Support Ping Utility Setting Up Remote Account Support Using the CLI CLI Access CLI Commands Command-Line Editing CLI Parameters and Options Sample show Commands Sample set Commands Sample file Commands Example CLI Commands Summary References
2-3
2-3 2-3 2-4 2-5 2-6 2-7 2-8 2-9 2-10 2-11 2-12 2-13 2-14 2-15 2-16 2-17 2-18 2-19 2-20 2-22 2-23 2-24 2-25 2-26 2-27 2-28 2-29 2-30 2-31 2-32 2-33 2-34 2-35 2-36
Performing General Administration

Overview Objectives Cisco Unified CallManager Administration Overview Accessing Cisco Unified CallManager Administration Cisco Unified CallManager Administration Main Page Expanded Menus Reference Cisco Unified CallManager Release 5.0 Applications Multilevel Administration Overview MLA Component Interaction Roles and User Groups Example Roles and User Groups Role Configuration Page User Groups User Group Configuration Page Creating a New Role and User Group Step 1: Add Custom Role Step 2a: Select Application Step 2b: Set Privileges Step 3: Add Users
2006 Cisco Systems, Inc. Implementing Cisco Unified CallManager Part I (CIPT1) v5.0
2-37
2-37 2-37 2-38 2-39 2-40 2-41 2-42 2-43 2-44 2-45 2-46 2-47 2-48 2-49 2-50 2-51 2-52 2-53 2-54
iii
Step 4: Create Custom User Groups Step 5: Add Users to User Groups Step 6: Assign Roles to User Groups Step 7: Verify Privileges Summary
2-55 2-56 2-57 2-59 2-62
Configuring Cisco Unified CallManager Release 5.0 Basic Settings

Overview Objectives Basic Settings Configuration Overview DNS vs. IP Addresses Service Activation Control Center Service Parameters Enterprise Parameters Summary
2-63
2-63 2-63 2-64 2-65 2-66 2-70 2-72 2-75 2-78
Implementing Disaster Recovery

Overview Objectives Disaster Recovery System Overview Cisco DRS Architecture Cisco DRS Components Features and Components Backed Up Cisco Disaster Recovery System GUI Access Cisco Disaster Recovery System CLI Access Performing a Backup Steps to Perform a Backup Step 1: Selecting Features to Back Up Step 2: Configuring a Storage Location Step 3a: Configuring a Scheduled Backup Step 3b: Starting a Manual Backup Step 4: Checking Backup Status or Canceling a Backup Performing a Restore Steps to Perform a Restore Step1: Choosing Storage Location Step 2: Choosing the Backup File Step 3: Choosing Features to be Restored Step 4: Choosing Nodes to be Restored Step 5: Checking Restore Status Executing CLI Commands List Tape Devices and Show Backup Files Backup Restore Status Summary References Module Summary Module Self-Check Module Self-Check Answer Key
2-79
2-79 2-79 2-80 2-81 2-82 2-83 2-84 2-85 2-86 2-87 2-88 2-89 2-90 2-91 2-92 2-93 2-94 2-95 2-96 2-97 2-98 2-99 2-100 2-101 2-102 2-103 2-104 2-105 2-105 2-106 2-107 2-111
Deployment of Cisco Unified CallManager Release 5.0 Endpoints

3-1
3-1 3-1
Configuring Cisco Catalyst Switches

Overview Objectives Cisco Catalyst Switch Role in IP Telephony Powering the Cisco IP Phone
3-3
3-3 3-3 3-4 3-5
iv
Types of PoE Delivery Cisco Prestandard Device Detection IEEE 802.3af Device Detection Cisco Catalyst Family of PoE Switches Configuring PoE Cisco Catalyst Switch: Show Inline Power Status Configuring Dual VLANs Configuring Voice VLANs Using the Cisco Catalyst Operating System Configuring Voice VLANs Using Native Cisco IOS Software Verifying Voice VLAN Configuration Using Cisco CatOS Verifying Voice VLAN Configuration Using Native Cisco IOS Software Configuring CoS Configuring CoS Summary References
3-6 3-8 3-9 3-10 3-12 3-13 3-14 3-16 3-17 3-18 3-19 3-20 3-21 3-22 3-23
Evaluating Cisco IP Telephony Endpoints

Overview Objectives Cisco IP Phone Overview Seamless, Native Support for SCCP and SIP SCCP Phones Midrange Cisco IP Phones Upper-End Cisco IP Phones Additional Cisco IP Telephony Endpoints SIP Endpoints SIP Endpoints: Basic SIP SIP Endpoints: Very Basic SIP SIP Endpoints: Advanced SIP SIP Phone Feature Issues SIP and SCCP Phone Configuration Comparison Cisco SIP Phone Startup Process Boot Sequence Differences Between Cisco SCCP and SIP Phones Third-Party SIP Phone Startup Process SIP Phones to Cisco Unified CallManager Interaction SIP and SCCP Configuration Similarities SIP and SCCP Configuration Differences Supported SIP Signaling Transports NTP Clock for SIP Phones DTMF Interworking: RFC 2833 and KPML Video Endpoints Cisco Video Endpoints Third-Party Video Endpoints Cisco IP Phone Codec Support Summary References
3-25
3-25 3-25 3-26 3-30 3-31 3-33 3-35 3-36 3-38 3-39 3-42 3-43 3-44 3-45 3-47 3-49 3-50 3-51 3-52 3-53 3-54 3-55 3-56 3-58 3-59 3-61 3-63 3-64 3-65
Configuring Cisco Unified CallManager to Support IP Phones

Overview Objectives Server Configuration Configuring Device Pools Example: Device Pool Configuration Cisco Unified CallManager Group Configuration Example: Cisco Unified CallManager Group Configuration Date/Time Group Configuration Region Configuration Example: Region Configuration Softkey Template Configuration
3-67
3-67 3-67 3-68 3-69 3-70 3-72 3-72 3-73 3-74 3-74 3-75
IP Phone Button Templates IP Phone Button Template Names Example: Naming a Phone Button Template Configuring IP Phone Autoregistration Device Defaults Manual IP Phone and Directory Number Configuration Directory Number Configuration DHCP Server Feature Steps to Configure DHCP Phone Support Step 1: Activate DHCP Monitor Service Step 2: Configure the DHCP server Step 3: Configure the DHCP Subnet DHCP Migration Considerations Summary References
3-76 3-77 3-77 3-78 3-80 3-81 3-83 3-84 3-85 3-86 3-87 3-89 3-90 3-91 3-92
Configuring SIP Endpoints

Overview Objectives Cisco Unified CallManager Release 5.0 SIP Phones Cisco Unified CallManager Release 5.0 SIP Features Cisco Unified CallManager Release 5.0 SIP Line-Side Support SIP Support In Cisco Unified CallManager Release 5.0 SIP Call Flow and Digit Collection SIP Call Flow: Called User Is Busy SIP Call Flow: Simple Call Hold Configuring Cisco SIP Phones Choosing the Protocol for Autoregistered Phones Choosing the Protocol for a Specific Phone SIP Profile SIP Profile Configuration Page SIP Phone Configuration Converting SCCP Phones to SIP Migrating Cisco IP Phone 7970 and 7971 from SCCP to SIP Adding Third-Party SIP Phones Digest Authentication for Phone Identification Third-Party SIP Phone Configuration Steps Third-Party Phone Configuration Adding SIP Dial Rules SIP Call Flow: Digit Collection Dial Rules and KPML Support in Cisco Unified CallManager 5.0 Dial Rules and KPML Interworking Dial Rules Configuration Overview SIP Phone Dial Rules Configuration Sequence Dial Rule Patterns and Phone Models Dial Rule Pattern Syntax Configuring SIP Dial Rules Add the SIP Dial Rules to the Phone Summary References
3-93
3-93 3-93 3-94 3-95 3-97 3-98 3-99 3-101 3-102 3-104 3-105 3-106 3-107 3-108 3-109 3-111 3-113 3-116 3-117 3-118 3-119 3-121 3-122 3-123 3-124 3-125 3-126 3-127 3-128 3-129 3-132 3-133 3-134
Managing Endpoints with Cisco Unified CallManager BAT

Overview Objectives Cisco Unified CallManager BAT Overview Cisco Unified CallManager BAT Components Cisco Unified CallManager BAT Support for Devices and Records Bulk Provisioning Service Cisco Unified CallManager BAT Configuration Process Configuring BAT Templates Uploading Data Input Files
vi Implementing Cisco Unified CallManager Part I (CIPT1) v5.0
3-135
3-135 3-135 3-136 3-139 3-140 3-141 3-143 3-144 3-147
Validating Data Input Files Verifying Validation Results Inserting IP Phones into Cisco Unified CallManager Database Verifying the Status of Insertion Creating CSV Files Scheduling Cisco Unified CallManager BAT Jobs Cisco TAPS Overview Autoregistration Options for TAPS Using TAPS TAPS Installation TAPS Activation Configuring Autoregistration Options for TAPS Summary References Module Summary Module Self-Check Module Self-Check Answer Key
3-148 3-149 3-150 3-152 3-153 3-158 3-161 3-162 3-164 3-165 3-166 3-167 3-169 3-170 3-171 3-173 3-178
vii
viii
CIPT1
Course Introduction
Overview
Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0 prepares you for installing, configuring, and maintaining a Cisco IP telephony solution. This course focuses primarily on Cisco Unified CallManager Release 5.0, the call routing and signaling component for the Cisco IP telephony solution. This course includes lab activities in which you will perform postinstallation tasks and configure Cisco Unified CallManager; configure gateways, gatekeepers, and switches; and build route plans to place intracluster and intercluster Cisco IP phone calls. You will also configure telephony class of service (calling restrictions) and numerous user telephone features, services, media resources, and applications.
Learner Skills and Knowledge

This subtopic lists the skills and knowledge that learners must possess to benefit fully from the course. The subtopic also includes recommended Cisco learning offerings that learners should first complete to benefit fully from this course.
Learner Skills and Knowledge

Working knowledge of fundamental terms and concepts of computer networking, to include LANs, WANs, and IP switching and routing. Ability to configure and operate Cisco routers and switches and to enable VLANs and DHCP. Fundamental knowledge of converged voice and data networks. Ability to configure voice interfaces on Cisco voice-enabled equipment for connection to traditional, nonpacketized telephony equipment, and to configure the call flows for POTS and VoIP dial peers.
2006 Cisco Systems, Inc. All rights reserved.
CIPT1 v5.03
Learner Skills and Knowledge (Cont.)

Cisco learning offerings: Interconnecting Cisco Network Devices (ICND) v2.3 Cisco CCNA certification recommended prerequisite Building Cisco Multilayer Switched Networks (BCMSN) v3.0 Cisco Voice over IP (CVOICE) v5.0
CIPT1 v5.04
Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0
Course Goal and Objectives

This topic describes the course goal and objectives.
Course Goal
To provide learners with the necessary knowledge and skills to install, configure, and maintain a Cisco IP Telephony solution based on Cisco Unified CallManager 5.0, the call routing and signaling component of the Cisco IP Telephony solution.
Implementing Cisco Unified CallManager Part 1
CIPT1 v5.05
Upon completing this course, you will be able to meet these objectives: Deploy a Cisco Unified CallManager Release 5.0 server in a cluster using a supported IP telephony deployment model Perform platform and general administration, and implement disaster recovery Deploy SCCP and SIP endpoints in a Cisco Unified CallManager Release 5.0 cluster Deploy a dial plan consisting of gateways, trunks, route plan elements, hunt groups, partitions, and calling search spaces Implement multiple site deployments with fallback telephony services in the remote branches, call admission control over the WAN bandwidth, and automated call rerouting through the PSTN when the WAN bandwidth is insufficient Configure Cisco Unified CallManager to enable end-user features and services
Course Introduction
Course Flow
This topic presents the suggested flow of the course materials.
Course Flow
Day 1
Course Introduction
Day 2
Day 3
Deployment of Cisco Unified CallManager Release 5.0 Endpoints Deployment of a Dial Plan on Cisco Unified CallManager Release 5.0
Day 4
Deployment of a Dial Plan on Cisco Unified CallManager Release 5.0 Implementation of Multiple Site Deployment
Day 5
Cisco Unified CallManager Features and Services
A Introduction to M Cisco Unified

CallManager Release 5.0
Lunch
Administration of Cisco Unified CallManager Release 5.0 Deployment of Cisco Unified CallManager Release 5.0 Endpoints Deployment of a Dial Plan on Cisco Unified CallManager Release 5.0 Implementation of Multiple Site Deployment Cisco Unified CallManager Features and Services
P M
Cisco Unified CallManager Features and Services
CIPT1 v5.06
The schedule reflects the recommended structure for this course. This structure allows enough time for the instructor to present the course information and for you to work through the lab activities. The exact timing of the subject materials and labs depends on the pace of your specific class.
Additional References
This topic presents the Cisco icons and symbols that are used in this course, as well as information on where to find additional technical references.
Cisco Icons and Symbols

Router Cisco Unified CallManager Camera PC/Video Network Cloud SRST-Enabled Router Phone
File Server
Voice Router
IP Phone Switch Router
PC
Switch
Gateway
Laptop
CIPT1 v5.07
Cisco Glossary of Terms

For additional information on Cisco terminology, refer to the Cisco Internetworking Terms and Acronyms glossary of terms at http://www.cisco.com/univercd/cc/td/doc/cisintwk/ita/index.htm.
Course Introduction
Your Training Curriculum

This topic presents the training curriculum for this course.
Cisco Career Certifications
CIPT1 v5.08
You are encouraged to join the Cisco Certification Community, a discussion forum open to anyone holding a valid Cisco Career Certification (such as Cisco CCIE, CCNA, CCDA, CCNP, CCDP, CCIP, CCVP, or CCSP). It provides a gathering place for Cisco certified professionals to share questions, suggestions, and information about Cisco Career Certification programs and other certification-related topics. For more information, visit www.cisco.com/go/certifications.
Cisco Career Certifications: CCVP

Expand Your Professional Options and Advance Your Career
Professional-level recognition in CCVP Expert
CCIE
Recommended Training Through Cisco Learning Partners
Quality of Service (QOS) Cisco Voice over IP (CVOICE) IP Telephony Troubleshooting (IPTT)
Professional
CCVP
Implementing Cisco Unified CallManager Part 1 (CIPT1) Implementing Cisco Unified CallManager Part 2 (CIPT2) Implementing Cisco Voice Gateways and Gatekeepers (GWGK)
Associate
CCNA
CCVP
www.cisco.com/go/certifications
CIPT1 v5.09
Course Introduction
Module 1
Get Started with Cisco Unified CallManager

Overview
Cisco Unified CallManager is the software-based, call-processing component of the Cisco Unified Communications solution. This module describes the servers that Cisco Unified CallManager supports, explores the available deployment models for using Cisco Unified CallManager in a Cisco Unified Communications solution, and reviews the Cisco Unified CallManager server installation process.
Module Objectives
Upon completing this module, you will be able to deploy a Cisco Unified CallManager Release 5.0 server in a cluster using a supported IP telephony deployment model. This ability includes being able to meet these objectives: Identify the functions of a Cisco Unified CallManager, and identify the hardware, software, and tasks required for a Cisco Unified CallManager installation Determine the optimum Cisco Unified CallManager cluster option and IP telephony deployment model for your enterprise Describe the Cisco Unified CallManager Release 5.0 installation framework and procedure
1-2
Lesson 1
Introducing Cisco Unified CallManager Release 5.0

Overview
A Cisco Unified Communications deployment relies on Cisco Unified CallManager for its callprocessing and call-routing functions. Understanding the role that Cisco Unified CallManager plays in a converged network from a system, software, and hardware perspective is necessary to successfully install and configure Cisco Unified CallManager. This lesson introduces the Cisco Unified Communications solution and describes Cisco Unified CallManager functions, hardware and software requirements, and the licensing model.
Objectives
Upon completing this lesson, you will be able to list the functions of Cisco Unified CallManager, and identify the hardware, software, and tasks required for a Cisco Unified CallManager installation and configuration. This ability includes being able to meet these objectives: Describe the overall Cisco Unified Communications strategy, systems architecture, and solution bundles Identify the primary Cisco Unified CallManager functions Describe the functions of the Cisco Unified CallManager Release 5.0 appliance model and implications for accessing the platform Describe the Cisco Unified CallManager Release 5.0 database architecture and topology Describe intracluster communication processes and protocols for the first node and subsequent nodes Identify the software that Cisco Unified CallManager Release 5.0 depends upon for its operating system, database, directory, and backup Identify the major features of each media convergence server platform on which Cisco Unified CallManager Release 5.0 is supported Explain the licensing model of Cisco Unified CallManager Release 5.0
Cisco Unified Communications Solution

This topic describes the layers and components of the Cisco Unified Communications solution.
Cisco Unified Communications Layers

Client Adaptive
Video Communications Client
Softphones
The Cisco Unified Communications solution provides the foundation for converged networks. The Cisco Unified Communications strategy encompasses voice, video, and data traffic within a single network infrastructure. Cisco Unified Communications equipment is capable of managing all three traffic types and interfacing with all standards-based network protocols in each network class. Cisco IP Communications represents a new way of delivering unified communications functionality to enterprise customers. Rather than delivering a collection of disjointed products with individual release dates, testing methodology, and documentation, Cisco Unified Communications is the first coordinated system release of an integrated set of products that are tested, documented, and supported as a system. The figure shows the four standard layers of the Cisco Unified Communications voice infrastructure model: the infrastructure layer, which lays the foundation for network components; the call-processing layer, which maintains PBX-like functions; the applications layer, which is where applications that provide additional network functionality reside; and the client layer, which is where end-user devices reside. The key points about the four standard layers are as follows: Infrastructure layer: The infrastructure consists of routers, switches, and voice gateways, and carries data between all network devices and applications. Call-processing layer: Call processing is physically independent from the infrastructure. Thus, a Cisco Unified CallManager in San Jose can process call control for a bearer channel in Chicago. Applications layer: Applications are physically independent from call-processing functions and the physical voice-processing infrastructure; that is, they may reside anywhere within the network.
1-4 Implementing Cisco Unified CallManager Part 1 (CIPT1) v5.0 2006 Cisco Systems, Inc.
Distributed Manageable
Phone
Applications
Voice Mail Presence Contact Center
Call Processing Open

Cisco Unified CallManager Directory
Infrastructure
Gateway Router Switch
CIPT1 v5.01-2
Client layer: The client layer brings applications to the user, whether the end device is a Cisco IP phone, a PC using a software-based phone, or a communications client or video terminal.
Cisco Unified Communications Network

This slide illustrates a network that uses the Cisco Unified Communications solution.
Cisco Unified Communications Network
Rich-media Conferencing Contact Center Voice Mail Presence Server 3rd Party Server
PSTN
IP WAN
SRST
CIPT1 v5.01-3
The infrastructure is built with switches and routers. Call processing is performed by Cisco Unified CallManager (appliance-based) or Cisco Unified CallManager Express (IOS-based). Applications are provided through separate servers such as voice mail, contact center, richmedia conferencing, Cisco Unified Presence Server, or third-party servers. Cisco IP phones, Cisco IP SoftPhones, video stations, and analog devices are shown as clients.
Get Started with Cisco Unified CallManager Release 5.0
1-5
Cisco Unified Communications Components

The goal of Cisco Unified Communications is to provide a complete IP communications solution for the customer. This solution is built from several components.
Cisco Unified Communications Components
IP
Cisco Unified Presence Server Cisco Emergency Responder Cisco Unified Contact Center Cisco Unified MeetingPlace Cisco Unified CallManager Third-Party Applications Cisco Unified Contact Center Express Desktop Applications Cisco IP Phones Cisco Unity Video
Cisco Unified Communications Solutions

2006 Cisco Systems, Inc. All rights reserved. CIPT1 v5.01-4
Cisco Unified Communications provides multiprotocol support for Skinny Client Control Protocol (SCCP), H.323, Media Gateway Control Protocol (MGCP), and Session Initiation Protocol (SIP). The Cisco Unified Communications solution covers all communications needs of an enterprise company. Voice mail or unified communications is provided through Cisco Unity (based on Microsoft Windows servers) or Cisco Unity Express (IOS-based). You can build contact centers of various sizes with Cisco Unified Contact Center and Cisco Unified Contact Center Express. A broad range of Cisco IP phones is available to cover various needs. Standard protocol interfaces are available to support third-party applications. For emergency calling, the Cisco Emergency Responder is available. Cisco Unified MeetingPlace is a large-scale conferencing server that also supports video integration. It is a full-featured product that integrates lecture-style conferences and full collaboration. For small to medium enterprises, you should use Cisco Unified MeetingPlace Express. It is the successor of Cisco Conference Connection. Cisco Emergency Responder enhances the existing emergency functionality offered by Cisco Unified CallManager. It ensures that Cisco Unified CallManager will send emergency calls to the public safety answering point (PSAP) that is appropriate for the location of the caller, and that the PSAP can identify the caller location and return the call if necessary. In addition, the system automatically tracks and updates equipment moves and changes. Deploying this capability helps ensure more effective compliance with legal or regulatory obligations, thereby reducing the risk of liability related to emergency calls. The Cisco Unified Presence Server collects information about the availability status and communications capabilities of a user, including whether or not the user is using a communications device, such as a phone, and whether the user has web collaboration or videoconferencing enabled on their system.
Cisco Unified CallManager Functions

This topic describes the functions that are provided by Cisco Unified CallManager.
Cisco Unified CallManager Functions

Call processing Signaling and device control Dial plan administration Phone feature administration Directory services Programming interface to external applications
CIPT1 v5.01-6
Cisco Unified CallManager extends enterprise telephony features and functions to packet telephony network devices. These packet telephony network devices include Cisco IP phones, media-processing devices, VoIP gateways, and multimedia applications. Additional data, voice, and video services, such as converged messaging, multimedia conferencing, collaborative contact centers, and interactive multimedia response systems, interact with the IP telephony solution through the Cisco Unified CallManager application programming interface (API). Cisco Unified CallManager provides these functions: Call processing: Call processing refers to the complete process of routing, originating, and terminating calls, including any billing and statistical collection processes. Signaling and device control: Cisco Unified CallManager sets up all of the signaling connections between call endpoints and directs devices such as phones, gateways, and conference bridges to establish and tear down streaming connections. Dial plan administration: The dial plan is a set of configurable lists that Cisco Unified CallManager uses to determine call routing. Cisco Unified CallManager provides the ability to create scalable dial plans for users. Phone feature administration: Cisco Unified CallManager extends services such as hold, transfer, forward, conference, speed dial, last-number redial, call park, and other features to IP phones and gateways.
1-7
Directory services: Cisco Unified CallManager uses its own database to store user information. You can authenticate users either locally or against an external directory. You can provision users by directory synchronization. With directory synchronization, you can automatically add users from the directory to the local database. Microsoft Active Directory (2000 and 2003), Netscape 4.x, iPlanet 5.1, and Sun ONE 5.2 are supported. Programming interface to external applications: Cisco Unified CallManager provides a programming interface to external applications such as Cisco IP SoftPhone, Cisco IP Communicator, Cisco Unified IP interactive voice response (IVR), Cisco Personal Assistant, and Cisco Unified CallManager Attendant Console.
Cisco Unified CallManager Functions (Cont.)

Cisco Unified CallManager
Signaling Protocol
Signaling Protocol
IP phone A
Real-Time Transport Protocol (RTP) Media Path
Cisco Unified CallManager performs call setup and maintenance tasks using a Signaling Protocol (Skinny/SIP). IP phones stream audio using RTP.
IP phone B
CIPT1 v5.01-7
Cisco Unified CallManager uses the SIP or the SCCP (Skinny) to communicate with Cisco IP phones for call setup and maintenance tasks. When the call is set up, Cisco IP phones communicate directly using Real-Time Transport Protocol (RTP) to carry the audio. You can better understand how Cisco Unified CallManager performs key functions by tracking a basic IP telephony call.
Example: Basic IP Telephony Call

In the figure, Party A (left telephone) wants to call Party B (right telephone). Party A picks up the handset and dials the number of Party B. In this environment, dialed digits are sent to Cisco Unified CallManager, the call-processing engine. Cisco Unified CallManager finds the address and determines where to route the call. Using the Skinny protocol, Cisco Unified CallManager signals the calling party over IP to initiate a ringback, and Party A hears ringing. Cisco Unified CallManager also signals the destination phone to initiate ringing. When Party B picks up the telephone, the RTP media path opens between the two stations. Party A or Party B may now initiate a conversation. The Cisco IP phones require no further communication with Cisco Unified CallManager until either Party A or Party B invokes a feature, such as call transfer, call conferencing, or call termination.
Cisco Unified CallManager Release 5.0 as an Appliance

This topic describes the new Cisco Unified CallManager appliance model that was introduced with Cisco Unified CallManager Release 5.0.
Cisco Unified CallManager as an Appliance

Complete hardware and software solution Cisco Unified CallManager servers get preinstalled with all software Supported on MCS servers Alternative operating system Improves installation and upgrade Increases security and reliability Interfaces provide access to system Administration via CLI and GUI Third-party access is through documented APIs Operates in a headless manner or in a headed manner
CIPT1 v5.01-8
Cisco Unified CallManager Release 5.0 works as an appliance on a non-Windows-based operating system. The appliance is essentially a closed system that only supports Cisco authorized applications and utilities. Key goals of the appliance model are to simplify the installation and upgrade of the box and to make transparent the underlying operating system and tools. An appliance-based model makes it possible for an administrator to install, implement, and manage a Cisco Unified CallManager deployment without having extensive knowledge or access to the underlying operating system. The Cisco Unified CallManager appliance has these functions: Complete hardware and software solution Cisco Unified CallManager servers are preinstalled with all software that is required to operate, maintain, secure, and manage a server or cluster of servers (including Cisco Security Agent) Supported on media convergence servers (MCSs).
Alternative operating system improves installation and upgrade and increases security and reliability You can upgrade Cisco Unified CallManager servers while they continue to process calls
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Interfaces provide access to system via either command-line interface (CLI) and GUI for administration purposes or through documented APIs for third-party access Outputs a variety of management parameters via a published interface to provide information to approved management applications such as, but not limited to, NetIQ Vivinet Manager, HP OpenView, and Integrated Research PROGNOSIS
Operates in a headless manner (without keyboard, mouse, or video graphics array [VGA] monitor support) or, in the case of some of the hardware platforms, in a headed manner (with keyboard, mouse, and monitor)
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Platform Access and Support

You must consider several points when you access the system using an appliance model.
Platform Access and Support

No root access Most debug interfaces are not directly accessible More Perfmon counters and alarms are added Customer can activate remote account support for specific time for Cisco TAC access Console application tools (show tech) have been added for when web is not available
Pre-5.0
5.0
Limited CLI access
CIPT1 v5.01-9
Root access to the file system is not permitted, but Cisco Unified CallManager Release 5.0 is designed with additional serviceability tools and performance counters so that root access should not be necessary except under extreme scenarios. In such cases, customers can optionally request a one-time encrypted administrative (root) password so that Cisco Technical Assistance Center (TAC) can access the system for a limited time to assist in problem diagnosis and resolution. Additionally, console application tools and other CLI commands are available when web access is not available. These are some details of the appliance model: Most debug interfaces not directly accessible No direct access to the file system or service control manager Normal serviceability methods are available for accessing trace files 4.X Perfmon counters and alarms were by Microsoft Structured Query Language (MS SQL) Server 5.X Perfmon counters and alarms flow through dbmon for central point of access
More Perfmon counters and alarms
Customer will be able to request encrypted administrative (root) password for TAC escalation and development engineers Console application tools (for example, the show tech command) have been added for times when web access to the GUI is not available
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Database Operation
This topic describes database operation in Cisco Unified CallManager Release 5.0.
Database Operation
Cisco Unified CallManager Cluster
First node
Informix Dynamic Server (IDS) Replication
IDS
IDS
CTI Manager
IDS IDS
MoH Server ICCS
IDS
IDS
TFTP Server
IDS IDS
SW Conf.
IDS
Call Processing Servers

Subsequent nodes
CIPT1 v5.01-10
Informix Dynamic Server (IDS)
The Cisco Unified CallManager service provides call routing, signaling, and media control for an IP telephony enterprise deployment. A cluster is a set of networked services that work together to provide the Cisco Unified CallManager service in addition to dedicated servers providing database, application, TFTP, and media services such as conferencing and music on hold. Clustering provides several benefits. It allows the network to scale to many thousands of endpoints, provides redundancy in case of network or server failures, and provides a central point of administration. To serve calls for devices, the Cisco Unified CallManager needs to retrieve settings for that device. These settings are stored in a database. The database is the repository for information such as service parameters, features, and the route plan.
Database Clustering
The first node in a cluster provides read and write access to its database, and subsequent nodes provide access to other databases in the cluster. In large installations, it is recommended that the first node resides on a separate server to prevent database interactions from impacting call processing. The subsequent node database servers maintain read-only copies of the database. You can have only one first node server and up to eight subsequent node servers per cluster. When you make database changes to the Cisco Unified CallManager configuration, the changes are written directly to the first node server. The first node maintains a TCP connection to the subsequent node, and when database changes occur, the first node replicates these changes (except for call detail records) to the connected subsequent node servers.
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When the first node server is off-line, the database automatically locks, and thus prevents further database changes. The IP telephony network will continue to operate, but you will not be able to add or configure any devices that are managed by Cisco Unified CallManager. The database itself in Cisco Unified CallManager Release 5.x has changed to use a new SQL engine: the Informix Dynamic Server by IBM.
Cisco Unified CallManager Clustering

The database replicates nearly all information in a star topology (one first node, many subsequent nodes). However, Cisco Unified CallManager nodes also use a second communication method to replicate run-time data in a mesh topology (every node updates every other node). This type of communication is used for dynamic information that changes more frequently than database changes. The primary use of this replication is to communicate newly registered phones, gateways, and digital signal processor (DSP) resources, so that optimum routing of calls between members of the cluster and the associated gateways occurs.
Note For more details on the two methods of intracluster communications, see http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_ guide_chapter09186a008063743a.html#wp1085323
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Cluster Communication
This topic describes intra-cluster communication processes and protocols.
Intra-Cluster Communication
Management Channel: UDP 8500 Marked as Default (0x00)
Changed from 1433 (SQL)
DB TRAFFIC: TCP 15001501
LICENSING SERVICE: TCP 5555
SDL/ICCS TRAFFIC: TCP 80018003
All ports are published on Cisco.com

: : :
Marked as CS3
These traffic types flow between the servers: Management channel: Used for setting up the cluster on TCP port number 8500. Database traffic: Cisco Unified CallManager Release 5.0 uses a different SQL engine. As a result, the port number used for database traffic has changed in Cisco Unified CallManager Release 5.0 to TCP 1500 and 1501. Licensing messages: A licensing service is added to the Cisco Unified CallManager cluster. If you add a phone on the subsequent node, it uses port number 5555 to inform the first node. All of these packets are marked as default, meaning that they are not marked and should remain as is: Call processing traffic: Uses the same port number as Cisco Unified CallManager 4.x; that is, TCP port 8001 to 8003. Call-processing traffic is priority traffic and should be marked as such so that it will use the priority queue. The figure addresses the most common communication messages. Other applications, such as Cisco Unified CallManager Assistant, Cisco Unified CallManager Extension Mobility, or any other Java applications, if enabled, also generate additional unique traffic between the servers. A complete list of the TCP and User Datagram Protocol (UDP) ports that Cisco Unified CallManager 5.0 uses is published on Cisco.com in a document called Cisco Unified CallManager TCP and UDP Port Usage. Review this document carefully as you plan the migration to Cisco Unified CallManager Release 5.0, to ensure the proper configuration of firewalls, access control lists (ACLs), and quality of service (QoS) features on a network to support the IP communications solution.
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Dynamic Firewall (iptables)

In Cisco Unified CallManager Release 5.0, Cisco added a dynamic firewall to secure the database ports.
Dynamic Firewall (iptables)

First node Subsequent node
Firewall
Subsequent node (DBports, ACCEPT)
Other (DBports, DENY)
1) Install first node using a security password. 2) Add subsequent node to the first node database. 3) Install subsequent node using the same security password. 4) Subsequent node establishes connection to first node on management channel. 5) First node verifies the subsequent nodes authenticity and adds its IP address to firewall (iptables). 6) Subsequent node allowed database access.
The dynamic firewall works according to these steps:

Step 1 Step 2 Step 3 Step 4
Install the first node with the security password. Add the subsequent node to the database of the first node before you install the subsequent node. Install the subsequent node using the same security password. When the subsequent node gets network connectivity, it tries to establish a connection to the first node over the management channel and sends either its Domain Name System (DNS) host name or its IP address. After the first node receives that packet, it verifies the IP address of the host name with the IP address of the host name of the subsequent node, which is already configured in the first node database. If a match is found, the first node adds a rule in its firewall table specifying that the subsequent node should be granted access to the database.
Step 5
The subsequent node can now access all of the first node database ports. This process is repeated for every subsequent node added to the cluster. A rule of last resort specifies that if anyone else tries to access the database, deny the access and drop the packets. In this way, only trusted sources are allowed access and all other access is denied.
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Cisco Unified CallManager Operating System, Database, and Supporting Applications

This topic describes software components of Cisco Unified CallManager and their purpose.
Operating System, Database, and Applications

Appliance (Linux-based operating system) Informix Dynamic Server Directory Synchronization Disaster Recovery System DHCP Server
CIPT1 v5.01-13
Since the introduction of the appliance model, the operating system of Cisco Unified CallManager is Linux-based. Informix Dynamic Server is used for the database in which users are stored. You can use external directories to add users to the first node database and you can authenticate a user against an external directory. The Cisco Disaster Recovery System (DRS) allows backing up and restoring the system. It is accessible from the CLI (direct console or SSH) and from a web browser and is fully integrated with Cisco Unified CallManager. No client installation is needed on the administrator PC. Cisco Unified CallManager provides Dynamic Host Configuration Protocol (DHCP) services to IP phones.
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Cisco Unified CallManager Servers

This topic describes Cisco 7800 Media Convergence Server platforms that operate with Cisco Unified CallManager Release 5.0.
Cisco 7800 Series MCS

Cisco 7800 MCS server platforms: 7815 Series 7825 Series 7835 Series 7845 Series Minimum hardware requirements for Cisco Unified CallManager Release 5.0: 2-GHz Processor 2 GB RAM 72-GB Harddisk For detailed model information check, Cisco.com
CIPT1 v5.01-14
Because voice networks should maintain a high uptime, you must install Cisco Unified CallManager on a server that meets Cisco configuration standards. For this reason, Cisco has collaborated with two server hardware manufacturers, Hewlett-Packard and IBM, to create MCSs. Hewlett-Packard and IBM designed these server hardware platforms specifically for Cisco voice applications. All of these servers are rack-mountable and do not include a monitor, mouse, or keyboard. Cisco designed the Cisco MCS for local setup, rack mounting, and remote administration. For a list of servers on which Cisco Unified CallManager Release 5.0 is supported, refer to the Cisco 7800 Series Media Convergence Servers data sheet at http://www.cisco.com/en/US/products/hw/voiceapp/ps378/products_data_sheets_list.html These are the minimum requirements for Cisco Unified CallManager: 2 GHz processor 2 GB RAM 72-GB hard disk
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Licensing Model
This topic describes the new licensing model that was introduced with Cisco Unified CallManager Release 5.0.
Licensing
License required for adding phones and Cisco Unified CallManager servers Upgrades do not need a new license; however, license is needed if additional phones or Cisco Unified CallManager servers are added First node is the only licensing server Cisco Unified CallManager cluster continues to work if licensing service is stopped
CIPT1 v5.01-15
Licensing is implemented in Cisco Unified CallManager Release 5.0 Administration to accurately track the number of devices that are connected to the Cisco Unified CallManager, including third-party SIP phones, and compare that number with the number of unit licenses that have been purchased. There are no temporary licenses. License enforcement occurs at the time of phone provisioning and Cisco Unified CallManager service activation. Customers who upgrade from Cisco Unified CallManager 4.x do not need to purchase new licenses unless additional phones or Cisco Unified CallManager servers are added. Details of licensing under upgrade scenarios are covered in more detail in subsequent topics. The first node is the only licensing server. The licensing server is the logical component that keeps track of the licenses purchased and the licenses used. If the first node fails, no new phones can register; however, existing phones still operate. If this happens, contact Cisco TAC and they will provide you with a license file.
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Licensing Compliance
Licensing compliance must be followed by devices such as Cisco IP phones, Cisco IP Communicator, analog telephone adapters (ATAs), and the Cisco Unified CallManager application software.
Licensing Compliance
Cisco Unified CallManager licensing compliance for devices and applications Device licenses The maximum number of provisioned devices (Cisco IP phones, video devices) in Cisco Unified CallManager database will be tracked and enforced. Route points and CTI ports are not enforced Application licenses The Cisco Unified CallManager software will be tied to a server via MAC address of the first node. Licenses created and distributed in accordance with Cisco FlexLM process
Device License
Application License
The Cisco FlexLM process handles the creation and distribution of licenses.
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Device License Units

Device licenses are sold in device license units.

Device licenses are sold in device license units. Cisco device license units Third-party device license units Cisco Unified CallManager tracks the number of units required by each device. Each device type (single line, multiple line, and so on) corresponds to a fixed number of units.

CIPT1 v5.01-17
These two types of product IDs are available: Cisco device license units Third-party device license units Cisco Unified CallManager tracks the number of units required by each device, as shown in the figure. Each device type corresponds to a fixed number of units. Units are perpetual and deviceindependent. The Cisco units are for Cisco devices only. The third-party units can be converted to Cisco units, but not vice versa.
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License File Request Process

The figure depicts the licensing fulfillment process from the time a customer orders a Cisco Unified CallManager until a license file is installed on the license server.
License File Request Process
Customer
Place Order PAK MAC Ship Web PAK
Cisco.com
E-Mail License Server Install
Generate License File
CIPT1 v5.01-18
The license file request process includes these steps:

Step 1 Step 2 Step 3 Step 4 Step 5 Step 6 Step 7
The customer places an order for Cisco Unified CallManager. The manufacturing database scans the Product Authorization Key (PAK) and records it against the sales order. The product (CD or paper Claim Certificate) is physically delivered to the customer. The customer registers the product at a Cisco.com or public web page and provides the MAC address of the first-node device that will become the license server. The license fulfillment infrastructure validates the PAK and the license key generator creates a license file. The license file is delivered via e-mail to the customer. The e-mail also contains instructions on how to install the license file. The customer installs the license file on the license server.
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License File Specifics

The license file contains information specific to the customer environment.
License File Specifics

The license file contains the following information: Number of Cisco Unified CallManager nodes licensed Version of the Cisco Unified CallManager supported Number of phone units licensed MAC address of the license server Only major releases will be taken into account for the version check. A 5% overdraft is allowed.
CIPT1 v5.01-19
The information in the license file includes: Number of Cisco Unified CallManager nodes licensed (application licenses) Version of the Cisco Unified CallManager supported Number of phone units licensed (device licenses) The number of licensed units in the license file will correspond to the number of unit licenses for all the phone types the customer has purchased. MAC address of the first-node license server where the license file will be installed
Note
The licenses specified in the license file can be used only within the cluster on which the license file is uploaded.
When the license file is uploaded, a version check is conducted. Only major releases are taken into account for the version check. Minor, maintenance, and service releases are not considered. When a major release upgrade occurs, a new license file must be loaded on the license server, with the version number of the new release. A five percent licensing overdraft is allowed. An overdraft condition occurs if the customer has more phones registered than licenses purchased. The licensing infrastructure in Cisco Unified CallManager allows customers to have five percent more phones registered than licenses purchased. This grace period enables customers to add phones when they run out of licenses, and to have time to purchase additional licenses.
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Example License File

The figure shows an example of a license file for a Cisco IP phone.
Example License File Permanent IP Phone Licenses

INCREMENT PHONE_UNIT cisco 5.0 permanent uncounted \ VENDOR_STRING=<Count>1000</Count><OrigMacId>000BCD4 EE59D</OrigMacId><LicFileVersion>1.0</L icFileVersion> \ HOSTID=000bcd4ee59d OVERDRAFT=50 \ NOTICE="<LicFileID>20050826140539162</LicFileID><LicLineID >2</LicLineID> \ <PAK></PAK>" SIGN="112D 17E4 A755 5EDC F616 0F2B B820 AA9C \ 0313 A36F B317 F359 1E08 5E15 E524 1915 66EA BC9F A82B CBC8 \ 4CAF 2930 017F D594 3E44 EBA3 04CD 01BF 38BA BF1B"
CIPT1 v5.01-20
Significant fields are highlighted and described as follows: INCREMENT PHONE_UNIT cisco 5.0 permanentIndicates a phone unit license file for a Cisco Unified CallManager Release 5.0. If this had been a Cisco Unified CallManager node license, the INCREMENT type would be Cisco Unified CallManager_NODE. There is no expiration date for this license, as indicated by the keyword permanent. This license file provides 1000 PHONE_UNIT licenses. The MAC address of the license server is 000BCD4EE59D. OVERDRAFT=50 indicates allowed overdraft (5% of 1000). The overdraft value is determined by Cisco.
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Obtaining Additional Licenses

To obtain additional licenses, use the same process as obtaining a new license file, as illustrated in the figure.
Obtaining Additional Licenses

Current License File 100 units
Order for addtional 100 units
New
100 units License Server 200 units
Obtaining additional licenses uses the same process as obtaining a new license file:
1. The customer places an order for the additional licenses for a license server (MAC address specified). 2. When the order is received, Cisco.com generates a license file with the additional count. 3. This new license file will have to be uploaded to the license server.
CIPT1 v5.01-21
License files are additive. For example, if you have an existing license file uploaded to Cisco Unified CallManager that contains 100 units, and you purchase another 100 units, the second license file that is generated will contain only 100 license units. When this license file is uploaded to Cisco Unified CallManager, the 100 units from the first license file is added to the second, giving the customer a total of 200 licenses.
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Migrating from Windows to Appliance

When you migrate from a Windows-based Cisco Unified CallManager to an appliance-based Cisco Unified CallManager, the licenses required for existing phones and existing Cisco Unified CallManager Services nodes are calculated, and an intermediate file containing these license counts is generated during the Cisco Unified CallManager migration process.
Migrating from Windows to Appliance

rm Inte te edia
Cisco.com
New
License Server
The licenses required are calculated during the Cisco Unified CallManager migration process and an intermediate file containing these license counts is generated.
Procedure to Upgrade:
1. Use View File option on the License File Upload window to view intermediate file. 2. Copy and paste the intermediate license file into the Cisco Unified CallManager License Upgrade window on Cisco.com to obtain the actual license file. 3. Upload the license file to the license server.
You can view this file by clicking View File in the License File Upload window on Cisco.com. Copy the intermediate license file content from the pop-up window that displays and paste it in the Cisco Unified CallManager License Upgrade window to obtain the actual license file.
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Licensing Functional Diagram

The figure shows the key components of the licensing implementation.
Licensing Functional Diagram

License Server Keeps track of licenses purchased and used License Manager Cisco Unified CallManager service acts as a broker between Cisco Unified CallManager applications that use licensing information and the License Server Cisco Unified CallManager
Request (1) Admin Subsystem Response (4) License Manager Grant or Deny (3)
Request (2) License Server
Alarm Subsystem
Async-Notification (for Deny)(5)
CIPT1 v5.01-23
The key licensing components are as follows: License server: The license server is the logical component that keeps track of the licenses purchased and the licenses used. When you request a license file, a MAC address is required to generate the license file. Once generated, the license file should be loaded on the machine with the matching MAC address, and the Cisco Unified CallManager node takes on the functionality of the license server. The license server must be run on the first node server. License Manager: A new service, License Manager (LicenseMgr), is implemented on Cisco Unified CallManager. This logical component acts as a broker between Cisco Unified CallManager applications that use licensing information and the license server. When the License Manager receives a request from the Cisco Unified CallManager application, it forwards the request to the license server, and responds back to the application after the request is processed completely.
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Licensing Functional Diagram (Cont.)

Admin subsystem Keeps information about the license units required for each phone type Provides license unit calculator Displays the total license capacity and the number of licenses in use Alarm subsystem provides alarms for following conditions: Overdraft License server down Insufficient licenses License file version mismatch
Request (1) Admin Subsystem Response (4) License Manager
Alarm Subsystem
Async-Notification (for Deny)(5)
Cisco Unified CallManager
CIPT1 v5.01-24
An administration subsystem and alarm subsystem complete the functional diagram. Details of these two subsystems are as follows: The administration subsystem provides these capabilities: Keeps information about the license units required for each phone type. The customer can view this information using a GUI. Supports a GUI tool that calculates the required number of phone unit licenses. The customer inputs phone types and the number of phones of each type that the customer wants to purchase. The output is the total number of licenses that the customer would need for the given configuration. Supports a GUI tool that displays the total license capacity and the number of licenses in use. The tool can also report the number of available licenses (total and per license server).
The alarm subsystem generates alarms that are routed to event logs and Simple Network Management Protocol (SNMP) traps to notify the administrator of these conditions: Overdraft: Occurs when an overdraft condition exists License server down: Occurs when the license manager cannot reach the license server Insufficient licenses: Occurs when the license server detects the fact that there are not sufficient licenses to fulfill the request and raises an alarm to notify the administrator Issues with license file: Occurs when there is a version mismatch between the license file and the Cisco Unified CallManager (license file version mismatch alarm), or when the number of licenses in the license file is less than the number of phones provisioned (license file insufficient licenses alarm), which can occur only in an upgrade to Cisco Unified CallManager Release 5.0.
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Calculating License Units

Cisco Unified CallManager includes a tool to calculate licenses for a given number of phones.
Calculating License Units
Calculate the number of phone unit licenses required for a specific configuration of type of phones and number of phones of each type.
Use this procedure to calculate the number of phone licenses required when the number of phone types and the total number of phones per phone type is entered:
Step 1
Choose System > License > License Unit Calculator. The License Unit Calculator window displays. The number of license units consumed per device displays corresponding to the node or device. In the Number of Devices column, update the number of devices needed, corresponding to each node or phone. Click Calculate. The total number of Cisco Unified CallManager node license units and phone license units display.
Step 2 Step 3
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Generating a License Unit Report

Another tool, the License Unit Report tool, generates a report listing the total number of license units authorized, the number of licenses used, and the number of units remaining.
Generating a License Unit Report
Displays the number of phone license units and number of node license units available
Use this procedure to generate a report of the number of licenses available:

Step 1 Step 2
Choose System > License > License Unit Report. The License Unit Report window displays. This window displays the number of phone licenses and number of node licenses available, in these categories: Units Authorized Units Used Units Remaining
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Uploading a License File

To upload a license file, use Cisco Unified CallManager Administration.
Uploading a License File

Step 1: System > Licensing > License File Upload
CIPT1 v5.01-27
Follow this procedure to upload a license file to the server:

Note Upload the license file only on the first node of a Cisco Unified CallManager cluster. The License File Upload page has a validation to allow the file upload only on the first node.
Step 1
Choose System > License > License File Upload. The License File Upload window displays. The Existing License Files drop-down list box displays the license files that are already uploaded to the server.
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Uploading a License File (Cont.)

Step 2: Click Upload License File Step 3: Click Browse to choose the license file from local directory, then click Upload
The license file should be saved to the PC that is used to access the Cisco CallManager Administration page.
Step 2 Step 3
To choose a new license file to upload, click Upload License File. The Upload File pop-up window displays. Click Browse and choose a license file to upload to the server.
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Step 4: Click Upload and Continue to proceed with the file validation
CIPT1 v5.01-29
Step 4 Step 5
Click Upload and Continue to proceed with the file validation. After the upload process is complete, the Upload Result file displays. Click Close. In the License File Upload window, the status of the uploaded file displays.
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(Optional): Click View File to view the uploaded file
CIPT1 v5.01-30
Once uploaded, you can click View File to view the uploaded file.
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Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
The Cisco Unified Communications solution provides the foundation for converged networks and includes the infrastructure, call-processing, applications, and client layers. Cisco Unified CallManager functions include call processing, signaling and device control, dial plan administration, phone feature administration, directory services, and a programming interface. Cisco Unified CallManager runs on an appliance beginning with Release 5.0. Cisco Unified CallManager uses an IDS database to replicate data from a first node to subsequent nodes within a cluster.
CIPT1 v5.01-31
Summary (Cont.)
Intra-cluster traffic includes management, signaling, and database communication. Cisco Unified CallManager appliances use their own operating system and provide services by additional applications, such as the IDS database, a disaster recovery system, a DHCP server, and directory synchronization. Cisco Unified CallManager hardware requirements include Cisco MCSs, designed specifically for Cisco voice applications by HP and IBM. A new phone and node unit licensing model is introduced. Tools are provided to calculate licenses for specific phone types and numbers and to determine the number of license units available.
CIPT1 v5.01-32
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References
For additional information, refer to these resources: Release Notes for a Cisco Unified CallManager Release 5.0 http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/rel_note/index.htm Cisco Unified Communications SRND Based on Cisco Unified CallManager Release 5.0 http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/srnd5_0/index.htm Licensing in Cisco Unified CallManager System Guide, Release 5.0(2): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/sys_ad/5_0_2/ccmsy s/a02licen.htm License Unit Report, License Unit Calculator, and License File Upload in Cisco Unified CallManager Administration Guide, Release 5.0(2): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/sys_ad/5_0_2/ccmcf g/index.htm
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Lesson 2
Evaluating Cisco Unified CallManager Release 5.0 Deployment Options

Overview
To ensure the same service availability that the traditional voice network provides, it is critical to build redundancy and failover capabilities into the IP telephony network design. The primary ways to achieve these capabilities are to cluster Cisco Unified CallManager servers and to follow recommended design and deployment practices.
Objectives
Upon completing this lesson, you will be able to determine the optimum Cisco Unified CallManager cluster option and IP telephony deployment model for your enterprise. This ability includes being able to meet these objectives: Identify supported IP telephony call-processing deployment models Identify the major characteristics and design guidelines of a single-site call-processing deployment model Identify the major characteristics and design guidelines of a multisite centralized callprocessing deployment model Identify the major characteristics and design guidelines of a multisite distributed callprocessing deployment model Identify the characteristics, limitations, and advantages of clustering over the IP WAN Explain redundancy requirements and restrictions for Cisco Unified CallManager Release 5.0 implementations
Call-Processing Deployment Models

This topic describes the supported Cisco Unified CallManager deployment models.
Call Processing Deployment Models
Applications Cisco Unified CallManager Cluster
Applications
PSTN IP WAN
Branch
Headquarters
Supported IP telephony deployment models:

Single-site deployment Multisite WAN with centralized call processing Multisite WAN with distributed call processing Clustering over the IP WAN
Each model differs in type of traffic that is carried over the WAN, location of the callprocessing agent, and size of the deployment. Cisco IP telephony supports these deployment models: Single-site Multisite with centralized call processing Multisite with distributed call processing Clustering over the IP WAN
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Single-Site Deployment
This topic describes the single-site Cisco Unified CallManager deployment model.
Single-Site Deployment
Cisco Unified CallManager servers, applications, and DSP resources at same physical location IP WAN (if one) used for data traffic only; PSTN used for all external calls Supports approximately 30,000 IP phones per cluster
PSTN
SIP/SCCP
CIPT1 v5.01-3
In a single-site deployment model, all Cisco Unified CallManager servers, applications, and digital signal processor (DSP) resources are located in the same physical location. You can implement multiple clusters inside a LAN or a metropolitan-area network (MAN) and interconnect them through intercluster trunks if you need to deploy more IP phones in a singlesite configuration. Gateway trunks that connect directly to the public switched telephone network (PSTN) handle external calls. If an IP WAN exists between sites, it is used to carry data traffic only; no telephony services are provided over the WAN. Use this model for a single campus or a site with fewer than 30,000 IP phones.
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Single-Site Deployment: Design Guidelines

Single-site deployment is a subset of the distributed and centralized call-processing model. Single-site deployment requires that, for future scalability, you adhere to the recommended best practices specific to the distributed and centralized call-processing model. When you develop a stable, single-site infrastructure that is based on a common infrastructure philosophy, you can easily expand the IP telephony system applications, such as video streaming and videoconferencing, to remote sites.
Single-Site: Design Guidelines

Understand the current calling patterns within the enterprise. Use the G.711 codec; DSP resources can be allocated to other functions, such as conferencing and MTP. OffNet calls should be diverted to the PSTN or sent to the legacy PBX. Choose a uniform gateway for PSTN use. Deploy the recommended network infrastructure. Do not oversubscribe the Cisco Unified CallManager and clustering capability.
CIPT1 v5.01-4
These are the guidelines for single-site deployments: You must understand the current calling patterns within the enterprise. How and where are users making calls? How many calls are intersite or interbranch versus intrasite? If calling patterns dictate that most calls are intrasite, use the single-site model to deploy IP telephony and make use of the relatively inexpensive PSTN. The single-site design also simplifies dial plans and avoids provisioning dedicated bandwidth for voice in the IP WAN. The G.711 coder-decoder (codec) should be used. The call will stay in the LAN, and G.711 is a simple mechanism for deployment. It does not require dedicated DSP resources for transcoding (which means converting between codec types, such as between G.711 and G.729), and older voice-mail systems may support only G.711. You can allocate these DSP resources to other functions, such as conferencing and Media Termination Point (MTP). Although the 64-kpbs per-call bandwidth that G.711 consumes is higher than that of all other codecs, it is not a concern in this design because the call is not traversing the WAN, in which bandwidth is generally limited. All OffNet calls will be diverted to the PSTN or sent to the legacy PBX for call routing if the PSTN resources are being shared during migratory deployments. Use Media Gateway Control Protocol (MGCP) gateways for the PSTN if H.323 functionality is not required. Centralize the gateway functions using H.323 gatekeepers when deploying multiple clusters, rather than using MGCP gateways.
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Deploy the recommended network infrastructure for high-availability, fault-tolerant infrastructure, connectivity options for telephones (in-line power), quality of service (QoS) mechanisms, and other services. Do not oversubscribe Cisco Unified CallManager to scale larger installations.
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Multisite WAN with Centralized Call Processing

This topic describes the multisite centralized call-processing deployment model with a Cisco Unified CallManager cluster at a central site and a connection to several remote sites through a QoS-enabled IP WAN.
Multisite WAN with Centralized Call Processing

Cisco Unified CallManager at central site; applications and DSP resources centralized or distributed IP WAN carries voice traffic and call control signaling Supports approximately 30,000 IP phones per cluster Call admission control (limit number of calls per site) SRST for remote branches AAR used if WAN bandwidth is exceeded
SIP/SCCP SIP/SCCP SIP/SCCP Cisco Unified CallManager Cluster
PSTN
IP WAN
CIPT1 v5.01-5
The multisite WAN model with centralized call processing consists of a single-call processing agent that provides services for many sites and uses the IP WAN to transport IP telephony traffic between the sites. The IP WAN also carries call control signaling between the central site and the remote sites. The figure illustrates a typical centralized call-processing deployment, with a Cisco Unified CallManager cluster as the call-processing agent at the central site and an IP WAN with QoS enabled to connect all the sites. The remote sites rely on the centralized Cisco Unified CallManager cluster to handle their call processing. Applications such as voice mail and interactive voice response (IVR) systems are typically centralized as well to reduce the overall costs of administration and maintenance. The WAN connectivity options include the following: Leased lines Frame Relay ATM ATM to Frame Relay service interworking (SIW) Routers that reside at WAN edges require QoS mechanisms, such as priority queuing and traffic shaping, to protect voice traffic from data traffic across the WAN (where bandwidth is typically scarce).
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To avoid oversubscribing the WAN links with voice traffic, causing deterioration of the quality of established calls, the network may need a call admission control scheme. With the introduction of Cisco CallManager Release 3.3, centralized call-processing models can take advantage of automated alternate routing (AAR) features. AAR allows Cisco Unified CallManager to dynamically reroute a call over the PSTN if the call exceeds the WAN bandwidth. You can provide PSTN access for the voice network through a variety of Cisco gateways. When the IP WAN is down, the users at remote branches can dial an access code and place their calls through the PSTN, or the Cisco Survivable Remote Site Telephony (SRST) feature that is available for Cisco IOS gateways can provide call processing during the outage. ISDN can also provide backup data connectivity during WAN failures; however, voice traffic should not use the ISDN links because ISDN interfaces do not support the required QoS features.
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Multisite WAN with Centralized Call Processing: Design Guidelines

Follow these best-practice guidelines when deploying a centralized call-processing model.
Multisite WAN with Centralized Call Processing: Design Guidelines

Minimize delay between Cisco Unified CallManager and remote locations to reduce voice cut-through delays. Use the locations mechanism in Cisco Unified CallManager to provide call admission control into and out of remote branches. SRST on the branch router limits remote offices to a maximum of 480 Cisco IP phones when using a Cisco 7200 Series router.
CIPT1 v5.01-6
Follow these guidelines when implementing the multisite WAN model with centralized call processing: Minimize delay between Cisco Unified CallManager and remote locations to reduce voice cut-through delays (also known as clipping). Use the locations mechanism in Cisco Unified CallManager to provide call admission control into and out of remote branches. See the lesson on Call Admission Control for details on how to apply this mechanism to the various WAN topologies. The locations can support a maximum of 30,000 IP phones when Cisco Unified CallManager runs on the largest supported server. In Cisco Unified CallManager Release 5.0, you can use Resource Reservation Protocol (RSVP)-based call admission control between locations. There is no limit to the number of IP phones at each individual remote branch. However, the capability that is provided by the SRST feature in the branch router limits remote branches to 480 Cisco IP phones on a Cisco 7200 Series router during failover. Smaller platforms have lower limits. If a distributed call-processing model is more suitable for the business needs of a customer, the choices include installing a local Cisco Unified CallManager server or running Cisco Unified CallManager Express on the branch router.
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Multisite WAN with Distributed Call Processing

This topic describes the multisite WAN with distributed call-processing deployment model.
Multisite WAN with Distributed Call Processing

Cisco Unified CallManager and applications located at each site IP WAN does not carry call control signaling Scales to hundreds of sites Transparent use of the PSTN if the IP WAN is unavailable
SIP/SCCP GK Cisco Unified CallManager Cluster
Gatekeeper
PSTN
IP WAN
SIP/SCCP
SIP/SCCP
Cisco Unified CallManager Clusters
CIPT1 v5.01-7
A multisite distributed call-processing deployment has one or more call-processing agents at each site, and each site has its own Cisco Unified CallManager cluster. You can trunk these sites together through an IP WAN. Depending on your network design, a distributed call-processing site may consist of the following: A single site with its own call-processing agent, which may be a Cisco Unified CallManager or a third-party call agent A centralized call-processing site (and all of its remote sites) that the network views as a single site for distributed call processing A legacy PBX with a VoIP gateway or a legacy PBX that is attached using a time-division multiplexing (TDM) interface to a VoIP gateway You can interconnect all distributed call-processing sites through an IP WAN. Cisco considers a site that is connected only through the PSTN to be a standalone site. The WAN connectivity options include the following: Multiprotocol Label Switching (MPLS) Virtual Private Network (VPN) Voice- and Video-Enabled VPN (V3PN) Leased lines Frame Relay ATM ATM to Frame Relay SIW
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Multisite distributed call processing allows each site to be completely self-contained. In the event of an IP WAN failure or insufficient bandwidth, the site does not lose call-processing service or functionality. Cisco Unified CallManager simply sends all calls between the sites across the PSTN. In summary, the main benefits of the multisite WAN with distributed call processing deployment model are as follows: Cost savings when you use the IP WAN for intersite calls Toll-bypass savings when you use remote gateways to drop off to the PSTN (known as tailend hop-off, or TEHO) No loss of functionality during an IP WAN failure Scalability to hundreds of sites
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Multisite Distributed Call Processing: Design Guidelines

The multisite WAN with distributed call-processing deployment model is a superset of the single-site and multisite WAN with centralized call-processing models.
Multisite Distributed Call Processing: Design Guidelines

Use HSRP gatekeeper pairs. Deploy a single WAN codec. Implement a logical hub-and-spoke topology for the gatekeeper. Gatekeeper networks scale to hundreds of sites.
CIPT1 v5.01-8
You should follow the best practices guidelines for single-site and multisite deployments in addition to those listed here, which are specific to this deployment model. The gatekeeper or Session Initiation Protocol (SIP) proxy servers are among the key elements in the multisite WAN with distributed call processing. Both provide dial plan resolution, with the gatekeeper also providing call admission control. A gatekeeper is an H.323 device that provides call admission control and E.164 dial plan resolution. Additional gatekeeper guidelines include the following: It is recommended that you use alternate gatekeeper support to provide a gatekeeper solution with high availability. It is also recommended that you use multiple gatekeepers to provide spatial redundancy within the network. It is recommended that you use a single WAN codec. This design makes capacity planning easy and does not require you to overprovision the IP WAN to allow for worst-case scenarios. Use a logical hub-and-spoke topology for the gatekeeper. A gatekeeper can manage the bandwidth into and out of a site or between zones within a site, but it is not aware of the topology. Gatekeeper networks can scale to hundreds of sites, and the design is limited only by the hub-and-spoke topology. Include information on SIP proxy servers. The following best practices apply to the use of SIP proxies: Provide adequate redundancy for the SIP proxies. Ensure that the SIP proxies have the capacity for the call rate and number of calls required in the network.
2006 Cisco Systems, Inc. Get Started with Cisco Unified CallManager Release 5.0 1-47
Clustering over the IP WAN

This topic describes clustering over the IP WAN.
Clustering over the IP WAN

First Node / TFTP
<40 ms Round-Trip Delay
IP WAN
SIP/SCCP
QoS Enabled BW
SIP/SCCP
Applications and Cisco Unified CallManagers of the same cluster distributed over the IP WAN IP WAN carries intracluster server communication and signaling Limited number of sites
Cisco supports Cisco Unified CallManager clusters over a WAN. Although there are stringent requirements, this design offers these advantages: Single point of administration for users for all sites within the cluster Feature transparency Shared line appearances Extension mobility within the cluster Unified dial plan The cluster design is useful for customers who require more functionality than the limited feature set offered by SRST. This network design also allows remote offices to support more Cisco IP phones than SRST in the event that the connection to the primary Cisco Unified CallManager is lost.
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Clustering over the IP WAN: Design Guidelines

Although the distributed single-cluster call-processing model offers some significant advantages, it must adhere to some strict design guidelines.
Clustering over the IP WAN: Design Guidelines

40-ms round-trip delay between any two Cisco Unified CallManager servers in the cluster Minimum 1.544 Mbps and 900 kbps for every 10,000 BHCAs within the cluster Up to eight small sites using the remote failover deployment model Failover across WAN supported (more bandwidth)
CIPT1 v5.01-10
The design guidelines for clustering over the IP WAN are as follows: Two Cisco Unified CallManager servers in a cluster must have a maximum round-trip delay of 40 ms between them. In comparison, high-quality voice guidelines dictate that one-way end-to-end delay should not exceed 150 ms. Because of this strict guideline, you can use this design only between closely connected, high-speed locations. For every 10,000 busy hour call attempts (BHCAs) within the cluster, you must support an additional 900 kbps of WAN bandwidth for intracluster run-time communication. The BHCA represents the number of call attempts made during the busiest hour of the day. Up to eight small sites are supported using the remote failover deployment model. Remote failover allows you to deploy the backup servers over the WAN. Using this deployment model, you may have up to eight sites with Cisco Unified CallManager subsequent nodes being backed up by Cisco Unified CallManager subsequent nodes at another site. SRST can function in this model but is not necessary. The telephones can failover across the WAN to other Cisco Unified CallManager servers. This design may require significant additional bandwidth, depending on the number of telephones at each location.
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Cisco Unified CallManager Release 5.0 Database Redundancy

This topic describes two cluster designs that provide call-processing redundancy.
1:1 Redundancy Design

Primary Secondary/ Backup
High availability (upgrade) Increased server count Simplified configuration

15,000 IP phones
Cisco MCS 7845 First Node and TFTP Server
7500 IP phones
Cisco MCS 7845 First Node and TFTP Server (Not Req. <1000) Primary 1 to 7500 Backup
30,000 IP phones
Backups
1 to 7500 7501 to 15000
Backups
1 to 7500 7501 to 15000 15001 to 22500 22501 to 30000
Backups
CIPT1 v5.01-11
In a 1:1 Cisco Unified CallManager redundancy deployment design, you can have a dedicated backup server for each primary server. This design guarantees that Cisco IP phone registrations will never overwhelm the backup servers, even if multiple primary servers fail. However, the 1:1 redundancy design considerably limits the maximum cluster size and is not cost-effective. Each cluster must also have a designated TFTP server. Depending on the number of devices that a server is supporting, you can combine this TFTP server functionality with the first-node or subsequent-node Cisco Unified CallManager servers, or you can deploy the TFTP functionality on a separate, standalone server. The TFTP server is responsible for delivering IP phone configuration files to each telephone, along with streamed media files, such as music on hold (MOH) and ring files; therefore, the TFTP server can experience a considerable network and processor load. In this example, a Cisco 7845 media convergence server (MCS) is used because each Cisco Unified CallManager server installed on that platform supports a maximum of 7500 Cisco IP phones. A single Cisco Unified CallManager is the primary server, with a secondary server acting as a dedicated backup. The primary or backup server can also serve as the TFTP server in smaller IP telephony deployments (fewer than 1000 IP phones). When you increase the number of IP phones, you must increase the number of Cisco Unified CallManager servers that are required to support the telephones. Some network engineers may consider the 1:1 redundancy design excessive, because a well-designed network is unlikely to lose more than one primary server at a time. With the low possibility of server loss and the increased server cost, many network engineers elect to use a 2:1 redundancy design.
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In a 2:1 Cisco Unified CallManager redundancy deployment design, you have a dedicated backup server for every two primary servers.

Primary Secondary/ Backup
Cost-efficient redundancy Service impacted during upgrade

15,000 IP phones
7500 IP phones
Cisco MCS 7845 First Node and TFTP Server (Not Req. <1000) Primary 1 to 7500 Backup
30,000 IP phones
Backup
1 to 7500 7501 to 15000
Backup
1 to 7500 7501 to 15000 15001 to 22500 22501 to 30000
Backup
CIPT1 v5.01-12
Although the 2:1 redundancy design offers some redundancy, there is the risk of overwhelming the backup server if multiple primary servers fail. In addition, upgrading the Cisco Unified CallManager servers can cause a temporary loss of service because you must reboot the Cisco Unified CallManager servers after the upgrade is complete. Network administrators use this 2:1 redundancy model in most IP telephony deployments because of the reduced server costs. If you are using a Cisco MCS 7845 (shown in the figure), that server is equipped with redundant, hot-swappable power supplies and hard drives. When you properly connect and configure these servers, it is unlikely that multiple primary servers will fail at the same time, which makes the 2:1 redundancy model a viable option for most businesses.
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Summary
Summary
Supported Cisco IP telephony deployment models are singlesite, multisite with centralized call processing, multisite with distributed call processing, and clustering over the IP WAN. In the single-site deployment model, the Cisco Unified CallManager applications and the DSP resources are at the same physical location; the PSTN handles all external calls. The multisite centralized model has a single call-processing agent; applications and DSP resources are centralized or distributed; and the IP WAN carries voice traffic and call control signaling between sites.
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Summary (Cont.)
The multisite distributed model has multiple independent sites, each with a call-processing agent, and the IP WAN carries voice traffic between sites but not call control signaling. Clustering over an IP WAN provides central administration, a unified dial plan, feature extension to all offices, support for more remote phones during failover, but places strict delay and bandwidth requirements on the WAN. Clusters provide redundancy. A 1:1 redundancy design offers highest availability but requires most resources.
CIPT1 v5.01-14
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Lesson 3
Installing Cisco Unified CallManager Release 5.0

Overview
Installing or upgrading software is a fundamental task that you need to perform to support the deployment of Cisco Unified CallManager Release 5.0. This lesson covers the new Cisco Unified CallManager Release 5.0 installation framework, installation requirements, and the procedures to perform a basic installation or apply service releases and patches.
Objectives
Upon completing this lesson, you will be able to describe the Cisco Unified CallManager Release 5.0 installation framework and procedures. This ability includes being able to meet these objectives: Define the Cisco Unified CallManager Release 5.0 installation framework Identify software and hardware requirements to install or upgrade to Cisco Unified CallManager Release 5.0 Perform the tasks that must be completed before installation Describe the process to perform a basic installation of the operating system and Cisco Unified CallManager application on the first node Describe the process to perform a basic installation on the subsequent nodes in the cluster Install service releases and patches while performing a basic installation View and retrieve log files created during installation Describe the tasks that must be performed after installation
Installation Framework
This topic describes the Cisco Unified CallManager Release 5.0 installation framework.
Installation Disc
One installation DVD includes:
Operating system Cisco Unified CallManager Release 5.0(x) application Hardware configuration is part of install process
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Only installation from DVD media is supported. The installation DVD or downloaded ISO file for Cisco Unified CallManager Release 5.0 includes the Linux operating system, the Cisco Unified CallManager Release 5.0 application, and the hardware configuration and recovery utility. There is no longer a separate Hardware Detection and Recovery disc. The installation DVD performs a hardware check to verify hardware requirements for the release. If any unsupported component is found, an applicable error message will be displayed and the system will be halted.
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Installation Methods
Two methods exist for installing Cisco Unified CallManager Release 5.0 and the operating system on which it runs.
Installation Methods
Full Install The operating system and Cisco Unified CallManager application are installed from scratch. Configuration information is prompted immediately before the operating system and application installation. Factory Install The operating system and Cisco Unified CallManager application are preloaded at the Cisco factory and shipped. Configuration is done when the system is first powered on.
CIPT1 v5.01-3
The first method is a full installation in which the customer inserts a DVD and loads the operating system and Cisco Unified CallManager Release 5.0 application. This method is primarily for customers who have an existing media convergence server (MCS) or equivalent platform. The second method is a factory installation in which the customer orders an MCS server platform, and the operating system and Cisco Unified CallManager Release 5.0 application are preloaded at the factory and shipped to the customer. This method is primarily for customers who order a new MCS platform.
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Installation Options
When you insert the installation DVD, you are presented with three installation options.
Installation Options
Three Installation Options
Option Basic Install Upgrade During Install Windows Upgrade Description Install operating system and Cisco Unified CallManager application software Basic install + install service releases and patches Upgrade from Release 4.0(x) and import database information from a Cisco Unified CallManager Release 4.x system using a file produced by the Data Migration Assistant tool
CIPT1 v5.01-4
The installation options are as follows: Basic Install: This option represents the basic installation and does not use any imported data. This type of installation generally starts by booting a system from an installation DVD or powering up a new system from the factory. Upgrade During Install: This option performs a basic installation on a system and also allows the system to be upgraded to a specific service release patch level before the completion of the basic installation. Selection of Upgrade during Install for a full installation will effectively perform a factory installation before prompting the installer for additional upgrade information. Windows Upgrade: This option upgrades a Windows-based Cisco Unified CallManager system to a Linux system and migrates data from an existing Windows-based Cisco Unified CallManager system. This installation method can be done on the same machine or a different machine from the Windows-based Cisco Unified CallManager machine. The Windows migration file can be saved to a variety of locations, including a remote hard drive or tape system. The CallManager then uploads the file from one of these locations during the upgrade process.
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System Requirements
This topic describes the hardware, browser, and software requirements for installing or upgrading to Cisco Unified CallManager Release 5.0.
System Requirements
Description Cisco MCS Platform Requirements Browser Requirements Cisco MCS servers with a -EVV or -IPC suffix Microsoft Internet Explorer Version 5.5 or later Netscape Navigator Version 7.1 or later
Windows Upgrade Cisco Unified CallManager Release 4.0 or 4.1 Software Requirements
CIPT1 v5.01-5
The figure identifies the system requirements for installing or upgrading to Cisco Unified CallManager Release 5.0, as follows: MCS Platform Requirements: Cisco Unified CallManager Release 5.0 only supports MCS servers with an -EVV or -IPC suffix. It does not support servers with ECS appended to the end of the model number. You can find specific MCS-supported hardware platforms and their Cisco-approved HP or IBM equivalents on http://www.cisco.com. Browser Requirements: You can access Cisco Unified CallManager Administration, Cisco Unified CallManager Serviceability, and Cisco IPT Administration by using the following browsers: Microsoft Internet Explorer Version 5.5 or later Netscape Navigator Version 7.1 or later
Windows Upgrade Software Requirements: Cisco Unified CallManager Release 4.0 or 4.1 are required to upgrade to Cisco Unified CallManager Release 5.0.
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NTP Strongly Recommended

Access to an external Network Time Protocol (NTP) clock source is also strongly recommended. The NTP provides a mechanism to synchronize time on computers across a network. The specification for NTP Version 3 is defined in RFC 1305. NTP uses Coordinated Universal Time (UTC) as the official standard for the current time.
NTP Strongly Recommended

Master Reference Clock First Node
Subsequent Nodes The first node synchronizes to the external master clock source. The subsequent nodes get their clock source from the first node.
When you install Cisco Unified CallManager Release 5.0 on the first node, you are strongly encouraged to enable NTP and reference an external NTP server to keep the system time accurate throughout the cluster. You can enter the IP address of up to five external NTP servers. If you are in an isolated environment such as a lab, it is acceptable to use the internal clock source of the first node as the master clock. In either case, subsequent nodes reference the first node for their time.
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Hardware Requirements
The figure shows the minimum requirements of the MCS or equivalent platform to run Cisco Unified CallManager Release 5.0.
Hardware Requirements
Minimum Requirements
Processor running at 2.0 GHz or greater Physical memory size must be 2 GB or greater Hard disk size must be 72 GB or larger
MCS 7815
MCS 7825
MCS 7835
MCS 7845
All Cisco MCS 7845 with more than 2 GHz of processor speed have 72-GB hard drive. New Cisco MCS 7835 has 72-GB hard drive and 2-GB memory. Older MCS 7835 have 1-GB memory and 36-GB hard drives and need to be upgraded. Cisco MCS 7815 and Cisco MCS 7825 need to be appropriately upgraded.
These servers support Cisco Unified CallManager Release 5.0 without any changes: MCS-7815I-I1-IPC3 MCS-7825-I1-IPC1 MCS-7825-H1-IPC1 MCS-7835-I1-IPC1 MCS-7835-H1-IPC1 MCS-7845-I1-IPC1 MCS-7845-H1-IPC1 MCS-7845H-2.4-EVV1 MCS-7845H-3.0-IPC1 To support Cisco Unified CallManager Release 5.0, these servers require additional memory: MCS-7815I-3000 MCS-7815I-I1-IPC1 MCS-7825I-3000 MCS-7825I-3.0-IPC1
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To support Cisco Unified CallManager Release 5.0, these servers must be upgraded to have at least a 72-GB hard drive and may also require additional memory: MCS-7815I-2000 MCS-7825H-2266 MCS-7825H-3000 MCS-7825H-2.2-EVV1 MCS-7835H-2.4-EVV1 MCS-7835I-2.4-EVV1 MCS-7835H-3.0-IPC1
Reasons for Hard Drive Space Expansion

The platform on which Cisco Unified CallManager runs requires 72 GB of hard disk space. This is an increase over the requirements for Cisco Unified CallManager Release 4.x.
Reasons for Hard Drive Space Expansion

Cisco Unified CallManager Release 5.0 is an appliance system.
Users do not have access to the operating system as with previous versions of Cisco Unified CallManager and cannot delete files if the hard drive is full.
Cisco Unified CallManager Release 5.0 installation partitions the hard drive for different file systems.
Log files may use the partition space needed for other components.
Cisco Unified CallManager Release 5.0 allows version switching.

Requires enough space for two Cisco Unified CallManager databases and two CDR databases
Recommendation:
Logs should periodically be offloaded to external servers because appliance systems do not have as much space for storing log files as Windows systems.
The figure shows why the hard disk space requirement has increased. Version switching is explained in more detail on the next page. Due to limited space, Cisco recommends that you should periodically offload log files to external servers. Appliance systems do not have as much space for storing log files as Windows systems.
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Dual Partitions Enable Version Switching

One reason for the increase in hard disk space requirements with Cisco Unified CallManager Release 5.0 is because this release allows version switching, or the ability to switch between two versions of the Cisco Unified CallManager application. In addition, you can install upgrade software on your server while the system continues to operate.
Dual Partitions Enable Version Switching

Dual partitions enable continued operation when you upgrade software. Upgrade software installs on the inactive partition. Activate the inactive partition and reboot system with new upgrade software. Current active partition becomes inactive and retains current old software until next upgrade. System always maintains two versions of software (does not apply to 4.x to 5.x upgrade).
Active Partition Inactive Partition 5.0(1) 5.0(2)
5.0(2) 5.0(1)
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Version switching is possible because two partitions exist on the system: an active, bootable partition and an inactive, bootable partition. The system boots up and operates entirely on the partition that is marked as the active partition. When you install upgrade software, you install the software on the inactive partition. The system continues to function normally while you install the software. When you are ready, you activate the inactive partition and reboot the system with the new upgrade software. The current active partition will then get identified as the inactive partition when the system restarts. The current (previous version) software remains in the inactive partition until the next upgrade. As a result, the system always maintains two complete versions of the software, including two Cisco Unified CallManager databases and 2 Call Detail Record (CDR) databases. If you encounter any problems with the new upgrade, you can always revert to the latest installed version. This applies only to Release 5.x and later releases of Cisco Unified CallManager software. The Release 5.x upgrade completely wipes out the hard disk. The initial installation of Release 5.x occupies one partition, and the next upgrade occupies the second partition.
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Preinstallation Tasks
This topic describes the tasks that must be completed before installing Cisco Unified CallManager Release 5.0.
Preinstallation Checklist
1. Verify that you meet the system requirements. 2. Gather required configuration information on the first and subsequent nodes. 3. Perform the recommended backup procedures on the Release 4.x first-node server. 4. Familiarize yourself with the navigation options within the installation wizards. 5. Configure the hardwareinstallation disc automatically handles the configuration.
CIPT1 v5.01-10
Before you install Cisco Unified CallManager Release 5.0, perform these preinstallation tasks as shown in the figure: Verify that you meet the software, browser, and hardware requirements. Gather required configuration information on the first and subsequent nodes. This configuration information is covered later in this topic. Perform the recommended backup procedures on the first-node server. Back up every database that is associated with your Cisco Unified CallManager server. Refer to Cisco IP Telephony Backup and Restore System (BARS) Administration Guide. Familiarize yourself with the navigation options within the installation wizards. These navigation options are covered later in this topic. Configure the hardware. You should upgrade the hard disks and install additional memory as required before you start the Cisco Unified CallManager Release 5.0 installation. The Cisco Unified CallManager Release 5.0 installation disc automatically configures the correct Redundant Array of Independent Disks (RAID) and BIOS settings.
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Important Notes
Before you proceed with the installation, consider the recommendations presented in the figure.
Important Notes
Designate the first-node server and subsequent nodes in the Cisco Unified CallManager cluster. Install the first node of the cluster before you install and configure subsequent nodes. If you use DHCP, you must reserve an IP address for each Cisco Unified CallManager server in the DHCP server scope. Install the same version of Cisco Unified CallManager on all the servers in a cluster. Cisco Unified CallManager Release 5.0(1) does not support third-party access to the CDR data. You must use the Cisco Unified CallManager CAR tool to access CDR data.
CIPT1 v5.01-11
The system can upload and process only software that Cisco Systems approves. You cannot install or use third-party or Windows-based software applications with Cisco Unified CallManager Release 5.0 that you may have been using with a previous version of Cisco Unified CallManager. If you choose to use Dynamic Host Configuration Protocol (DHCP), you must reserve an IP address for each Cisco Unified CallManager server in the DHCP server scope. If you do not reserve IP addresses through the DHCP server scope, the DHCP server may assign a different address to the Cisco Unified CallManager server if the server is disconnected from, and then reconnected to, the network. This change in IP address can render Cisco Unified CallManager inoperable.
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Configuration Data
The figure shows the configuration information that you will need to complete installation of the Cisco Unified CallManager Release 5.0.
Configuration Data
Configuration
DHCP Host Name IP Address
Able to Change Entry After Installation

YesCLI > set network dhcp No
YesUse Platform Administration > Settings > IP > Ethernet or CLI > set network ip addr YesUse Platform Administration > IP Mask Settings > IP > Ethernet or CLI > set network ip mask YesUse Platform Administration > Gateway Address Settings > IP > Ethernet or CLI > set network gateway
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Gather this information for first and subsequent nodes before starting the installation. Some settings cannot be changed after the installation wizard completes. Other parameters can be changed after installation is complete, either from the Cisco IP Telephony Platform Administration pages or from the command-line interface (CLI) via set commands. The Cisco IP Telephony Platform Administration page is a GUI application introduced in Cisco Unified CallManager Release 5.0 to manage the IP telephony platform. Configuration worksheets are available for you to view or print out in the Installing Cisco Unified CallManager Release 5.0 documentation.
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Configuration Data (Cont.)

Configuration Data
DNS Enable DNS Primary DNS Secondary Time Zone Domain Master Administrator ID Password

No YesCLI > set network dns primary YesCLI > set network dns secondary YesCLI > set timezone YesCLI > set network domain No, but you can add new accounts with CLI > set account YesCLI > set password admin
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The figure shows additional configuration information that you will need to complete the Cisco Unified CallManager Release 5.0 installation.
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Configuration Data (Cont.)

Configuration Data
SMTP NTP Server IP Address Security Password Organization Unit Location State Country Application User Password

YesCLI > set smtp host YesUse Platform Administration > Settings > NTP Servers YesCLI > set password security Yes to allCLI > set web-security
YesCLI > set password
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The figure shows additional configuration information that you will need to complete the Cisco Unified CallManager Release 5.0 installation. The application user password is the default password that will be used by all application users. CCMAdministrator is the default username for accessing Cisco Unified CallManager Administration and Cisco Unified CallManager Serviceability. When accessing Cisco Unified CallManager Administration or Cisco Unified CallManager Serviceability for the first time, you have to log on using username CCMAdministrator and the application user password that you entered during the installation. You can change application user passwords individually from Cisco Unified CallManager Administration, including the CCMAdministrator password.
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Navigating Within the Installation Wizard

An installation wizard queries the installer user for platform-specific information and Cisco Unified CallManager-specific configuration information.
Navigating Within the Installation Wizard

Key Combination Tab Alt-Tab Spacebar Up or down arrow Space bar to choose Back (when available) Space bar to choose Help (when available)
Action Move to the next field Move to the previous field Choose an option Scroll up or down in a list Go to the previous window Get help information on a window
CIPT1 v5.01-15
The table in the figure provides instructions on how to navigate within the installation wizard.
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Hardware Configuration
Hardware configuration is integrated within the Cisco Unified CallManager installation process.
Hardware Configuration
Hardware configuration is integrated with the Cisco Unified CallManager installation process. The installation disc automatically configures the correct BIOS and RAID settings. The installation process checks for correct hardware configuration, unsupported platforms, and minimum hardware requirements.
CIPT1 v5.01-16
The system installer automatically detects and configures the correct BIOS and RAID settings. The installation process also checks for unsupported platforms and checks that the minimum hardware requirements are met. Refer to Installing Cisco Unified CallManagerConfiguring the Hardware section for the release you are installing for BIOS and RAID configuration settings. The URL for the Cisco Unified CallManager Release 5.0(2) is as follows: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/install/instcall/cmins502. htm#wp59573
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Performing a Basic Installation: First Node

This topic describes the process for performing a basic installation of the operating system and Cisco Unified CallManager Release 5.0 application on the first node.
Starting the Installation

1. Boot the server with the installation DVD. 2. Verify the checksum for the DVD against the checksum shown on Cisco.com. 3. The Overwrite Hard Drive window indicates the software version on your hard drive and the DVD (if a previous installation). If you choose Yes, all existing data on your hard drive gets overwritten and destroyed.
You must install Cisco Unified CallManager Release 5.0 on the first node before installing it on any subsequent nodes. Installation starts the same way for all three installation options: Insert the installation disc in the DVD drive and reboot the server. As mentioned, a hardware check is performed to determine if the correct hardware is installed, and then the RAID and BIOS settings are configured. After the hardware is configured, the installation process performs a media check of the image on the DVD to ensure that the image is error-free before installation. Verify that the checksum displayed on the Media Check matches the checksum for the release on Cisco.com. If the media fails the Media Check, either download another copy from Cisco.com or obtain another disc directly from Cisco Systems. If the result is pass, you can continue with the installation. The Overwrite Hard Drive window indicates the software version on your hard drive and the version on the DVD. Choosing No here halts the installation. Choosing Yes overwrites the hard drive. This message is only seen when there has not been a previous version of software installed on the hard drive.
Caution If you choose Yes on the Overwrite Hard Drive window, all existing data on your hard drive is overwritten and destroyed.
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Basic Installation (Platform Configuration)

After the hard drive is formatted, the Platform Installation Wizard window displays.
Basic Installation (Platform Configuration)

Platform Installation Wizard screen:
CIPT1 v5.01-18
The Platform Installation Wizard guides you through the installation process and gathers the required information. Review this window to familiarize yourself with navigating within the Platform Installation Wizard, and follow these guidelines: If you choose Proceed, the Product Installation Configuration window displays immediately before any software is copied or installed. If you choose Skip, the software is first transferred to your hard drive and the Product Installation Configuration window displays later. After the software is transferred to the hard drive, the system shuts down. At the next bootup, the system will display the Installation Configuration window. This is the same state that you find on a factoryinstalled system, in which the software is preloaded but no configuration has been done. When the preloaded system boots up, the configuration dialog is completely skipped if a Universal Serial Bus (USB) drive with a configuration file that includes all configuration parameters is found.
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Basic Installation (Product Installation)

The Product Installation Configuration window enables you to choose from one of the three installation types.
Basic Installation (Product Installation)

Choose Basic Install for the installation type:
CIPT1 v5.01-19
If you want to install the software version that is on your DVD, choose Basic Install. If you want to upgrade to a later service release of the software during installation, choose Upgrade During Install. The next several pages show a sample installation sequence when the Basic Install option is chosen. The Upgrade During Install option is covered later.
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Basic Installation (Timezone)

After selecting the Basic Install option, the Timezone Configuration window is displayed.
Basic Installation (Timezone)

Choose timezone:
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At the Timezone Configuration window, choose the appropriate time zone for the server, and then choose OK.
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Basic Installation (Auto Negotiation)

The next window is the Auto Negotiation Configuration window.
Basic Installation (Auto Negotiation)

If
Set speed and duplex to highest speed supported on the switch and Cisco Unified CallManager server
At the Auto Negotiation Configuration window, choose Yes to enable the server to automatically negotiate the Ethernet network interface card (NIC) speed and duplex settings with the connected switch port, or choose No to statically configure speed and duplex settings. As with Cisco Unified CallManager Release 4.x, set the NIC to the highest speed supported on both the server and the switch and set the duplex to Full. Auto-detect can cause signaling delays, poor music on hold (MOH) quality, and poor quality on software-based conferences. Also, set the attached switch port to match the server speed and duplex to avoid speed and duplex mismatches.
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Basic Installation (DHCP)

Next, the DHCP Configuration window displays.
Basic Installation (DHCP)

If
Starts network configuration
CIPT1 v5.01-22
It is recommended that you either statically enter IP information or reserve an IP address in the DHCP scope specifically for the Cisco Unified CallManager. Choosing No disables DHCP and opens the Static Network Configuration page, where you enter these items: Host Name: A host name represents an alias that is assigned to an IP address to identify it. Enter a host name that is unique to your network. The host name can be up to 64 characters and can contain alphanumeric characters and hyphens. If DHCP is set to no, this is a mandatory field. IP Address: Enter the unique IP address of the machine you are configuring. If DHCP is set to no, this is a mandatory field. IP Mask: Enter the subnet mask. If DHCP is set to no, this is a mandatory field. Gateway Address: Enter the unique IP address of the default gateway that will forward packets to other networks. If you do not have a gateway, you must still fill in this field by setting it to 255.255.255.255. Not having a gateway may limit you to only being able to communicate with devices on your subnet.
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Basic Installation (DNS)

Next, you have to configure Domain Name System (DNS) settings.
Basic Installation (DNS)

If
CIPT1 v5.01-23
A DNS server represents a device that resolves a host name into an IP address or an IP address into a host name. If you do not have a DNS server, enter No. If DNS is not enabled, you should only enter IP addresses (not host names) for all network devices in your Cisco IP telephony network.
Note Using DNS is not recommended.
Field definitions for DNS settings are as follows: Primary DNS: Cisco Unified CallManager contacts this DNS server first when attempting to resolve host names. Enter the IP address of the DNS server that you want to specify as the primary DNS server. If DNS is set to yes, this is a mandatory field. Secondary DNS: When a primary DNS server fails, Cisco Unified CallManager will attempt to connect to the secondary DNS server. In this optional field, enter the IP address of the secondary DNS. Domain: This field represents the name of the domain in which this machine is located. If DNS is set to yes, this is a mandatory field.
Note DNS activation status is set at installation. After installation, you cannot change the activation status of DNS; however, you can change the DNS addresses. To change DNS activation status, you must reinstall Cisco Unified CallManager.
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Basic Installation (Account Login)

The next window is the Administrator Login Configuration window.
Basic Installation (Account Login)

Used for accessing Platform Administration, Disaster Recovery, and CLI
Entries are case-sensitive. Strong passwords are enforced.

The Administrator ID field specifies the name that you want to assign to the Administrator account that is used for accessing Platform Administration, Disaster Recovery, and the CLI. Ensure that the name is unique; it can contain lowercase, alphanumeric characters, hyphens, and underscores. It must start with a lowercase alphanumeric character. All entries are casesensitive. You should record this mandatory field for use when logging in to the CLI on the platform or into the Cisco IPT Platform Administration.
Note You cannot change the Administrator ID field after installation.
The Password field specifies the password that you use for logging into the CLI on the platform and for logging into Cisco IPT Platform Administration. Ensure that the password is at least six characters long; it can contain alphanumeric characters, hyphens, and underscores. Strong passwords are enforced. For example, the password cannot contain any dictionary words or an abundance of repeated characters. The password cisco will be rejected due to a violation of a dictionary word rule. The password cisc_ccm1 will be rejected due a violation of the too-many-repeated-characters (c) rule. You can change the password for this account after installation via the CLI using the set password admin command.
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Basic Installation (First Node)

Next, you are asked whether the server is the first node or a subsequent node, in the Cisco CallManager Cluster Configuration window.
Basic Installation (First Node)

Designates the node as the first node or subsequent node
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To configure the server as the first Cisco Unified CallManager node, choose Yes. To configure the server as a subsequent node in the cluster, choose No.
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Basic Installation (NTP External Source)

The Network Time Protocol Client Configuration window displays.
Basic Installation (NTP External Source)

If
NTP configuration only displays on the first node.
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To configure this node as a client of another NTP server, choose Yes. When the Network Time Protocol Client Configuration window displays, enter the IP address for the external NTP servers you want to use.
Note When you install Cisco Unified CallManager Release 5.0 on the first node in a production environment, you are strongly encouraged to enable NTP and reference an external NTP server to keep the system time accurate throughout the cluster.
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Basic Installation (NTP Internal Source)

Another option at the Network Time Protocol Client Configuration window is to click No.
Basic Installation (NTP Internal Source)

If
CIPT1 v5.01-27
In this case, set the first node to use its internal hardware clock and serve subsequent nodes in the network. Accept the values displayed on the Hardware Clock Configuration window or set the date and local time for the time zone that you choose. Enter the hours based on a 24-hour format. Again, this configuration is not recommended in a production network connected to the Internet but is acceptable in a predeployment lab environment.
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Basic Installation (Database Access Security)

The Database Access Security Configuration window displays.
Basic Installation (Database Access Security)

Sets the password that the databases use throughout the cluster to communicate. Must be the same on all nodes in the cluster.
CIPT1 v5.01-28
The Database Access Security Configuration window requires you to enter the database access security password. The database access security password must be the same on all nodes in the cluster. This password will be used when subsequent nodes authenticate themselves to the first node.
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Basic Installation (SMTP Host)

Next, you will see the SMTP Host Configuration window. You will be prompted if you want to configure Simple Mail Transfer Protocol (SMTP) on the server. If you want to use SMTP, click Yes.
Basic Installation (SMTP Host)

If
Enter the host name or IP address for the SMTP server used for outbound e-mail.
CIPT1 v5.01-29
To configure an SMTP host that is used for outbound e-mail, enter the host name or IP address for the SMTP server. The host name contains alphanumeric characters, hyphens, or periods and must start with an alphanumeric character. You must fill in this field if you plan to use electronic notification. If not, you can leave it blank.
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Basic Installation (Platform Configuration Confirmation)

Finally, you will see a window that signals the end of the Platform Configuration Wizard.
Basic Installation (Platform Configuration Confirmation)

End of the Platform Configuration Wizard. After you select OK, you cannot return to this set of windows without a reinstall.
CIPT1 v5.01-30
If you choose OK, you cannot access any of the previous windows without a reinstall. If uncertain about any of your configurations, you can choose Back to review any configuration window.
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Basic Installation (Application User)

Now the Application User Password window will display.
Basic Installation (Application User)

Sets the CCMAdministrator password used when logging into Cisco CallManager Administration and Serviceability GUIs. Also sets all other predefined application user passwords.
CIPT1 v5.01-31
In the Application User Password Configuration window, you can set the password for logging in to Cisco Unified CallManager Administration with the Cisco Unified CallManagerAdministrator username, and for logging in to other predefined applications as these users: CallManagerSysUser WDSysUser CallManagerQRTSysUser IPMASysUser WDSecureSysUser CallManagerQRTSecureSysUser IPMASecureSysUser You can change the application user password for each application through Cisco Unified CallManager Administration after the installation is complete.
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Basic Installation (Installation Complete)

This window shows a successful Cisco Unified CallManager Release 5.0 installation.
Basic Installation (Installation Complete)
This screen signals a successful Cisco Unified CallManager Release 5.0 installation. Log in if desired with Administrator login ID and password to access CLI commands.
From this window, you can log in to the CLI using the Administrator ID and password that was defined at installation (not the CCMAdministrator password used for accessing Cisco Unified CallManager Administration). From the CLI, you can perform a number of basic platform functions. The Cisco IP telephony Platform Administration GUI application also makes these functions available. Typically you would use the CLI only when a problem occurs while you are using the Cisco IP Telephony Platform Administration interface. CLI commands and the Cisco IP Telephony Platform Administration GUI application are covered later.
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Logon Window for Platform Administration

You should now be able to use the GUI to access Cisco Unified CallManager Administration pages.
Logon Window for Platform Administration
http://server-name/iptplatform, where server-name is the name or IP address of the Cisco Unified CallManager. You can now log on to Cisco Unified CallManager Administration, Serviceability, and Cisco IP Telephony Platform Administration.
CIPT1 v5.01-33
To log in to the Platform Administration and Disaster Recovery pages, use the Administrator ID and password. To log in to the Cisco Unified CallManager Administration and Cisco Unified CallManager Serviceability pages, use the Cisco Unified CallManager Administrator username and password.
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Interface Access and Log On

You can also log on to Cisco Unified CallManager using a CLI.
Interface Access and Logon

login as: administrator administrator@192.168.1.10's password: Welcome to the Platform Command Line Interface (version 1.1) admin: show* set* delete* unset* file* utils* run* admin:
Administrator = CLI, Cisco IP Telephony Platform Administration, and Disaster Recovery System
CCMAdministrator = Cisco Unified CallManager Administration and Serviceability Web GUI

The CLI is used for platform administration using a platform administration account. Platform account usernames and passwords are stored on the Linux system in the file system. The other key interface is the web GUI. The web GUI is used to perform all configuration tasks in the Cisco Unified CallManager Administration and Serviceability interfaces. CCMAdministrator accounts are stored in the database.
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Performing a Basic Installation: Subsequent Nodes

This topic describes the process for performing a basic install on a subsequent node. Most of the steps for configuring the subsequent node are the same as those for configuring the first node. Therefore, only windows with configuration differences are shown. Choose No when the First Node Configuration window appears to configure this node as the subsequent node.
Basic Installation (Subsequent Node)

If
Choose No to configure this node as a subsequent node.
The first node should be configured before any subsequent nodes.
CIPT1 v5.01-35
The first node must be configured before any subsequent nodes, be operational, and must be accessible to the subsequent nodes. Also, the same version of Cisco Unified CallManager must be installed on the first node and subsequent nodes, and the subsequent nodes must have connectivity to the first node. When you perform a fresh installation of Cisco Unified CallManager, you must define any subsequent servers (nodes) in the Cisco Unified CallManager Administration Server Configuration window before you can install the Cisco Unified CallManager software on each subsequent server.
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Basic Installation (Access to First Node)

The First Node Access Configuration window allows you to enter connectivity information to reach the first node.
Basic Installation (Access to First Node)

Enter connectivity information to reach the first node. The Security Password field is the database access security password, and must match the first node password.
CIPT1 v5.01-36
Enter connectivity information to reach the first node as follows: Host Name: Enter the host name of the first Cisco Unified CallManager node in the cluster. IP Address: Enter the IP address of the first Cisco Unified CallManager node in the cluster. Security Password: Enter the security password that you entered for the first Cisco Unified CallManager node. You must enter the same security password for each server in the cluster.
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Basic Installation (NTP)

The NTP configuration is done automatically on subsequent nodes.
Basic Installation (NTP)

On subsequent nodes, the NTP client automatically points to the first nodes time.
CIPT1 v5.01-37
Subsequent nodes are automatically configured as an NTP client so that they set their hardware clock to the time of the first node.
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Performing an Upgrade During Installation

This topic describes the process for installing service releases and patches while performing a basic installation.
Upgrade During Installation

Upgrade during Install option
Performs a basic install plus allows the system to be upgraded to a specific patch level Obtains upgrade patch from a remote source or local DVD source Installs multiple patches if desired
CIPT1 v5.01-38
Upgrade during Install is a method of performing a basic install on a system but allowing the system to be upgraded to a specific patch level before completion of the basic installation. The Upgrade during Install option is useful for when you have a base Cisco Unified CallManager Release 5.0 installed on a DVD, and then go to a Cisco.com server or DVD to upgrade on top of the partial install with the actual desired release. For example, you have Release 5.0(1) on your system and you want to apply the upgrade patch to Release 5.0(2). This option performs a basic install of the operating system and application, and then prompts you to enter the location of the patch file, either a remote source via FTP or Secure FTP (SFTP), or from a local DVD drive. You can install one patch using this installation option or multiple patches. When you choose the Upgrade during Install option, the install process starts with a basic install, performing a media check followed by a prompt to overwrite the hard drive, because, as with other install options, the hard drive is reformatted before installing the image. As before, the platform installation completes, followed by the application installation. After the image is successfully installed, the Platform Configuration Wizard window displays. Proceed and choose the Upgrade during Install installation option again. Next, you will be prompted for the platform configuration information, similar to the Basic Install.
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Upgrade During Installation (Retrieval Mechanism)

The Install Upgrade Retrieval Mechanism Configuration window displays.
Upgrade During Installation (Retrieval Mechanism)

Choose the mechanism for retrieving the upgrade file:
SFTP FTP Local
The next windows show the procedure for configuring the network settings before you can retrieve the file for upgrading from a remote source.
Choose the upgrade retrieval mechanism that you want to use to retrieve the upgrade file: SFTP: Retrieves the upgrade file from a remote server using the Secure FTP FTP: Retrieves the upgrade file from a remote server using FTP Local: Retrieves the upgrade file from a local CD or DVD
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Upgrade During Installation (Remote Source - DHCP)

If you choose to upgrade through an FTP or SFTP connection to a remote server, you must first configure the network settings.
Upgrade During Installation (Remote Source - DHCP)

If you choose SFTP or FTP, the DHCP Configuration window displays.
CIPT1 v5.01-40
Configuring the network settings for an upgrade during install is the same procedure as for a basic install, as follows: The Auto Negotiation Configuration window allows you to automatically set the speed and duplex settings of the Ethernet NIC using automatic negotiation. The DHCP Configuration window enables you to choose to either set up static network IP addresses for the node and gateway or to use DHCP. The DNS Client Configuration window allows you to choose to enable DNS and enter DNS client information.
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Upgrade During Installation (Remote Patch Access Information)

After the system configures the network interface and checks for connectivity, the Remote Patch Configuration window displays.
Upgrade During Installation (Remote Patch Access Information)

Enter the location and login information for the remote file server.
CIPT1 v5.01-41
To retrieve the remote patch, enter the location and login information for the remote file server. After restarting the network stack, the system connects to the remote server and retrieves a list of available upgrade patches.
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Upgrade During Installation (Remote Patch Selection)

The Install Upgrade Patch Selection window displays.
Upgrade During Installation (Remote Patch Selection)

Choose the upgrade patch you want to download.
CIPT1 v5.01-42
Choose the upgrade patch you want to download. The system downloads, unpacks, and installs the patch and then restarts the system running the upgraded software version. After the system restarts, the Preexisting Configuration Information window displays. If you have preexisting configuration information stored on a floppy disc or a USB key, or both, insert the disc or the USB key now and choose Continue. The installation wizard will read the configuration information during the installation process.
Note Preexisting configuration information includes files created by the Standalone Configuration Wizard. The wizard speeds up the installation by creating a configuration file with all necessary parameters in the correct XML format. The file is then typically stored on a diskette or a USB memory key. Standalone Configuration Wizard runs only on Linux.
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Upgrade During Installation (Local Source)

If you want to install the patch from a local drive, you must download the appropriate patch file from Cisco.com and copy the file to a CD or DVD. Then choose Local in the Install Upgrade Retrieval Mechanism Configuration window and click OK.
Upgrade During Installation (Local Source)

The next windows show the procedure for upgrading from a local source.
CIPT1 v5.01-43
Because of the size of the patch files, you will need to copy it to a DVD in most cases. The patch-file name has the following format:
cisco-ipt-k9-patchX.X.X.X-X.tar.gz
where X.X.X.X-X represents the release and build number. Do not untar or unzip the patch file before you install it because the system will not recognize it as a valid file. The system automatically unzips the TAR file. Do not rename the TAR file because the system looks for a file with an extension .tar or .gz.
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Upgrade During Installation (Patch Location)

The Local Patch Configuration window displays. Enter the patch directory and patch name, if required, and choose OK.
Upgrade During Installation (Patch Location)

Enter the patch directory and patch name (optional) of the local disc that contains the upgrade patch(es).
CIPT1 v5.01-44
You only need to enter the patch directory when the patch is not stored in the root directory of the CD or DVD. The system searches through the specified directory for a valid archive.
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Examining the Installation Log Files

This topic describes how to view and retrieve the installation log files.
Examining the Installation Log Files

You can obtain install logs via CLI or RTMT*.
CLI Command file list install file view install <log file name> Description Obtain a list of install log files View the log file
Install problems can generally be addressed by the install logs and the security logs. /common/log/install Contains the install.log files /root/.security/logs Contains log files generated as the result of certificate and key generation
*RTMT=Cisco Unified CallManager Real-Time Monitoring Tool
If you encounter problems with the installation, you can obtain and examine the install log files by entering the following commands in the CLI: file list install file view install <log file name> You can also view logs using the Cisco Unified CallManager Real-Time Monitoring Tool (RTMT). For more information on using and installing the Cisco Unified CallManager RTMT, refer to the Cisco Unified CallManager Serviceability Administration Guide, Release 5.0. RTMT is covered in part 2 of the CIPT training course.
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Perform Postinstallation Tasks

This topic describes tasks you should perform after the installation.
Postinstallation Tasks
1. Log in as the CCMAdministrator by using a web browser, and change the password. 2. Activate Cisco Unified CallManager services. 3. Configure the backup settings. 4. Add new locales. 5. Implement security. 6. Add additional Cisco Unified CallManager nodes to the cluster.
CIPT1 v5.01-46
After installing Cisco Unified CallManager on your server, you must set some configuration parameters for Cisco Unified CallManager and perform other postinstallation tasks before you can begin using it. Perform these tasks for the server that you install before other servers in the cluster are installed: Log in as the Cisco Unified CallManager Application User and change the Application User passwords. Activate Cisco Unified CallManager feature services that you want to run. Before you activate feature services, you must perform required preactivation tasks; for example, before you can activate a service on a server, you must add the server to Cisco Unified CallManager Administration (System > CallManager). Configure the backup settings. Remember to back up your Cisco Unified CallManager data daily. The locale, English_United_States, installs automatically on the server; however, you can add new locales to the server, if required. Cisco recommends that you implement authentication and encryption in your Cisco IP telephony network. If necessary, you can add subsequent Cisco Unified CallManager nodes to the cluster. See Installing Cisco Unified CallManager Release 5.0 for more detailed information on these postinstallation tasks.
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Summary
Summary
Cisco Unified CallManager Release 5.0 can be installed at the factory or on premises, with options to install from scratch or upgrade from Windows. The minimum requirements to install Cisco Unified CallManager Release 5.0 include (two) 72-GB hard drives, 2 GB RAM, and a 2-GHz processor. Before installing Cisco Unified CallManager Release 5.0, verify system requirements, gather required configuration, and back up the Release 4.x first node. Perform a basic installation to install the operating system and Cisco Unified CallManager application on the first node. Install subsequent nodes after the first node is configured. Choose Upgrade during Install to upgrade to a specific patch or service release. Finally, log in to Cisco Unified CallManager Administration, start services, and configure backup settings.
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Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Cisco Unified CallManager is the central component of the Cisco Unified Communications solution that covers the whole range of IP communications. There are four call-processing deployment models. These models differ based on whether call processing is centralized or distributed and whether the cluster is located within a single site or dispersed across multiple sites. Deploying Cisco Unified CallManager in a Cisco IP telephony solution entails selecting the Cisco MCS hardware platform, installing the Cisco Unified CallManager software, and performing postinstallation procedures.
CIPT1 v5.01-1
This module describes key factors to consider when you use a Cisco Unified CallManager server in an IP telephony deployment. The module describes the role that Cisco Unified CallManager plays in the overall Cisco Unified Communications solution, and the Cisco Unified CallManager hardware and software requirements. Also, the module describes the four call-processing deployment models and how Cisco Unified CallManager clusters provide redundancy and failover. Finally, the module describes the Cisco Unified CallManager installation and upgrade process, and postinstallation procedures.
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Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and solutions are found in the Module Self-Check Answer Key. Q1) Which two of these are layers of the Cisco Unified Communications solution? (Choose two.) (Source: Introducing Cisco Unified CallManager Release 5.0) A) B) C) D) E) Q2) applications presentation call processing media transport
Which three of these are Cisco Unified CallManager functions? (Choose three.) (Source: Introducing Cisco Unified CallManager Release 5.0) A) B) C) D) E) F) packet routing signaling and device control dial plan administration phone feature administration storing voice mails providing call center functionality
Q3)
Which three of these are advantages of the appliance operating system in Cisco Unified CallManager Release 5.0? (Choose three.) (Source: Introducing Cisco Unified CallManager Release 5.0) A) B) C) D) E) F) lower cost improved installation and upgrades easier administration increased security plug-and-play increased reliability
Q4)
Which database is used by Cisco Unified CallManager Release 5.0? (Source: Introducing Cisco Unified CallManager Release 5.0) A) B) C) D) E) Informix Dynamic Server Microsoft SQL 7 Microsoft SQL 2000 Oracle MSJET DB
Q5)
Which of these is not an example of server intracluster communication? (Source: Introducing Cisco Unified CallManager Release 5.0) A) B) C) D) SDL signaling database traffic licensing service music on hold media transfer
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Q6)
Which two of these applications are supported on Cisco Unified CallManager servers? (Choose two.) (Source: Introducing Cisco Unified CallManager Release 5.0) A) B) C) D) E) Directory Synchronization Microsoft Windows management center Disaster Recovery System DNS server IIS
Q7)
List the minimum hardware requirements for MCS platforms required by Cisco Unified CallManager Release 5.0. (Source: Introducing Cisco Unified CallManager Release 5.0)
Q8)
What is a licensing overdraft and by what percentage is it allowed? (Source: Introducing Cisco Unified CallManager Release 5.0)
Q9)
Which three of these are IP telephony deployment models supported by Cisco? (Choose three.) (Source: Evaluating Cisco Unified CallManager Release 5.0 Deployment Options) A) B) C) D) E) F) a single-site with one call-processing agent two clusters in active-backup mode multisites with centralized call processing two load-balancing clusters multisites, each with its own call-processing agent a single-cluster with distributed call processing
Q10)
Which codec is recommended in a single-site Cisco Unified CallManager deployment? (Source: Evaluating Cisco Unified CallManager Release 5.0 Deployment Options) A) B) C) D) G.721 G.711 G.723 G.729
Q11)
Which statement is true about a multisite WAN with centralized call-processing Cisco Unified CallManager deployment? (Source: Evaluating Cisco Unified CallManager Release 5.0 Deployment Options) A) B) C) D) The IP WAN carries voice traffic but no call-control signaling. The IP WAN is used for data only. The IP WAN carries voice traffic and call-control signaling. The IP WAN carries call-control signaling but no voice traffic.
Q12)
Which Cisco Unified CallManager deployment model offers the highest scalability? (Source: Evaluating Cisco Unified CallManager Release 5.0 Deployment Options) A) B) C) D) multisite WAN with centralized call processing multisite WAN with distributed call processing single-site with one call-processing agent single-cluster with distributed call processing
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Q13)
Which two of these are features of Cisco Unified CallManager clustering over the WAN? (Choose two.) (Source: Evaluating Cisco Unified CallManager Release 5.0 Deployment Options) A) B) C) D) E) feature extension to offices robustness in high-delay environments distributed administration unified dial-plan highest scalability
Q14)
A 1:1 redundancy design offers _____. (Source: Evaluating Cisco Unified CallManager Release 5.0 Deployment Options) A) B) C) D) maximum uptime; however, no more than a 40-ms round-trip delay can exist between servers increased redundancy; however, the increased server cost is often prohibitive some redundancy; however, a server reboot is required after an upgrade high availability; however, you may overwhelm the backup servers
Q15)
Which of these items is not an installation option of Cisco Unified CallManager Release 5.0? (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) basic install Windows upgrade appliance upgrade upgrade during install
Q16)
Which of these features is enabled by active and inactive partitions? (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) software upgrade with active call processing version switching reversal to a known state in case of failure offload of call processing
Q17)
Which three of these tasks are preinstallation tasks for Cisco Unified CallManager Release 5.0? (Choose three.) (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) E) F) Verify system requirements. Synchronize time in the network. Disconnect components that may fail. Gather required configuration information on the first and subsequent nodes. Perform the recommended backup procedures on all Release 4.x servers (publisher and subscribers) Run the Hardware Check Wizard.
Q18)
Which password is not set when doing a basic install on the first node? (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) E) cluster security passphrase administrator password database access security password application user password Cisco Unified CallManager CAR administrator password
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Q19)
Which password has to be provided when installing a subsequent node (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) E) cluster security passphrase administrator password database access security password application user password Cisco Unified CallManager CAR administrator password
Q20)
Which three of these options can be used to retrieve an upgrade file when performing an upgrade during install? (Choose three.) (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) E) F) SFTP HTTP SMTP TFTP Local FTP
Q21)
Which CLI command is used to list installation log files? (Source: Installing Cisco Unified CallManager Release 5.0)
Q22)
After installation of Cisco Unified CallManager, you have to _____ Cisco Unified CallManager services. (Source: Installing Cisco Unified CallManager Release 5.0) A) B) C) D) activate enable restart configure
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Module Self-Check Answer Key

Q1) Q2) Q3) Q4) Q5) Q6) Q7) Q8) Q9) Q10) Q11) Q12) Q13) Q14) Q15) Q16) Q17) Q18) Q19) Q20) Q21) Q22) A, C B, C, D B, D, F A D A, C 2-GHz processor, 2 GB RAM, and 72-GB hard disk An overdraft is a condition in which more phones register to Cisco Unified CallManager than license units purchased. Cisco Unified CallManager allows a five percent overdraft. A, C, E B C B A, D B C B A, D, E A A A, E, F File list install A
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Module 2

Overview
Performing system administration and configuring basic settings are the first important tasks when deploying Cisco Unified CallManager. Being familiar with the backup and restore procedures and doing regular backups is essential for disaster recovery. This module describes the different administration methods and features, provides information about how to access them, discusses the configuration of basic settings, and explains how to back up and restore the system.
Module Objectives
Upon completing this module, you will be able to perform platform and general administration of Cisco Unified CallManager Release 5.0, and implement disaster recovery. This ability includes being able to meet these objectives: Configure and manage the Cisco IP telephony platform Access Cisco Unified CallManager Release 5.0 administration and integrated applications, navigate the menu structure, and enable MLA to provide multiple levels of security to Cisco Unified CallManager administration Configure Cisco Unified CallManager Release 5.0 basic settings to activate services, and use the Cisco Unified CallManager IP address instead of its host name Back up and restore data for a Cisco Unified CallManager Release 5.0 cluster
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Lesson 1
Administering the Cisco IP Telephony Platform

Overview
On the Cisco Unified CallManager appliance, access to the local operating system is limited. Therefore, platform-related administration is integrated into the Cisco Unified CallManager Release 5.0 application. The integrated application that performs numerous platform tasks is called the Cisco IP Telephony Platform Administration. This lesson describes the various tasks that you can perform using Cisco IP Telephony Platform Administration.
Note In Cisco Unified CallManager Release 5.0(3), the Cisco IP Telephony Platform Administration name changed to Cisco Unified Communications Operating System Administration. Functionally, the two versions are equivalent.
Objectives
Upon completing this lesson, you will be able to manage the Cisco IP Telephony platform. This ability includes being able to meet these objectives: Identify the basic functions of the Cisco IP Telephony Platform Administration pages and how to access them Check the status of various platform components View and update IP, NTP, and SMTP settings Switch software versions and shut down the server Manage security certificates and display IPsec policies Use the Cisco IP Telephony Administration GUI to upgrade the software version running on the platform or add locales, dial plans, and TFTP server files Use the ping interface and set up a limited-time remote account that Cisco support personnel can use to access the system Use the CLI to perform basic platform functions
Cisco IP Telephony Platform Administration Overview

This topic describes the basic functions of Cisco IP telephony and how to access it.
Cisco IP Telephony Platform Administration Functions

Check software, hardware, CPU, and memory status Check and update IP addresses Ping other network devices Manage NTP servers Upgrade system software and options Switch versions, and restart the system
Platform administration functions can be accessed using GUI and CLI commands.
The Cisco IP Telephony Platform Administration is an intuitive application that allows you to configure and manage the Cisco IP telephony platform. The figure lists typical tasks that you can perform using the Platform Administration pages. You can access the Platform Administration interface in two waysthrough a web GUI or through a command-line interface (CLI). Most of your time will be spent on the GUI interface. Using the CLI interface to manage the platform provides a subset of the functions that the GUI provides. The primary use of the CLI interface is to enable you to get the GUI working again.
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Accessing the GUI

There are two ways to access the Cisco IP Telephony Platform Administration GUI.
Accessing the GUI

https://server-name/iptplatform
Username is administrative account as entered during install Password is the same as entered during install
Or access from Cisco CallManager Administration main page:
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You can use either of these options to access the Cisco IP Telephony Platform Administration GUI: From a web browser directly: Enter the URL: https://<server-name>/iptplatform From Cisco Unified CallManager Administration: Log in to Cisco Unified CallManager Administration. From the Navigation menu in the upper, right corner of the Cisco Unified CallManager Administration window, choose Platform Administration and click Go. In either case, enter Administrator for the username and enter the password that you assigned to the Administrator account at the time of install. Keep in mind that this is not the CCMAdministrator account password that is also set during install. You can assign the same password to the Administrator account and the CCMAdministrator account.
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GUIMain Page
The Cisco IP Telephony Platform Administration page has several menus as shown in the figure.
GUIMain Page
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Menu functions are explained in detail in subsequent pages.
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Expanded Menus Reference

Each menu has several menu items as shown in the figure.
Help Menu Also
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The most important menu items are discussed in detail in separate topics of this course.
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Checking Platform Status

This topic describes the basic functions of the Show menu to check the status of various platform components.
Show Menu
ClusterDisplays host name, IP address, alias name, and node function HardwareDisplays the model, number and type of processors, total memory LogsRefers administrator to install RTMT* NetworkDisplays DHCP, MAC address, NIC speed/duplex, IP address/mask SoftwareDisplays active and inactive software versions, and installed software options SystemDisplays host name, time zone, CPU utilization, memory usage, disk space
*RTMT=Cisco Unified CallManager Real-Time Monitoring Tool
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The Show menu enables you to view information about the cluster, hardware, network, software, and system.
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Showing Hardware, Software, and System Status

The figure shows sample screen output from Show Hardware (top left), Show Software (top right), and Show System (bottom).
Showing Hardware, Software, and System Status
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The Hardware Status screen shows that the hardware platform is a media convergence server (MCS) 7835H with Intel 3.06 Ghz CPU, and 2 GB of main memory. The Detailed Report window shows the serial number and HP product name. Although not shown, the Detailed Report window also shows the BIOS version and manufacturer, number of process slots and active processors, and a host of other platform-related settings. The Software Packages screen shows the Cisco Unified CallManager versions running on the active and inactive partitions, and any software options that are installed, such as locales. On the system in which this screen capture was taken, a full install of Cisco Unified CallManager Release 5.0(1) was performed, so no prior software release is maintained in the inactive partition. The System Status screen shows some parameters such as host name, date and time, and the time zone that were set at installation. It also shows important run-time statistics about the CPU utilization, memory capacity, and disk space consumption by partition.
Note The figure presents a subset of the functions available under the Show menu.
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Viewing and Updating Platform Settings

This topic describes the basic functions of the Settings menu to view and update IP, NTP, and SMTP settings.
Settings Menu
Display or change IP address of the system or the first node Add or delete an NTP server or change IP address or host name Modify SMTP host name or IP address Change the system date and time
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The Settings menu enables you to change settings related the IP network, Network Time Protocol (NTP) server, Simple Mail Transfer Protocol (SMTP) host, and the system date and time.
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Changing IP Settings
The figure shows sample screen output from Settings > IP > Ethernet (top) and Settings > IP > Publisher pages (bottom).
Changing IP Settings
Warns you to change the IP addresses in System > Server first Enable or disable DHCP, and change IP address, mask, and default gateway.
Change IP address of First node on Subsequent node (read only on First node).
From the Ethernet settings page, you can change the status of the Dynamic Host Configuration Protocol (DHCP) setting (from Disabled to Enabled, or vice versa), and modify the IP address, mask, and default gateway used by this system. A Warning message indicates that if you use the IP address (instead of the host name) in Cisco Unified CallManager Administration under the System > Servers menu, then you must change the IP address there first before you change it in Platform Administration. Otherwise, call processing will fail.
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Changing NTP, SMTP, and Time Settings

The figure shows screens for NTP Server Settings, SMTP Settings, and Time.
Changing NTP, SMTP, and Time Settings

Add or delete NTP servers for use by first node. Read-only for subscribers
SMTP host name or IP needed to receive e-mails from the system.
Modify system time on first node if not using an NTP server

The Settings > NTP Servers page allows you to specify NTP servers for use by the publisher. This setting is only valid for the publisher. For subscribers, this screen is read-only, because subscribers always get their time from the publisher. The Settings > SMTP page allows you to specify the IP address or host name of the SMTP server used for mail. This setting is used if you want the system to send you e-mail if certain situations occur, such as an expiring certificate. If a host is defined, a status message is available that will indicate if the specified SMTP host is reachable. The Settings > Time page is only valid for the publisher not using an NTP server. If NTP servers are configured for the publisher, this page will pop up a message stating Cannot set system time because NTP is currently configured. Remove NTP settings to allow access to this page. Again, this is because the subscribers always get their time from the publisher.
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Switching Versions and Restarting the Server

This topic describes the Restart menu to switch software versions and shut down the server.
Restart Menu
Restart the system on the current partition without switching versions. Shut down the system. Shut down the system that is running on the active disk partition, and then automatically restart the system using the software version on the inactive partition.
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The Restart menu enables you to restart the system on the current version in the active partition, shut down the system, or switch versions by restarting the system using the software version on the inactive partition.
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Restarting and Switching Software Versions

The figure shows sample screen output from Restart > Current Version (top) and Restart > Switch Versions (bottom) pages.
Restarting and Switching Software Versions

Restart current version or cancel to abort.
Switch software versions to boot under inactive partition.
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Because these screen captures were taken from a system with a fresh install and not an upgrade, the system warning indicates that there is not an inactive version on the inactive partition and therefore versions cannot be switched. You can also shut down the system without rebooting. When you choose the Restart > Shutdown menu, the Shutdown System page issues a warning message to ensure that you want to shut down. Press Shutdown to continue or Cancel to abort.
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Managing Security Certificates and IPsec

This topic describes the Security menu for managing security certificates and displaying IPsec policies.
Security Menu
Manage certificates, CTLs, and CSRs Display, upload, download, delete, and regenerate certificates. View and change certificate expiry monitoring and notification. Manage IPsec policies Display or update existing IPsec policies. Set up new IPsec policies and associations.
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The Security menu enables you to manage certificate operations and IPsec policies. With the Certificate Management menu, you can display, delete, and regenerate certificates, Certificate Trust Lists (CTLs), and Certificate Signing Requests (CSRs). You can download these to your PC or upload them from your PC. You can also view when the certificate is about to expire, and enable or disable e-mail notification. In public key infrastructure systems, a CSR is a message sent from an applicant to a certificate authority in order to apply for a digital identity certificate. Before creating a CSR, the applicant first generates a key pair, keeping the private key secret. The CSR contains information identifying the applicant (such as a directory name in the case of an X.509 certificate), and the public key chosen by the applicant. The corresponding private key is not included in the CSR, but is used to digitally sign the entire request. The CSR may be accompanied by other credentials or proofs of identity required by the certificate authority, and the certificate authority may contact the applicant for further information. If the request is successful, the certificate authority will send back a digitally signed identity certificate, signed with the private key of the certificate authority. The IPsec Management menu enables you to display or update existing IPsec policies and set up new policies. IPsec Management is covered in more detail in the Implementing Cisco Unified CallManager Part 2 (CIPT2) v5.0 training course.
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Display Certificate
From the Certificate Management menu, you can display your Own Certificates or Trust Certificates.
Display Certificate
Own Certificates are certificates that belong to Cisco Unified CallManager (Cisco Unified CallManager holds the private key for that certificate). Trust Certificates are certificates for other devices that Cisco Unified CallManager has in its certificate store, such as phone MICs.
Own Certificates are certificates that belong to Cisco Unified CallManager (Cisco Unified CallManager holds the private key for that certificate), such as Cisco Certificate Authority Proxy Function (CAPF) and CallManager service certificates. Trust Certificates are certificates for other devices that Cisco Unified CallManager has in its certificate store. Examples of such certificates are phone manufacturing installed certificates (MICs) or certificate authority-rooted certificates.
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Display Cisco Unified CallManager Certificate

The figure shows a portion of the Cisco Unified CallManager certificate (one of the Own certificates).
Display Cisco Unified CallManager Certificate
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Notice that the certificate contains the subject, in this case the name of the Cisco Unified CallManager (EAST3A), the signature algorithm (SHA1withRSA), the validate dates of the certificate, serial number, and other information. You should write down the fingerprints of the certificate immediately after installation when accessing Cisco Unified CallManager Serviceability over a trusted network (for instance, from the same LAN where Cisco Unified CallManager is located). When you access Cisco Unified CallManager web pages from other locations of your network, you can compare the fingerprints of the received certificate against the fingerprints that you wrote down. If they are equal, you can be sure that no falsified certificate is presented by an attacker attempting a manin-the-middle attack.
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Upgrading Software or Adding Options

This topic describes the Software Upgrades menu functions for upgrading the software version running on the platform, or adding locales, dial plans, and TFTP server files.
Software Upgrades Menu

Install or upgrade software such as new versions of Cisco Unified CallManager or CSA updates Install options Locales, phone firmware, gateway firmware, and dial plans Upload custom phone rings, callback tones, and phone backgrounds for TFTP download Can install software from local or remote source
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The Software Upgrades menu enables you to upgrade the application software, install software options, and upload files to the TFTP server for the phones to download. Options are software packages that provide extra functionality and are installed separately from the normal Cisco Unified CallManager install. An example of an option is Cisco IP Telephony Locale Installers.
Note Installing international dial plans is a two-step process. The first step is to install the software package on the Cisco Unified operating system Administration Software Upgrades page. The second step is to install the international dial plan in Cisco Unified CallManager Administration. This topic relates to the first step only.
When an administrator wants to install a new option on the Cisco Unified CallManager, the administrator downloads the appropriate software package from Cisco.com to their FTP or Secure FTP (SFTP) server. Alternatively, the administrator can download the software package and burn it on a CD or DVD and insert the CD or DVD into the system. The next step is to go to the Platform Administration pages, Software Upgrades > Install/Upgrade menu. From that page, you have a choice of using local media to install the option or using a remote SFTP server. The second task you can perform from the Software Upgrade menu is to upload TFTP server files. Use this option to upload various device files for use by the phones. The TFTP server files that you can upload include custom phone rings, callback tones, and phone backgrounds.
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Backing Out of an Upgrade

The software upgrade process also enables you to back out of an upgrade for any reason and revert the system to use the prior installed software version.
Backing Out of an Upgrade

Active Partition 5.0(1) 5.0(2) Inactive Partition 1
Install upgrade software on inactive partition.
4
Previous version of software is now the active partition.
5.0(1) 5.0(2)
Data Migration 5.0(2) 5.0(1)
2
Switch versions and reboot. Upgraded software in active partition.
3
If a problem occurs, repeat Step 2.
Any configuration changes made in Step 2 are lost.

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Recall that with Cisco Unified CallManager, there is an active and an inactive partition. When you install software, it installs on the inactive partition. To activate the newer version, you simply switch versions using the Restart > Switch Versions function of the Cisco IP Telephony Platform Administration. This automatically restarts the system with the newer version of the software. During this process, the upgraded software becomes the active partition, and your current (old) software becomes the inactive partition. Your configuration information migrates automatically to the upgraded version in the active partition. Locales are not migrated as part of the upgrade but must be reinstalled. Phone and gateway firmware loads do migrate.
Note The above paragraph describes the manual upgrade process. You can also optionally install upgrade software, and reboot automatically to the upgraded software version.
If for any reason you decide to back out of the upgrade, repeat the version-switching process, and when you restart the system, the inactive partition that contains the older version of the software becomes the active partition. Now the newer version is maintained in the active partition, but any configuration changes that you made since upgrading the software will be lost.
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Installing or Upgrading Software Options

You can install or upgrade software from a CD (assuming the image will fit) or DVD or from a remote source.
Installing or Upgrading Software Options

1. Download software package to media or SFTP server. 2. Check MD5 checksum against original file on Cisco.com. 3. Choose option to install from local or remote source and complete required fields. 4. Can automatically reboot to upgrade version or manually reboot.
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The steps to perform a software upgrade using both sources are as follows:
Upgrading from CD/DVD

To upgrade from a CD or DVD, follow this procedure:
Step 1
Download the appropriate upgrade file from Cisco.com. Do not unzip or untar the file. If you do, the system will not be able to read the upgrade files. Copy the upgrade file to a writeble CD or DVD. Insert the new CD or DVD into the disc drive on the local server that is to be upgraded. Choose Software Upgrades > Install/Upgrade. For the software location source, choose DVD/CD. If you burned the patch file to a subdirectory on the CD or DVD, enter the path in the Directory field. To continue the upgrade process, click Next. Choose the upgrade version that you want to install and click Next. In the next window, monitor the progress of the download, which includes the file name and the number of megabytes that are getting transferred.
Step 2 Step 3
Step 4 Step 5 Step 6
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Step 10
When the download completes, the Checksum window displays. Verify the checksum value against the checksum for the downloaded file as shown on Cisco.com.
The two checksum values must match to ensure the authenticity and integrity of the upgrade file. If the checksum values do not match, download a fresh version of the file from Cisco.com and try the upgrade again.
Caution
Step 11
After determining that the checksums match, click Next to proceed with the software upgrade. A Warning window displays the current and upgrade software versions. To continue with the software upgrade, click Next. The Post Installation Options window displays. Choose whether you want the system to automatically reboot to the upgraded partition after installing the upgrade software: To install the upgrade and automatically reboot to the upgraded partition, choose Reboot to upgraded partition. To install the upgrade and then manually reboot to the upgraded partition at a later time, choose Do not reboot after upgrade.
Step 12
Step 13
Click Upgrade. The Upgrade Status window displays with the Upgrade log. When the installation completes, click Finish.
Step 14
If you choose to upgrade the software and then manually reboot to the upgraded partition at a later time (the second option in Step 12), then choose the Restart > Switch Versions option and the system restarts, running the upgraded software.
Upgrading from Remote Server

To upgrade from a remote source, choose Remote File System for the Software Location Source. Enter the directory name for the software upgrade, if the file is not stored in the root directory of the FTP or SFTP server. Enter the host name or IP address of the remote server from which software will be downloaded, the name of a user who is configured on the remote server, the password that is configured for this user on the remote server, and choose SFTP or FTP for the download protocol. The rest of the upgrade procedure is the same as with a DVD or CD.
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Uploading TFTP Files

You can use the Upload TFTP Server File option to upload various files for use by the phones to the server.
Uploading TFTP Files
Current uploaded files
Upload new files Use the CLI command file list tftp to see the files in the TFTP directory and file get tftp to get a copy of a file in the TFTP directory.
Files that you can upload include custom phone rings, callback tones, and backgrounds. This option uploads files only to the specific server to which you connected, and other nodes in the cluster do not get upgraded. To upload a file, click Browse and then choose the file that you want to upload. To start the upload, click Upload File.
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Checking Network Connectivity and Setting Up Remote Support

This topic describes the basic functions of the Services menu in checking network connectivity and setting up a limited-time remote access account.
Services Menu
Ping another system on the network. Set up a remote account that Cisco support personnel can use to access the system for a specified period of time that you control.
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The Services menu enables you to ping other network devices and set up remote support.
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Ping Utility
The Ping GUI of the services function provides all the options to verify network connectivity, including an option to validate IPsec.
Ping Utility
Specify ping destination, time between packets sent, packet size, and count. Set Validate IPsec option to Yes if you want to verify that the link is secure.
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When you specify multiple pings, the ping command does not display the ping timestamp in real time. It displays results after the number of pings that you specified is complete.
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Setting Up Remote Account Support

The figure depicts the process to obtain remote account support from the Cisco Technical Assistant Center (TAC).
Setting Up Remote Account Support

1
Remote account configured using platform web interface (https://A.B.C.D/iptplatform)
4
LPBFXNCTQL
Cisco TAC
Customer
SSH with following credentials: Username: ciscoaccount & Password: LPBFXNCTQL

When the Cisco TAC escalation or development engineer determines that root file system access is necessary to resolve your problem, the engineer will request that you set up a limitedduration account so that the engineer can securely access the system. The procedure for granting remote access to the Cisco engineer is as follows:
Step 1
From the Cisco IP Telephony Platform Administration window, navigate to Services > Remote Support. If no remote support account is configured, click Add. Enter an account name for the remote account and the account life in days. Click Save. In the figure, ciscoaccount is the account name, and the account duration is 30 days.
Note
The account name must be at least six-characters long and all lowercase, alphabetic characters.
Step 2
The Remote Support Status window displays and shows the account name and generates a passphrase for this account. The time and date when the account will expire and the decoder version will also display. Call the Cisco TAC and inform the engineer working on your case of the account name you entered when you created the account and the passphrase. The TAC escalation or development engineer generates a password out of the passphrase you supply, which the engineer will use to remotely access the root system using a secure connection (Secure Shell Protocol [SSH]).
Step 3
Step 4
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Using the CLI

This topic describes how to use the CLI to perform basic platform functions.
CLI Overview
Platform CLI contains simple one-line commands Basic function is get the web interface working Multiple CLI (and GUI) accounts can be created via CLI Multiple levels of access CLI can be accessed by using a directly connected terminal or by using SSH Logon name and password to access the CLI are the same as entered for the Administrator account in the install wizard Name of the master Administrator account CANNOT be changed after install; the password can be changed
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Cisco IP Platform Administration offers CLI access in addition to GUI access. The CLI interface provides a subset of the functions that the GUI provides, so normally, you only use the CLI if there is a problem accessing the GUI. There are too many CLI commands to cover in this training. Refer to the Cisco IP Telephony Platform Administration Guide, Command-Line Interface chapter for the complete list of commands.
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CLI Access
You can access the CLI directly from the console or remotely using SSH.
CLI Access
Username and password are case-sensitive
Access via SSH client using adminname@hostname-or-ip or direct access
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Remote Access
From a remote system, use SSH to connect securely to the Cisco IP Telephony Platform. In your SSH client, enter: ssh adminname@hostname-or-ip where adminname specifies the administrator ID and hostname-or-ip specifies the host name or the host IP address. Both of these parameters are defined during installation. For example, ssh admin@ipt-1.
Direct Connection
From a direct connection, you automatically receive this prompt:
ipt-1 login:
where ipt-1 represents the host name of the system. Enter the administrator ID username and password that was defined during installation. The CLI prompt displays. The prompt represents the administrator ID; for example:
admin:
You can now use any CLI command.
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CLI Commands
The categories of CLI commands and a description of their function are shown in the figure.
CLI Commands
Name show set delete unset file utils run Description Show system parameters. Set system parameters and enable functionality. Delete parameters. Disable functionality. Manage files on the system. Use system utilities. Start processes on the system.
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More examples from several command types are shown in the associated lab task.
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Command-Line Editing
The figure shows navigating and editing features for command-line editing.
Command Line Editing

Command
Tab ? Left/Right Arrow Home/End Up/Down Arrow Control-C Delete/Rub
Description
Command completion and subcommands Command syntax and sub-commands Move cursor within command Command history Abort commands Delete character
Commands can have mandatory and nonmandatory parameters, and command options. Help is available for all commands.
The line-editing features provided by the Cisco Unified CallManager CLI are similar to the features of Cisco IOS.
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CLI Parameters and Options

An example of using a question mark (?) to obtain command help is shown in the figure.
CLI Parameters and Options

The syntax will show which parameters are required and if there are any options.
admin:utils network host ? Syntax: utils network host name [server serv][options] name mandatory name of the host or address to resolve servers optional the name of an alternate name server to use options optional page,detail admin:utils network host
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If you type a question mark after a valid command, the complete command syntax is displayed. The command in the example has a mandatory parameter (name) and two command options (servers and options). You can repeat the command without the ? to accomplish the input of the mandatory parameter and any options.
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Sample show Commands

Show commands enable you to display information about the system. The figure lists some of them.
Sample show Commands

Name
show status show process show network show hardware
Description
Displays basic platform status including host name, product version, CPU usage, and memory and disk usage Displays process and load information Displays network information, including Ethernet, routing, and active Internet connections Displays platform hardware information including, platform, serial number, BIOS build level, and active processors Displays detailed technical information about the system. Generates a lot of output Lists current administrator accounts, except the master administrator account Many more show commands are available.
show tech show account
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For a complete list of show commands and their options, refer to the Cisco IP Telephony Platform Administration Guide, Command Line Interface chapter, or type the command followed by ? at the command prompt.
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Sample set Commands

The figure lists some of the set commands, which enable you to configure or change settings.
Sample set Commands
Name set account set logging set password set trace
Description Sets up a new account on the platform Enables or disables logging Changes the administrator and security passwords Sets trace activity for the specified task.
More set commands are available.

For a complete list of set commands and their options, refer to the Cisco IP Telephony Platform Administration Guide, Command Line Interface chapter, or type the command followed by ? at the command prompt.
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Sample file Commands

The figure lists some of the file commands that enable you to perform many file operations.
Sample file Commands

Name
file delete file dump file get file list file view
Description
Deletes one or more files Dumps the contents of a file to the screen in hex format Sends the file to another system by using SFTP Lists the log files in an available log directory in ASCII format Displays the contents of a file
More file commands are available.

For a complete list of file commands and their options, refer to the Cisco IP Telephony Platform Administration Guide, Command Line Interface chapter, or type the command followed by ? at the command prompt.
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Example CLI Commands

Example CLI commands and their uses are shown in the figure
Example CLI Commands

admin: file list activelog cm/trace/ccm/sdi Shows a list of all SDI logs admin: file search activelog/messages authentication failure Looks for failed login attempts the server admin: utils network capture eth0 src 10.10.10.1 Watches all traffic coming from 10.10.10.1 (good for debug sessions) admin: show tech all Prints all high-level system information for TAC to review before troubleshooting
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Summary
Summary
The Cisco IP Telephony Platform Administration allows you to configure and manage the Cisco IP Telephony platform and can be accessed by a GUI or CLI. The Show menu allows you to check the status of various platform components. The Settings menu allows you to view and update IP, NTP, and SMTP settings. The Restart menu allows you to restart current version, switch software versions, and shut down the server.
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Summary
The Security menu allows you to manage security certificates and display or update IPsec policies. The Software Upgrades menu allows you to upgrade the software version running on the platform or add locales, dial plans, and TFTP server files. The Services menu allows you to ping network devices and set up a limited-time remote account for Cisco support personnel.
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References
For additional information, refer to these resources: Cisco IP Telephony Platform Administration Guide, Release 5.0 http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/iptp_adm/index.htm
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Lesson 2
Performing General Administration

Overview
Configure Cisco Unified CallManager features by using Cisco Unified CallManager Administration. In contrast to the Cisco IP Telephony Platform Administration, Cisco Unified CallManager Administration only allows GUI access. This lesson describes how to access the administration pages, lists all available menu items, and provides information about applications that are installed as part of the Cisco Unified CallManager installation. The lesson describes multilevel administration, which allows granular access control to Cisco Unified CallManager Administration functions.
Objectives
Upon completing this lesson, you will be able to access Cisco Unified CallManager Administration and integrated applications, navigate menu structure, and enable multilevel administration access (MLA) to provide multiple levels of security to Cisco Unified CallManager Administration. This ability includes being able to meet these objectives: Describe how to access the Cisco Unified CallManager Administration page Explain the applications installed as part of the Cisco Unified CallManager Release 5.0 installation Describe how MLA provides multiple levels of security to Cisco Unified CallManager Administration Describe how roles and user groups provide multiple levels of security to Cisco Unified CallManager Administration and to other applications Create a new role and a new user group, and assign a privilege level to the members in the user group
Cisco Unified CallManager Administration Overview

This topic describes functions of Cisco Unified CallManager Administration, explains how to access the GUI, and lists all available menu items.
Cisco Unified CallManager Administration Functions

System configuration (Cisco Unified CallManager groups, device pools, regions, locations, etc.) Call-routing configuration Media resource configuration (conferences, transcoders, MOH, etc.) Voice-mail configuration Device configuration User Management
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Cisco Unified CallManager Administration provides these functions: System configuration: Cisco Unified CallManager system configuration includes global settings such as Cisco Unified CallManager groups, device pools, regions, and locations. These settings can be applied to devices. Call-routing configuration: Call routing refers to the complete process of routing, originating, and terminating calls, including any billing and statistical collection processes. Media resource configuration: Cisco Unified CallManager controls access to features that require media resources such as conferences, transcoders, music on hold (MOH), etc. Voice-mail configuration: Cisco Unified CallManager allows access to voice-mail systems via configuration of voice-mail ports. Device configuration: This includes the configuration of all devices that can interact with Cisco Unified CallManager. Such devices are phones, gateway, gatekeepers, and trunks. User Management: Several features of Cisco Unified CallManager are user-based. Cisco Unified CallManager has its own user database to allow per-user access control.
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Accessing Cisco Unified CallManager Administration

You can access Cisco Unified CallManager Administration from a web browser.
Accessing Cisco Unified CallManager Administration

https://server-name/ccmadmin
Username is CCMAdministrator Password is the same as entered during installation

Or access from Cisco IPT Platform Administration page:
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To access the Cisco Unified CallManager Administration page, browse to https://<Cisco Unified CallManager server IP or name>/ccmadmin, as shown in the figure. If you are already connected to another GUI-based Cisco Unified CallManager administration page, such as the Cisco IP Telephony Platform Administration or the Disaster Recovery System, you can use the Navigation drop-down box located at the top right corner of the window, as shown in the figure. The default username for logging on to the Cisco Unified CallManager Administration is CCMAdministrator. Use the password that you set during the installation of Cisco Unified CallManager.
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Cisco Unified CallManager Administration Main Page

The Cisco Unified CallManager Administration page has several menus as shown in the figure.
Cisco CallManager Administration Main Page
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After a successful logon, a page is displayed that reveals the system version. Menu functions are explained in detail in subsequent pages.
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The figure shows the individual items under each menu category.
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The most important menu items are discussed in detail in separate topics of this training course.
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Cisco Unified CallManager Release 5.0 Applications

This topic describes applications that are installed during the installation routing of Cisco Unified CallManager.
Cisco Unified CallManager Release 5.0 Applications

Bulk Administration: for bulk inserts, updates, and removals Serviceability: to manage services Disaster Recovery System: to back up and restore servers Platform Administration: for basic system administration DHCP service Management tools Cisco Unified CallManager RTMT* Cisco Unified CallManager CAR* Dialed Number Analyzer
*RTMT=Cisco Unified CallManager Real-Time Monitoring Tool *CAR=Cisco Unified CallManager CDR Analysis and Reporting
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Cisco Unified CallManager includes these applications: Cisco Bulk Administration Tool (BAT): Cisco BAT is used for bulk inserts, updates, and removals of configuration database entries. It simplifies mass deployments or changes that affect many devices. Cisco Unified CallManager Serviceability tool: Cisco Unified CallManager serviceability is used to mange services and for system management such as alarm configuration or troubleshooting. Cisco Disaster Recovery System (DRS): DRS is used to backup and restore Cisco Unified CallManager servers. Cisco IP Telephony Platform Administration: Cisco IP Telephony Platform Administration is used for basic system administration. DHCP service: Cisco Unified CallManager provides a Dynamic Host Configuration Protocol (DHCP) service that you can use to assign IP addresses in small deployments with a backup DHCP server. Management tools: Several Cisco Unified CallManager management tools are available, including the Cisco Unified CallManager Real-Time Monitoring Tool (RTMT), which allows monitoring of the Cisco Unified CallManager servers; the Cisco Unified CallManager Call Detail Record (CDR) Analysis and Reporting tool (CAR); and the Dialed Number Analyzer, a tool that allows dial-plan analysis.
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Multilevel Administration Overview

This topic describes Cisco Unified CallManager multilevel administration access (MLA).
MLA Overview
Multiple levels of Cisco Unified CallManager Administration privileges are available with MLA. MLA is based on two components: User groups End users are assigned to user groups. Roles Each role refers to one application. Each application has one or more resources (static list). Per role, access privileges are configured per. application resource. Roles are assigned to user groups.
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Multilevel administration access (MLA) allows multiple levels of Cisco Unified CallManager Administration privileges. The CCMAdministrator username always has full permissions.
Note The CCMAdministrator username has full permissions for Cisco Unified CallManager Administration only. It does not have any permissions for platform administration. For platform administration, a different username must be used. The platform administration username and password are configured during the installation. The CCMAdministrator password is also set during installation.
Cisco Unified CallManager Release 5.0 MLA is based on these two main components: User Groups: First, you create individual users. In Cisco Unified CallManager Release 5.0, there are application users and end users. Application users are usernames used by application processes, while end usernames are used by persons. Each end user is assigned to one or more user groups. A user group can (and usually will) have more than one user assigned. Roles: Each role refers to one application (such as Cisco Unified CallManager Administration or Cisco Unified CallManager Serviceability). More than one role can refer to the same application. Each application has a static list of application resources, or application functions. For each role, set access privileges per application resource. For most application resources, the available privilege options are none, read, or update. At the end, roles are assigned to user groups. You can assign multiple roles to one user group. The same role can be assigned to more than one user group. Therefore, an end user has all the permissions that have been configured for the roles that are assigned to the user group.
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MLA Component Interaction

The figure illustrates how the different components of MLA interact.
MLA Component Interaction

Users n:n User Groups n:n Roles 1:1 Applications
Application1 Role1 User2 Group1 Role2 User3 Group2 Role3 User4 Application2 Resource1 Resource2 Resource3 Resource4
1 : 1 Privileges
User1
Resource1 Resource2 Resource3 Application1 Resource1 Resource2 Resource3
read (none) read, update
read, update read (none)
read read (none) read, update

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The diagram shows four users (User1 to User4) and two user groups (Group1 and Group2). User1 and User2 are assigned to Group1, User3 is assigned to both groups, and User4 is assigned to Group2. There are three roles (Role1 to Role3). Role1 is assigned to Group1, Role2 is assigned to both groups, and Role3 is assigned to Group2. Role1 and Role2 both refer to Application1. Application1 has three application resources (Resource1 to Resource3). Role1 and Role2 have different privileges assigned to resources of Application1. Role3 refers to Application2 and has privileges assigned to the four application resources (Resource1 to Resource4) of Application2.
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Roles and User Groups Example

In versions of Cisco Unified CallManager before Release 5.0, access privileges are assigned directly to individual users. In Cisco Unified CallManager Release 5.0, access privileges are assigned to application resources.
Roles and User Groups Example

User Group CCM Superusers User John Doe User Jane Smith Role Standard CCMADMIN Administration Application Resource Privilege Update Call Park CCM Administration page AAR Group page CallManager Group page CallManager CCM Group web Administration pages
CCM Read-Only User Kim Lu User Tom Adams
Standard CCMADMIN Read-Only
Read-Only
Privileges are assigned to application resources (not to users, as in releases before Release 5.0). Roles consist of application resources and are assigned to user groups. User groups consist of users.
Note: This is not the configuration order.
Resources in Cisco Unified CallManager Release 5.0 are the web pages associated with a function, such as the Call Park web pages, the automated alternate routing (AAR) group web pages, and the CallManager group web pages. Resources have privileges. For each role that is associated with an application, the privilege levels apply for access to each of the resources or web pages. The access privileges can be read or update. The read access privilege limits access to windows to read operations only. Buttons such as Insert, Delete, Update, and Reset do not display. The update privilege allows operations such as insert, delete, update, and reset, as well as executive functions to start or stop a process or service from the Cisco Unified CallManager Administration and Serviceability windows. The figure shows an example of how user groups, roles, applications, resources, and privileges interact. The Standard CCMADMIN administration role includes the application Cisco Unified CallManager Administration. The application Cisco Unified CallManager Administration consists of resources such as the Call Park web pages, AAR group web pages, and Cisco CallManager group web pages. The privilege level assigned to the Call Park web pages is update for users in the user group CCM Superusers, of which John Doe is a member user.
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Roles and User Groups

This topic describes standard roles and standard user groups and explains how you can add custom roles and custom user groups.
Roles
Standard roles exist; custom roles can be created. Standard roles cannot be deleted. Create new roles or copy and modify existing roles.
Cisco Unified CallManager has several standard roles, which exist by default and cannot be deleted. You can add custom roles, or copy and edit existing roles. There is an enterprise parameter called Effective Access Privileges For Overlapping User Groups and Roles, which determines the effective privileges that a user has in case of conflicting values for the same application resource. This is possible because more than one role can be assigned to a user group, and the user can be a member of multiple user groups. There are two options: Maximum or Minimum. If the enterprise parameter is set to maximum, the user will have the highest privilege that is configured on any applicable role. If the enterprise parameter is set to minimum, the user will have only the best possible privilege that is permitted on all applicable roles.
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Role Configuration Page

The figure shows the content of the Role Configuration page.
Role Configuration Page
Selected Application
Role Name
Configured Privilege per Application Resource
Roles consist of application resources (web pages) with access privileges.

When adding a new role, you have to select the application, configure a role name and then set privileges for each resource of the selected application.
Note If you check update, you get read privileges even if read is not checked
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User Groups
Cisco Unified CallManager has several standard user groups, as shown in the figure.
User Groups
Standard user groups exist; custom user groups can be created. Standard user groups cannot be deleted. Create a new user group or copy and modify an existing group one.
Standard user groups exist by default and cannot be deleted. You can add custom user groups or copy and edit existing user groups. To copy a user group, you may use the search function to find the respective user group name, and then click the respective Copy button in the column on the right side of the window. You will be prompted for the name of the destination user group. To edit a custom user group, click on the group and you will have the option to edit the group.
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User Group Configuration Page

The figure shows the content of the User Group Configuration page.
User Group Configuration Page
Add end users and application users to user group.

To add a new user group, click Add End Users to Group and another window (Find and List Users) appears. From this window, you can search for users and select the users that you want to add to the user group.
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Creating a New Role and User Group

This topic describes how to configure MLA for a new user.
MLA Configuration Steps

1. Add custom roles (or use default) 2. For added custom roles: a) Choose the application to associate with the role b) Assign privileges to application resources 3. Create users 4. Create custom user groups (or use default) 5. Add users to user groups 6. Assign roles to user groups 7. Verify privileges
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To configure MLA for a new user, perform these steps:

Step 1 Step 2
Add custom roles (or use default). For added custom roles:
a) Choose the application to associate with the role. b) Assign privileges to application resources.
Step 3 Step 4 Step 5 Step 6 Step 7
Create users. Create custom user groups (or use default). Add users to user groups. Assign roles to user groups. Verify privileges.
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Step 1: Add Custom Role

In order to add a custom role, first go to User Management > Role in Cisco Unified CallManager Administration.
Step 1: Add Custom Role
Click Add New
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The Find and List Roles window displays, as shown in the figure. To add a new custom role, click Add New.
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Step 2a: Select Application

You will see the Role Configuration window, as shown in the figure.
Step 2a: Select Application
Select application
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In the Role Configuration window, select the application to associate with the role.
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Step 2b: Set Privileges

After selecting the application, the window will be extended as shown in the figure.
Step 2b: Set Privileges
Enter role name
Set privilege for each application resource
Click Save
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Enter a name and a description (optional) for the role and configure the privileges for each application resource. In the example, the role name is CCM Read Dial Plan and read access has been permitted only for the Dial Plan resource. Click Save when you have finished the role configuration.
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Step 3: Add Users

In order to add a new user, first go to User Management > End User in Cisco Unified CallManager Administration.
Step 3: Add Users
Click Add New
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The Find and List Users window displays, as shown in the figure. To add a new user, click Add New.
Step 3: Add Users (Cont.)
Enter user ID and other user details
Click Save
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In the End User Configuration window, enter the User ID of the new user and populate other fields as desired. Then click Save. In the example shown in the figure, user andy is created.
Step 4: Create Custom User Groups

In order to add a new user group, first go to User Management > User Group in Cisco Unified CallManager Administration.
Step 4: Create Custom User Groups
Click Add New
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The Find and List User Groups window displays, as shown in the figure. To add a new user group, click Add New.
Step 4: Create Custom User Groups (Cont.)
Enter user group name
Click Save
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In the User Group Configuration window, enter the group name and click Save. In the example shown in the figure, user group Call Routing Operators is created.
2006 Cisco Systems, Inc. Administration of Cisco Unified CallManager Release 5.0 2-55
Step 5: Add Users to User Groups

In this step, you add a user to a user group.
Step 5: Add Users to User Groups
Click Add End Users to Group
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To add a user to the newly created group, click Add End Users to Group in the User Group Configuration window, as shown in the figure.
Step 5: Add Users to User Groups (Cont.)
Select users
Click Add Selected
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In the Find and List Users window, select all users that you want to add to the user group and click Add Selected, as shown in the figure. In this example, user andy is added to the user group.
Step 6: Assign Roles to User Groups

In order to assign roles to user groups, go to User Management > Role in Cisco Unified CallManager Administration.
Step 6: Assign Roles to User Groups
Click user group
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The Find and List User Groups window displays, as shown in the figure. Click the user group that you want to assign roles to. In the example shown in the figure, the Call Routing Operators group is chosen.
Step 6: Assign Roles to User Groups (Cont.)
Click Assign Role to Group
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In the User Group Configuration window, click Assign Role to Group, as shown in the figure.
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Select Roles
Click Add Selected
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In the Find and List Roles window, select all roles that you want to assign to the user group and click Add Selected, as shown in the figure. In this example, the role CCM Read Dial Plan is added to the user group.
Verify assigned roles
Click Save
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The User Group Configuration window displays the list of assigned roles, as shown in the figure. Click Save to save the role assignment.
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Step 7: Verify Privileges

In order to verify the privileges of a user, go to User Management > User Group in Cisco Unified CallManager Administration.
Step 7: Verify Privileges
Click user group
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The Find and List User Groups window displays, as shown in the figure. Click on any user group that the user whose privileges you want to verify is a member of. In the example shown in the figure, the Call Routing Operators user group is chosen.
Note If you do not know which groups the user whose privileges should be verified is a member of, you can go to User Management > End User and check the user group memberships of the user on the End User Configuration page of that particular user.
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Step 7: Verify Privileges (Cont.)
Locate user
Click information button
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In the User Group Configuration window, all members of the user group are listed. In the Permission column, there is an information symbol (i). Click the symbol that is in the row of the user whose privileges you want to verify. In the example shown in the figure, the information symbol of user andy is clicked.
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Step 7: Verify Privileges (Cont.)
Verify permissions
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The User Privilege window displays all applications and application resources and the corresponding permissions for the previously selected user. In the example shown in the figure, the only permission of the chosen user is to read the Dial Plan of Cisco Unified CallManager Administration.
Note An alternative way to verify the privileges of a user is to access the User Configuration page, select the User Privilege Report option under the Related Links located on the top right corner of the screen and click Go.
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Summary
Summary
Cisco Unified CallManager Administration consists of the dial plan, device, and user configuration. Applications that are automatically installed during the installation of Cisco Unified CallManager include Cisco Unified CallManager BAT, Cisco Unified CallManager Serviceability, Disaster Recovery System, and Platform Administration. MLA allows multiple levels of Cisco Unified CallManager administration. User privileges are based on roles and user groups. MLA configuration includes user and user group configuration, role configuration, and the mapping of roles to user groups.
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Lesson 3
Configuring Cisco Unified CallManager Release 5.0 Basic Settings

Overview
Cisco Unified CallManager configuration includes basic settings plus specific settings that depend on the features and settings used. This lesson describes how basic settings on Cisco Unified CallManager Release 5.0 are configured in order to enable the system and prepare Cisco Unified CallManager for endpoint deployment.
Objectives
Upon completing this lesson, you will be able to configure Cisco Unified CallManager Release 5.0 basic settings to activate services and use the IP address instead of the host name for Cisco Unified CallManager. This ability includes being able to meet these objectives: Describe the basic settings configuration in Cisco Unified CallManager Release 5.0 Explain the advantages and disadvantages of using either DNS or IP in Cisco Unified CallManager Release 5.0 server name configuration Describe how to enable Cisco Unified CallManager Release 5.0 services through a web interface Describe the service parameters during a basic configuration of Cisco Unified CallManager Release 5.0 Describe the enterprise parameters during a basic configuration of Cisco Unified CallManager Release 5.0
Basic Settings Configuration Overview

This topic describes the basic settings that you need to configure to enable Cisco Unified CallManager.
Basic Settings Configuration Overview

Cisco Unified CallManager Release 5.0 should have a basic configuration before any device is configured:
Make DNS vs. IP considerations. Enable Cisco Unified CallManager, TFTP server, and CTI manager services. Configure service and enterprise parameters to meet customer criteria.
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Before any device is added to Cisco Unified CallManager Release 5.0, you must make a basic initial configuration, including these considerations: You must consider whether to use DNS or an IP address, decide what better fits into the customer environment, and then configure the system according to that decision. Enable basic services such as Cisco Unified CallManager, a TFTP server, and a computer telephony integration (CTI) manager. Configure service and enterprise parameters to meet the customer environment.
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DNS vs. IP Addresses

This topic describes the pros and cons of using DNS names versus IP addresses in Cisco Unified CallManager deployments.
DNS vs. IP
Cisco Unified CallManager Release 5.0 can use DNS names or IP addresses for system address values.
Advantages of IP Addresses Advantages of DNS
No need of DNS server Reduces potential source of failure Simplifies troubleshooting
Simplified management because of names instead of numbers Easier IP address changes because of name-based IP paths
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You can configure Cisco Unified CallManager Release 5.0 based on IP addresses or DNS names. Consider the following when making the decision to use IP addresses versus DNS: Advantages of IP Addresses: No DNS server is needed, eliminating one potential source of failures within an IP network. Eliminating potential sources of failure automatically simplifies troubleshooting. A reduction of network traffic because no extra DNS is needed. Management is simplified when using DNS because names are easier to remember and handle then IP addresses. Changing IP addresses within the network is easier when using DNS because only the DNS server settings need to be changed.
Advantages of DNS:
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Service Activation
This topic describes how to activate, deactivate, restart, and verify Cisco Unified CallManager services.
Service Activation
To enable Cisco Unified CallManager Release 5.0 services, perform the following tasks:
Access the Cisco CallManager Administration Serviceability Tool. Go to Tools > Service Activation. Select your server. Enable the necessary services. Go to Tools > Control Center Feature Services and select your server. Verify that the configured services are up and running.
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Cisco CallManager Administration Serviceability Tool, a web-based troubleshooting tool for Cisco Unified CallManager, provides a Service Activation window that allows you to activate, deactivate, and view services. The services are grouped into two categories: feature services, which correspond to Cisco Unified CallManager features, and network services, which are for more generic use. Most services are inactive by default, to save the CPU and memory resources of the server. You should enable only the services that are needed. Follow these steps to enable services on Cisco Unified CallManager Release 5.0:
Step 1 Step 2 Step 3 Step 4 Step 5
Access the Cisco Unified CallManager Administration Serviceability Tool. Go to Tools > Service Activation. Select your server. Enable the necessary services. To verify that the configured services are enabled, go to Tools > Control Center Feature Services and select your server.
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Service Activation (Cont.)
Select Cisco CallManager Serviceability and click Go
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To access the Cisco Unified CallManager Administration Serviceability Tool, select the corresponding link from the navigation drop-down menu within the Cisco Unified CallManager Administration page and click Go.
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Go to Service Activation
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As shown in the figure, go to Tools > Service Activation in the Cisco Unified CallManager Serviceability window to go to the service activation feature.
Select the appropriate server.
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First, you must select the server on which the services should be activated or deactivated.
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Refresh page Reset to default for single-server operation.
Save and perform settings
Select the services that should be activated. Deselect the services that should be deactivated.
Configured status
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After selecting a server, the Service Activation page for the corresponding server within the Cisco Unified CallManager Release 5.0 cluster is displayed. From here, you can activate and deactivate services on the previously selected Cisco Unified CallManager Release 5.0 server. You may activate or deactivate as many services as you want at the same time. Some services depend on other services, and the dependent services automatically get activated together with the selected service. After you select the services you wish to activate or deactivate, click the Save and Perform Settings button. The server may pause while the services are being activated or deactivated. When the screen refreshes, verify that the services you selected for activation or deactivation are in the correct state. You may have to click on the Refresh button to review the changes as well.
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Control Center
To start, stop, restart, and monitor services, use the Control Center pages.
Control Center
Go to Control Center Feature Services
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In Cisco Unified CallManager Release 5.0 and later, the Control Center is split into Feature Services and Network Services. To get to these Control Center pages, go to Tools > Control Center Feature Services or Tools > Control Center Network Services.
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Control Center (Cont.)
Stop, start, restart selected service
Select service to start, stop or restart
Configured status Actual status
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Use the Control Center to start, stop, or restart a service by selecting the service and then clicking to the appropriate button. On the control center, you can verify the activation status (activated or deactivated) and the actual status (started or stopped) of a service.
Note Only activated services can be started, stopped, or restarted from the Control Center. If you want to launch a service which is currently deactivated, you have to activate it first by using the Service Activation page.
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Service Parameters
This topic describes service parameters and their configuration, and lists some examples.
Service Parameters
Service parameters on Cisco Unified CallManager Release 5.0 can be changed to fit special needs, such as:
T302 timer to speed up dialing SIP retry counts to relieve overloaded SIP proxy servers Enable call detail records Enable MGCP gateway overlap receiving Define extension mobility maximum login time Define attendant console username Change extended functions connecting port Define voice media streaming application codecs
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Service parameters for Cisco Unified CallManager allow you to configure different services on selected servers. You can view a list of parameters and their descriptions by clicking the question mark button in the Service Parameters Configuration window. You can view the list with a particular parameter by clicking that parameter. Service parameters can only be updated when a service is running. If you try to modify the parameters when a service is deactivated, the changes are discarded. Therefore, before you begin to change any service values, ensure that the service is active on the specified server. The slide lists some examples of parameters that can be influenced by service parameters.
Caution Some changes to service parameters may cause system failure. Cisco recommends that you do not make any changes to service parameters unless you fully understand the feature that you are changing or unless the Cisco Technical Assistance Center (TAC) specifies the changes.
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Service Parameters (Cont.)
Select Server
Select Service
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To access the service parameters, choose Tools > Control Center Feature Services. To configure service parameters on Cisco Unified CallManager Release 5.0, select the Cisco Unified CallManager server for which the settings should be made. Then select the service from the drop-down menu. Each service has its own service parameter configuration page within Cisco Unified CallManager Administration.
Note The list of services also includes the activation status (active or inactive) of the services.
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Service Parameters (Cont.)
Verify/configure service settings
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Each service has its individual service parameters. The default value for each service parameter is displayed in the column Suggested Value, located on the right.
Tip To easily find out which parameters are modified to a different value than the suggested value, select Parameters for All Servers from the Related Links drop-down menu, which gives you a list of all values within the cluster. Then select Modified Parameters for All Servers from the Related Links drop-down menu on that new page to hide all parameters that use suggested values.
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Enterprise Parameters
This topic describes enterprise parameters and their configuration, and lists some examples.
Enterprise Parameters
Enterprise parameters are used to define cluster-wide system settings, such as:
Default phone protocol (SCCP or SIP) Cisco Unified CallManager user web-page content System URLs used by phones
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Enterprise parameters provide default settings that apply to all devices and services in the same cluster. When you install a new Cisco Unified CallManager, it uses enterprise parameters to set its initial values. You cannot add or delete enterprise parameters, but you can update existing enterprise parameters.
Note Many enterprise parameters rarely require change. Do not change an enterprise parameter unless you fully understand the feature that you are changing or unless the Cisco Technical Assistance Center (TAC) specifies the change.
Enterprise parameters that are commonly changed are: Default phone protocol (Skinny Client Control Protocol [SCCP] or Session Initiation Protocol [SIP]) Cisco Unified CallManager user web-page content System URLs used by phones
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Enterprise Parameters (Cont.)
Select protocol for auto registration
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To change the default protocol for autoregistered phones, use the drop-down menu and select SIP or SCCP.

Select settings that should be available to phone users
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The figure shows a section from the Enterprise Parameters Configuration page, containing Cisco Unified CallManager user parameters and their default values.
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Change the name to an IP address if DNS services are not used
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The figure shows Cisco Unified CallManager phone URL parameters and their default values. If DNS services are not used, the host name in URL links must be changed to the IP address of the server. In the figure, the first three URLs are already corrected, while the fourth URL still needs to be changed.
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Summary
Summary
Cisco Unified CallManager Release 5.0 needs a basic configuration before any device is added. Use IP addresses for Cisco Unified CallManager Release 5.0 to avoid dependencies on DNS servers. Services are activated through service activation within the Cisco CallManager Administration Serviceability Tool. Service parameters are used to configure Cisco Unified CallManager services to apply individual needs. Enterprise parameters are used to define cluster-wide parameter values.
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Lesson 4
Implementing Disaster Recovery

Overview
Disaster recovery solutions ensure that, in the event of a catastrophic failure, you can quickly restore mission-critical systems to normal operation with minimal or no interruption in normal business activities. You should regularly back up all Cisco Unified CallManager nodes in a cluster in order to mitigate the impact of a catastrophic failure of one or many nodes, or so that you can restore data to a pre-upgrade state. Cisco Unified CallManager includes integrated disaster-recovery tools that enable you to back up all nodes in a Cisco Unified CallManager cluster on a predefined schedule, and restore data to individual nodes as necessary. This lesson describes how to use the Cisco Disaster Recovery System GUI and command-line interface (CLI) to perform backup and restore operations to Cisco Unified CallManager Release 5.0 clusters.
Objectives
Upon completing this lesson, you will be able to back up and restore data for a Cisco Unified CallManager Release 5.0 cluster using the GUI and CLI. This ability includes being able to meet these objectives: Describe the capabilities, architecture, components, and access methods of the Cisco DRS Perform an immediate and scheduled backup using the Cisco DRS GUI Perform a restore operation using the Cisco DRS GUI Define the CLI commands to execute a backup and restore
Disaster Recovery System Overview

This topic describes the purpose, capabilities, and components of the Cisco Disaster Recovery System (DRS), including which files it backs up.
Cisco Disaster Recovery System Overview

Cisco DRS provides backup and restore functionality for Cisco Unified CallManager Release 5.0 (similar to BARS in Release 4.x). Cisco DRS performs cluster-level backups and allows you to choose which nodes are restored. Cisco DRS supports regularly scheduled automatic or user-invoked data backups. Cisco DRS archives backups to a physical tape drive or remote SFTP server. Cisco DRS provides a GUI and CLI interface for performing backup and restore tasks.
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The Cisco DRS, which you can enable from Cisco Unified CallManager Administration Release 5.0, provides full data backup and restore capabilities for all servers in a Cisco Unified CallManager cluster. The Cisco DRS performs a cluster-level backup, which means that it collects backups for all servers in a Cisco Unified CallManager cluster to a central location and archives the backup data to a physical storage device.
Note In Cisco Unified CallManager Release 4.x environments, you use the Cisco Unified CallManager Backup and Restore System (BARS), which provides similar backup and restore capability, although on a per-node basis.
Cisco DRS allows you to perform regularly-scheduled automatic or user-invoked data backups. Cisco DRS supports only one backup to be scheduled at a time.
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Cisco DRS Architecture

Cisco DRS uses a distributed system architecture for performing backup and restore functions.
Cisco DRS Architecture

Local Agent
SFTP
Remote Share
Master Agent
Tape
Cisco Unified CallManager Release 5.0 Cluster
First Node
TCP/IP
Local Agent Subscriber 1
DRS HTML GUI and CLI Tools
Local Agent Subscriber 2
Network
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The Cisco DRS distributed architecture consists of two key functions, the Master Agent and the Local Agent. The Master Agent coordinates backup and restore activity with all the Local Agents. Cisco DRS provides a GUI or CLI for performing backup and restore tasks. You perform backup and restore tasks on the Master Agent on the first node, and the Master Agent communicates to the Local Agents to execute the backup or restore. Backups are archived to a physical tape drive or remote Secure FTP (SFTP) server. The system automatically activates both the Master Agent and the Local Agent on all nodes in the cluster. However, you can only access the Master Agent functions on the first node of the cluster.
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Cisco DRS Components

The Cisco DRS consists of two access methods (GUI and CLI) and two agents (Master Agent and Local Agent), as shown in the figure.
Cisco DRS Components

Master Agent: Stores configuration Controls backup and restore Communicates with archive devices (tape or SFTP server) Local Agent: Executes backup and restore tasks GUI: Provides web-based administration tool CLI: Provides command-line interface to DRS (subset of GUI Administration page)
The Master Agent is automatically activated on the first node. Local Agents are automatically activated on all nodes in the cluster.
Master Agent
The Master Agent performs these duties: Stores systemwide component registration information. Maintains a complete set of scheduled tasks in the Cisco Unified CallManager database. When it receives updates from the user interface, the Master Agent sends executable tasks to the applicable Local Agents, as scheduled. (Local Agents execute immediate-backup tasks without delay.) Access the Master Agent through the Cisco DRS user interface to perform activities such as scheduling backups, adding a new backup task for a specific server or a defined cluster, updating or reviewing an existing entry, displaying status of executed tasks, and performing system restoration. The Master Agent stores backup sets on a locally attached tape drive or a remote network location.
Local Agents
Each server in a Cisco Unified CallManager cluster, including the server that contains the Master Agent, must have its own Local Agent to perform backup and restore functions for its server. By default, a local agent automatically gets activated on each node of the cluster.
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Features and Components Backed Up

For Cisco Unified CallManager Release 5.0, you can back up and restore the features and subcomponents shown in the figure.
Features and Components Backed Up

Features
Cisco CallManager
Components
Cisco Unified CallManager Release 5.0 database Serviceability Tool Music on hold (MOH) Cisco Emergency Responder Bulk Administration Tool (BAT) Phone device files
CDR_CAR
Call Detail Record (CDR) CDR Analysis and Reporting (CAR)
Cisco DRS backs up features. By selecting a feature, all subcomponents of that feature are backed up.
For each feature that you choose, the system automatically backs up all of its subcomponents.
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Cisco Disaster Recovery System GUI Access

To obtain GUI access to the Cisco DRS, choose Disaster Recovery System from the Navigation drop-down menu in the upper, right corner of Cisco Unified CallManager Administration window.
Cisco Disaster Recovery System GUI Access
Choose Disaster Recovery System from the Navigation menu in the upper, right corner of Cisco CallManager Administration window.
Log in using Administrator username and associated password set at installation or changed with CLI.
Log in to the Disaster Recovery System by using the same Administrator username and password that you use for Platform Administration.
Note You set the Administrator username and password during Cisco Unified CallManager installation.
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Cisco Disaster Recovery System CLI Access

For CLI access to the Cisco DRS, use the console or connect to Cisco Unified CallManager by Secure Shell Protocol (SSH).
Cisco Disaster Recovery System CLI Access

Access CLI on the console or by SSH Log in using Administrator username and associated password set at installation or changed with CLI All disaster recovery commands start with utils disaster_recovery
admin:utils utils utils utils utils utils utils utils
disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery
? status cancel_backup show_tapeid show_registration show_backupfiles* backup* restore*
admin:utils disaster_recovery
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Log in to the CLI by using the same Administrator username and password that you use for Platform Administration. After successful login, you have access to all CLI-based Cisco DRS commands, which start with utils disaster_recovery.
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Performing a Backup
This topic describes how to perform an immediate and scheduled backup using the Cisco Disaster Recovery System.
Backup Menu
Configure backup parameters Start manual backup View backup information
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The figure shows the Cisco DRS Backup menu. The top three menu items enable you to configure features that you want to back up, the storage location where the backup files will be stored, and the scheduling engine to initiate tasks at user-specified times. The Manual Backup menu item enables you to access submenus to start a manual on-demand backup. The History menu item enables you to see the last 20 backup jobs. The Current Status menu item enables you to check the status of the current backup job and cancel the current backup job.
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Steps to Perform a Backup

A Cisco DRS backup procedure consists of four major steps.
Steps to Perform a Backup

1. Configure the features to back up. 2. Configure a storage location. 3. Configure a backup: a) Scheduled backup b) Manual backup 4. Check the status or cancel a backup.
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The steps to perform a backup are as follows:

Step 1
Configure the features to back up. Before you can run a backup job, you must choose the features that you want to back up. Configure a storage location. You must choose the physical location where you want to store the backup file. Configure a scheduled backup or start a manual backup. You can perform an immediate manual backup or configure a regularly-scheduled backup for the cluster. Check the status of the backup. While a backup is running, you can check the status of the current backup job, or cancel a backup.
Step 2
Step 3
Step 4
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Step 1: Selecting Features to Back Up

Before you can schedule or start a backup job, you must configure the features that you want to back up.
Step 1: Selecting Features to Back Up
Select feature
1a
Save changes
1b
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After you log in to the Cisco DRS, navigate to Backup > Configure Cluster Features. The options are to back up Cisco Unified CallManager data (CCM) and Call Detail Record Analysis and Reporting (CDR_CAR). From the list of available features, choose the feature or features that you want to include in the backup and click Save. You must choose at least one feature.
Note Changing a backup feature changes it for both manual and scheduled backups.
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Step 2: Configuring a Storage Location

Next, configure the location where you want the backup file to be stored. To access the Storage Location window, navigate to Backup > Storage Location.
Step 2: Configuring a Storage Location
Configure backup destination
2a
Save changes 2b
In the Storage Location window, choose the number of backup files to store on a network
directory (or leave at default value).
Then choose the storage destination option and enter the appropriate field values: Tape Device: Stores the backup file on a locally attached tape drive. Choose the appropriate tape device from the drop-down list. Network Directory: Stores the backup file on a networked drive accessed through an SFTP connection. Enter this required information:
Note
Server name: Name or IP address of the network server Path name: Path name for the directory where you want to store the backup file User name: Valid username for an account on the remote system Password: Valid password for the account on the remote system
You must have access to an SFTP server to configure a network directory location.
Click Save to update the settings.
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Step 3a: Configuring a Scheduled Backup

You can schedule a backup to start at a specified date and time or start an immediate backup.
Step 3a: Configuring a Scheduled Backup
Define start time and date
3b
Define backup frequency
3c
Save schedule 3d
Enable scheduler
3a
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The figure shows the backup scheduler GUI. To access this window, navigate to Backup > Scheduler and if the schedule is not enabled, click the Enable Scheduler button. Choose the date and time when you want the backup to begin. Choose the frequency at which you want the backup to occur: Once, Daily, Weekly, or Monthly. If you choose Weekly, you can also choose the days of the week when the backup will occur. Save the settings. The next backup occurs automatically at the time that you set.
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Step 3b: Starting a Manual Backup

To perform a manual backup, rather than a scheduled backup, log in to the Disaster Recovery System and navigate to Backup > Manual Backup.
Step 3b: Starting a Manual Backup
Manually start backup
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At the Manual Backup window, ensure that the features that you want to back up are selected. If they are not selected, navigate to Backup > Configure Features. Click Start Backup to immediately begin the backup.
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Step 4: Checking Backup Status or Canceling a Backup

Navigate to Backup > Current Status to check the status of the current backup job.
Step 4: Checking Backup Status or Canceling a Backup

Refresh to view actual status
4
Cancel backup
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From the Backup Status window, you can also view the backup log file or cancel the current backup.
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Performing a Restore
This topic describes how to perform a restore operation using the Cisco Disaster Recovery System GUI.
Restore Menu
Start system restore View restore information
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The figure shows the Cisco DRS Restore menu. The top menu item accesses the Restore Wizard. The Restore Wizard walks you through the steps required to restore a backup file. From the Restore menu, you can also see the last 20 restore jobs, and check the status of the current restore job.
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Steps to Perform a Restore

A Cisco DRS restore procedure consists of five steps, as shown in the figure.
Steps to Perform a Restore

1. Choose storage location. 2. Choose the backup file. 3. Choose features. 4. Choose nodes. 5. Check the status of the restore.
To restore an entire cluster, you must first restore the first node and then restore the subsequent nodes. To restore the first node, perform a fresh installation of Cisco Unified CallManager Release 5.0, then proceed with steps to restore.
If a major failure occurs or you upgrade hardware, you may need to restore all nodes in the cluster. To restore an entire cluster, you must first restore the first node or publisher and then restore the subsequent nodes or subscribers. To restore the first node, perform a fresh installation of Cisco Unified CallManager Release 5.0. After the first node is restored, use this procedure to restore subsequent nodes:
Step 1
Choose storage location. You must first choose the storage location from which you want to restore a backup file. Choose the backup file. From a list of available files, choose the backup file that you want to restore. Choose features. From the list of available features, choose the features that you want to restore. Choose nodes. If the feature was backed up from multiple nodes, you must choose the nodes that you want to restore. Check the status of the restore. While the restore process is running, you can check the status of the current restore job.
Before you restore Cisco Unified CallManager, ensure that the Cisco Unified CallManager version that is installed on the server matches the version of the backup file that you want to restore.
Step 2
Step 3
Step 4
Step 5
Caution
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Step1: Choosing Storage Location

From the Cisco DRS main menu, navigate to Restore > Restore Wizard to start the Restore Wizard.
Step 1: Choosing Storage Location
Select backup file source to restore
1a
Click Next
1b
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At the first Restore Wizard window, you must choose the storage location from which you want to restore the file and enter the required information for the chosen storage location. These parameters are the same as you entered when you configured the backup storage location. Click Next after you enter the required data.
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Step 2: Choosing the Backup File

At the next window, select the backup file as shown in the figure.
Step 2: Choosing the Backup File
Select backup file to restore
2a
Click Next
2b
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Choose the backup file that you want to restore. To restore subsequent nodes in the cluster, you must choose the same backup file that you used to restore the first node. Click Next after you select the file.
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Step 3: Choosing Features to be Restored

The next window allows you to select the features to be restored.
Step 3: Choosing Features to be Restored
Select feature(s)
3a
Click Next
3b
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From the list of features, choose the features that you want to restore. The features displayed are the features that you chose when you backed up the system. Click Next after making your selection.
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Step 4: Choosing Nodes to be Restored

After selecting the features that you want to restore, you will see a warning that the restore will irreversibly overwrite the data destination.
Step 4: Choosing Nodes to be Restored
Start restore
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Click Restore to confirm that you want to start restoring the data. You will be prompted to choose the feature and node that you want to restore.
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Step 5: Checking Restore Status

You can check the status of an active restore process under Restore > Status.
Step 5: Checking Restore Status
Refresh to view actual status
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The status of an active restore procedure is displayed as shown in the figure. At the displayed status page, you can click the Refresh button to update the screen.
Note After the restore completes successfully, the server must be rebooted.
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Executing CLI Commands

This topic describes the CLI commands to execute a backup and restore operation.
Disaster Recovery CLI Commands

The following disaster recovery commands are available on the Cisco Unified CallManager Release 5.0 CLI:
admin:utils utils utils utils utils utils utils utils disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery disaster_recovery ? status cancel_backup show_tapeid show_registration show_backupfiles* backup* restore*
admin:utils disaster_recovery
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The Cisco DRS also provides command-line access to a subset of backup and restore functions. Normally, you will use the CLI to perform disaster recovery options only when the GUI is not accessible.
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List Tape Devices and Show Backup Files

You can use the CLI to list tape devices and show backup files.
List Tape Devices and Show Backup Files

admin: utils disaster_recovery show_tapeid
Displays a list of available tape devices on the computer running the Master Agent Device name must be used when backing up or restoring from tape The format of the name will be /dev/nst0.
admin: utils disaster_recovery show_backupfiles tape tapeid utils disaster_recovery show_backupfiles network path servername userid
Lists all backup sets on tape or network SFTP server The output of this command is required by the restore command
When you back up or restore from tape, you need the tape identification name. You can obtain tape information with the utils disaster_recovery show_tapeid command. When you restore from tape or a network server, you are prompted for the name of the existing backup file. You can display the names of the existing backup files with the utils disaster_recovery show_backupfiles command.
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Backup
The utils disaster_recovery backup command starts a manual backup by using the configuration set up in the Cisco Disaster Recovery System interface.
Backup
admin: utils disaster_recovery backup tape tapeid utils disaster_recovery backup network path servername userid
Initiates a backup of all features to tape or network SFTP server. The <tapeid> parameter is obtained from the show_tapeid command. When backing up to the network, the CLI will ask the user for a password.
admin: disaster_recovery cancel_backup Y
Cancels an active backup after the current component completes its backup
The tapeid parameter is obtained from the show_tapeid command. When backing up to the network, the CLI will prompt the user to provide a password. You cannot supply the password from the command line. The utils disaster_recovery cancel_backup command will cancel an active backup after the current component completes its backup.
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Restore
The utils disaster_recovery restore command is used to start a restore.
Restore
admin: utils disaster_recovery restore tape server tarfile tapeid utils disaster_recovery restore network server tarfilename path servername userid
This command is used to restore all features to a specific server. The tapeid parameter can be obtained from the show_tapeid command. The first server parameter indicates which server should be restored.
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The utils disaster_recovery restore command requires parameters for backup location, filename, and nodes to restore.
Note For debugging purposes, backup and restore logs are stored at /common/drf/log as follows: - Overall backup and restore log is <yyyy-mm-dd-hh-mm-ss>_[b/r].log - Component log name format: <yyyy-mm-dd-hh-mmss>_[b/r]_server_feature_component.log where b = backup and r = restore
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Status
The utils disaster_recovery status command is used to obtain status information for active operations.
Status
admin: utils disaster_recovery status restore utils disaster_recovery status backup
Displays current backup or restore status.
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The utils disaster_recovery status command requires the parameter restore to see the status of a restore operation and backup to see the status of a backup operation.
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Summary
Summary
Cisco Unified CallManager Release 5.0 uses the Cisco Disaster Recovery System, a distributed architecture, for backup and recovery purposes. You can perform an immediate or scheduled backup of select features using the Backup menu, and check backup status and history. You can perform a restore of one node or all nodes, select features to restore using the Restore menu, and check restore status and history. The CLI offers backup and restore commands in case the GUI is not operational.
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References
For additional information, refer to these resources: Disaster Recovery System Administration Guides for Cisco Unified CallManager Administration Release 5.0: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/drs/index.htm
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Module Summary
Summary
Cisco IP Telephony Platform Administration is used to configure system settings and to perform the initial setup of the Cisco Unified CallManager appliance. CLI is used if there is a problem accessing GUI. General administration of Cisco Unified CallManager is done using the Cisco Unified CallManager Administration GUI. MLA allows assigning different administration privileges to users. Basic Cisco Unified CallManager settings include activation of services on Cisco Unified CallManager servers, and global configuration settings such as service and enterprise parameters. Cisco Unified CallManager uses the integrated Disaster Recovery System for backup and restore operations.
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This module describes Cisco Unified CallManager Administration, including Cisco IP Telephony Platform Administration and use of the CLI and the GUI. The module describes how to access the Cisco Unified CallManager Administration, lists all available menu items and describes Cisco Unified CallManager multilevel administration. In addition, the module describes Cisco Unified CallManager service activation, enterprise and service parameters, and the Cisco Unified CallManager Disaster Recovery System.
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Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and solutions are found in the Module Self-Check Answer Key. Q1) Which two of these functions are supported by Cisco IP Telephony Platform Administration? (Choose two.) (Source: Administering the Cisco IP Telephony Platform) A) B) C) D) E) Q2) connecting to other network devices via Telnet switching versions pinging other network devices Cisco Unified CallManager group configuration configuring IP phones
In Cisco IP Telephony Platform Administration, the Show > _____ menu item displays information about CPU utilization, memory usage, and disk space. (Source: Administering the Cisco IP Telephony Platform) A) B) C) D) System Software Hardware Network
Q3)
NTP servers can only be added at the _____. (Source: Administering the Cisco IP Telephony Platform) Which of these statements is true about restarting the system and switching versions? (Source: Administering the Cisco IP Telephony Platform) A) B) C) D) You can switch versions on the fly. You must restart the server to switch versions. You can shut down the system in a way that it will run the other version after the next power on. System shutdown, restart, and switching versions are available in the Cisco CallManager Serviceability GUI. Trust certificates are certificates belonging to the local host. Cisco Unified CallManager knows the corresponding private keys. Trust certificates are the CA root certificates. Cisco Unified CallManager does not know the corresponding private keys. Trust certificates are certificates of other devices. Cisco Unified CallManager does not know the corresponding private keys. Trust certificates are preconfigured certificates of trust.
Q4)
Q5)
What are trust certificates? (Source: Administering the Cisco IP Telephony Platform) A) B) C) D)
Q6)
Which protocol can you use to upload custom phone rings, callback tones, and phone backgrounds to Cisco Unified CallManager? (Source: Administering the Cisco IP Telephony Platform) A) B) C) D) FTP SFTP TFTP SSH
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Q7)
When setting up remote support, what information needs to be provided to the TAC escalation or development engineer? (Source: Administering the Cisco IP Telephony Platform) A) B) C) D) account name and password account name and passphrase passphrase only password only
Q8)
What are two methods for accessing the Cisco IP Telephony Platform CLI? (Choose two.) (Source: Administering the Cisco IP Telephony Platform) A) B) C) D) E) F) Telnet HTTP HTTPS SSH Console SFTP
Q9)
Which two of these functions are supported by Cisco CallManager Administration? (Choose two.) (Source: Performing General Administration) A) B) C) D) E) call routing configuration remote installation of subsequent cluster nodes pinging other network devices user management switching versions
Q10) Q11)
The _____ is used for bulk updates. (Source: Performing General Administration) Which two of these statements are true about multilevel administration? (Choose two.) (Source: Performing General Administration) A) B) C) D) E) MLA supports different usernames for Cisco IP Telephony Platform Administration and for Cisco Unified CallManager Administration. Cisco Unified CallManager Release 5.0 MLA uses functional groups. One role can be assigned to multiple user groups. MLA uses the concepts of functions and scopes to define the user privileges. Different roles can refer to the same application.
Q12)
Standard roles and user groups _____ be deleted. (Source: Performing General Administration) How do you verify the privileges of a user? (Source: Performing General Administration) A) B) C) D) Go to the user list and click the information button of the corresponding user. Go to any group that the user is a member of. At the list of users, click the information button at the corresponding user. Go to User Management > Privilege Report and select the corresponding user from the list. The only way to verify privileges of a user is to log in with the user ID and to try to access the required functions.
Q13)
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Q14)
Which two of these are basic settings that should be performed before adding devices? (Choose two.) (Source: Configuring Cisco Unified CallManager Release 5.0 Basic Settings) A) B) C) D) E) configuration of IP addresses versus DNS usage creating partitions Cisco Unified CallManager services activation dial-plan configuration changing versions
Q15)
When using DNS, IP address changes are _____ (Source: Configuring Cisco Unified CallManager Release 5.0 Basic Settings) A) B) C) D) simpler impossible more complex required
Q16)
Only _____ services can be activated and deactivated using Cisco CallManager Serviceability. (Source: Configuring Cisco Unified CallManager Release 5.0 Basic Settings) A) B) C) D) network feature user enterprise
Q17)
Which parameters do you have to specify before you can set service parameters? (Source: Configuring Cisco Unified CallManager Release 5.0 Basic Settings) A) B) C) D) Cisco Unified CallManager server and service Cisco Unified CallManager service Cisco Unified CallManager server Cisco Unified CallManager server, service, and application resource
Q18)
Which two of these are enterprise parameters? (Choose two.) (Source: Configuring Cisco Unified CallManager Release 5.0 Basic Settings) A) B) C) D) E) Cisco Unified CallManager user web-page redirection default phone protocol for autoregistration system URLs used by phones enable Call Detail Records media services allocated to conferencing
Q19)
What are two functions of the Cisco DRS Local Agent? (Choose two.) (Source: Implementing Disaster Recovery) A) B) C) D) E) control backup and restore tasks communicate with local archive devices (tape) execute backup tasks communicate with remote archive devices (SFTP) execute restore tasks tape, network directory, and DVD DVD and network directory local file, tape, and network directory tape and network directory
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Q20)
What are valid storage locations? (Source: Implementing Disaster Recovery) A) B) C) D)
Q21)
Which features can you select for a restore operation? (Source: Implementing Disaster Recovery) A) B) C) D) CCM, CDR_CAR CCM, CDR_CAR, DCD CCM, DCD CCM, DCD_CDR
Q22)
Which command can you use to perform a backup operation using the CLI? (Source: Implementing Disaster Recovery) A) B) C) D) backup recovery recovery backup utils recovery utils disaster_recovery backup
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Q1) Q2) Q3) Q4) Q5) Q6) Q7) Q8) Q9) Q10) Q11) Q12) Q13) Q14) Q15) Q16) Q17) Q18) Q19) Q20) Q21) Q22) B, C A Publisher or first node B C C B D, E A, D Bulk Administration Tool C, E cannot B A, C A B A B, C C, E D A D
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Module 3

Overview
There are many components to consider when you place Cisco IP phone calls within a cluster or between clusters. This module describes how to configure the Cisco Catalyst switch to power Cisco IP phones and support voice traffic. The module also describes the various Cisco IP phone models in a Cisco IP telephony network and how to configure Cisco Unified CallManager to support IP phones, using either Skinny Client Control Protocol (SCCP) or Session Initiation Protocol (SIP). Because manual configuration of a large number of endpoints or users can be a time-consuming process, the module explains how to automate additions, updates, or deletions using Cisco Unified CallManager Bulk Administration Tool (BAT) to achieve faster administrative operations.
Module Objectives
Upon completing this module, you will be able to deploy SCCP and SIP endpoints in a Cisco Unified CallManager Release 5.0 cluster. This ability includes being able to meet these objectives: Implement a Cisco Catalyst switch in an IP telephony network to provide inline power to Cisco IP phones, prioritize voice traffic, and create a separate voice VLAN for IP phones Distinguish between the various Cisco IP endpoints, including IP phones, video endpoints, conference stations, and PC-based phones Configure Cisco Unified CallManager system settings, and use autoregistration and manual configuration to add Cisco IP phones to the Cisco Unified CallManager database and assign a directory number Add and configure SIP phones, convert a Cisco SCCP phone to Cisco SIP, and apply dialing rules Use Cisco Unified CallManager BAT and TAPS to bulk-add and autoregister Cisco IP phones, users, and ports in an IP telephony network
3-2
Lesson 1
Configuring Cisco Catalyst Switches

Overview
Deploying IP telephony requires planning how IP phones will be powered and how the voice network will be combined with the data network, while ensuring that data traffic does not degrade the quality of voice calls. Cisco Catalyst switches provide three features that aid an IP telephony deployment: inline power, voice VLANs, and class of service (CoS). Using the Cisco Catalyst switch to power Cisco IP phones can save on wiring costs and simplify management. Enabling multiple VLANs in a single port and placing voice packets in one VLAN and data in another VLAN saves money by reducing the number of switch ports. Extending CoS to an IP phone improves voice quality by ensuring that voice packets receive priority over data. This lesson describes the three major functions that Cisco Catalyst switches perform in an IP telephony network, and how to configure a Cisco Catalyst switch to enable these functions.
Objectives
Upon completing this lesson, you will be able to implement a Cisco Catalyst switch in an IP telephony network to provide Power over Ethernet (PoE) to IP phones, prioritize voice traffic, and create a separate voice VLAN for IP phones This ability includes being able to meet these objectives: Identify the functions that Cisco Catalyst switches perform in a Cisco IP telephony solution Describe the three options for powering Cisco IP phones Describe the two types of PoE that Cisco Catalyst switches provide Identify the Cisco Catalyst switches that support PoE Identify the commands to configure PoE on Cisco Catalyst switches Configure dual VLANs on a single port on a Cisco Catalyst switch so that the IP phones reside in a separate VLAN Configure CoS on Cisco Catalyst switches so that voice traffic has priority over data traffic as it travels throughout the network
Cisco Catalyst Switch Role in IP Telephony

This topic describes the role of Cisco Catalyst switches in the IP telephony infrastructure.
Cisco Catalyst Switch Role in IP Telephony

Supplies inline power to IP phones Supports voice and data VLANs on a single port Prioritizes voice traffic with CoS marking
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Cisco voice-capable Catalyst switches provide three primary features to assist you with your IP telephony deployment: Inline power: Inline power capabilities allow a Cisco Catalyst switch to send power through the Ethernet copper to a Cisco IP phone or other inline power-compatible device (such as wireless access points) without the need for an external power supply. Inline power is also referred to as PoE. Inline power was developed in 2000 by Cisco to support the emerging IP telephony solution. Auxiliary VLAN support: Auxiliary VLAN support allows a switch to support multiple VLANs on a single port. You can connect one or more network devices to the back of a Cisco IP phone because some Cisco IP phones have built-in switches. Auxiliary VLANs allow you to place the IP phone, and the devices that are attached through the IP phone, on separate VLANs. CoS marking: CoS marking is data link layer (Layer 2) marking that is used to prioritize network traffic. Prioritizing voice traffic is critical in IP telephony networks. If voice traffic is not given priority, poor voice quality may result because voice frames wait in the switch queue behind large data frames. The switch can use existing CoS marking to prioritize network traffic and can also classify and mark traffic that it receives.
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Powering the Cisco IP Phone

This topic describes the three options for powering Cisco IP phones.
Three Ways to Power Cisco IP Phones

Power over Ethernet (PoE): Needs PoE line cards or PoE ports for Cisco Catalyst switches Delivers 48V DC over data pairs (pins 1, 2, 3, and 6) Midspan power injection: Needs external power source equipment Delivers 48V DC over spare pairs (pins 4, 5, 7, and 8) Wall power: Needs DC converter to connect a Cisco IP phone to a wall outlet
AC Source
110 V AC Wall Power to 48 V DC Converter
Power
Power Injector No Power Power
CIPT1 v5.03-3
Most Cisco IP phone models are capable of using these three options for power: PoE: With PoE, the phone plugs into the data jack that connects to the switch, and the user PC in turn connects to the IP phone. Power-sourcing equipment (PSE), such as Cisco Catalyst PoE-capable modular and fixed-configuration switches, insert power over an Ethernet cable to a powered device, for example, an IP phone or 802.11 wireless access point. Midspan power injection: Because many switches do not support PoE, the powered device must support a midspan power source. This midspan device sits between the LAN switch and the powered device and inserts power on the Ethernet cable to the powered device. A technical difference between a midspan and inline power mechanism is that, in the midspan injection, power is delivered on the spare pairs (pins 4, 5, 7, and 8). An example of a midspan PSE is a Cisco Catalyst Inline Power Patch Panel. Wall power: Wall power needs a DC converter to connect an IP phone to a wall outlet.
Note You must order a wall power supply separately from a Cisco IP phone.
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Types of PoE Delivery

This topic describes the two types of PoE delivery that Cisco Catalyst switches can provide.
Two Types of PoE Delivery

Cisco original implementation:
Provides 48V DC at up to 6.3-to-7.7W per port over data pins 1, 2, 3, and 6. Supports most Cisco devices (Cisco IP phones and wireless access points). Uses a Cisco proprietary method of determining if an attached device requires power. Power is delivered only to devices that require power.
IEEE 802.3af Power over Ethernet:

Specifies 48V DC at up to 15.4W per port over data pins 1, 2, 3, and 6 or spare pins 4, 5, 7, and 8. Enables a new range of Ethernet-powered devices because of increased power. Standardizes the method of determining if an attached device requires power. Power is delivered only to devices that require power. Has several optional elements, including power classification.
CIPT1 v5.03-5
Cisco provides two types of inline power delivery: the Cisco original implementation and the IEEE 802.3af PoE standard. You can refer to both inline power types as PoE, as follows: Cisco original implementation PoE: Cisco was the first to develop PoE. The original Cisco (prestandard) implementation is characterized as follows: Provides 48V DC at up to 6.3 to 7.7W per port over data pins 1, 2, 3, and 6. Supports most Cisco devices (IP phones and wireless access points). Uses a Cisco proprietary method of determining whether an attached device requires power. Power is delivered only to devices that require power.
IEEE 802.3af PoE: Since first developing PoE, Cisco has been driving the evolution of this technology toward standardization by working with the IEEE and member vendors to create a standards-based means of providing power from an Ethernet switch port. The IEEE 802.3af committee has ratified this capability. The IEEE 802.3af standard supports these features: Specifies 48V DC at up to 15.4W per port over data pins 1, 2, 3, and 6 or the spare pins 4, 5, 7, and 8 (a PSE can use data pins or spare pins, but not both). Enables a new range of Ethernet-powered devices that consume additional power.
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Standardizes the method of determining whether an attached device requires power. Power is delivered only to devices that require power. The 802.3af PoE type has several optional elements, such as power classification, in which powered devices can optionally support a signature that defines the maximum power requirement. PSE that supports power classification reads this signature and budgets the correct amount of power per powered device, which will likely be significantly less than the maximum allowed power.
Without power classification, the switch reserves the full 15.4W of power for every device. This behavior may result in oversubscription of the available power supplies, so that some devices will not be powered even though there is sufficient power available. Power classification defines these five classes: 0 (default): 15.4W reserved 1: 4W 2: 7W 3: 15.4W 4: Reserved for future expansion All Cisco IEEE 802.3af-compliant switches support power classification. The Cisco Power Calculator is an online tool that enables you to calculate the power supply requirements for a specific PoE configuration. The Cisco Power Calculator is available to registered Cisco.com users at http://www.cisco.com/en/US/netsol/ns340/ns394/ns147/ns412/networking_solutions_package.h tml.
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Cisco Prestandard Device Detection

The figure illustrates how a Cisco prestandard Catalyst switch detects a Cisco IP phone, wireless access point, or other inline power-capable device.
Cisco Prestandard Device Detection
Pin3
Cisco Prestandard Implementation Powered Device Port Rx FLP Tx
FLP Switch It is an inline device.
Pin6 Pin1 Pin2
CIPT1 v5.03-6
When a switch port that is configured for inline power detects a connected device, the switch sends an Ethernet Fast Link Pulse (FLP) to the device. The Cisco powered device (IP phone) loops the FLP back to the switch to indicate its inline power capability. The switch then delivers 48V DC PoE (inline) power to the IP phone or other inline power-capable endpoint.
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IEEE 802.3af Device Detection

The figure illustrates how a Cisco Catalyst IEEE 802.3af-compliant switch detects a Cisco IP phone, wireless access point, or other inline power-capable device.
IEEE 802.3af Device Detection
IEEE 802.3af PSE
IEEE 802.3af Powered Device
2.8V to 10V
Pin3 Pin6 Pin1 Pin2
Switch
It is an IEEE powered device.
Detect Voltage
Rx Tx
25K Ohm Resistor
CIPT1 v5.03-7
The PSE, Cisco Catalyst switch, detects a powered device by applying a voltage in the range of 2.8V to 10V on the cable and then looks for a 25K ohm signature resistor. Compliant powered devices must support this resistance method. If the appropriate resistance is found, the Cisco Catalyst switch delivers power.
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Cisco Catalyst Family of PoE Switches

This topic describes the Cisco Catalyst PoE switch models.
Cisco Catalyst Family of PoE Switches

Cisco Catalyst 6500 Cisco Catalyst 4500 Cisco Catalyst 3750
Cisco EtherSwitch Network Module* Cisco Catalyst 3560
*Cisco prestandard PoE only

The Cisco Catalyst LAN switching portfolio is the industry-leading family of intelligent switching solutions delivering a robust range of security and quality of service (QoS) capabilities. The Cisco Catalyst switch portfolio allows organizations to enable new business applications and integrate new technologies such as wireless and IP telephony into their network infrastructure. These are the switches in the Cisco Catalyst family: Cisco Catalyst modular switching: The Cisco Catalyst 6500 Series delivers a 96-port 10BASE-T/100BASE-T line card and 48-port 10BASE-T/100BASE-T and 10BASE-T/100BASE-T/1000BASE-T line cards. The Catalyst 6500 Series offers a modular PoE daughter card architecture for the 96-port card and the 48-port 10/100/1000 card. The Cisco Catalyst 4500 Series delivers 48-port 10/100 and 10/100/1000 line cards. All line cards support both IEEE 802.3af and Cisco prestandard inline power. The Cisco Catalyst modular chassis switches can deliver 15.4W per port for all 48 ports on a module simultaneously. Cisco Catalyst stackable switching: The Cisco Catalyst 3750 Series offers 48- and 24-port Fast Ethernet switches that comply with IEEE 802.3af and Cisco prestandard PoE. The Cisco Catalyst 3560 Series offers 48- and 24-port Fast Ethernet switches that support both the industry standard and Cisco standard PoE.
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Cisco EtherSwitch modules: The Cisco 36- and 16-port 10/100 EtherSwitch modules for Cisco 2600, 2800, 3700, and 3800 Series routers offer branch office customers the option to integrate switching and routing in one platform. These modules can support Cisco prestandard PoE and provide straightforward configuration, easy deployment, and integrated management in a single platform. The Cisco 2600 Series requires a separate external PoE power supply; the Cisco 3700 Series integrates the power supply. This table lists the Cisco Catalyst PoE options. Catalyst Switch PoE Options
Cisco Catalyst 6500 PoE Configuration Options IEEE 802.3afCompliant Cisco Prestandard PoE 48-, 96-port 10/100 or 48-port 10/100/1000 Yes Yes Cisco Catalyst 4500 48-port 10/100 or 10/100/1000 Yes Yes Cisco Catalyst 3750 24-, 48-port 10/100 Cisco Catalyst 3560 24-, 48-port 10/100 Cisco EtherSwitch Module 16-, 36-port 10/100
Yes Yes
Yes Yes
No Yes
Note
Cisco does not offer an IEEE 802.3af midspan injection product. The 48-port Cisco Catalyst Inline Power Patch Panel supports Cisco prestandard PoE.
Note
The switches that are listed here also support multiple VLANs per port and CoS.
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Configuring PoE
This topic describes the configuration of PoE on Cisco Catalyst switches.
Cisco Catalyst Switch: Configuring PoE
Cisco CatOS: CatOS>(enable) set port inlinepower <mod/port> ? auto Port inline power auto mode off Port inline power off mode Native Cisco IOS software: CSCOIOS(config-if)# power inline <auto/never>
CIPT1 v5.03-9
Use the set port inlinepower command on a switch that is running the Cisco Catalyst operating system. The two modes are auto and off. In the off mode, the switch does not power up the port even if an unpowered IP phone is connected. In the auto mode, the switch powers up the port only if the switching module has discovered the IP phone. Examples of devices running a Cisco Catalyst operating system include the Cisco Catalyst 6500, 4500, and 4000 Series. Use the power inline command on switches that are running native Cisco IOS software (examples include the Catalyst 6500, 4500, 3750, and 3560 switches). The powered devicediscovery algorithm is operational in the auto mode. The powered device-discovery algorithm is disabled in the never mode. Other modes exist for allocating power, depending on the version of Cisco IOS software, for example, the ability to allocate power on a per-port basis with the allocation milliwatt mode.
Note The Catalyst 6500 Series can run either Cisco Catalyst operating system software or native Cisco IOS software if the switch supervisor engine has a Multilayer Switch Feature Card (MSFC). Otherwise, these switches can run only Cisco Catalyst software. The Cisco Catalyst 4500 and 4000 Series can also run Cisco Catalyst software or native Cisco IOS software, depending on the supervisor engine. Generally, late-edition supervisor engines run native Cisco IOS software; however, you should check the product documentation to determine the supervisor engine and the operating system that is supported on your specific model.
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Cisco Catalyst Switch: Show Inline Power Status

The figure shows how to display the status of inline power on a Cisco Catalyst switch.
Cisco Catalyst Switch: Show Inline Power Status

show port inline power 7 Default Inline Power allocation per port: 10.000 Watts (0.23 Amps @42V) Total inline power drawn by module 7: 75.60 Watts (1.80 Amps @42V) Port InlinePowered PowerAllocated Admin Oper Detected mWatt mA @42V -----------------------------------7/1 auto off no 0 0 7/2 auto on yes 6300 150 7/3 auto on yes 6300 150 7/4 auto off no 0 0 7/5 auto off no 0 0 7/6 auto off no 0 0 7/7 auto off no 0 0
show power inline Interface Admin -------------FastEthernet9/1 auto FastEthernet9/2 auto FastEthernet9/3 auto
Oper ---on on off
Power ( mWatt ) --------------6300 6300 0
Device -----Cisco 6500 IP Phone Cisco 6500 IP Phone n/a

CIPT1 v5.03-10
Use the show port inline power command, as shown in the figure, to view the power allocated on Cisco Catalyst switches. The switch shows the default allocated power as 10W and the inline power status of every port. The Inline Power Syntax Descriptions table provides a brief description of the syntax output. Inline Power Syntax Descriptions
Column Heading Port Inline Powered Admin Oper Power Allocated Detected mWatt mA @42V Identifies whether power is detected Identifies the milliwatts supplied on a given port Identifies the milliamps at 42V supplied on a given port (the actual voltage is 48V) Identifies the port configuration from using the set inlinepower mod/port [auto | off] command Identifies whether the inline power is operational Description Identifies the port number on the module
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Configuring Dual VLANs

This topic describes the configuration of a multi-VLAN access port.
A Multi-VLAN Access Port

An access port able to handle two VLANs Access (data) VLAN and voice (auxiliary) VLAN
Access Port Tagged 802.1Q
Untagged 802.3
CIPT1 v5.03-11
In a traditional switched scenario, all data devices typically reside on data VLANs. You may need a separate voice VLAN when you combine the voice network into the data network. The Cisco Catalyst software command-line interface (CLI) refers to this new voice VLAN as the auxiliary VLAN for configuration purposes. You can use the new voice VLAN to represent Cisco IP phones. Although you can think of it as a voice VLAN, in the future, other types of nondata devices will reside in the voice VLAN. The placement of nondata devices (such as IP phones) in a voice VLAN makes it easier for customers to automate the process of deploying IP phones. IP phones will boot and reside in the voice VLAN if you configure the switch to support them, just as data devices boot and reside in the access (data) VLAN. The IP phone communicates with the switch via Cisco Discovery Protocol when it powers up. The switch provides the telephone with the appropriate VLAN ID. You can implement multiple VLANs on the same port by configuring an access port. A tagging mechanism must exist to distinguish among VLANs on the same port. 802.1Q is the IEEE standard for tagging frames with a VLAN ID number. The IP phone sends tagged 802.1q frames. The PC sends untagged frames and the switch adds the access VLAN tag before forwarding to the network. When a switch receives a frame from the network destined for the PC, it will remove the access VLAN tag before forwarding the frame to the PC. These are some advantages to implementing dual VLANs: A dual-VLAN solution allows for the scalability of the network from an addressing perspective. IP subnets usually have more than 50 percent (often more than 80 percent) of their IP addresses allocated. A separate VLAN (separate IP subnet) to carry the voice traffic allows you to introduce a large number of new devices, such as IP phones, into the network without extensive modifications to the IP addressing scheme.
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A dual-VLAN solution allows for the logical separation of data and voice traffic, which have different characteristics. This separation allows the network to handle these two traffic types individually. A dual-VLAN solution allows you to connect two devices to the switch using only one physical port and one Ethernet cable between the wiring closet and the IP phone or PC location.
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Configuring Voice VLANs Using the Cisco Catalyst Operating System

This subtopic shows how to configure voice VLANs on Cisco Catalyst using a Catalyst operating system.
Configuring Voice VLANs Using Cisco Catalyst Operating System

Syntax: set port auxiliaryvlan mod[1 port]{vlan|untagged| dot1p|none} vlan = 1 to 1000
Example: Console>(enable) set port auxiliaryvlan 2/1-3 222 Auxiliaryvlan 222 configuration successful. AuxiliaryVlan AuxVlanStatus Mod/Ports ------------- ------------- ---------------------222 active 1/2,2/1-3
CIPT1 v5.03-12
Configure auxiliary VLAN ports in Cisco Catalyst software 5.5 and above using the set port auxiliaryvlan command to configure the auxiliary VLAN ports: set port auxiliaryvlan mod[/port] {vlan | untagged | dot1p | none} Syntax Description
Parameter mod[/port] vlan untagged dot1p Description Number of the module and (optional) ports Number of the VLAN; valid values are from 1 to 1000 Keyword to specify that the Cisco IP Phone 7960 sends untagged packets without 802.1p priority Keyword to specify that the Cisco IP Phone 7960 sends packets with 802.1p priority
The voice VLAN ID in this example is set to the value of 222 for ports 2/1 through 2/3. The switch instructs the IP phone to reside in VLAN 222 when it powers up. You can use this command to set the voice VLAN ID on a per-port basis, for a range of ports, or for an entire module.
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Configuring Voice VLANs Using Native Cisco IOS Software

This subtopic shows how to configure voice VLANs on Cisco Catalyst using native Cisco IOS software.
Configuring Voice VLANs Using Native Cisco IOS Software
Example: Console(config)#interface FastEthernet0/1 Console(config-if)#switchport mode access Console(config-if)#switchport voice vlan 261 Console(config-if)#switchport access vlan 262 Console(config-if)#spanning-tree portfast
CIPT1 v5.03-13
Use the commands shown in the figure to configure voice and data VLANs on the single-port interface of a switch that is running native Cisco IOS software. These commands apply the same functionality as setting a port to use an auxiliary VLAN on a Cisco Catalyst switch that is running Cisco Catalyst software. Catalyst Switch Voice Interface Commands
Command switchport mode access switchport voice vlan voice-VLAN_ID Description Configures the switchport to be an access (non-trunkjng) port Configures the switchport with the voice VLAN (261 in this example) to be used for voice traffic. The range is 1 to 4094. By default, the IP phone forwards the voice traffic with an IEEE 802.1Q priority of 5. Configures the interface as a static access port with the access VLAN ID (262 in this example); the range is 1 to 4094. Causes a port to enter the spanning-tree forwarding state immediately, bypassing the listening and learning states. You can use PortFast on switch ports that are connected to a single workstation or server (as opposed to another switch or network device) to allow those devices to connect to the network immediately.
switchport access vlan data_VLAN_ID spanning-tree portfast
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Verifying Voice VLAN Configuration Using Cisco CatOS

This subtopic shows how to verify voice VLAN configuration on a Catalyst switch that uses the Catalyst operating system.
Verifying Voice VLAN Configuration Using Cisco Catalyst Operating System

Console> (enable)show port auxiliaryvlan 222 AuxiliaryVlan AuxVlanStatus Mod/Ports ------------- ------------- ---------222 active 1/2,2/1-3 Console> (enable)
Console> (enable)show port 2/1 ... Port AuxiliaryVlan AuxVlan-Status ----- ------------- -------------2/1 222 active ...
CIPT1 v5.03-14
You can check the status of the auxiliary VLAN on a port or module in one of these two ways: Use the show port auxiliaryvlan vlan-id command to show the status of the auxiliary VLAN and the module and ports where it is active. Use the show port [module[/port]] command to show the module, port, auxiliary VLAN, and the status of the port.
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Verifying Voice VLAN Configuration Using Native Cisco IOS Software

This subtopic shows how to verify voice VLAN configuration on a Catalyst switch that uses native Cisco IOS software.
Verifying Voice VLAN Configuration Using Native Cisco IOS Software

Class-1-Switch#sh interfaces fa0/4 switchport Name: Fa0/4 Switchport: Enabled Administrative Mode: static access Operational Mode: static access Administrative Trunking Encapsulation: negotiate Operational Trunking Encapsulation: native Negotiation of Trunking: Off Access Mode VLAN: 262 (VLAN0262) Trunking Native Mode VLAN: 1 (default) Voice VLAN: 261 (VLAN0261) . . .
CIPT1 v5.03-15
You can verify your voice VLAN configuration on the Cisco Catalyst switches that are running native Cisco IOS software by using the show interfaces mod/port switchport command.
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Configuring CoS
This topic describes the configuration of CoS when a PC and a Cisco IP phone share the same switch port.
Extending QoS to the Cisco IP Phone

This feature allows the extension of the trusted boundary to the Cisco IP phone.
Phone VLAN = 200 CoS = 5 PC VLAN = 3 CoS = 7
Access port
IP Phone: IP Subnet B PC is not trusted CoS set to 0 (normal) PC is not trusted CoS set to 2 PC is trusted
CoS = 0 CoS = 2 CoS = 7
CoS = 7 CoS = 7 CoS = 7
Desktop PC: IP Subnet A
CIPT1 v5.03-16
CoS is a data link layer marking that you can use to classify traffic as it passes through a switch. You should ensure that voice traffic has priority as it travels throughout your network because it is extremely sensitive to delay. Cisco IP phones send all voice packets tagged with CoS 5 by default, which is the highest level of CoS that is recommended for user traffic. The multi-VLAN port also receives packets from any device (PC or workstation) that is connected to the access port of the IP phone. The attached device, if it is not in the native VLAN, can send packets with a CoS equal to or higher than the packets that are being sent by the IP phone, which can cause severe voice quality problems on your IP telephony network. An attached device can do this only if the device is not in the native VLAN. Cisco Catalyst switches have the ability to extend the boundary of trust to the IP phone. You can use the switch to instruct the IP phone to accept the CoS value of frames that are arriving from connected devices (trust) and allow the CoS to remain unchanged. Alternatively, you can choose not to trust the attached device and set the CoS to 0 or set the CoS to a configured value that you determine. The Cisco Catalyst switch uses Cisco Discovery Protocol to send configuration information to the IP phone. The switch sends an additional Cisco Discovery Protocol packet to the IP phone whenever there is a change in the CoS configuration. The switch uses its queues, which are available on a per-port basis, to buffer incoming and outgoing frames. The switch can use the CoS values to place the frames in the appropriate queues. Voice frames should be placed in the priority queue for minimal delay.
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Configuring CoS
You can configure the switch with different QoS settings on a per-port basis.
Configuring CoS
Cisco Catalyst Operating System
set port qos mod/port cos cos-value

Sets the default value for all packets that have arrived through an untrusted port
set port qos mod/ports... trust-ext {trusted | untrusted}

Allows you to trust or not trust (set to 0) the CoS assigned to the device attached to the Cisco IP phone
Native Cisco IOS
Switch(config)#interface FastEthernet0/1 Switch(config-if)#switchport priority extend {cos value | trust}

Chooses to modify, ignore, or trust the CoS of the attached device
Use the set port qos mod/port cos command to set the default value for all packets that have arrived through an untrusted port. The cos-value specifies the CoS value for a port; valid values are from 0 to 7. Seven is the highest priority. The default is 0 (not trusted). For example, the command set port qos 3/1 cos 3 sets the CoS value to 3 on port 3/1. Use the set port qos mod/ports... trust-ext {trusted | untrusted} command to either extend trust to the PC by specifying that all traffic received through the access port passes through the phone switch unchanged, or to not trust the port and change the CoS value to 0. Use the switchport priority extend interface configuration command to set a port priority for the incoming frames received by the IP phone connected to the specified port. The cos-value is used to set the IP phone port to override the priority received from the PC or attached device. Valid values for the cos-value are from 0 to 7. Seven is the highest priority. The default is 0 (not trusted). Alternatively, the trust keyword causes the IP phone port to trust the priority received from the PC or attached device; that is, to not change the CoS value. The Implementing Cisco Quality of Service (QOS) course provides more information about voice QoS theory and configuration.
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Summary
Summary
Cisco voice-capable switches support three primary feature sets that can assist with an IP telephony deployment: POE, dual VLANs, and CoS. Most Cisco IP phone models are capable of using three options for power: inline power, external power, and wall power. Two types of inline power delivery are the Cisco prestandard implementation and IEEE 802.3af PoE. These differ in the amount of power supplied, how powered devices are discovered, and optional enhancements.
CIPT1 v5.03-18
Summary (Cont.)
The Cisco Catalyst 6500, 4500, 3750, and 3560 Series switches support 802.3af and Cisco PoE, dual VLANs, and CoS. Configure inline power using the set port inline power command (Catalyst operating system) or power inline (native Cisco IOS software) command. Using dual VLANs on a single-port Cisco Catalyst switch improves network scalability when you combine a voice network into a data network. When a PC and a Cisco IP phone share the same switch port, you can use the CoS on Cisco Catalyst switch models to classify circuits so that voice packets have priority over data packets.
CIPT1 v5.03-19
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References
For additional information, refer to these resources: Cisco Multilayer LAN Switches documentation. http://www.cisco.com/univercd/cc/td/doc/product/lan/index.htm Cisco Systems, Inc., Understanding IP Phone In-Line Power Provisioning on the Catalyst 6500/6000 Switch. http://www.cisco.com/warp/public/788/AVVID/cat6k_inline_pwr.html#second
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Lesson 2
Evaluating Cisco IP Telephony Endpoints

Overview
An important task in implementing and supporting an IP telephony deployment is managing the end-user devices, or endpoints. You should be able to distinguish between various Cisco IP telephony end-user devices that you may encounter during the course of deploying and administering a Cisco IP telephony network. In addition, understanding the boot and registration communication between a Cisco IP phone and Cisco Unified CallManager is important for understanding normal voice network operations and for troubleshooting purposes. This lesson describes the various models of Cisco IP phones and how they work within a Cisco IP telephony solution. You will learn the basic features of Cisco IP phones and conference stations; the IP phone power-up and registration process; and the audio coders-decoders (codecs) that are supported by Cisco IP phones. The lesson also describes video endpoints that Cisco Unified CallManager supports for video telephony.
Objectives
Upon completing this lesson, you will be able to distinguish between the various Cisco IP endpoints that include IP phones, video endpoints, conference stations, and PC-based phones. This ability includes being able to meet these objectives: Describe the basic features of Cisco IP phones List entry-level, midrange, upper-end Cisco IP phones, and additional Cisco IP telephony endpoints Describe SIP endpoint integration into Cisco Unified CallManager Release 5.0 Compare Cisco SIP phone configuration and bootup differences and similarities to SCCP phones Explain video endpoints in Cisco Unified CallManager video telephony Identify the major audio codecs supported by Cisco IP phones
Cisco IP Phone Overview

This topic describes the basic features of Cisco IP phones.
Cisco IP Phone Overview
Cisco IP phones have the following enhancements:

Display-based User customization Inline PoE Support of the G.711 and G.729 audio codecs
CIPT1 v5.03-2
To the user, the telephone is the most visible component of the voice communications network. Cisco IP phones are next-generation, intelligent communication devices that deliver essential business communications. Fully programmable, the growing family of Cisco IP phones provides the most frequently used business features in phones. Most Cisco IP phones provide these enhancements: Display-based user interface Straightforward user customization Inline Power over Ethernet (PoE) Support for the G.711 and G.729 audio codecs Each Cisco IP phone provides toll-quality audio and does not require a companion PC. Because it is an IP-based phone, you can install it on any LAN segment in the enterprise, regardless of whether it connects to Cisco CallManager over the LAN or WAN.
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Cisco IP Phone Overview (Cont.)
CIPT1 v5.03-3
Cisco IP phones are full-featured telephones that can be plugged directly into a VoIP network and can be used very much like a standard PBX telephone. Phone terminals can attach to the existing data network infrastructure, using 10BASE-T, 100BASE-T, or 1000BASE-T interfaces on an Ethernet switch. When used with a voice-capable Ethernet switch that understands type of service (ToS) bits and can prioritize VoIP traffic, the phones eliminate the need for a traditional proprietary telephone set and key system, or PBX. The Cisco IP phone also supports an adjustable ring tone, and a hearing-aid compatible handset or a headset. The table identifies the buttons on a Cisco IP phone.
No. 1 Name Handset with indicator light LCD screen Description The light strip at the top of the handset blinks when the phone rings and also functions as a Message Waiting Indicator (MWI) when there is a voice message. Displays information about the Cisco IP phone, such as the time, date, phone number, caller ID, line and call status, and the soft key tabs. The screen is 4.25 x 3 inches (10.79 x 7.62 cm) and has an adjustable contrast. Indicates the Cisco IP phone model. Opens a new line or speed-dials the number on the LCD screen. Phones in the Cisco IP Phone 7960 Series have six line or speeddial buttons, and phones in the 7940 Series have two buttons. Allows adjustment of the angle of the phone base. Provides access to call histories and directories.
3 4
Cisco IP phone model type Line or speed-dial button
5 6
Footstand adjustment Directories button
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No. 7
Name i or ? button
Description Provides online help for selected keys or features and network statistics about the active call. Pressing the button and then the up or down scroll key displays a descriptor of the key. For example, pressing the i or ? button and then the up or down scroll key displays a screen that instructs you how to scroll up and down on the LCD. Provides access to phone settings such as contrast and ring sound, network configurations, and status information. Toggles the speaker on or off.
Settings button
Speaker button
10
Mute button
Toggles the mute on or off.
11
Headset button
Toggles the headset on or off.
12
Volume button
Increases or decreases the volume for the handset, headset, or speakerphone (depending upon which is currently active). Also controls the ringer volume (if the handset is in its cradle) and the LCD screen contrast. Provides access to available phone services.
13
Services button
14
Messages button
Provides access to a message system, if available.
15
Navigation button
Allows scrolling through text and selection of features displayed on the LCD screen.
16 17
Dial pad Soft keys
Functions exactly like the dial pad on a traditional telephone. Activates any functions displayed on the corresponding LCD screen tabs. Soft keys point to feature options displayed along the bottom of the LCD screen.
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The table shows the connections on the back of the Cisco IP phone.
No. 1 Description AC/DC adapter port (DC 48V) for power connector. For redundancy, you can use the AC adapter even if you are using inline power from Cisco Catalyst switches. Cisco IP phones use the inline power or external power source. If either the inline power or the external power goes down, the phone can switch entirely to the other power source. Power supply with AC plug. Power cable with wall socket plug for connecting to power. Network port RJ-45 connector, to connect the phone to the network supporting 10-Mbps, 100-Mbps, or 1-Gbps half- or full-duplex Ethernet connections to external devices. To avoid collisions, use full-duplex mode. You must use a straight-through cable on this port. The phone can also obtain inline power from the Cisco Catalyst switch over this connection. Access port RJ-45 connector, to connect a network device, such as a computer, to the phone supporting 10-Mbps, 100-Mbps, or 1-Gbps half- or full-duplex Ethernet connections to external devices. To avoid collisions, use full-duplex mode. You should use a straight-through cable on this port. Handset port for connecting a handset. Headset port for connecting a headset. Enables the headset. The phone supports a fouror six-wire headset jack. The volume and mute controls also adjust volume to the earpiece and mute the speech path of the headset. The headset activation key is located on the front of the phone. The phone supports the following Plantronics four- or six-wire headsets: Tristar Monaural, Encore Monaural H91, and Encore Binaural H101. When the headset is used, an amplifier is not required. However, a coil cord is required to connect the headset to the headset port on the back of your Cisco IP 7960G/7940G Phone. For information on ordering compatible headsets and coil cords for the Cisco IP phones, go to http://getheadsets.com/cisco or http://vxicorp.com/cisco.
2 3 4
6 7
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Seamless, Native Support for SCCP and SIP

You can configure an IP phone for Skinny Client Control Protocol (SCCP) or Session Initiation Protocol (SIP).
Seamless, Native Support for SCCP and SIP

Cisco Unified CallManager Release 5.0
Existing SCCP Phones Existing Cisco SIP Phones
SCCP SIP
Enhanced SCCP Phones

Third-Party SIP Phones
Advanced Cisco SIP Phones

CIPT1 v5.03-4
These Cisco Unified IP phones are controlled by Cisco Unified CallManager using SCCP: Cisco Unified IP Phone model 7900 Family Cisco Unified IP Video Phone model 7985 Cisco Unified IP Phone model 7914 Expansion Module Cisco IP Conference Station 7935 and 7936 Cisco IP Phone model 30 VIP Cisco IP Phone model 12 series Cisco IP Communicator Cisco Unified Personal Communicator Cisco Unified CallManager supports the SIP protocol on these Cisco Unified IP phone models: Cisco Unified IP Phone 7970/71 Cisco Unified IP Phone 7960/61 Cisco Unified IP Phone 7940/41 Cisco Unified IP Phone 7911 Cisco Unified IP Phone 7905/12 Cisco Unified Personal Communicator
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SCCP Phones
This topic describes entry-level, midrange, upper-end Cisco Unified IP phones, and additional Cisco IP Telephony endpoints.
Entry-Level Cisco IP Phones

Cisco Unified IP Phone 7902G Cisco Unified IP Phone 7905G Cisco Unified IP Phone 7911G Cisco Unified IP Phone 7912G
Basic-featured Cisco IP phones for low-to-medium telephone use Single line/directory number Display-based (except Cisco 7902G) Message waiting indicator
CIPT1 v5.03-5
Cisco produces several entry-level IP phones for a variety of business functions. Depending on user requirements, these IP phones function well for employees or for use in public areas such as lobbies or break rooms. Entry-level Cisco phones provide these common features: Display-based user interface (except Cisco Unified IP Phone 7902G) G.711 and G.729 codecs Single line (directory number) Cisco inline power, powered patch panel, or local power option support via a power cube (the same power supply as the Cisco Unified IP Phone 7910, 7940, or 7960) Visual Message Waiting Indicator (MWI) No speakerphone or headset port
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Here is a brief description of the major features of each entry-level Cisco IP phone: Cisco Unified IP Phone 7902G: The Cisco Unified IP Phone 7902G is a single-line, entrylevel, no-display business phone with fixed feature keys that provide one-touch access to redial, transfer, conference, and voice-mail access features. The phone has a single RJ-45 connection (no internal Ethernet switch). Cisco Unified IP Phone 7905G and Cisco Unified IP Phone 7912G: The Cisco Unified IP Phone 7905G provides single-line access and four interactive softkeys that guide you through call features and functions via the pixel-based LCD. Use this IP phone for employees who do not need a full-featured phone, or for a common area such as a hallway, manufacturing floor, break room, reception space, or office cubicle. The Cisco Unified IP Phone 7912G includes an integrated Ethernet switch that provides LAN connectivity to a colocated PC. Cisco Unified IP Phone 7905G and Cisco Unified IP Phone 7912G support SCCP, H.323v2 (Cisco Unified IP Phone 7905G only), and SIP. Cisco Unified IP Phone 7911G: The Cisco Unified IP Phone 7911G fills the communication needs for low to moderate telephone traffic, such as employees working in cubicles, retail workers, and classrooms. The Cisco Unified IP Phone 7911G supports only a single line, has two 10/100BASE-T Ethernet connections, and supports SCCP and SIP.
Note For more details on Cisco IP phones, visit Cisco.com.
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Midrange Cisco IP Phones

This subtopic describes the midrange Cisco Unified IP phones family.
Midrange Cisco IP Phones

Cisco Unified IP Phone 7940G Cisco Unified IP Phone 7941G(G-GE) Cisco Unified IP Phone 7960G Cisco Unified IP Phone 7961G(G-GE)
Full-featured Cisco IP phones for medium-to-high telephone use Multiline Large pixel-based displays Integral switches Built-in headsets and high-quality speakerphones Multiprotocol-capable (SCCP, SIP, MGCP)
Cisco designed the Unified IP Phones 7940G, 7941G (G-GE), 7960G, and 7961G(G-GE) to meet the demand for a corporate-level, full-featured IP phone for medium-to-high telephone use. Features that are common to these phones are as follows: Multiline capability Large pixel-based displays, which allow for the inclusion of eXtensible Markup Language (XML) and future features Integrated two-port 10/100-Mbps Ethernet switches Built-in headset connections and quality full-duplex speakerphones (phones do not come with a headset) Information key for online help with features A minimum of 24 user-adjustable ring tones Adjustable foot stand (flat to 60 degrees) and basic or wall mounting SCCP, Media Gateway Control Protocol (MGCP), and SIP support XML service support An EIA/TIA-232 port for options, such as line expansion and security access
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Additional details of these phone models follow: Cisco Unified IP Phone 7940G: The Cisco Unified IP Phone 7940G is for medium traffic. Cisco Unified IP Phone 7941G: The Cisco Unified IP Phone 7941G is a full-featured enhanced business IP phone that addresses communication needs of sales professionals. The Cisco Unified IP Phone 7941G can have up to two directory numbers and includes two 10/100BASE-T Ethernet connections, and supports SCCP and SIP. This phone includes a high-resolution, four-bit grayscale display for enhancing feature usage and XML applications, as well as for enabling support for double-byte languages. Cisco Unified IP Phone 7941G-GE:The Cisco Unified IP Phone 7941G-GE can have up to two directory numbers and is the equivalent of the Cisco Unified IP Phone 7941G with the exception that it includes two 10/100/1000BASE-T Ethernet connections. The addition of gigabit throughput capability allows for high bit-rate and bandwidth-intensive applications on a colocated PC. Cisco Unified IP Phone 7960G: The Cisco Unified IP Phone 7960G continues to address the communications needs of the professional worker in an enclosed office environment with a high or busy amount of phone traffic. It has access to six telephone lines or a combination of lines and direct access to telephony features. Cisco Unified IP Phone 7961G: The Cisco Unified IP Phone 7961G is a full-featured, enhanced IP phone designed to meet the needs of managers and administrative assistants. A built-in headset port and an integrated Ethernet switch are standard with the Cisco Unified IP Phone 7961G. The phone also features a best-of-class large, high-resolution grayscale pixel-based LCD. The crisp graphic capability allows for the inclusion of high-value XML applications and double-byte languages. Cisco Unified IP Phone 7961G-GE: Providing unconstrained bandwidth to desktop applications, the Cisco Unified IP Phone 7961G-GE delivers the latest technology in Gigabit Ethernet VoIP telephony, bringing network data with its Gigabit Ethernet port for integration with a PC or desktop server. The features are identical to the Cisco Unified IP Phone 7961G.
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Upper-End Cisco IP Phones

This subtopic describes the upper-end Cisco Unified IP phones family.
Upper-End Cisco IP Phones

Cisco Unified IP Phone 7970G Cisco Unified IP Phone 7971G-GE
Addresses needs of executives Large color touch-sensitive, pixel-based displays bring applications in full, vivid colors to users without PCs Eight telephone lines, or combinations of lines and direct access to telephony features Five interactive softkeys Built-in headsets and high-quality, hands-free speakerphones Multiprotocol-capable (SCCP, SIP, MGCP)
Here is a brief description of the major features of upper-end Cisco IP phones: Cisco Unified IP Phone 7970G: The Cisco Unified IP Phone 7970G is a prestigious device exhibiting the latest technology and advancements in IP telephony. The Cisco Unified IP Phone 7970G addresses both the needs of the executive and brings network data and applicationsin full, vivid colorto users without PCs. Eight telephone lines, a highquality, hands-free speakerphone, and a built-in headset connection are included. With both a color display and touch screen, the Cisco Unified IP Phone 7970G delivers powerful applications and network data to the desktop. Cisco Unified IP Phone 7971G-GE: The first phone to provide unconstrained bandwidth to desktop applications, the Cisco Unified IP Phone 7971G-GE delivers the latest technology and advancements in Gigabit Ethernet VoIP telephony. It addresses the needs of an executive or major decisionmaker, and brings network data and applications to users quickly with its Gigabit Ethernet port for integration to a PC or desktop server. This stateof-the-art Gigabit Ethernet IP phone is feature-identical to the Cisco Unified IP Phone 7970G.
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Additional Cisco IP Telephony Endpoints

This subtopic describes the features and functions of additional Cisco IP telephony endpoints.
Additional Cisco IP Telephony Endpoints
Cisco Unified IP Phone 7985G
Cisco Unified IP Conference Station 7936
Cisco IP Communicator
Cisco ATA 186

Cisco Unified 7914 Expansion Module
Cisco Unified Wireless IP Phone 7920

CIPT1 v5.03-8
Cisco provides a complete portfolio of IP endpoints to meet business needs for conferencing, wireless voice communications, PC-based voice calls, and connecting analog phones to the VoIP network. These products are as follows: Cisco Unified IP Phone 7985G: The Cisco Unified IP Phone 7985G is a personal desktop video phone for the Cisco Unified Communications system. Offering executives and managers a productivity-enhancing tool that makes instant, face-to-face communication possible from their offices, the Cisco Unified IP Phone 7985G has all the components to enable a video callcamera, LCD screen, speaker, keypad, and a handsetincorporated into one easy-to-use unit. Cisco Unified IP Conference Station 7936: The Cisco Unified IP Conference Station 7936 is a full-featured, IP-based, full-duplex, hands-free conference phone for use on desktops, in offices, and in small-to-medium conference rooms. The Cisco Unified IP Conference Station 7936 offers external microphone ports, an optional external microphone kit, audio-tuned speaker grill, and a backlit LCD display.
Note For more details on additional Cisco IP telephony endpoints, see Cisco.com.
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Cisco IP Communicator: The Cisco IP Communicator is a Microsoft Windows softwarebased application that delivers enhanced telephony support through personal computers. This application provides computers with the functionality of Cisco IP phones and provides high-quality voice calls on the road, in the office, or from wherever users may have access to the corporate network. Cisco Analog Telephone Adaptor (ATA) 186: The Cisco ATA 186 interfaces regular telephones with your IP-based telephony network. This adapter is useful for customers that have existing analog devices, such as fax machines or telephones that they do not want to replace after they have migrated to VoIP. The Cisco ATA 186 provides two voice ports, each with its own independent telephone number, and a single 10BASE-T Ethernet port for network connectivity. Cisco Unified IP Phone 7914 Expansion Module: The Cisco Unified IP Phone 7914 Expansion Module extends the capabilities of the Cisco Unified IP Phone 7960 with additional buttons and an LCD display. The Cisco Unified IP Phone 7914 helps administrative assistants and others who must manage a number of calls beyond the six-line capability of the Cisco Unified IP Phone 7960. This expansion module enables you to add 14 buttons to the existing 6 buttons of the Cisco Unified IP Phone 7960, increasing the total number of buttons to 20 with one module or 34 with two modules. You can use up to two Cisco Unified IP Phone 7914 Expansion Modules with a Cisco Unified IP Phone 7960. Cisco Unified Wireless IP Phone 7920: The Cisco Unified Wireless IP Phone 7920 is an easy-to-use IEEE 802.11b wireless IP phone that provides comprehensive voice communications in conjunction with Cisco Unified CallManager and Cisco access points.
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SIP Endpoints
This topic describes SIP endpoint integration into Cisco Unified CallManager.
Overview of SIP Phone Models
Cisco Unified IP Phone 7905/7912
Cisco Unified IP Phone 7940/7960
Cisco ATA 186
Third-Party SIP phones
Basic SIP phones

Cisco SIP Phone 7911 Cisco SIP Phone 7941/7961
Very basic SIP phones

Cisco SIP Phone 7970/7971 Unified IP Communicator
Cisco Unified Cisco Unified Cisco Unified Cisco Unified IP IP Phone 7911 IP Phone 7941/7961 IP Phone 7970/7971 Communicator
Advanced SIP phones

Cisco Unified CallManager Release 5.0 has the ability to allow all of the types of SIP phones shown in the figure to interoperate and register to the same Cisco Unified CallManager. There are three categories of SIP phones that Cisco Unified CallManager 5.0 supports, as follows: Very basic SIP phones support the minimum features for SIP compliance outlined in RFC 3261, the core SIP standard. This category also includes third-party soft SIP phones. Basic SIP phones are the Cisco Unified IP Phones 7905, 7912, 7940, and 7960. These phones support the core SIP standards plus several additional features. You can migrate these phones from SCCP to SIP, but when you do, you lose some features because these phones do not have enough memory to run all of the features that the newer advanced Cisco SIP phones have. Advanced SIP phones support all the features of the basic SIP phones, but use SIP extensions to try to achieve feature parity with SCCP phones. (In Cisco Unified CallManager 5.0, feature parity is nearly achieved. The missing features are covered later). SIP was designed to be highly extensible. The Cisco implementation of SIP is 100 percent standards-compliant. All protocols leave certain areas open to interpretation or are vendor implementation-specific. In the case of SIP, Cisco implements certain extensions to the SIP protocol to achieve SCCP feature parity. In some cases, Cisco has driven these improvements back into the standards community, and in other cases, these features are specific to Cisco Unified CallManager environments and remain Cisco intellectual property.
Note Cisco IP phones are designed to recognize whether they are attached to a Cisco Unified CallManager, or automatically fall back to basic standards-only operation mode.
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SIP Endpoints: Basic SIP

This subtopic describes basic SIP features and capabilities of Cisco SIP IP phones.
SIP Endpoints: Basic SIP

Minimum required features for Cisco Unified IP Phones 7905 and 7912 are available with firmware version 8.0:
Registration redundancy Failover to SRST Reset/restart from Cisco Unified CallManager Configuration and local dial plan provisioned from Cisco Unified CallManager administration pages Encrypted configuration files Call preservation
CIPT1 v5.03-10
Cisco introduces new phone firmware on the Cisco Unified IP Phones 7905 and 7912. The firmware release 1.3(x) is renumbered 8.0 for consistency. The 8.0 firmware release adds the minimum required features to operate in a Cisco Unified CallManager environment using the SIP standard RFCs, and the following extensions to the SIP standard that Cisco has implemented: Registration redundancy to register up to five Cisco Unified CallManager systems including Survivable Remote Site Telephony (SRST) Failover to SRST if Cisco Unified CallManager is not reachable, due to network issues, system outage, and so on Reset and restart from Cisco Unified CallManager to make configuration changes take effect Entire configuration and dial-plan provisioning completed on Cisco Unified CallManager administration page Encrypted configuration files to enhance security within the network Call preservation to provide call stability
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SIP Endpoints: Basic SIP (Cont.)

These required features for Cisco Unified IP Phones 7940 and 7960 are available with software version 8.0:
Enhancement to release 7.5: Version adds better RFC compliance (3261 (SIP), 3264 (Offer/Answer), and 3311(UPDATE)) Registration redundancy Reset/Restart from Cisco Unified CallManager Failover to SRST Configuration and local dial plan provisioned from Cisco Unified CallManager administration pages Encrypted configuration files
CIPT1 v5.03-11
Firmware version 8.0 for Cisco Unified IP Phones 7940 and 7960 includes more functionality than the firmware for Phones 7905 and 7912. The version 8.0 SIP firmware release adds the minimum required features to operate in a Cisco Unified CallManager environment, and also improves the following: Compared to the SIP version 7.5, version 8.0 adds better RFC compliance to better support third-party SIP proxy and SIP registrar servers. Those updates include enhancements for RFC 3261 (SIP), RFC 3264 (An Offer/Answer Model), and RFC 3311 (UPDATE Method). Registration redundancy to register up to five Cisco Unified CallManager systems including SRST. Failover to SRST if Cisco Unified CallManager is unreachable. Reset and restart from Cisco Unified CallManager to apply configuration changes. Entire configuration and dial plan provisioning is done on Cisco Unified CallManager administration page. Encrypted configuration files to enhance security within the network.
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SIP Endpoints: Basic SIP (Cont.)

These required features for Cisco Unified IP Phones 7940 and 7960 are available with software version 8.0:
Limited scope compared to the Cisco Unified IP Phones 7911, 7941, 7961, 7970, 7971 models SIP implementation: Music on Hold Display name updates via Remote-Party-ID End-user feature interaction and user operation is different between SCCP and SIP on the Cisco Unified IP Phones 7940 and 7960: Look and Feel of existing firmware 7.5.
CIPT1 v5.03-12
Compared to the Cisco Unified IP Phones 7970 and 79x1 SIP implementations, the Cisco Unified IP Phones 7940 and 7960 SIP implementation is limited, for example, with music on hold (MOH) support and display names update via Remote-Party-ID. For an end user, the interaction and user operation is different between the SIP phone software version and the SCCP software version. The SIP phone version has the same look and feel as the SIP firmware version 7.5 supported with the Cisco Unified CallManager Release 4.1.
Note Differences between SCCP and SIP on the Cisco Unified IP Phones 7940 and 7960 will be discussed later.
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SIP Endpoints: Very Basic SIP

This subtopic describes the Cisco Unified CallManager support for very basic SIP features on third-party SIP IP phones.
SIP Endpoints: Very Basic SIP

Third-party SIP phones:
Cisco Unified CallManager Release 5.0 supports Cisco SIP phones as well as third-party SIP phones such as the following:
Hardphones Softphones
CIPT1 v5.03-13
Cisco anticipates some of its customers seeking to integrate certain third-party SIP-based products with their Cisco IP Communication call control systems. Understanding the desire of customers to integrate these third-party products with Cisco Unified CallManager, Cisco is participating in an independent third-party testing and interoperability verification process being offered by tekVizion. This independent service has been established to enable third-party vendors to test and verify the interoperability of their endpoints with Cisco Unified CallManager. Because SIP is an Internet Engineering Task Force (IETF) RFC, and therefore a global standard, Cisco now also supports third-party SIP phones, including hardphones and softphones.
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SIP Endpoints: Advanced SIP

This subtopic describes advanced SIP features of Cisco Unified IP phones.
SIP Endpoints: Advanced SIP

SIP firmware Version 8.0 introduces SIP support on the Cisco Unified IP Phones 7911, 7941, 7961, 7970, 7971: Java architecture currently used in the SCCP phones was leveraged for its user interface control and platform infrastructure. Cisco Unified IP Phones 7940 and 7960: SIP stack and call feature is ported and enhanced for the new phones. From an end-user perspective, feature interaction and user interface operation is nearly identical between SCCP and SIP: Majority of the SCCP features have been ported to SIP on the Cisco Unified IP Phones 7911, 7941, 7961, 7970, 7971
CIPT1 v5.03-14
The firmware version 8.0 introduces SIP support on the Cisco Unified IP Phones 7911, 7941, 7961, 7970, and 7971. The most extensive feature implementation for Cisco SIP phones is included in the SIP software version for 7970 and 79x1 Phone models. A portable Java software architecture provides user interface and feature consistency across SCCP and SIP on these phone models. From an end-user perspective, feature interaction and user interface operation are nearly identical between SCCP and SIP on the Cisco Unified IP Phones 7911, 7941, 7961, 7970, and 7971. The feature deficit of SIP versus SCCP is minimal. As mentioned, the majority of SCCP features have been ported to SIP on the Cisco Unified IP Phones 7911, 7941, 7961, 7970, and 7971 (100 percent parity in a future release).
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SIP Phone Feature Issues

This subtopic describes general Cisco SIP IP phone features.
SIP Phone Feature Issues

What is not supported with the Cisco Unfied IP Phones 7905, 7912, 7940, and 7960?
Anything requiring CMXML 3.1 (IPPM, IPMA, EM) Digit-by-digit dialing using Key Pad Markup Language The 7914 sidecar module Cisco Unified Video Advantage CTI call control Configurable softkeys Ad-Hoc Conference B2BUA mode (external conferencing resources) User interface consistency with the SCCP phones
CIPT1 v5.03-15
These functionalities are not supported with the Cisco SIP phone firmware version 8.0 on Cisco Unified IP Phones 7905, 7912, 7940, and 7960: Anything that requires Content Manager XML (CMXML) 3.1, such as IP Phone Messenger (IPPM), Cisco IP Manager Assistant (IPMA), or extension mobility Digit-by-digit dialing using Keypad Markup Language Cisco Unified IP Phone 7914 Expansion Module Cisco Unified Video Advantage (CVTA) video telephony Computer telephony integration (CTI) call control from external applications Configurable softkeys Ad-Hoc conference back-to-back user agent (B2BUA) mode (external conferencing resources) No user interface consistency with the SCCP phones
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SIP and SCCP Phone Configuration Comparison

This topic compares Cisco SIP phones to SCCP phones in terms of configuration and bootup.
Cisco SCCP IP Phone Startup Process

DHCP Cisco Unified CallManager Cisco TFTP
6 4 1 2
1. 2. 3. 4. 5. 6. Cisco IP phone obtains power from the switch Cisco IP phone loads locally stored image Switch provides VLAN information to Cisco IP phone Phone sends DHCP request; receives IP information and TFTP server address Cisco IP phone gets configuration from TFTP server Cisco IP phone registers with Cisco Unified CallManager server
CIPT1 v5.03-16
When connecting to the VoIP network, the Cisco Unified IP phone goes through a standard startup process consisting of several steps. Depending on your specific network configuration, not all of these steps may occur on your Cisco Unified IP phone:
Step 1
Obtaining Power from the Switch. You can connect the Cisco Unified IP phone to a Cisco Catalyst switch with one of the modules that provides power to the phone. If you use this optional configuration, the phone receives phantom power and powers up when you connect the Cisco Unified IP phone to the switch. The phone then sends Cisco Discovery Protocol (CDP) notifications to the switch indicating that it is ready to receive CDP packets and indicating the power requirement for the phone. The switch allocates power and sends it over the network cable. Loading the Stored Phone Image. The Cisco Unified IP phone has nonvolatile Flash memory in which it stores firmware images and user-defined preferences. At startup, the phone runs a bootstrap loader that loads a phone image stored in Flash memory. Using this image, the phone initializes its software and hardware. Configuring VLAN. If the Cisco Unified IP phone is connected to a Cisco Catalyst switch, the switch next informs the phone of the voice VLAN defined on the switch. The phone needs to know its VLAN membership before it can proceed with the Dynamic Host Configuration Protocol (DHCP) request for an IP address. Obtaining an IP Address. If the Cisco Unified IP phone is using DHCP to obtain an IP address, the phone queries the DHCP server to obtain an IP address. DHCP also informs the IP phone about how to reach the TFTP server (DHCP Option 150). If you are not using DHCP in your network, you must assign a static IP address and TFTP server address to each IP phone locally.
Step 2
Step 3
Step 4
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Step 5
Requesting the Configuration File and the Profile File. The TFTP server has configuration files and profile files. A configuration file includes parameters for connecting to Cisco Unified CallManager and information about which image load a phone should be running. A profile file contains various parameters and values for phone and network settings. The IP phone requests first its SEP<mac>.cnf.xml file from the TFTP server. If the SEP<mac>.cnf.xml file is not found on the server, then the phone requests the XMLDefault.cnf.xml file. From that file, the IP phone obtains its list of Cisco Unified CallManagers and then registers to the primary server. Registering on Cisco Unified CallManager. The configuration file defines how the Cisco Unified IP phone communicates with Cisco Unified CallManager. After obtaining the file from the TFTP server, the phone attempts to make a TCP connection to the highest priority Cisco Unified CallManager on the list. If the phone was manually added to the database, Cisco Unified CallManager identifies the phone. If the phone was not manually added to the database and autoregistration is enabled in Cisco Unified CallManager, the phone attempts to autoregister itself in the Cisco Unified CallManager database. Cisco Unified CallManager informs devices using .cnf or .cnf.xml format configuration files of their load ID. Devices using .xml format configuration files receive the load ID in the configuration file.
Step 6
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Cisco SIP Phone Startup Process

This subtopic describes the startup process of a Cisco SIP IP phone.
Cisco SIP Phone Startup Process

Cisco SIP phone Cisco TFTP Cisco Unified CallManager
1. CTL File (If present) 2. SEP<mac>.cnf.xml 3. XMLDefault.cnf.xml 4. Loads File 5. Dial rules (Optional) 6. Establish Connection 7a. Register 7b. 200 OK 8. Localization Files 9. Soft Keys 10. Custom Ringers
CIPT1 v5.03-17
The steps the SIP phone takes to get its power from the switch and its IP address information from the DHCP router are the same as the steps for a SCCP-controlled IP phone. These steps occur after an SIP phone obtains an IP address and information about how to reach a TFTP server:
Step 1
The SIP phone boots and performs Certificate Trust List (CTL) file processing. The CTL file contains a set of self-signed certificates containing public keys for all the servers that the phone is expected to trust. Autoregistration will work only if the Cisco Unified CallManager is not running in secure mode. The SIP phone requests its SEP<mac>.cnf file from the Cisco TFTP server. If a SIP phone is new, this file will not be found, because the phone is not currently configured in the Cisco Unified CallManager database. The SIP phone downloads the default configuration file XMLDefault.cnf.xml from the TFTP server. This configuration file contains system-wide configuration parameters, including the location of the Cisco Unified CallManager of the SIP phone. For autoregistration to work for SIP, this file also contains a parameter called auto_registration_name. If this parameter is blank, then the SIP phone will not attempt to autoregister. If this parameter is not blank, the SIP phone will attempt to autoregister if it finishes the boot sequence and still does not have any legitimate directory number lines configured. The SIP phone requests the .Loads file, if one was specified in the default configuration file, to see what image the phone should be running. If the .Loads file specifies an image that is different from the one contained in the SIP phone, the SIP phone attempts to obtain the new images from the Cisco TFTP server. If the image is downloaded and verified successfully, the SIP phone will reboot.
Step 2
Step 3
Step 4
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Step 5
The next step is to establish the connection to the Cisco Unified CallManager working as a SIP proxy. The default SIP configuration file indicates whether the SIP phone should connect using User Datagram Protocol (UDP) or TCP. If the SIP phone does not have any directory number lines provisioned, but it does have the Cisco Unified CallManager IP address and port, the phone will check the auto_registration_name parameter. If the parameter contains a name, that name is used as the directory number line in the SIP Register message sent to the SIP proxy. Upon receiving this message, the Cisco Unified CallManager should do the following: Identify that the special autoregistration name has been used. Create an entry in the database for the new phone based on the current autoregistration settings. Generate the SEP<mac>.cnf.xml file for the new phone. Accept the registration with the 200 OK Response. Reset the registered phone using the reset notify mechanism. The phone will automatically reset and reboot.
Step 6
The bootup procedure for the legacy Cisco SIP phones (Cisco Unified IP Phones 7960 and 7940) is slightly different from the procedure used by the advanced Cisco SIP phones that is shown on the figure. The legacy Cisco SIP phones first download the SIPdefault.cnf file. This file contains the default configuration parameters shared by all SIP phones that use this TFTP server. Then, the Cisco SIP phone continues requesting the SIP<mac>.cnf file.
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Boot Sequence Differences Between Cisco SCCP and SIP Phones

This subtopic lists the differences between a Cisco SCCP bootup and Cisco SIP phones bootup.
Boot Sequence Differences Between Cisco SCCP and SIP Phones

The boot sequences for SIP are very similar to SCCP. There are three main differences:
SEP<mac>.cnf.xml: The SIP phones get all of their configuration from the config file. Therefore, the SEP<mac>.cnf.xml file is much larger for SIP than for SCCP. Dialplan file (optional): The SIP phones can download and use local dial plans. Softkey file: The SIP phones download their softkey sets in this XML file.
CIPT1 v5.03-18
The boot sequences for SIP phones are very similar to those used for SCCP phones. There are these three main differences: SEP<mac>.cnf.xml: The SIP phones get all of their configuration from the config file. Therefore, the SEP<mac>.cnf.xml file is much larger for SIP than for SCCP. Dialplan file (optional): The SIP phones can download and use local dial plans. Softkey file: The SIP phones download their softkey sets in this XML file.
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Third-Party SIP Phone Startup Process

This subtopic describes the startup process of third-party SIP phones to work with Cisco Unified CallManager.
Third-Party SIP Phone Startup Process

Third-party SIP Phone Cisco Unified CallManager
SIP REGISTER Authorization: Digest username=mperry",realm="ccmsipline, nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5", uri="sip:172.18.197.224",algorithm=MD5, response="126c0643a4923359ab59d4f53494552e" 1. 2. 3. SIP OK mperry looked up in the database mperry associated with the SIP phone configuration of this SIP phone
CIPT1 v5.03-19
You do not configure third-party SIP phones by using the Cisco Unified CallManager TFTP server. Instead, configure them by using the native phone configuration mechanism, which is usually a web page or a TFTP file. You must keep the device and line configuration in the Cisco Unified CallManager database synchronized with the native phone configuration (for example, extension 1002 on the phone and 1002 in Cisco Unified CallManager). Additionally, if the directory number of a line is changed, you must ensure that it is changed in both Cisco Unified CallManager Administration and in the native phone configuration mechanism. Third-party SIP phones do not send a MAC address; they must identify themselves by using digest authentication. For this purpose, the SIP REGISTER message includes the following header:
Authorization: Digest username=mperry,realm=ccmsipline,nonce=GBauADss2qoWr6k9y3 hGGVDAqnLfoLk5,uri=sip:172.18.197.224,algorithm=MD5,respons e=126c0643a4923359ab59d4f53494552e
To enable digest authentication for third-party SIP phones, you must create a SIP phone Security Profile. On the SIP phone Security Profile Configuration window, check the Enable Digest Authentication check box. After the security profile is configured, you must assign that security profile to the SIP phone by using the Phone Configuration window. If the Enable Digest Authentication check box is not checked, Cisco Unified CallManager will use digest authentication for purposes of identifying the phone by the end user ID (mperry), and it will not verify the digest password. If the check box is checked, Cisco Unified CallManager will verify the password.
Note More information on IP phone security is provided in the CIPT2 course.
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SIP Phones to Cisco Unified CallManager Interaction

The table provides a comparison overview of the configuration differences between Cisco SIP IP phones and third-party SIP phones.
SIP Phones to Cisco Unified CallManager Interaction

Integrated Sends with MAC CTFTP address Cisco SIP Phone 7905/12 Cisco SIP Phone 7940/60 Cisco SIP Phone 797x, 7941, 7961 Third-Party
1
Downloads softkey file
Downloads dialplan file
Supports Cisco Unified CallManager failover/ fallback
Supports Reset/ Restart
Yes Yes
Yes Yes
No No
Yes1 Yes
Yes Yes
Yes Yes
Yes No
Yes No2
Yes No
Yes No
Yes No2
Yes No2
2 In
Part of the configuration file. the future, third-party phones may add this support. Failover can be supported today via DNS SRV.
CIPT1 v5.03-20
When you configure third-party SIP phones, the Cisco Unified CallManager database is updated when you use Cisco Unified CallManager Administration. You must also perform these configuration checks on the third-party SIP phone: Proxy address in the phone should be the IP address or the fully qualified domain name (FQDN) of Cisco Unified CallManager Directory numbers in the phone should match the directory numbers that are configured for the device in Cisco Unified CallManager Administration Digest user ID (sometimes referred to as Authorization ID) in the phone should match the Digest User ID in Cisco Unified CallManager Administration Consult the documentation that came with the third-party SIP phone for more information.
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SIP and SCCP Configuration Similarities

This subtopic compares Cisco SIP phone configuration and bootup to SCCP phones.
SIP and SCCP Configuration Similarities

SIP phones are configured similarly in Cisco Unified CallManager Administration to SCCP phones (devices and lines). The protocol is specified when a new phone type is added. Cisco SIP phones can be reset and restarted from the Phone Administration page (but not third-party phones). Much of the device configuration is the same. All of the line configuration is the same.
CIPT1 v5.03-21
The configuration of SIP and SCCP phones in Cisco Unified CallManager Administration is very similar from a device and line standpoint. When you add a phone, you specify the protocol type (SCCP or SIP) after you specify the phone model. You can reset and restart Cisco SIP phones from the phone administration page, as with SCCP phones. However, third-party phones do not understand the reset and restart commands, so you will need to reset these phones manually (until a SIP mechanism is created to reset and restart phones).
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SIP and SCCP Configuration Differences

Some minor configuration differences exist between Cisco SIP and SCCP phones.
SIP vs. SCCP Configuration Differences
SIP Phones All configuration via TFTP (device pool, line configuration, softkey, dial plan) Can have local dial plan
SCCP Phones Configuration via TFTP (Cisco Unified CallManager group, SRST, load information), and softkey templates from SCCP Do not have local dial plan
Register via UDP, TCP, or TLS Register via TCP or TLS
CIPT1 v5.03-22
SCCP phones get device pool configuration information from the TFTP server and get the remaining configuration information from SCCP messages, including Cisco Unified CallManager group (primary, secondary, tertiary), SRST, load information, and so on. SIP phones get all configuration information from the TFTP server, including device pool, line configuration, softkey, and dial-plan information. SCCP phones do not have a local dial plan, whereas SIP phones can have a local dial plan. If configured, the dial plan is either part of the configuration file or a separate file, depending on the phone model. SCCP phones register by using TCP or Transport Layer Security (TLS), and SIP phones register by using UDP, TCP, or TLS.
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Supported SIP Signaling Transports

Three transport protocols are available for SIP phones on the Cisco Unified IP Phones 7911, 7941, 7961, 7970, and 7971: UDP, TCP, or TLS. SCCP phones have TCP or TLS support only.
Supported SIP Signaling Transports

Three transport methods are available in SIP: UDP, TCP, or TLS.
Cisco IP Phone 7905/7912 Cisco IP Phone 7940/7960 Cisco IP Phone 7911/41/61/7x
UDP TCP TLS
In SCCP, UDP is not an option.
CIPT1 v5.03-23
There are advantages and disadvantages of using TCP or UDP. With TCP, session establishment (SYN, SYN ACK, ACK) must occur before data transfer, whereas UDP is besteffort, so data packets are merely sent and retransmitted if a response is not received. Therefore, UDP is a lighter-weight protocol. If TCP has an established session, if the primary Cisco Unified CallManager fails to acknowledge messages from the phone, the phone will recognize the failure more quickly and failover to the secondary Cisco Unified CallManager. Failover will still occur with UDP, but it will take longer. Due to the lighter weight of UDP, it might be a be better choice for large (over 30,000 phones) networks.
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NTP Clock for SIP Phones

This subtopic describes how to use Network Time Protocol (NTP) clocking with Cisco SIP Unified IP phones.
NTP Clock for SIP Phones

SCCP phones derive their clock from Cisco Unified CallManager via SCCP signaling messages. Cisco 8.0 SIP phones derive their clock from one of the following sources, in priority order: 1. Primary NTP server 2. Secondary NTP server 3. Broadcast NTP server 4. Date header in 200 OK response from Cisco Unified CallManager to their REGISTER message NTP servers are defined in Cisco Unified CallManager Administration and propagated to the phone TFTP configuration files.
CIPT1 v5.03-24
A very important recommendation is to use an NTP server for the clock source in your Cisco Unified CallManager deployments. SCCP phones derive their clock from Cisco Unified CallManager through SCCP signaling messages. Cisco SIP phones running firmware version 8.0, however, derive their clock from an NTP server. In addition, referencing an NTP server for a clock is important to synchronize trace files as they traverse from phone to Cisco Unified CallManager to gateway, and so on, so it is advised to have all these devices reference a common clock source. Cisco 8.0 SIP phones derive their clock from one of these sources, in priority order: Primary NTP server Secondary NTP server Broadcast NTP server Date header in 200 OK response from Cisco Unified CallManager to their REGISTER message The NTP servers are defined during Cisco Unified CallManager installation and can be changed using Platform Administration pages. The addresses of the NTP servers that the phones should use are sent to the phones in their TFTP configuration file. If no server is found, the last option is for the phone, in the case of an SIP phone, to use the date and time found in the date header of the 200 OK response that the phone receives from Cisco Unified CallManager in response to the phone REGISTER message. This might not be the best time to use because the Cisco Unified CallManager may be in a different time zone. For this reason, synchronizing to an NTP clock source is strongly recommended. Neither Cisco Unified CallManager, nor SIP phones, use secure NTP.
2006 Cisco Systems, Inc. Deployment of Cisco Unified CallManager Release 5.0 Endpoints 3-55
DTMF Interworking: RFC 2833 and KPML

This subtopic describes how to interwork dual tone multifrequency (DTMF) in Cisco Unified CallManager networks.
DTMF Interworking: RFC 2833 and KPML

Cisco Unified CallManager Release 4.x and earlier versions pass DTMF messages in signaling (out-of-band). Most SIP devices communicate DTMF tones in the media channel (in-band) according to RFC 2833. This is why in Release 4.x, you had to allocate an MTP for every SIP call. The IETF now has an out-of-band method for SIP, called KPML. Cisco Unified CallManager Release 5.0 introduces KPML support on SIP and RFC 2833 support on SCCP, MGCP, and H.323. This significantly reduces (but does not eliminate) the number of scenarios in which an MTP must be allocated. KPML is always preferred over RFC 2833 on line-side. Cisco Unity, MeetingPlace, MeetingPlace Express, Cisco IOS gateways, and many others support KPML CTI applications do not yet support KPML in Release 5.0
CIPT1 v5.03-25
In Cisco Unified CallManager Release 4.x and earlier versions, all Cisco Unified CallManager protocols, including SCCP, MGCP, H.323, Telephony Application Programming Interface (TAPI), and Java Telephony Application Programming Interface (JTAPI), use out-of-band signaling messages to communicate DTMF. However, most SIP devices on the market use an in-band DTMF method, defined in RFC 2833. RFC 2833 communicates DTMF digits inside the Real-Time Transport Protocol (RTP) stream using a different payload type. Therefore, in Cisco Unified CallManager Release 4.x, you had to allocate a Media Termination Point (MTP) for every SIP call. The MTP converts the DTMF signaling from in-band to RFC 2833 and vice versa, from one call leg to the other, if, for example, an H.323 gateway calls an SIP device. The IETF invented an out-of-band method for SIP called Key Pad Markup Language or Key Press Markup Language (KPML). Cisco helped author this standard. KPML is a SIP message that communicates DTMF events. Cisco Unified CallManager Release 5.0 introduces KPML support on the SIP side and RFC 2833 support on SCCP, MGCP, and H.323 intercluster trunks, so there exists a high probability of having a common DTMF method when a device using one DTMF method communicates with another. This enhancement reduces the number of scenarios in which an MTP needs to be allocated. So, for example, if a SIP device that only supports RFC 2833 calls an SCCP phone, the SCCP phone supports both methods so it will use RFC 2833 to talk to the SIP device and no MTP is invoked. This enhancement requires firmware version 8.0 on the SCCP phones.
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Most of the Cisco products have been upgraded to RFC 2833, including Cisco Unity, Cisco Unified MeetingPlace, Cisco Unified MeetingPlace Express, and Cisco IOS gateways. However, one important exception exists: CTI applications do not yet support RFC 2833; therefore, applications such as Cisco Unified CallManager Attendant Console, Cisco Unified IP Contact Center (Cisco IPCC) Express, and others (those that use JTAPI and TAPI) do not support RFC 2833 in Release 5.0. If one of these applications communicates with a SIP device that only supports RFC 2833, Cisco Unified CallManager invokes an MTP (as it does in Release 4.x) to handle the DTMF translation for that call. KPML is also used for digit collection when dialing a SIP phone, in addition to DTMF, and is covered later in this lesson.
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Video Endpoints
This topic describes video endpoints to support Cisco Unified CallManager video telephony.
Cisco Unified CallManager Video Telephony
Supports video and video-related features, such as far-end camera control (FECC) Supports multiple logical channels that are needed to allow the transmission of video streams Transmits midcall, media-related messages that are needed for video Supports H.323, SCCP, and SIP Enhances locations and regions to provide bandwidth management Provides serviceability information, such as Call Detail Records (CDRs), about video calls
Cisco Unified CallManager supports video telephony and thus unifies the world of voice and video calls. Video endpoints use Cisco Unified CallManager call-handling features and access a unified voice and video solution for dialing and connecting video calls. The Cisco Unified CallManager video telephony solution offers these features: Supports video and video-related features, such as far-end camera control (FECC) Supports multiple logical channels that are needed to allow the transmission of video streams Transmits midcall, media-related messages that are needed for video (that is, transmits commands or indications that are needed for video calls) Supports H.323, SCCP, and SIP Enhances locations and regions to provide bandwidth management Provides serviceability information, such as Call Detail Records (CDRs), about video calls Cisco Unified CallManager Release 5.0 supports these types of video-enabled endpoints: Cisco Unified Video Advantage associated with Cisco IP phones running SCCP Cisco IP Video Phone 7985 Tandberg 2000 MXP, 1500 MXP, 1000 MXP, 770 MXP, 550 MXP, T-1000, or T-550 models running SCCP Sony PCS-1, PCS-TL30, or PCS-TL50 models running SCCP H.323 and SIP clients (Polycom, Sony, PictureTel, EyeBeam, Tandberg, VCON, VTEL, Microsoft NetMeeting, and others)
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Cisco Video Endpoints

This subtopic describes Cisco video endpoints supported by Cisco Unified CallManager.
Cisco Video Endpoints
Cisco Unified Video Advantage
Cisco IP Phone 7985G
Windows-based application and USB camera Associates with a Cisco Unified IP Phone 7940, 7941, 7960, 7961, 7970, or 7971 running SCCP
Personal desktop standalone video phone It has 8.4-inch color LCD screen and an embedded video camera
CIPT1 v5.03-27
SCCP video endpoints register directly with Cisco Unified CallManager and download their configurations via TFTP. They support many features and supplementary services, including hold, transfer, conference, park, pickup and group pickup, music on hold, shared line appearances, mappable softkeys, call forwarding (busy, no answer, and unconditional), and much more. At the moment, these two Cisco video endpoints are supported: Cisco Unified Video Advantage: Cisco Unified Video Advantage is a Windows-based application and USB camera that you can install on a Windows 2000 or Windows XP personal computer. When the PC is physically connected to the PC port on a Cisco Unified IP Phone 7940, 7941, 7960, 7961, 7970, or 7971 running SCCP, the Unified Video Advantage application associates with the phone, thus enabling users to operate their phones as they always have but with the added benefit of video. You can control which IP phones allow this association to take place by toggling the Video Capabilities: Enabled/Disabled setting on the IP phone configuration page in Cisco Unified CallManager Administration. When this feature is enabled, an icon representing a camera appears in the bottom right-hand corner of the IP phone display. By default, Unified Video Advantage is disabled. You can also use the Bulk Administration Tool to modify this setting on many phones at once. Note that the PC Port: Enabled/Disabled setting must also be enabled for Cisco Unified Video Advantage to work; however, the PC Access to Voice VLAN setting does not have to be enabled. To achieve the association, Cisco Unified Video Advantage installs a CDP driver onto the Ethernet interface of the PC. CDP enables the PC and the IP phone to discover each other automatically, which means that you do not have to configure anything on the PC or the IP phone in order for Unified Video Advantage to work. You can, therefore, plug the PC into any IP phone that is video-enabled and automatically associate with it.
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Cisco IP Video Phone 7985G: The Cisco IP Video Phone 7985G is a personal desktop video phone. Unlike Unified Video Advantage, which is an application that runs on a PC, the Cisco IP Video Phone 7985G is a standalone phone with integrated video features. The phone has an 8.4-inch color LCD screen and an embedded video camera for making video calls. The phone supports up to eight line appearances, has two 10/100BASE-T Ethernet connections, and has buttons for Directories, Messages, Settings, and Services. Like other Cisco IP phones, the Cisco IP Video Phone 7985G uses CDP to learn VLAN and CoS information from the attached switch to use in 802.1p and 802.1Q markings. The table lists the codecs supported by Cisco Unified Video Advantage and Cisco IP Video Phone 7985G.
Codec or Feature H.261 H.263/H.263+ H.264 H.711 G.722 G.722.1 G.723.1 G.728 G.729 Cisco Wideband Cisco Unified Video Advantage No Yes No Yes No No No No Yes Yes Cisco IP Video Phone 7985G Yes Yes Yes Yes Yes No No No Yes No
Note
More details on Cisco video telephony is provided in the CIPT2 class.
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Third-Party Video Endpoints

This subtopic describes third-party video endpoints supported by Cisco Unified CallManager.
Third Party Video Endpoints
Sony PCS-TL50
TANDBERG 1500 MXP
Cisco Unified CallManager Release 5.0 supports third-party video-enabled endpoints:

Sony PCS-1, PCS-TL30, or PCS-TL50 models running SCCP Tandberg 2000 MXP, 1500 MXP, 1000 MXP, 770 MXP, 550 MXP, T-1000, or T-550 models running SCCP
CIPT1 v5.03-28
Two manufacturers of video endpoints, Sony and Tandberg, currently have the following products that support SCCP: Sony PCS-1 and PCS-TL50 Tandberg T-1000 and T-550 SCCP on both the Sony and Tandberg endpoints is modeled after SCCP on the Cisco Unified IP Phone 7940. Most features found on the Cisco Unified IP Phone 7940 user interface are also supported on the Sony endpoints as well as the Tandberg endpoints, including multiple line appearances, softkeys, and buttons for Directories, Messages, Settings, Services, and so forth. The Sony and Tandberg endpoints also support the Option 150 field in DHCP to discover the IP address of the TFTP server, and they download their configurations from the TFTP server. However, software upgrades of the Sony and Tandberg endpoints are not done via TFTP. Instead, the customer must manually upgrade each endpoint using tools provided by the vendor. (Tandberg uses an FTP method, while Sony uses FTP or a physical memory stick.) The Sony and Tandberg endpoints register with up to three Cisco Unified CallManager servers and will failover to its secondary or tertiary servers if its primary server becomes unreachable. While the Sony and Tandberg endpoints support softkey functionality similar to that of the Cisco Unified IP Phone 7940 and 7960, the exact feature support differs between vendor and model. Check the manufacturer documentation for supported features. Features that are currently known to be missing on some platforms include these features: Messages button Directories (placed calls, received calls, missed calls, and corporate directory) Settings and Services buttons Some XML services (such as Extension Mobility)
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Because the Sony and Tandberg endpoints use SCCP, dialing a video call from an endpoint is similar to dialing an audio call from a Cisco IP phone. If users are familiar with Cisco IP phones, they should also find the Sony and Tandberg endpoints very intuitive to use. The main difference in the user interface is that the Sony and Tandberg endpoints do not have a button keypad or a handset like those on a phone. Instead, the remote control is used to access features and to dial numbers on the Sony and Tandberg endpoints. These are the major features of the Sony PCS Video Communication Terminal: Ideal for small meeting rooms and executive offices Completely integrated unit: 20-inch wide-screen LCD, camera with privacy shutter, speakers, and microphones Simultaneously operates as a videoconference terminal and PC monitor Intuitive and user-friendly GUI Easy-to-use Remote Commander unit Picture-in-Picture, Picture-and-Picture High-quality audio and video with bandwidth up to 1920 kBps Interoperable with H.323 and H.320 via Cisco Unified CallManager Call Forwarding, Call Transfer, Call Pickup, Hold, Park, Caller ID These are the major features of Tandberg 2000 MXP Videoconferencing Terminal: Designed for offices and small meeting rooms 23-inch wide-screen LCD, wireless remote control, PTZ Camera, microphone, and cables Wide-screen LCD (16:9), wide-angle screen, XGA resolution, auto or manual brightness Full range of Cisco IP telephony functions, including Directory, Voice Mail and Call Control, as supported by Cisco Unified CallManager Softkey design that replicates the IP phone experience H.264 support for high-quality video over lower bandwidth Extensions set up through Cisco Unified CallManager web administration tools PBX functions with H.323 endpoints (H.323 systems cannot initiate these services) Software upgradeable, to grow with the capabilities of Cisco Unified CallManager Supports XML applications, which deliver a range of information services to the desktop Interoperable with H.323 video communication using Cisco Unified CallManager Interoperable with H.320 (ISDN) video communication, using an H.323/H.320 video gateway and Cisco Unified CallManager. Codec support for third-party SCCP endpoints varies by vendor, model, and software version. Check the vendor product documentation for the supported codecs.
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Cisco IP Phone Codec Support

This topic describes major audio codecs supported by Cisco IP phones.
Cisco IP Phone Codec Support

Audio codecs:
Potentially able to compress audio signals Cisco IP phones support: G.711: 64 kbps G.729: 8 kbps
CIPT1 v5.03-29
Before a VoIP device is able to stream audio, the analog audio signal must be converted to a digital format. This is accomplished by an audio codec, which digitizes the audio input at the transmitting end and converts the digital stream back to analog audio at the receiving end. Because converted audio streams can consume a significant amount of bandwidth, many of the audio codecs also provide a level of compression, which can considerably reduce the bandwidth that they consume. However, compression can cause degraded voice quality, which is why the different audio codecs offer different levels of compression. The International Telecommunication Union Telecommunication Standardization Sector (ITUT) standards committee specifies several standards (called recommendations) for audio codecs. Cisco IP phones natively support two primary codecs: G.711 and G.729. The G.711 and G.729 codecs deliver relatively equal sound quality (both considered toll quality), with G.711 scoring slightly higher than G.729 in a mean opinion score (MOS) test. Because of this similarity, some network administrators choose to operate an entirely G.729-based network, while others choose to implement G.729 over the WAN and G.711 on the LAN. Although this configuration is ideal for many network environments, you may eventually encounter a codec mismatch. A codec mismatch occurs when two devices cannot negotiate a common codec or when the network administrator has forbidden the use of their common codec, such as using G.711 over the WAN. Regardless of the cause, you now have a need for transcoding. Transcoding resources perform conversions between the audio codecs. These resources are often costly and can introduce significant delay and quality degradation into your IP telephony network. When designing a voice network, you should attempt to limit the amount of transcoding that takes place between devices.
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Summary
Summary
Cisco IP phones are display-based, support customization, have inline power, and you can configure them for SCCP or SIP. Entry-level Cisco IP phones include the Cisco Unified IP Phone 7902G, 7905G, 7911G, and 7912G. Midrange Cisco IP phones include the Cisco Unified IP Phone 7940G, 7941G/G-GE, 7960G, 7961G/G-GE. Upper-end Cisco IP phones include the Cisco Unified IP Phone 7970G, 7971G/GE. Additional IP telephony devices include the Cisco Unified IP Phone 7985G, Cisco Unified IP Conference Station 7936, Cisco IP Communicator, Cisco ATA 186 and 188, and Cisco Unified IP Phone 7914 Expansion Module. Cisco Unified CallManager supports SIP IP phones of the three feature levels: Basic SIP for Cisco SIP Phones 7905, 7912, 7940, and 7960
Summary (Cont.)
Advanced SIP for Cisco SIP Phones 7911, 7941, 7961, 7970/71, and Cisco IP Communicator Very basic SIP for all supported third-party SIP phones An IP phone follows a specific process each time it boots up. In Cisco CallManager Administration, SIP phones are configured similarly to SCCP phones but some minor configuration differences exist. Cisco Unified CallManager supports video telephony on Cisco video endpoints such as Cisco Unified Video Advantage, Cisco Unified IP Phone 7985 or third-party video endpoints such as Tandberg, Sony, and others. Audio codecs convert analog voice signals to a digital stream and compress the output to save bandwidth. Cisco IP phones support the G.711 and G.729 codecs.
CIPT1 v5.03-31
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References
For additional information, refer to these resources: Cisco Systems, Inc., Cisco 7900 Series Unified IP phones support documentation http://www.cisco.com/en/US/customer/products/hw/phones/ps379/tsd_products_support_se ries_home.html Cisco Systems, Inc., Cisco Unified CallManager System Guide, Release 5.0(1), Cisco IP Phones http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_c hapter09186a008037e321.html#wp1066139 Cisco Systems, Inc., Cisco Unified IP Phones and Services Documentation http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/index.htm
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Lesson 3
Configuring Cisco Unified CallManager to Support IP Phones

Overview
Configuring Cisco Unified CallManager to support Cisco IP phones is a critical part of implementing a Cisco IP telephony network. This lesson prepares you for the initial IP phone setup and configuration and for the maintenance of existing IP phones. This lesson describes the Cisco Unified CallManager configuration to support Cisco IP phones. You will learn how to configure Cisco Unified CallManager to manually and automatically add IP phones and assign directory numbers. You will also learn how to configure device pools to provide a convenient way to define a set of common characteristics that can be assigned to devices, such as date or time zone, codec use, and other functionalities.
Objectives
Upon completing this lesson, you will be able to configure Cisco Unified CallManager to support Cisco IP phones, including server configuration, device pools, phone button templates, and directory numbers. This ability includes being able to meet these objectives: Configure Cisco Unified CallManager to eliminate IP phone reliance on DNS Configure device pools in Cisco Unified CallManager to define sets of common characteristics for IP phones Use Cisco Unified CallManager default phone button templates and create new templates to assign a common button configuration to a large number of IP phones Manually add and configure IP phones in Cisco Unified CallManager and assign directory numbers Configure Cisco Unified CallManager to support IP phone autoregistration to automatically issue directory numbers to new IP phones Configure Cisco Unified CallManager to be a DHCP server that provides IP configuration to phones
Server Configuration
This topic describes server configuration in Cisco Unified CallManager Administration.
Removing DNS Reliance
Enables Cisco IP phones and other Cisco Unified CallManager-controlled devices to contact the Cisco Unified CallManager without resolving a DNS name
Changing the name of the selected server to the IP address of the server in the Cisco Unified CallManager Administration window is the first step in configuring Cisco Unified CallManager to support Cisco IP phones. Renaming the server to the IP address has these benefits: It allows IP phones and other devices to find Cisco Unified CallManager on the network without having to query the Domain Name System (DNS) server to help resolve the server name to an IP address. It prevents the IP telephony network from failing if the IP phones lose the connection to the DNS server. It decreases the time that is required when a device attempts to contact Cisco Unified CallManager. Complete these steps to eliminate DNS reliance:
Step 1
In Cisco Unified CallManager Administration, choose System > Server. The Find and List Servers window appears.
Step 2
Click a server name. The Server Configuration window appears.
Step 3
Remove the host name and enter the IP address for the server in the Host Name/IP Address field. Click Save.
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Configuring Device Pools

This topic describes device pool configuration.
Device Pool Configuration
CIPT1 v5.03-3
Device pools provide a convenient way to define a set of common characteristics that can be assigned to devices, such as IP phones, instead of assigning individual characteristics to individual phones. Device pools enable you to simply assign a phone to a device pool so that the phone automatically inherits the common configuration items. You must configure the device pool for Cisco IP phones before adding the IP phones to the network. To create a new device pool, you must first create these minimal mandatory components, or use default settings where applicable: Cisco Unified CallManager group Date/time group Region Softkey template Cisco Survivable Remote Site Telephony (SRST) reference: The SRST Reference field allows you to specify the IP address of the Cisco SRST router. Cisco SRST enables routers to provide call-handling support for Cisco IP phones when they lose their connection to remote Cisco Unified CallManager installations or when the WAN connection is down. The device pool combines all of the individual configurations that you have created into a single entity. You will eventually assign this entity to individual devices, such as IP phones. This process will configure these devices with most of the configuration elements that they need to operate efficiently in your IP telephony network. Complete these steps to create the device pool:
Step 1 Step 2
Choose System > Device Pool. The Find and List Device Pools window opens. Click the Add New button to open the Device Pool Configuration window.
Deployment of Cisco Unified CallManager Release 5.0 Endpoints 3-69
Step 3
Choose, at a minimum, the Cisco Unified CallManager Group, Date/Time Group, Region, and a Softkey Template.
Example: Device Pool Configuration

Configuring 100 different Cisco IP phones to use the Arizona time zone, the English language, and the classical-style music on hold (MOH) option is tedious. Instead, you can create a device pool, named Arizona, that configures the correct time zone, language, and MOH, and you can make a single assignment of the Arizona device pool to each IP phone. The Device Pool Configuration Items table shows the device characteristics that you can specify for a device pool. You must configure these items before you configure a device pool if you want to choose the items for the device pool. Device Pool Configuration Items
Field Device Pool Name* Cisco Unified CallManager Group* Description Describes a name for the device pool. Chooses a redundancy group for the device pool. This redundancy group can contain a maximum of three redundant Cisco Unified CallManager servers. Assigns the correct time zone to the device. Determines the coder-decoder (codec) selection used by the device, depending on the end location of the call. Defines the type and order of the softkeys that are displayed on the LCD of a Cisco IP phone. Configures SRST and chooses the gateway that will support the device if the connection to the Cisco Unified CallManager is lost. Defines whom an IP phone is able to call if it autoregisters with the Cisco Unified CallManager. Assigns media resource support to a device for functions such as conferencing, transcoding, or MOH. Chooses the audio that Cisco Unified CallManager should play when a user presses the Transfer or Conference button on the Cisco IP phone. Chooses the audio that Cisco Unified CallManager should play when a user presses the Hold button on the Cisco IP phone. Defines the tones and cadences that the device uses. Defines the language that the device uses. Defines the amount of time that the IP phone monitors its connection to Cisco Unified CallManager before it unregisters from SRST and reregisters to Cisco Unified CallManager. This is to ensure that the link is stable (not flapping). The default for the enterprise parameter specifies 120 seconds, which can be modified on a device-pool basis or left at the default value.
Date/Time Group* Region* Softkey Template* SRST Reference*
Calling Search Space for Auto-Registration Media Resource Group List Network Hold MOH Audio Source
User Hold MOH Audio Source Network Locale User Locale Connection Monitor Duration
Note
An asterisk (*) Indicates a required field.
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If you make changes to a device pool, you must reset the devices in that device pool before the changes will take effect. You cannot delete a device pool that has been assigned to any device or one that is used for device defaults configuration. To find out which devices are using the device pool, click the Dependency Records link in the Device Pool Configuration window. If you try to delete a device pool that is in use, an error message is displayed. Before deleting a device pool that is currently in use, you must perform one of these tasks: Update the devices to assign them to a different device pool. Delete the devices that are assigned to the device pool that you want to delete.
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Cisco Unified CallManager Group Configuration

This figure describes how to configure a Cisco Unified CallManager group.
Cisco Unified CallManager Group Configuration
CIPT1 v5.03-4
A Cisco Unified CallManager group specifies a prioritized list of Cisco Unified CallManager servers to register to, with a maximum of three in the list. The first Cisco Unified CallManager in the list serves as the primary Cisco Unified CallManager for devices that are assigned to that group. The other members of the group serve as the secondary and tertiary backups. Changes to the Cisco Unified CallManager group affect the configuration file that is given to Cisco IP phones by the TFTP server when the IP phones initially boot.
Example: Cisco Unified CallManager Group Configuration

In the figure, the Cisco Unified CallManager group, called West has two Cisco Unified CallManager servers. You assign the Cisco Unified CallManager group to a device pool, and you then assign this device pool to the Cisco IP phone. The IP phone uses the Arizona Cisco Unified CallManager as its primary Cisco Unified CallManager, the California Cisco Unified CallManager as its secondary. You may want to assign another Cisco Unified CallManager, for example Washington, as its tertiary. Cisco Unified CallManager Administration will present an error message if you attempt to add a fourth Cisco Unified CallManager (for example, the Soporifix Cisco Unified CallManager) to the list. Checking the Auto-registration Cisco Unified CallManager Group check box enables the Cisco Unified CallManager to place any new IP phones that autoregister (IP phones that are added to the network without manual administrative configuration) into this group by default. Complete these steps to configure a Cisco Unified CallManager group:
Step 1 Step 2
Choose System > Cisco Unified CallManager Group. The default group that was created by Cisco Unified CallManager during the installation appears. Click Add New button to create a new Cisco Unified CallManager group.
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Step 3
Move the existing Cisco Unified CallManager servers using the Up and Down arrows between the panes and change the order of Cisco Unified CallManager servers using the Up and Down arrows on the right side.
Date/Time Group Configuration

This figure shows how to configure Cisco Unified CallManager Date and Time Group.
Date/Time Group Configuration
CIPT1 v5.03-5
Date and time groups define time zones for the various devices that are connected to Cisco Unified CallManager. You can assign each device to only one device pool. As a result, the device has only one date/time group. Cisco Unified CallManager has a default date/time group called CMLocal. The CMLocal date/time group synchronizes to the active date and time of the operating system on the Cisco Unified CallManager server. You can change the settings for CMLocal after installing Cisco Unified CallManager. Complete these steps to configure the date/time group:
Step 1 Step 2 Note
Choose System > Date/Time Group. The default CMLocal group appears. Choose Add a New button to insert additional date/time groups as required.
For a worldwide distribution of Cisco IP phones, you may want to create one named date/time group for each of the 24 time zones.
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Region Configuration
This figure shows how to configure Region.
Region Configuration
CIPT1 v5.03-6
When you create a region, you specify the audio codec that can be used for calls between devices (such as IP phones) within that region and between that region and other regions. As of Cisco Unified CallManager Release 4.0, you can also specify the video call bandwidth. You can create regions to modify the codec selection for any reason; however, most network administrators create regions based on geographical areas. The default voice codec for all calls through Cisco Unified CallManager is G.711. If you do not plan to use any other voice codec, you do not need to use regions. The system will use the default region.
Example: Region Configuration

In the figure shown here, there are the following regions: California, Default, and Washington; the configuration of the Washington region is displayed. If a device that is assigned to the Washington region calls another device in the Washington region, the devices use the G.711 codec. However, if a device assigned to the Washington region calls a device that is assigned to the California region, the devices use the G.729 codec. Cisco Unified CallManager creates the Default region during the installation process. You can rename the Default region to a more logical name to avoid confusion, or you can ignore it and use only regions that you have created. Complete these steps to configure a region:
Choose System > Region. The default region that was created during the Cisco Unified CallManager installation appears. Click Add New button to configure the regions. Give the new region a unique name. Click Save. Choose the codec and video bandwidth as appropriate between the regions.
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Softkey Template Configuration

This figure shows how to configure Softkey template.
Softkey Template Configuration
CIPT1 v5.03-7
The Softkey Template Configuration window allows you to manage the on-screen softkeys that the Cisco IP phones support (such as Cisco Unified IP Phones 7960 and 7940). You can configure these softkeys with many Cisco Unified CallManager functions and features. Applications that support softkeys can have one or more standard softkey templates associated with them; for example, Cisco Unified IP Manager Assistant (IPMA) has the Standard IPMA Assistant, the Standard IPMA Manager, and the Standard IPMA Manager Shared Mode softkey templates associated with it. You cannot modify standard softkey templates. First you copy the template and then modify the copy. Choose Device > Device Settings > Softkey Templates to access the Softkey Template Configuration window in Cisco Unified CallManager Administration.
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IP Phone Button Templates

This topic discusses the configuration and application of the Cisco IP phone button templates.
IP Phone Button Templates

Default Cisco Unified IP Phone 7960 template is 2 lines, 4 speed dials Prepare for anything; configure all possible combinations: 1 line, 5 speed dials 3 lines, 3 speed dials 4 lines, 2 speed dials 5 lines, 1 speed dial 6 lines, 0 speed dials
CIPT1 v5.03-8
Creating and using templates provides a fast way to assign a common button configuration to a large number of Cisco IP phones. Cisco Unified CallManager includes several default phone button templates. When adding IP phones, you can assign one of these templates to each IP phone or assign custom templates that you have created. You must assign at least one line per IP phone; usually this line is button 1. Depending on the Cisco IP phone model, you can assign additional lines. IP phones generally have several features, such as speed dial and call forwarding, assigned to the remaining buttons. Before adding any IP phones to the system, create phone button templates with all of the possible combinations for all IP phone models. An IP phone model may have various combinations; for example, a Cisco IP Phone 7960 supports six lines and can use these phone button template combinations: One line, five speed-dial buttons Two lines, four speed-dial buttons (default) Three lines, three speed-dial buttons Four lines, two speed-dial buttons Five lines, one speed-dial button Six lines, no speed dial button
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IP Phone Button Template Names

This subtopic suggests names for IP phone button templates.
IP Phone Button Template Names

Use the model number, line and speed-dial settings in the name: 7960 1-5 7960 3-3 Template updates affect the IP phones that use that template. Renaming a template does not affect the IP phones using that template. Cannot delete a template assigned to one or more devices.
Create easily recognizable naming conventions for the phone button template. A suggested best practice is to use the model number of the Cisco IP phone followed by the number of lines and speed dials. For example, a phone button template named 7960 1-5 would indicate a Cisco Unified IP Phone 7960 with one line and five speed-dial buttons. To create a template, copy an existing template and assign a unique name to the template. You can make changes to the default templates that are included with Cisco Unified CallManager or to the custom templates that you have created. You can rename existing templates and modify them to create new templates. You can also update custom templates to add or remove features, lines, or speed-dial buttons. When you update a template, the change affects all of the IP phones that use it. Renaming a template does not affect the IP phones that use that template. All Cisco IP phones that use a template continue to use that template after you rename it. You can delete IP phone templates that are not currently assigned to any IP phone in your system. You cannot delete a template that is assigned to one or more devices. Currently, there is not an easy way to query whether a template is in use or not. Before you can delete a template, you must reassign all of the Cisco IP phones that are using the template to a different phone button template.
Example: Naming a Phone Button Template

If you have a special request or only a limited number of users who need three lines and three speed-dial buttons on the Cisco Unified IP Phone 7960G, you can create a phone button template that supports three lines and three speed dials, name the template 7960 3-3, and quickly assign the phone button template 7960 3-3 to the IP phones.
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Configuring IP Phone Autoregistration

This topic describes how to configure Cisco Unified CallManager for autoregistering Cisco IP phones.
Autoregistration Configuration
CIPT1 v5.03-10
Autoregistration allows Cisco Unified CallManager to issue extension numbers to new IP phones, which is similar to the way in which the Dynamic Host Configuration Protocol (DHCP) server issues IP addresses. When a new IP phone boots and attempts to register with Cisco Unified CallManager for the first time, Cisco Unified CallManager issues an extension number from a configured range. After Cisco Unified CallManager issues the extension, it records the extension number to the MAC address mapping in the Informix database. Although autoregistration simplifies the process of deploying a new IP telephony network, it is an option that is available only in some new IP telephony deployments. Because administrators deploy most IP telephony networks as a migration from a PBX environment, users have existing telephone extensions. These existing telephone extensions typically map to direct inward dialing (DID) numbers from the public switched telephone network (PSTN) and cannot change. Therefore, these IP telephony deployments usually use manual configuration rather than autoregistration. You should carefully evaluate autoregistration before implementing it because its use can pose a security risk to the network. Autoregistration allows anyone with physical assess to the voice network to connect an IP phone and use it, whether authorized or not. For this reason, many organizations, as part of their security policy, disable the use of autoregistration or use autoregistration in a secure staging environment for initial Cisco Unified CallManager configuration. Complete these steps to configure the Cisco Unified CallManager server to support autoregistration:
Step 1
From Cisco Unified CallManager Administration, choose System > Cisco Unified CallManager.
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Click Find and then from the list of Cisco Unified CallManager servers, choose the server that you want to support autoregistration. Under the Auto-Registration Information section, enter the appropriate directory number range in the Starting and Ending Directory Number fields. Ensure that the Auto-Registration Disabled on this Cisco Unified CallManager check box is unchecked. Click Save.
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Device Defaults
Use device defaults to set the default characteristics of each type of device that registers with a Cisco Unified CallManager.
Device Defaults
CIPT1 v5.03-11
The device defaults for a device type apply to all autoregistered devices of that type within a Cisco Unified CallManager cluster. You can set these device defaults for each device type to which they apply: Load information: Lists the firmware load that is used with a particular type of hardware device Device pool: Allows you to choose the device pool that is associated with each type of device Phone template: Indicates the phone button template that is used by each type of device When a device autoregisters to a Cisco Unified CallManager, it acquires the device default settings for its device type. After a device registers, you can update its configuration individually to change the device settings. Installing Cisco Unified CallManager automatically sets device defaults. You cannot create new device defaults or delete existing ones, but you can change the default settings. Complete these steps to update the device defaults:
In Cisco Unified CallManager Administration, choose Device > Device Settings > Device Defaults to open the Device Defaults Configuration window. In the Device Defaults Configuration window, modify the appropriate settings for the device that you want to change. Click Save to save the changes in the Cisco Unified CallManager configuration database.
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Manual IP Phone and Directory Number Configuration

This topic describes manual configuration of Cisco IP phones and directory numbers in Cisco Unified CallManager Administration.
Manual IP Phone Configuration
Phones are identified using MAC addresses, obtained from: Sticker on the phone Settings button Mandatory settings: Device Pool Phone Button Template Common Phone Profile Location Built-in Bridge Privacy
CIPT1 v5.03-12
Manually adding new IP phones to the network is often tedious, but it can constitute a large part of day-to-day voice network management. The Cisco Bulk Administration Tool (BAT) allows you to add a large number of IP phones to the Cisco Unified CallManager database at once, but BAT is not appropriate for adding or modifying a single IP phone for a new employee. Cisco Unified CallManager uses the IP phone MAC address to track the phone in the voice network, and ties all IP phone configuration settings to the IP phone MAC address. Before you can perform any configuration on a Cisco IP phone through Cisco Unified CallManager, you must find the MAC address of that IP phone. Use these guidelines to locate a MAC address: You can find the MAC address in the text and Universal Product Code (UPC) form, which is imprinted on the shipping box for the IP phone. Some administrators use bar code scanners to simplify the process of adding multiple IP phones. You can also find the MAC address in the text and UPC form on the back of the IP phone, on a sticker near the bottom. If you boot the IP phone, you can press the Settings button on the face of the phone. Use the arrow keys to navigate, and choose Network Configuration. The MAC address will be displayed on line 3 of the network configuration.
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You can continue the Cisco IP phone configuration on the Cisco Unified CallManager after you have the MAC address of the IP phone, as follows:
In Cisco Unified CallManager Administration, choose Device > Phone to open the Find and List Phones window. Click Add New button. From the Phone Type drop-down menu, choose the type of IP phone that you want to add. Click Next. Choose the SCCP protocol used to control the IP phone from the drop-down menu. Click Next. At a minimum, you must configure these fields: MAC Address Device Pool Phone Button Template Common Phone Profile Location Built-in Bridge Privacy Click Save. Cisco Unified CallManager prompts you to reset the new IP phone.

Step 8 Step 9
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Directory Number Configuration

The figure shows how to configure the directory number for an IP phone.
Directory Number Configuration
CIPT1 v5.03-13
Follow this procedure to configure a directory number for the manually added IP phone:
Start from the IP phone configuration window from the previous page. In the Associated Information column on the left, click on the Line [x] Add a new DN link to configure a directory number. When the Directory Number Configuration window appears, enter the directory number of the IP phone in the appropriate field. Click Save.
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DHCP Server Feature

This topic describes the DHCP server feature of Cisco Unified CallManager.
DHCP Server Feature Support Overview

DHCP server support is added to Cisco Unified CallManager Release 5.0. Provides a subset of Windows 2000 Server DHCP functionality but sufficient for IP phone purposes. Provides addresses for IP phones, not for all devices in the network. One DHCP server per Cisco Unified CallManager cluster is permitted. Multiple subnets can be configured for each server. Clusters can share the same DHCP server if not geographically separated. DHCP server is a standalone server; no backup server exists.
CIPT1 v5.03-14
DHCP server support is added to Cisco Unified CallManager Release 5.0. DHCP enables Cisco IP telephones, connected to either customer data or a voice Ethernet network, to dynamically obtain their IP addresses and configuration information. The DHCP server feature in Cisco Unified CallManager Release 5.0 replaces the DHCP server service on Windows-based Cisco Unified CallManager nodes, which was part of the Windows Administrative Tools. While not fully equivalent to the Windows 2000 DHCP server, the DHCP server feature in Cisco Unified CallManager Release 5.0 provides sufficient functionality to provide IP address information to IP phones. In environments that require a scalable DHCP solution, enable DHCP service on Cisco IOS routers.
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Steps to Configure DHCP Phone Support

The figure shows the steps to configure the DHCP feature.
Steps to Configure DHCP Phone Support

1. Activate DHCP Monitor Service 2. Add and configure the DHCP server 3. Configure the DHCP subnet
CIPT1 v5.03-15
Follow these steps to configure the DHCP server on a CallManager node:

Activate DHCP Monitor Service. This step is optional but highly recommended. Add and configure the DHCP server. Configure the DHCP subnet. Only two subnets can be configured: a primary and a secondary.
If DHCP services are already in use in your network and there is sufficient capacity for multiple ranges, using the already deployed method may be simpler and easier. This is due to the limited capabilities of the CallManager-based DHCP service and possible implementation differences with the existing solution.
Note
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Step 1: Activate DHCP Monitor Service

The figure shows how to activate DHCP Monitor Service.
Step 1: Activate DHCP Monitor Service
More services are available.

The Cisco DHCP Monitor Service monitors IP address changes for IP phones in the database. When a change is detected, it modifies the /etc/dhcpd.conf file and restarts the DHCP daemon. You can activate and deactivate Cisco DHCP Monitor Service by using the Service Activation window of Cisco Unified CallManager (Go to Tools > Service Activation, then choose the desired server). The DHCP Monitor Service starts automatically after it is activated by using Cisco Unified CallManager Serviceability.
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Step 2: Configure the DHCP server

The figure shows how to configure a DHCP server on Cisco Unified CallManager.
Step 2: Configure the DHCP server

Designate a node to be the DHCP server from the drop-down menu. Designate DNS server IP used. Option 150 specifies the IP addresses of the TFTP servers to the phones.
All other required entries default to 0.
CIPT1 v5.03-17
To add and configure the DHCP server, follow this procedure:

Choose System > DHCP > DHCP Server from Cisco Unified CallManager Administration. Click Add New. Then you will see the window shown in this figure. Enter the host name of the Cisco Unified CallManager server that should run the DHCP service. That server must use DNS services.
You must update the DNS server with the appropriate Cisco Unified CallManager name and address information before using that information here.
Note
Optionally, provide the DNS address, possibly a primary and secondary address, that will be provided to the DHCP clients. Specify the primary TFTP server IP address (Option 150) from which the phones will retrieve the needed files Optionally, enter the address of the bootstrap server that is used in the phone bootstrap process. You can use the IP address of the TFTP server or the default value to a DHCP server address if the server supplies the next bootstrap service. Optionally, specify the domain name that you should use when resolving host name via the DNS. You can enable the IP phones to access the TFTP server using DHCP option 66. Use this field to identify a TFTP server. You can configure only one DNS name or an IP address in this parameter.
Step 7 Step 8
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Step 9 Step 10
Optionally, specify timeout in seconds for Address Resolution Protocol (ARP) cache entries. Specify the time as a 32-bit unsigned integer. You may configure the IP address lease time. The DHCP server uses the information in this field to specify the lease time that it is willing to offer. Specify the time in seconds as a 32-bit unsigned integer. Optionally, specify the renewal time. This field specifies the time interval from address assignment until the client transitions to the RENEWING state. You may also configure rebinding time. This field specifies the time interval from address assignment until the client transitions to the REBINDING state. Specify the value in units of seconds and as a 32-bit unsigned integer. Click Save.
Step 11 Step 12
Step 13
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Step 3: Configure the DHCP Subnet

The figure shows how to configure a DHCP subnet.
Step 3: Configure the DHCP Subnet

Choose the DHCP server from the drop-down menu. Configure subnet number, primary address range, mask, and other information that you want DCHP to provide to the phones.
Configure secondary address range as necessary (useful if you have a block of reserved addresses to exclude between primary and secondary range).
CIPT1 v5.03-18
The DCHP subnet configured in Cisco Unified CallManager is a similar concept to the DHCP scope configured under Windows 2000. The subnet (or scope) defines the address range to be provided, the mask, domain name, address lease times, and so on. To add and configure a DHCP subnet, follow this procedure:
Choose System > DHCP > DHCP Subnet. Click Add New. Choose the DHCP Server, and enter the Subnet IP Address, the start IP address of the first range of IP addresses to be assigned, the end IP address of the first range of IP addresses to be assigned, the mask, and optional parameters as desired. Click Save.
The timers in the DHCP subnet configuration have precedence over the timers in DHCP server configuration for the subnet configured.
Step 4 Note
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DHCP Migration Considerations

This subtopic describes the migration of the DHCP when upgrading from Cisco Unified CallManager 4.x to 5.0.
DHCP Migration Considerations

When upgrading from Release 4.x to 5.0:
There is no migration provided from Windows 2000-based DHCP configuration to Linux-based DHCP configuration when using Windows Upgrade installation type. The administrator will need to reprovision the Release 5.0 server, use the DHCP Server feature on the Cisco IOS router or switch, or use an alternative mechanism.
CIPT1 v5.03-19
When Cisco Unified CallManager is migrated from Release 4.x to 5.0, and it worked as a DHCP server, the DHCP configuration must be migrated manually because there is no automatic DHCP configuration transfer from Cisco Unified CallManager 4.x to 5.0. A DHCP server in Release 4.x works at the Windows 2000 server operating system level, while in Release 5.0, the DHCP server is part of the Cisco Unified CallManager itself.
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Summary
Summary
Assigning an IP address to Cisco Unified CallManager eliminates IP phone reliance on DNS. Device pools simplify configuration by allowing you to define sets of common characteristics for devices. Creating and using IP phone button templates provides a fast way to assign a common button configuration to a large number of IP phones.
CIPT1 v5.03-20
Summary (Cont.)
Using IP phone autoregistration allows you to automatically issue extension numbers and a default configuration to new phones. Manually configuring the IP phone is appropriate when you want to add or modify a single IP phone or a few IP phones. Cisco Unified CallManager Release 5.0 adds integrated DHCP server functionality to IP phones.
CIPT1 v5.03-21
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References
For additional information, refer to these resources: Cisco Systems, Inc. Cisco Unified CallManager System Guide, Release 5.0(1), System Configuration Overview and System-Level Configuration Settings. http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_b ook09186a008037cff7.html Cisco Systems, Inc. Cisco Unified CallManager Administration Guide, Release 5.0(2). http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_b ook09186a0080624b45.html
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Lesson 4
Configuring SIP Endpoints

Overview
For many customers, the addition of Session Initiation Protocol (SIP) endpoint support and SIP trunk enhancements in Cisco Unified CallManager Release 5.0 is a key factor in the decision to migrate from a Cisco Unified CallManager Release 4.x environment to Release 5.0. The new standards-based SIP capabilities offer organizations the freedom to incorporate a host of new endpoint devices to meet business demands as well as SIP-enabled enterprise applications. Nearly all Cisco IP phones support SIP connectivity to Cisco Unified CallManager Release 5.0, as do endpoints from third-party manufacturers in the Cisco third-party SIP Verify Program and endpoints to RFC 3261 and other Internet Engineering Task Force (IETF) SIP standards. SIP is a highly extensible protocol, and Cisco has implemented extensions to the SIP protocol to try to achieve Skinny Client Control Protocol (SCCP) feature parity in newer Cisco IP phones. As a result, differences in phone features, configuration, and user dialing experiences exist with IP phones, from a very basic third-party SIP phone that only supports the minimal specifications to comply with RFC 3261 to new Cisco IP phones that go well beyond meeting 100 percent standards-compliance. This lesson describes SIP endpoint configuration, how to migrate a Cisco SCCP-controlled IP phone to SIP protocol, and how to add third-party SIP phones to Cisco Unified CallManager.
Objectives
Upon completing this lesson, you will be able to configure Cisco Unified CallManager to work with Cisco and third-party SIP phones. This ability includes being able to meet these objectives: Identify SIP phone support in Cisco Unified CallManager Release 5.0 Define SIP line-side support in Cisco Unified CallManager Release 5.0 Review a basic SIP call flow and describe SIP digit-collection behavior and the impacts on end users Add and configure Cisco SIP phones Convert a Cisco SCCP phone to a Cisco SIP phone Add third-party SIP phones to Cisco Unified CallManager Add a local SIP dial rule in Cisco Unified CallManager to provide a consistent dialing experience between SIP and SCCP phones
Cisco Unified CallManager Release 5.0 SIP Phones

This topic describes SIP phone support, functions, and features in Cisco Unified CallManager Release 5.0.
Cisco Unified CallManager Release 5.0 SIP Phones

Cisco Unified CallManager Release 5.0 offers SIP support for these Cisco IP phone models:
Cisco Unified IP Phone 7970/71 Cisco Unified IP Phone 7960/61 Cisco Unified IP Phone 7940/41 Cisco Unified IP Phone 7911 Cisco Unified IP Phone 7906 Cisco Unified IP Phone 7905/12
CIPT1 v5.03-2
Cisco Unified CallManager introduced support for the SIP protocol on all major IP phones. The supported Cisco IP phone models are as follows: Cisco Unified IP Phones 7970 and 7971 Cisco Unified IP Phones 7960 and 7961 Cisco Unified IP Phones 7940 and 7941 Cisco Unified IP Phone 7911 Cisco Unified IP Phone 7906 Cisco Unified IP Phones 7905 and 7912
Note Cisco Unified IP Phone 7905 is end-of-sale as of July 2006.
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Cisco Unified CallManager Release 5.0 SIP Features

Cisco Unified CallManager Release 5.0 supports many functions and features for the SIP protocol.
Cisco Unified CallManager Release 5.0 SIP Features

Functions and features supported for SIP calls:
Basic calls between SIP endpoints and Cisco Unified CallManager DTMF relay between SIP endpoints and Cisco Unified CallManager Supplementary services that are initiated if an MTP is allocated Ringback tone during blind transfer Supplementary services that are initiated by SIP endpoint Enhanced call identification services
CIPT1 v5.03-3
Cisco Unified CallManager supports these functions and features for SIP calls: Basic calls between SIP endpoints and Cisco Unified CallManager: There are three basic functions compliant with SIP RFC 3261: Basic Outgoing Call: You can initiate outgoing calls from any SIP endpoint, such as an IP phone or a gateway, to a SIP device registered at Cisco Unified CallManager. The device answering the call triggers media establishment. Basic Incoming Call: Any SIP endpoint, including SIP IP phones, fax devices, or SIP gateways can accept incoming calls. The answering device triggers media establishment. Use of Early Media: This basic SIP feature provides call progress signaling. While the public switched telephone network (PSTN) provides inband progress information to signal early media (such as a ring tone or a busy signal), SIP does not have the same functionality. The originating party includes session definition protocol (SDP) information, such as codec usage, IP address, and port number, in the outgoing INVITE message. When early media needs to be delivered to SIP endpoints prior to connection, Cisco Unified CallManager sends an SDP 183 Session Progress message.
Dual tone multifrequency (DTMF) relay between SIP endpoints and Cisco Unified CallManager: Each digit dialed with tone dialing is assigned a unique pair of frequencies. Voice compression of these tones with a low-rate codec can cause DTMF signal loss or distortion. Therefore, DTMF tones are separated from the voice bearer stream and sent as signaling indications in SIP signaling channel. Media Termination Points (MTPs) are now dynamically allocated, if needed, based on the DTMF methods used on each endpoint.
2006 Cisco Systems, Inc. Deployment of Cisco Unified CallManager Release 5.0 Endpoints 3-95
Supplementary services initiated if an MTP is allocated: The system supports all supplementary services that the SCCP endpoint initiates during a SIP call. Cisco Unified CallManager internally manages SCCP endpoints without affecting the connecting SIP device. Any changes to the original connecting information get updated with re-INVITE or UPDATE messages that use the Remote-Party-ID header. Ringback tone during blind transfer: For SCCP-initiated blind transfers, Cisco Unified CallManager needs to generate tones or ringback after a call has already connected. In other words, Cisco Unified CallManager provides a media announcement for blind transfers. A blind transfer works when the transferring phone connects the caller to a destination line before the target of the transfer answers the call. A blind transfer differs from a consultative, or attended, transfer, in which one transferring party either connects the caller to a ringing phone (ringback received) or speaks with the third party before connecting the caller to the third party. Blind transfers that are initiated from an SCCP IP phone allow ringback to the original, connected SIP device user. To accomplish ringback, Cisco Unified CallManager uses an annunciator software device that is often located with an MTP device. Supplementary services initiated by a SIP endpointSIP-initiated call transfer, call hold, and call forward. Cisco Unified CallManager supports SIP-initiated call transfer. It also supports call hold and call retrieve that a SIP device or a Cisco Unified CallManager device initiates. For example, when an SCCP IP phone user retrieves a call that was placed on hold by another user, Cisco Unified CallManager sends a re-INVITE message to the SIP proxy. Cisco Unified CallManager supports a call forward that a SIP device initiates or that a Cisco Unified CallManager device initiates. Cisco Unified CallManager processes call forwarding redirection requests from SIP devices. No SIP redirection messages are used for call forwarding initiated by Cisco Unified CallManager. Enhanced call identification services: Cisco Unified CallManager provides flexible configuration options to provide enhanced call identification services either on a call-bycall basis, or a statically preconfigured basis for each SIP signaling interface. These are the enhanced call identification services: Line identification services: Calling Line Identification Presentation (CLIP) and Calling Line Identification Restriction (CLIR): These supplementary services allow or restrict the phone number of the originating caller on a call-by-call basis. You can configure Cisco Unified CallManager to allow or restrict the display of the phone number of the calling party on the phone display of the called party. Connected Line Identification Presentation (COLP) and connected Line Identification Restriction (COLR): COLP and COLR allow or restrict the phone number of the called party. You can configure Cisco Unified CallManager to allow or restrict the display of the phone number of the connected party on the phone display of the calling party. Name Identification Services: Calling Name Presentation (CNIP) and Restriction (CNIR): CNIP and CNIR allow or restrict the name of the originating caller. You can configure Cisco Unified CallManager to allow or restrict the display of the name of the calling party on the phone display of the called party. Connected Name Presentation (CONP) and Restriction (CONR): CONP and CONR allow or restrict the phone number of the called party. You can configure Cisco Unified CallManager to allow or restrict the display of the phone number of the connected party on the phone display of the calling party.
Cisco Unified CallManager Release 5.0 SIP LineSide Support

This topic defines SIP line-side support in Cisco Unified CallManager Release 5.0.
SIP Support In Cisco Unified CallManager Release 4.X

Cisco Unified CallManager Release 4.x
Gateways SIP Trunk
Co nf/X co de
SIP Network
SIP Proxy Server
DSP Resources
Rich-Media Conferencing Soft Phones CTI Apps Cisco and Third-Party Phones
Video Endpoints
SIP Apps
Unified Messaging
SIP Gateways SIP SoftClients SIP Phones

SCCP MGCP H.323 CTI SIP
CIPT1 v5.03-4
SIP support was limited to trunk-side interface and required a proxy server to connect to the SIP network for call control.
SIP support in Cisco Unified CallManager Release 4.x was limited to a SIP-trunk signaling interface between the Cisco Unified CallManager Release 4.x network and the SIP network. Endpoints registered to Cisco Unified CallManager using SCCP and an external SIP trunk connected to a SIP proxy server that interfaced to the SIP clients and SIP gateways to provide call control. Only basic audio calls were supported.
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SIP Support In Cisco Unified CallManager Release 5.0

This subtopic describes SIP line support in Cisco Unified CallManager Release 5.0.
SIP Support In Cisco Unified CallManager Release 5.0

Cisco Unified CallManager Release 5.0
Gateways MeetingPlace/ MP Express
CCME
Carriers / Other PBXs CallManager Release 5.0
Cisco Unified Presence Server
Unity-Unity Connection Soft Phones

Cisco IP Video Endpoints Communicator
Microsoft LCS IBM Sametime SCCP MGCP H.323 CTI SIP/SIMPLE/KPML
CTI Apps
Cisco and Third-Party Phones
Native SIP, SIMPLE, support on both line-side and trunk-side is added for audio and video calls.
CIPT1 v5.03-5
Cisco Unified CallManager Release 5.0 integrates native SIP and SIP for Instant Messaging and Presence Logical Extensions (SIMPLE) support on both line-side and trunk-side interfaces (for both audio and video calls) for a variety of endpoint types, gateways, and applications. SIMPLE defines how you can use SIP to build instant messaging and presence. Cisco Unified CallManager offers integrated presence on phones and applicationsKeypad Markup Language (KPML) and RFC 2833 support for DTMF; Transport Layer Security (TLS) and digest authentication for security; and protocol interworking between SIP, H.323, Media Gateway Control Protocol (MGCP), SCCP, Telephony Application Programming Interface (TAPI)/ Java Telephony Application Programming Interface (JTAPI); Resource Reservation Protocol (RSVP) support for topology-aware Call Admission Control, and other features. No longer is a Cisco SIP proxy server required to connect to a SIP network.
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SIP Call Flow and Digit Collection

This topic describes basic SIP call flows and describes SIP digit collection behavior.
SIP Call Flow: Successful Call Setup and Disconnect

Cisco SIP Phone Cisco SIP Phone
1 INVITE 2 100 Trying 3 180 Ringing 4 200 OK 5 ACK 6 2-way RTP channel 7 BYE 8 200 OK
CIPT1 v5.03-6
The figure illustrates a successful Cisco SIP phone-to-Cisco SIP phone call setup and disconnect. In this scenario, the two end users are User A and User B. The call flow is as follows: 1. User A calls User B. 2. User B answers the call. 3. User B hangs up. The following describes in detail the call flow for this call:
Step 1
The Cisco SIP phone A sends a SIP INVITE request. The SIP INVITE request contains the following: The IP address of the Cisco SIP phone is inserted in the Request-uniform resource identifier (URI) field. Cisco SIP phone A is identified as the call session initiator in the From field. A unique numeric identifier is assigned to the call and is inserted in the Call-ID field. The transaction number within a single call leg is identified in the CSeq field. The Command Sequence (CSeq) header field contains an integer and a method name. The CSeq number is initialized at the start of a call, incremented for each new request within a dialog, and is a traditional sequence number. The CSeq is used to distinguish a retransmission from a new request. Session definition protocol (SDP) specifies that User A is ready to receive.
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The port on which the Cisco SIP phone A is prepared to receive the Real-Time Transport Protocol (RTP) data is specified using SDP.
Step 2
The Cisco SIP phone B sends a SIP 100 Trying response to Cisco SIP phone A. The 100 Trying response indicates that the INVITE request has been received by the Cisco SIP phone B. The Cisco SIP phone B sends a SIP 180 Ringing response to the Cisco SIP phone A. The 180 Ringing response indicates that the user is being alerted. The Cisco SIP phone B sends a SIP 200 OK response to the Cisco SIP phone A. The 200 OK response notifies the Cisco SIP phone A that the connection has been made. The Cisco SIP phone A sends a SIP ACK to the Cisco SIP phone B. The ACK confirms that Cisco SIP phone A has received the 200 OK response. The call session is now active. The two-way RTP channel is created and users have two-way voice connection. User B terminates the call session and the Cisco SIP phone B sends a SIP BYE request to Cisco SIP phone A. The BYE request indicates that User B wants to release the call. The Cisco SIP phone A sends a SIP 200 OK response to the Cisco SIP phone B. The 200 OK response notifies the phone that Cisco SIP phone A has received the BYE request.
Step 6 Step 7
Step 8
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SIP Call Flow: Called User Is Busy

The figure shows SIP call flow when a called user is busy and cannot answer the call.
SIP Call Flow: Called User Is Busy

1 INVITE 2 100 Trying 3 486 Busy Here 4 ACK
CIPT1 v5.03-7
The figure illustrates an unsuccessful call in which User A initiates a call to User B, but User B is on the phone at the moment and is unable or unwilling to take another call. The following describes in detail the call flow for this call:
Step 1
Cisco SIP phone A sends a SIP INVITE request. The SIP INVITE request contains the following: The IP address of the Cisco SIP phone is inserted in the Request-URI field. Cisco SIP phone A is identified as the call session initiator in the From field. A unique numeric identifier is assigned to the call and is inserted in the Call-ID field. The transaction number within a single call leg is identified in the CSeq field. The media capability that User A is ready to receive is specified using SDP. The port on which Cisco SIP phone A is prepared to receive the RTP data is specified using SDP.
Step 2
Cisco SIP phone B sends a SIP 100 Trying response to Cisco SIP phone A. The 100 Trying response indicates that the INVITE request has been received by Cisco SIP phone B. Cisco SIP phone B sends a SIP 486 Busy Here response to Cisco SIP phone A. The 486 Busy Here response is a client error response that indicates that User B was successfully contacted, but User B was not willing or was unable to take the call. Cisco SIP phone A sends a SIP ACK to Cisco SIP phone B. The ACK confirms that User A has received the 486 Busy Here response. The call session attempt is now being terminated.
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Step 3
Step 4
SIP Call Flow: Simple Call Hold

The figure illustrates a successful call between Cisco SIP phones in which one of the participants places the other on hold and then returns to the call.
SIP Call Flow: Simple Call Hold

1 Call setup procedure 2 2-way RTP channel 3 INVITE (c=IN IP4 0.0.0.0) 4 200 OK 5 ACK 6 RTP channel torn down 7 INVITE (c=IN IP4 user B IP) 8 200 OK 9 ACK 10 2-way RTP channel reestablished
CIPT1 v5.03-8
In this call flow scenario, the two end users are User A and User B. User A and User B are both using Cisco SIP phones, which are connected via an IP network. The call flow scenario is as follows: 1. User A calls User B. 2. User B answers the call. 3. User B places User A on hold. 4. User B takes User A off hold. 5. The call continues. The following describes in detail the call flow for this call:
The call is successfully established. The two-way RTP channel is created and users have two-way voice connection. Cisco SIP phone B sends a midcall INVITE to Cisco SIP phone A with new SDP session parameters (IP address), which are used to place the call on hold.
Call_ID=1 SDP: c=IN IP4 0.0.0.0
The c= SDP field of the SIP INVITE contains an 0.0.0.0. This places the call on hold.
Step 4 Step 5
Cisco SIP phone A sends a SIP 200 OK response to Cisco SIP phone B. Cisco SIP phone B sends a SIP ACK to Cisco SIP phone A. The ACK confirms that Cisco SIP phone B has received the 200 OK response from Cisco SIP phone A.
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Step 6 Step 7
The RTP channel has been torn down and resources released. Cisco SIP phone B sends a mid-call INVITE to Cisco SIP IP phone A with the same call ID as the previous INVITE and new SDP session parameters (IP address), which are used to reestablish the call.
Call_ID=1 SDP: c=IN IP4 181.23.250.2
To reestablish the call between phone A and phone B, the IP address of phone B is inserted into the c= SDP field.
Cisco SIP phone A sends a SIP 200 OK response to Cisco SIP phone B. Cisco SIP phone B sends a SIP ACK to Cisco SIP phone A. The ACK confirms that Cisco SIP phone B has received the 200 OK response from Cisco SIP phone A. A two-way RTP channel is reestablished between Cisco SIP phone A and Cisco SIP phone B.
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Configuring Cisco SIP Phones

This topic describes how to add and configure Cisco SIP phones in Cisco Unified CallManager Release 5.0.
Configuring SIP Phones

Provisioning a Cisco SIP phone is just like provisioning an SCCP phone. The protocol choice (SIP or SCCP) that you choose when you add the phone automatically dictates what firmware filename gets specified in the phone configuration file.
Cisco Unified CallManager Device Defaults
CIPT1 v5.03-9
Provisioning a Cisco SIP phone is just like provisioning an SCCP phone. The Device Defaults window (Device > Device Settings > Device Defaults) includes both a SIP and SCCP default firmware load. When you add a phone in Cisco Unified CallManager Release 5.0, in addition to specifying the phone model, you now also specify the protocol. The protocol chosen (SIP or SCCP) automatically dictates what firmware filename gets specified in the phone configuration file.
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Choosing the Protocol for Autoregistered Phones

The figure shows how to choose between SCCP and SIP protocols for autoregistered phones.
Choosing the Protocol for Auto-Registered Phones
Protocol choice dictates what firmware filename gets specified in the default configuration file for each phone model. When set to SIP, only applies to phones that can run SIP. SCCP-only phone models will still auto-register using SCCP.
The Enterprise Parameters Configuration page provides a new option to choose the phone protocol to use for phone autoregistration.
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Choosing the Protocol for a Specific Phone

The figure shows how to choose between SCCP and SIP protocols for a specific phone.
Choosing the Protocol for a Specific Phone
CIPT1 v5.03-11
When you manually add a phone, choose the protocol type.
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SIP Profile
This subtopic defines SIP profiles.
SIP Profile
Provides SIP-specific configuration information for SIP phones and SIP trunks Assigned to SIP phones from the Phone Configuration window Comes with a standard SIP profile that you can use, or copy and modify to create a new profile
CIPT1 v5.03-12
A SIP profile consists of a set of SIP attributes that are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, call pickup URI, and so on. The profiles contain some standard entries that cannot be deleted or changed.
Note A SIP URI consists of a call destination configured with a user@host format, such as xten3@CompB.cisco.com or 2085017328@10.21.91.156:5060.
A default SIP profile exists called the Standard SIP Profile that you can assign to SIP phones on the SIP phone Configuration page. You can also copy and modify the Standard SIP Profile or create a new SIP profile from scratch.
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SIP Profile Configuration Page

This figure shows how to configure SIP profiles.
SIP Profile Configuration Page

Standard SIP Profile
SIP profiles provide SIP-specific phone settings such as default telephony event payload type, registration and keepalive timers, and media ports.
CIPT1 v5.03-13
SIP profiles provide specific SIP information for the phone such as default telephony event payload type, registration and keepalive timers, media ports, and dynamic Domain Name Service (DNS) server addresses. Assigning a SIP profile is a required part of adding a Cisco SIP phone. Cisco Unified CallManager provides a Standard SIP Profile (shown in the figure) with default parameters that work in the majority of environments. If a conflict or issue arises with the default setting, you can create a new SIP profile by copying and renaming the Standard SIP Profile and making the required changes. A detailed discussion of all the parameters on the SIP Profile Configuration page is beyond the scope of this course. The learner is referred to Cisco Unified CallManager System Guides and Administration Guides.
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SIP Phone Configuration

This subtopic describes how to configure Cisco SIP phone in Cisco Unified CallManager.
SIP Phone Configuration Page

Standard SIP phone button template (2 lines and 4 speed-dial buttons) Standard Common Phone Profile is the same for SIP or SCCP phones. Enabled by default.
CIPT1 v5.03-14
The figure shows the SIP Phone Configuration window for a Cisco IP Phone 7961. A Standard SIP phone button template exists that configures two lines and four speed dials on the phone. A Standard Common Phone Profile is a required setting to add both SCCP and SIP phones. A Standard Common Phone Profile is provided (shown in the figure), that, by default, does not require a password to unlock the local phone and allows end users to change the background image setting. There are no other settings that are associated with the Common Phone Profile for Cisco Unified CallManager Release 5.0(2).
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SIP Phone Configuration Page (Cont.)
SIP phone Security Profile Settings
You must also choose a SIP Phone Security Profile and SIP Profile to assign to the phone. Optionally, assign SIP dial rules, if configured, and other parameters as desired.
Additional SIP Phone Configuration window settings are shown in the figure. You must also choose a SIP phone Security Profile (settings shown in the figure) and a SIP Profile (either the standard SIP Profile template or a custom one that you create) to assign to the phone. Optionally, assign SIP dial rules (covered later), if configured, and other parameters as desired.
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Converting SCCP Phones to SIP

This topic describes how to convert a Cisco SCCP phone to a Cisco SIP phone.
Converting Cisco SCCP Phones 7940 and 7960 to SIP

1. Download the files from Cisco SIP Phone 7940/7960 software depository. 2. Rename the file SIPDefaultGeneric.cnf to SIPDefault.cnf. 3. Rename the file SIPConfigGeneric.cnf to SIPmac_address.cnf, for each phone. 4. Reset the phones and ensure that they get IP address, gateway address, and TFTP server address.
CIPT1 v5.03-16
Cisco IP Phone 7940 and 7960 models can support the SCCP to run with either Cisco Unified CallManager, SIP, or MGCP, but not more than one simultaneously. This is possible because they load different firmware versions at bootup. This functionality is transparent to the end user, and you enable it through changes to the basic text-based configuration files that the phones download from a TFTP server. Complete these steps in order to enable SIP functionality:
Step 1
Download these files from Cisco SIP Phone 7940/7960 software depository at http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 and place them in the root directory of your TFTP server (tftpboot on a UNIX machine): P0S30100.bin: This is the SIP image. Download the file in binary format, in order to ensure that it is not corrupted. P0S3xxyy.bin (SIP Release 2.3 or earlier): The xx variable is the version number, and yy is the sub-version number. P0S3-xx-y-zz.bin (SIP Release 3.0 and later): The xx variable is the major version number, y is the minor version number, and zz is the sub-version number. SIP Release 5.0 and later: After you install this version, you are not able to revert back to versions earlier than Release 5.0. You can still change from SCCP images to SIP images, but they both must be Release 5.0 or later. For more information on this, refer to Release Notes for Cisco SIP Phone 7940/7960 Release 5.0.
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Note
Select Settings > Status > Firmware Versions to verify which image the phone uses. Different phone models use different processors. The fourth digit can help determine the model of phone for which the file is used.
OS79XX.TXT: This file tells the Cisco IP Phone 7940 and 7960 which binary file to download from the TFTP server. This file is case-sensitive and must only contain the name of the file that you want to load, without the .bin extension. For example, if you attempt to load the SIP version 2.3 software, it must contain only the line P0S30203. If you try to load versions 3.0 and later, the file name must be in the format P0S3-xx-y-zz. For example, if you attempt to load the SIP version 7.1 software, OS79XX.TXT must contain the line P0S3-07-1-00. The binary referenced here must also be present in the TFTP root directory. Without this file, the phone does not know which file it needs to retrieve in order to replace its existing software. SIPDefaultGeneric.cnf: This file is an example of a default configuration file, and contains configuration information relevant to all phones. SIPConfigGeneric.cnf: This file is similar to the previous one, except that it contains information relevant to a specific phone instead of to all phones. RINGLIST.DAT: Lists audio files that are the custom ring type options for the phones. The audio files listed in the RINGLIST.DAT file must also be in the root directory of the TFTP server. ringer1.pcm: This file is a sample ring tone that is used by the Cisco IP Phone 7940 and 7960.
Step 2 Step 3
Rename the file SIPDefaultGeneric.cnf to SIPDefault.cnf (used for global parameters on all phones). Rename the file SIPConfigGeneric.cnf to SIPmac_address.cnf, for each phone (for example, SIP002094D245CB.cnf). The MAC address must be specified in capital letters and the extension (.cnf) must be in lower case. The MAC address of the phone can be found on the sticker that is located on the bottom of the phone, or it can be found through the phone LCD screen (choose Settings > Network Configuration > MAC Address).
Note Step 4
Allow read and write file permissions on the TFTP server for the files.
Unplug the power cord or Ethernet cord (if inline power is used) in order to reset the phones. Ensure that the phones can find the TFTP server. Manually configure the phone IP address, gateway address, and TFTP server address; or configure the phone network settings from the Dynamic Host Configuration Protocol (DHCP) server. It is recommended that you not use the TFTP server on the Cisco Unified CallManager if you have a TFTP server in your current system.
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Migrating Cisco IP Phone 7970 and 7971 from SCCP to SIP

This subtopic describes how to migrate Cisco IP phones from SCCP to SIP using Cisco Unified CallManager Bulk Administration Tool (BAT).
Migrating Cisco IP Phone 7970 and 7971 from SCCP to SIP
BAT used to convert a Cisco IP phone that is in use from SCCP to SIP
CIPT1 v5.03-17
An easier, and more scalable way to convert Cisco IP phones from SCCP to SIP is provided by the Cisco Unified CallManager BAT. This method supports all phone models and is described below. A new, unused phone is set for SCCP by default. To convert this phone to SIP, perform these steps:
Step 1
Take one of these actions: To autoregister the phone, set the Auto Registration Phone Protocol parameter in Cisco Unified CallManager Administration to SIP. To provision the phone using the Cisco Unified CallManager BAT, choose the appropriate phone model and choose SIP from the BAT. To provision the phone manually, make the appropriate changes for SIP on the Phone Configuration page in Cisco Unified CallManager Administration.
Step 2 Step 3
If you are not using DHCP in your network, configure the network parameters for the phone. Power cycle the phone.
You can use the Cisco Unified CallManager BAT to convert a phone that is in use in your network from SCCP to SIP:
Step 1
To access BAT from Cisco Unified CallManager Administration, choose Bulk Administration > Phones > Migrate Phones > SCCP to SIP.
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Step 2
From the first Find Phone where drop-down list box, choose one of the following criteria: Device Name Description Directory Number Calling Search Space Device Pool Device Type Call Pickup Group LSC Status Authentication String Location Phone Load Name Security Profile From the second Find Phone where drop-down list box, choose one of the following criteria: begins with contains is exactly ends with is empty is not empty Specify the appropriate search text, if applicable.
To find all phones that are registered in the database, click Find without entering any search text.
Step 3
Step 4 Note
Step 5
To further define your query, check the check box corresponding to Search Within Results using drop-down list box. You can choose AND or OR to add multiple filters. Repeat Steps 2 through 4. Click Find. A list of discovered phones displays. Click Next. Choose the phone template from the drop-down list box.
SCCP to SIP migration will pick up only SIP protocol specific default values from this template during migration. It will not pick any other value from the template.
Step 6 Step 7 Step 8 Note
Step 9 Step 10
In the Job Information area, enter the Job description. Click the Run Immediately radio button to migrate phone records immediately, or click Run Later to migrate at a later time.
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Step 11 Step 12
Click Submit to create a job for migrating the phone records. After submitting a job for migrating phones from SCCP to SIP, make sure that you reset these phones. Reset phones using Bulk Administration > Phones > Reset/Restart Phones > Query. Use the Job Scheduler option in the Bulk Administration main menu to schedule or activate this job.
Step 13
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Adding Third-Party SIP Phones

This topic describes how to add third-party SIP phones to Cisco Unified CallManager.
Third-Party Phone Configuration

Cisco Unified CallManager Release 5.0 supports RFC 3261compliant SIP phones from third-party companies. Two categories of third-party SIP phones in Cisco Unified CallManager: Basic supports one line and consumes three license units. Advanced supports up to eight lines and video, and consumes six license units. Third-party SIP phones do not send their MAC address to Cisco Unified CallManager in their REGISTER message. Therefore, you must use Digest Authentication with third-party SIP phones. Configuration is performed on Cisco Unified CallManager and on the phone itself.
CIPT1 v5.03-18
Cisco Unified CallManager Release 5.0 supports Cisco SIP phones as well as RFC 3261compliant SIP phones from third-party companies. Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication. How to configure digest authentication is covered later in this lesson.
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Digest Authentication for Phone Identification

The figure shows how digest authentication works to identify third-party phones.
Digest Authentication for Phone Identification
Database Cisco Unified CallManager
CC M
Respond with device/line config
Device config (mac) Find associated device End User config block
Line config (1001)
Query End User config block REGISTER 1001 username=block
Third-Party SIP Phone
DN = 1001 AuthID = block MAC = 123456789ABC
CIPT1 v5.03-19
The third-party sends the REGISTER message to Cisco Unified CallManager with the username block in the message. The Cisco Unified CallManager service does a database lookup query on the end user block. If username block is found in the database, the database will locate the device and respond back to the Cisco Unified CallManager service with the phone device configuration and any associated line configuration. The phone will be allowed to register to Cisco Unified CallManager.
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Third-Party SIP Phone Configuration Steps

The figure lists the steps to add and configure a third-party phone in Cisco Unified CallManager.
Third-Party SIP Phone Configuration Steps

1. Configure the end user in Cisco Unified CallManager. 2. Configure the device in Cisco Unified CallManager. 3. Associate the device to the end user. 4. Configure the phone with the end user ID.
CIPT1 v5.03-20
Follow these steps to add and configure a third-party phone in Cisco Unified CallManager:
Configure the end user in Cisco Unified CallManager. Configure the device in Cisco Unified CallManager. Associate the device to the end user. Configure the phone with the end user ID.
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The figure shows the first three steps to add a third-party phone.

1. Configure the end user. 2. Select the end user ID in Digest User drop-down list in Phone Configuration.
3. Be sure that there is only one device associated with the end user ID.
. . .
CIPT1 v5.03-21
Follow this procedure to add a third-party phone:

Step 1 Step 2
Add the end user in Cisco Unified CallManager Administration. Add the SIP phone, and in the Protocol Specific Information pane of the Phone Configuration window, choose the end user that you configured in Step 1 from the Digest User drop-down list. Associate the device to the end user. Ensure that only one device is associated to the end user ID or registration will fail.
Step 3
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Third-Party Phone Configuration (Cont.)

4. Configure the authorization ID on the phone and ensure that the phone is enabled to use it.
The proxy address should be the Cisco Unified CallManager IP address or name.
block
If the phone uses a directory number instead of an authorization ID, the User ID in the End User Configuration in Cisco Unified CallManager must also be the directory number.
The fourth step to add a third-party phone takes place on the phone itself. You must configure the authorization ID on the phone and ensure that the phone is enabled to use it. Some thirdparty phones have a separate parameter that enables the phone to use the authorization ID. The phone in the figure is one such phone. Ensure that this setting is enabled if it exists. In the proxy address field of the third-party phone, you must also specify the IP address or Fully Qualified Domain Name of Cisco Unified CallManager. Most SIP phones allow both HTTP and TFTP configuration. The third-party SIP phone in the figure is using HTTP. HTTP is generally easier for a small number of phones, and TFTP is better when managing large banks of phones. Some phones do not have a separate authorization ID configuration, but use the directory number instead. For these phones, the User ID field in the End User Configuration window in Cisco Unified CallManager must contain the directory number. Most problems with third-party phones result from configuration mismatches between the phone and the Cisco Unified CallManager. Carefully assigning these values, and understanding if your third-party phone uses the directory number or authorization ID will reduce registration errors.
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Adding SIP Dial Rules

This topic describes SIP dial rules and KPML in Cisco Unified CallManager, which provide a consistent dialing experience for users, independent of the phone protocol.
Impact of Dial Rules or KPML on User Experience

Third-Party or Cisco SIP Phone Without SIP Dial Rules or KPML User experience is like that of a cell phone: 1. User enters digits into keypad 2. User presses Dial softkey 3. Phone sends digits and call is placed Cisco SIP Phone with Dial Rules or KPML User experience is like that of an SCCP phone: 1. User enters digits into keypad 2. When digits match a valid pattern, the phone sends digits automatically and call is placed without user having to press the Dial softkey
KPML enables digit-by-digit dialing. Dial rules tell the Cisco SIP IP Phone when enough digits are collected before call processing takes place. Both enable a user to dial a destination without pressing the Dial key.
When you make a call on an SCCP phone, Cisco Unified CallManager collects digits one at a time. When you press 1, for example, the phone sends the digit immediately to Cisco Unified CallManager. As soon as Cisco Unified CallManager collects enough digits to match a route pattern (or the interdigit timeout expires after no more digits are entered), Cisco Unified CallManager knows how to route the call and does so. Without KPML or local dial rules, SIP phones behave like a cell phone in which you enter all the digits and press the Send button, but with a SIP phone, you press the Dial button and the phone generates an INVITE message with the dialed digits.
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SIP Call Flow: Digit Collection

This subtopic describes digit collection using SIP.
SIP Call Flow: Digit Collection

KPML can also be used for digit collection when dialing on a Cisco SIP phone.
SCCP phone Digit 1 Digit 0
Cisco Unified CallManager performs digit-by-digit collection and routes the call as soon as enough digits are collected. Cisco Unified CallManager performs digit-by-digit collection and routes the call as soon as enough digits are collected; Performance impacts on Cisco Unified CallManager in terms of call signaling bandwidth. Local dial rules are configured in Cisco Unified CallManager and downloaded to the phone; Phone collects all digits and sends them to Cisco Unified CallManager.
CIPT1 v5.03-24
INVITE 1 SIP Phone Using KPML 100 Trying SUBSCRIBE 200 OK NOTIFY (0) 200 OK UNSUBSCRIBE 200 OK
SIP Phone with Local Dial Map
INVITE 10 100 Trying 200 OK
There are two methods to accomplish the same behavior with SIP phones as with SCCP phones: KPML and SIP local dial plans. KPML creates a considerable overhead. The SIP phone using KPML sends the SIP INVITE message with one digit, then the Cisco Unified CallManager sends the 100 Trying and KPML Subscribe to ask for more digits and then the SIP phone using KPML will send one digit at a time with a Notify message, and so on. There is significantly more signaling to accomplish the same behavior as the SCCP phone, so this consumes additional bandwidth and creates extra work for Cisco Unified CallManager to process more SIP messages if supporting KPML. If you have a large network (30,000 phones), and you want to use KPML, you might encounter scalability issues.
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Dial Rules and KPML Support in Cisco Unified CallManager 5.0

The figure shows the Cisco Unified CallManager support for dial rules.
Dial Rules and KPML Support in Cisco Unified CallManager Release 5.0
Dial rules support in Cisco Unified CallManager Release 5.0: Local dial rules are called SIP Dial Rules and are configured in Cisco Unified CallManager Release 5.0. Cisco 8.0 SIP phones with firmware version 8.0 download dial rules from TFTP server. Due to syntax differences, one set of dial rules must be defined for Cisco IP Phone 7940, 7960, 7911, 7941, 7961, and 797x models, while another set for Cisco IP Phone 7905 and 7912 models. KPML support in Cisco Unified CallManager Release 5.0: KMPL is implemented as per draft-ietf-sipping-kpml. KMPL is only supported on Cisco IP Phone 7911, 7941, 7961, and 797x models; Cisco IP Phone 7905, 7912, 7940, and 7960 models must use local dial rules. If the third-party SIP phone does not support KPML, you must use locally-defined digit maps.
Only Cisco IP Phone 7911, 7941, 7961, and 797x models support KPML. Other SIP phones can only use local dial rules configured on Cisco Unified CallManager.
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Dial Rules and KPML Interworking

The figure shows the behavior of phones that have either KPML enabled and SIP dial rules defined, one or the other, or neither.
Dial Rules and KPML Interworking

Is KPML Enabled? No Are SIP Dial Rules Defined? No User Must Press Dial Softkey No Yes
CIPT1 v5.03-26
Yes
Are SIP Dial Rules Defined? No
Yes
Send En-bloc INVITE
Yes
Were Enough Digits Received? No
Yes
Call Is Routed
Send En-bloc INVITE
SUBSCRIBE To KPML
Were Enough Digits Received?

Reorder Tone
If a phone has neither KPML enabled nor SIP dial rules defined, the user must press the Dial softkey to send the digits. If enough digits are received, the call is routed. If not, the user receives a reorder tone. The call-processing logic checks to see if KPML is enabled and if SIP dial rules are defined. If local dial rules are not defined, Cisco Unified CallManager sends a SUBSCRIBE message to KPML to collect the digits, and then routes the call. If KPML is enabled and local dial rules are defined, the phone sends all the digits in the INVITE message (en-bloc). If not enough digits are received, Cisco Unified CallManager sends a SUBSCRIBE message to KPML to collect the remaining digits (one-by-one) and then routes the call. So, for phones that support KPML, even if the phone has a local dial rules file configured and sends an en-bloc INVITE message, if more digits are required by Cisco Unified CallManager after receiving the INVITE message, Cisco Unified CallManager will fall back to KMPL to collect the rest of the digits.
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Dial Rules Configuration Overview

This subtopic provides an overview of dial rules configuration in Cisco Unified CallManager.
Dial Rules Configuration Overview

Dial rules enable the Cisco SIP phone to know when enough digits are collected before call processing takes place. Dial rule syntax is slightly different on Cisco IP Phone 7905/12 models than it is on Cisco IP Phone 797x, 7941, 7961, 7911 models. Dial rule syntax varies considerably on third-Party SIP phones, and some do not support dial rules.
CIPT1 v5.03-27
Dial rules enable the Cisco SIP phone to know when enough digits are collected before call processing takes place.
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SIP Phone Dial Rules Configuration Sequence

The figure shows the sequence followed when configuring SIP phone dial rules.
SIP Phone Dial Rules Configuration Sequence

Cisco Unified CallManager 1 2 TFTP Server 3 Cisco IP Phone 7970 4
7970_DialRules
DR7970_DialRules.xml
1. The administrator configures the SIP dial rules and associates them with the SIP phone. 2. Cisco Unified CallManager notifies the TFTP server of the change. 3. The TFTP server rebuilds the dial rules configuration file of the phone. 4. The administrator resets or restarts the phone and the dial rules file is downloaded to the phone.
Follow these steps to configure the SIP phone dial rules sequence:
Configure the SIP dial rules and associate the dial rules with the SIP phone. The Cisco Unified CallManager database sends a change notification to the TFTP server, which triggers the TFTP server to build a new set of files for the SIP phone. The TFTP server rebuilds the dial rules configuration file or the configuration file for the SIP phone. When all the updates to the dial rules have been made to the Cisco Unified CallManager database, click the Reset or Restart button to apply the change to the phone.
These configuration files are generated based on the SIP phone type: Model 7970/71, 7961, 7941, 7911: DR<dialrules>.xml Model 7960/40: <dialprule>.xml Model 7905: Parameter in ld<mac> Model 7912: Parameter in gk<mac>
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Dial Rule Patterns and Phone Models

This table compares Cisco SIP phones in terms of how they support KPML and which dial rule pattern they use.
Dial Rule Patterns and Phone Models

Supports KMPL Cisco IP Phone 7905/12 Cisco IP Phone 7940/60 Cisco IP Phone 797x, 7941, 7961, 7911 No No Uses Which Dial Rule Pattern
7905_7912 7940_7960_OTHER
Yes
7940_7960_OTHER
Even if the phone supports KMPL, the recommendation is to use SIP dial rules to increase the performance of Cisco Unified CallManager.
Cisco SIP Phone 7911, 7941, 7961, 7970, and 7971 models use the 7940_7960_OTHER dial rules patterns. These phones support KPML, which allows the digits to be sent to Cisco Unified CallManager without requiring you to press the Dial softkey. To increase the performance of Cisco Unified CallManager (increasing the number of calls that get processed), Cisco recommends that you configure SIP dial rules. Cisco SIP Phone 7940 and 7960 models use the 7940_7960_OTHER dial rules patterns and do not support KPML. If you do not configure a SIP dial plan for these phones, you must press the Dial softkey or wait a specified time before digits are sent to Cisco Unified CallManager for processing. This extra step delays the processing of the actual call. Cisco SIP Phone 7905 and 7912 models use the 7905_7912 dial rules patterns and do not support KPML. If you do not configure a SIP dial plan for these phones, you must press the Dial softkey or wait a specified time before digits are sent to Cisco Unified CallManager for processing. This extra step delays the processing of the actual call.
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Dial Rule Pattern Syntax

The figure shows the pattern formats and their meaning for the two dial rules.
Dial Rule Pattern Syntax

Dial Rule Pattern Format
Period (.) Asterisk (*)
Meaning
Matches any digit Matches one or more characters Acts as the terminating key
7940_7960_OTHER 7905_7912
Pound sign (#)
On 7905_7912 pattern, you must precede the # (or other terminating key) with a > Causes the phone to generate a secondary dial tone More digits can be entered; hyphen must appear at the end Sets the timeout to n seconds
Comma (,)
7905_7912 Variances
Hyphen (-) tn
Example: 6,..... will match 6, play secondary dial tone, then match any 5-digit directory number.
The pattern formats and meanings for the 7940_7960_OTHER pattern are as follows: Period (.) matches any character. Asterisk (*) matches one or more characters. The * gets processed as a wildcard character. You can override this by preceding the * with a backward slash (\) escape sequence, which results in the sequence \*. The phone automatically strips the \ so that it does not appear in the outgoing dial string. When * is received as a dial digit, it gets matched by the wildcard characters * and period (.). Comma (,) causes the phone to generate a secondary dial tone. Example: 7.... will match any 4-digit directory number that starts with 7. 8,..... will match 8, play a secondary dial tone, then match any 5-digit directory number. The pattern formats and meanings for the 7905_7912 pattern are as follows: Period (.) matches any digit. Hyphen (-) means more digits can be entered. If this character is needed, it must appear at the end of an individual rule. For example, 1408t5- is legal, but 1408t5-3... is illegal. Pound sign (#) acts as the terminating key, and termination can be applied only after matching hits >#. So >* means that the terminating character specifies the asterisk (*); that is, the terminating key must follow the greater-than sign (>). Characters "tn" set the timeout to n seconds.
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Configuring SIP Dial Rules

The figure shows the high-level steps to configure SIP dial rules.
Configuring SIP Dial Rules

1. Choose Call Routing > Dial Rules > SIP Dial Rules 2. Choose the dial pattern depending on the phone type
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Follow these steps to configure SIP dial rules:

Step 1 Step 2
Choose Call Routing > Dial Rules > SIP Dial Rules in the Cisco Unified CallManager menu. Choose the dial pattern.
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Configuring SIP Dial Rules (Cont.)

Dial parameters:
Pattern = Digits Timeout = Number of seconds to wait after match to dial 0 means dial immediately User = Tag added to the dialed number. IP and phone are valid values (not casesensitive). Use only for 7940_7960_OTHER dial rules.
The figure shows a simple example of a basic dial rule for North America for the Cisco IP Phone 7940, 7960, 7911, 7941, 7961, and 797x models. Obviously, dial rules can be more complex in other countries. The first pattern has the description Operator and pattern value 0. If the phone sees a pattern of 0, route the call immediately to the internal attendant. The second pattern External International has the pattern 9,011* (the , signaling Cisco Unified CallManager to play dial tone after the digit 9 is pressed). The * indicates a variable length pattern, so after five seconds with no more digits received, the phone generates the INVITE with the dialed digits. The local dial plan is not a complete dial plan. It does not tell the phone where to route the call. It simply tells the phone how many digits to collect before considering it a valid pattern and what the timeout value is. Dial rules must be manually configured, but doing so does not take long (5 or 10 minutes). Cisco Unified CallManager does not do any prepopulation of dial rules based on route plans. It would not be practical given the complexity of route plans that include translation patterns, route filters, partitions and calling search spaces, valid local extensions, and national numbering plan, and so on to derive the local dial rules that a given organization needs. Following are definitions of the options available from the dial parameters drop-down list box: Pattern: Use this parameter for 7905_7912 and 7940_7960_OTHER dial rules. Button: This parameter specifies the dial pattern to which a line button applies. If a user initiates a call on line button 1, only the dial pattern that is specified for Button 1 applies. If this optional parameter is not configured, the dial pattern applies to all lines. It only applies to the Cisco SIP Phone 7940, 7941, 7960, 7961, 7970, and 7971 models. The administrator must enter a button number as the value. The button number corresponds to the order of the buttons on the side of the screen that is on the phone, from top to bottom, with 1 being on top. The Cisco SIP Phone 7940 and 7941 models have two line buttons, the Cisco IP Phone 7960 and 7961 models have six line buttons, and the Cisco IP Phone 7970 and 7971 models have eight line buttons.
Timeout: This parameter specifies the time, in seconds, before the system times out and dials the number as entered by the user. To have the number dial immediately, specify 0. Use this parameter only for 7940_7960_OTHER dial rules. User: This parameter represents the tag that automatically gets added to the dialed number. Valid values include IP and Phone for this tag, which is not case sensitive. Use this parameter only for 7940_7960_OTHER dial rules.
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Add the SIP Dial Rules to the Phone

The figure shows how to add the configured SIP dial rule to the phone.
Add the SIP Dial Rules to the Phone

Add SIP dial rule in the Protocol Specific Information pane of the Phone Configuration window.
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After you define the dial rule, the second step is to assign the SIP dial rule to the phone in the Protocol Specific Information pane of the Phone Configuration window.
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Summary
Summary
Cisco Unified CallManager supports SIP on Cisco IP Phone 7970/71, 7960/61, 7940/41, 7911, and 7905/12 models as well as selected third-party phones. Cisco Unified CallManager integrates native SIP (RFC 2833) and SIMPLE support on both line-side and trunk-side interfaces for a variety of endpoints, gateways, and applications. External SIP proxy server is no longer required. DTMF and KPML can be used for digit collection when dialing on a SIP phone. Cisco Unified CallManager collects digits either one at a time, or, more efficiently, using SIP dial rules. Provisioning a Cisco SIP phone is just like provisioning a Cisco IP phone controlled by SCCP.
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Summary (Cont.)
SIP profile provides SIP-specific configuration information for SIP phones and SIP trunks. Conversion of a Cisco SCCP phone to a Cisco SIP phone is done either by manual replacement of TFTP configuration files, or by using Cisco Unified CallManager BAT. Third-party SIP phones do not send their MAC address at boot time; all the configuration is performed on Cisco Unified CallManager and on the phone itself. When you configure a SIP dial plan and associate it with a Cisco SIP phone, TFTP server builds a new set of files for the Cisco SIP phone. Once the you initiate the phone reset, the new dial rules are applied to the phone.
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References
For additional information, refer to these resources: Cisco Unified CallManager Administration Guide, Release 5.0 http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/sys_ad/5_0_2/ccmcf g/index.htm: Configuring Non-Cisco SIP phones Trunk Configuration SIP Phone Security Profile Configuration SIP Dial Rules Configuration SIP Route Pattern Configuration SIP Profile Configuration Common Phone Profile Configuration Application User Configuration
Cisco Unified CallManager System Guide, Release 5.0 http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/5_0/sys_ad/5_0_2/ccmsy s/index.htm: Dial Rules Overview Application Users and End Users Understanding Session Initiation Protocol (SIP) Understanding Cisco Unified CallManager Trunk Types Cisco Unified IP phones
Internet Engineering Task Force (IETF), collection of relevant RFC standards: www.ietf.org
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Lesson 5
Managing Endpoints with Cisco Unified CallManager BAT

Overview
Adding, updating, and deleting phones, users, and gateway ports are important functions in the day-to-day activities of a Cisco Unified CallManager administrator. When you use Cisco Unified CallManager Administration, each database transaction requires an individual manual operation. Manually adding and configuring large numbers of these entities can be timeconsuming and tedious. The Cisco Unified CallManager Bulk Administration Tool (BAT) automates the process and achieves faster add, update, and delete operations so that you can focus on more business- and network-critical activities. This lesson describes Cisco Unified CallManager BAT and Tool for Auto-Registered Phones Support (TAPS). Cisco Unified CallManager BAT is a web-based application for Cisco Unified CallManager that allows you to add, update, or delete a large number of similar telephones, users, or ports at the same time. TAPS works in conjunction with Cisco Unified CallManager BAT to update MAC addresses and download a predefined configuration for new phones.
Objectives
Upon completing this lesson, you will be able to use Cisco Unified CallManager BAT and TAPS to bulk-add and autoregister Cisco IP phones, users, and ports in an IP telephony network. This ability includes being able to meet these objectives: Identify the major Cisco Unified CallManager BAT features and components of the Cisco Unified CallManager BAT application Use the Cisco Unified CallManager BAT Wizard to perform bulk configuration tasks Identify the two ways to create CSV files for importing data into Cisco Unified CallManager BAT Describe how to schedule bulk transactions Describe the purpose of TAPS Install, activate, and use TAPS to autoregister Cisco IP phones
Cisco Unified CallManager BAT Overview

This topic gives an overview of the Cisco Unified CallManager Bulk Administration Tool (BAT).
Cisco Unified CallManager BAT Overview

Cisco Unified CallManager BAT allows management of many devices/records within a short period of time.
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The Cisco Unified CallManager BAT, a web-based application, performs bulk transactions to the Cisco Unified CallManager database. Cisco Unified CallManager BAT lets you add, update, or delete a large number of similar phones, users, or ports at the same time. When you use Cisco Unified CallManager Administration, each database transaction requires an individual manual operation, while Cisco Unified CallManager BAT automates the process and achieves faster add, update, and delete operations. It is therefore an ideal tool for administrators who manage many devices within their voice network. For example, if you need to roll out a new site and therefore have to add 150 phones using nearly the same configuration, it is much easier to use Cisco Unified CallManager BAT than to configure all of them manually.
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Cisco Unified CallManager BAT Overview (Cont.)

The Cisco Unified CallManager Release 5.0 BAT characteristics:
Integrated with the Cisco Unified CallManager Administration pages and available by default (no plug-in required) All pages are available under the Bulk Administration menu Same look and feel as the Cisco Unified CallManager Administration pages Supports internationalization MLA support for Cisco Unified CallManager BAT pages TAPS is now part of bulk administration
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Cisco Unified CallManager BAT uses a completely new developed user interface. Cisco Unified CallManager BAT is now part of the Cisco Unified CallManager Administration page and the user interface has the following characteristics: No plug-in is required, as required for earlier versions of Cisco Unified CallManager BAT. Cisco Unified CallManager BAT now comes as a preinstalled function of Cisco Unified CallManager. All pages are available under the Bulk Administration menu of the Cisco Unified CallManager Administration page. The Cisco Unified CallManager BAT user interface pages have the same look and feel as the Cisco Unified CallManager Administration pages. Cisco Unified CallManager BAT now supports internationalization. As Cisco Unified CallManager BAT is now part of the Cisco Unified CallManager Administration pages, it fully supports multilevel administration access (MLA). TAPS, which was a plug-in that required Cisco Customer Response Solutions (CRS) in earlier versions, is now fully integrated into Cisco Unified CallManager and is part of bulk administration.
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Cisco Unified CallManager BAT Overview (Cont.)
Manage config files
Manage devices Schedule configuration Manage TAPS service
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The Cisco Unified CallManager Bulk Administration menu contains these functionalities: Managing Files: Use this section to manage Cisco Unified CallManager BAT files. The menu allows you to upload, download, or delete files used for Cisco Unified CallManager BAT. Managing Devices: This section contains menus and submenus for each device that can be configured using Cisco Unified CallManager BAT. These items appear on the menus: Phones: Used to manage phones Users: Used to manage end users Phones & Users: Used to insert new phones and indicate users who should own those phones Managers/Assistants: Used to add or delete Cisco Unified IP Manager Assistants (IPMAs) and generate IPMA reports User Device Profiles: Used to manage user device profiles Gateways: Used to add or delete gateways and generate gateway reports Forced Authorization Codes: Used to add or delete forced authorization codes Client Matter Codes: Used to add or delete client matter codes Call Pickup Group: Used to add or delete call pickup groups
Scheduling Jobs: To schedule Cisco Unified CallManager BAT jobs, use the Job Scheduler, which allows you to prepare configuration changes and apply the changes at a predefined time. Use this section to start, cancel, delete, and verify scheduled jobs configured via the manage device section menus. TAPS: Use the TAPS section to manage integrated TAPS service configuration on Cisco Unified CallManager.
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Cisco Unified CallManager BAT Components

This subtopic describes Cisco Unified CallManager BAT functional components.
Cisco Unified CallManager BAT Components

Cisco Unified CallManager BAT administration consists of these features:
Cisco Unified CallManager BAT templates are used to define general settings that fit all of the devices that should be added. CSV files are used to define devices and record specific settings that should be bulk configured. Adding, updating, and deleting devices and records is done automatically based on queries and CSVs. Additions, updates, and deletions can be scheduled to be performed at a defined time.
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To fully support administrators in managing their Cisco Unified CallManager, BAT includes these functional components: Cisco Unified CallManager BAT templates are used to define general settings that fit all of the devices that should be added, such as phone button templates, security profiles, locations, or regions. Comma-separated values (CSV) files are used to define devices and record specific settings that should be bulk configured and cannot be defined within the template, such as directory numbers and names. Adding, updating, and deleting devices and records is done automatically based on selections from CSV values or queries. You can schedule additions, updates, and deletions to be performed at a defined time.
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Cisco Unified CallManager BAT Support for Devices and Records

This subtopic describes how Cisco Unified CallManager BAT supports devices and records.
Cisco Unified CallManager BAT Support for Devices and Records

Cisco Unified CallManager BAT can be used on Cisco Unified CallManager Release 5.0 to add, update, and delete these devices and records:
Cisco IP phones Voice gateways CTI ports Users User device profiles IPMAs Ports on a Cisco Catalyst 6000 FXS Analog Interface Module Cisco VG200 series analog gateways and ports Forced authorization codes Client matter codes Call pickup groups
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You can use Cisco Unified CallManager BAT to work with these types of devices and records: Add, update, and delete Cisco IP phones, including voice gateway phones, computer telephony integration (CTI) ports, and H.323 clients, plus you can migrate phones from Skinny Client Control Protocol (SCCP) to Session Initiation Protocol (SIP) Add, update, and delete users Add, update, and delete User Device Profiles Add, update, and delete Cisco IPMA Add, update, and delete ports on a Cisco Catalyst 6000 FXS Analog Interface Module Add or delete Cisco VG200 analog gateways and ports Add or delete forced authorization codes Add or delete client matter codes Add or delete call pickup groups You can also work with these devices in combination with user information. For example, when you add CTI ports and users, Cisco Unified CallManager BAT allows you to "Enable CTI Application Use." This saves time when you are adding users who have applications that require a CTI port, such as Cisco IP SoftPhone.
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Bulk Provisioning Service

Bulk Provisioning Service (BPS) administers and maintains all jobs that are submitted through Bulk Administration menu of Cisco Unified CallManager Administration.
Bulk Provisioning Service

Bulk Provisioning Service is used by Cisco Unified CallManager BAT to perform tasks. Bulk Provisioning Service is listed under database services in the service activation pages. Service should be activated for scheduled jobs to be executed. Can be activated only on the first node in a cluster.
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You can start Bulk Provisioning Service from the Cisco Unified CallManager Serviceability tool. Therefore, when using the BPS, note the following: On Cisco Unified CallManager service activation page, BPS is listed under database services. BPS should be activated for scheduled jobs that are going to be executed. Otherwise, the jobs cannot be performed. BPS can be activated only on the first node in a Cisco Unified CallManager cluster, because the first node is the server that handles the entire configuration within the cluster. The BPS Server service parameter determines if the service is activated on a particular server or not.
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Bulk Provisioning Service (Cont.)
Activate bulk provisioning service
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To enable BPS on Cisco Unified CallManager, choose Cisco CallManager Serviceability > Tools > Service Activation and select Cisco Bulk Provisioning Service from the Database and Admin Services list.
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Cisco Unified CallManager BAT Configuration Process

This topic describes the bulk configuration process.
Cisco Unified CallManager BAT Configuration Process Overview

The Cisco Unified CallManager BAT configuration process includes these steps:
Step 1: Configure Cisco Unified CallManager BAT template. Step 2: Create the CSV data input file. Step 3: Validate the data input files. Step 4: Insert the devices into the Cisco Unified CallManager database.
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Cisco Unified CallManager BAT uses a multistep process to prepare the bulk configuration transaction. The Cisco Unified CallManager BAT configuration process includes these tasks:
Step 1
Configure the template for data input. You can create Cisco Unified CallManager BAT templates for these types of device options: Phones: All Cisco IP phone models and Cisco ATA 186, Cisco voice gateway chalice (VGC) phones, CTI ports, and H.323 clients Gateways: Cisco VG200 and ports for the Cisco Catalyst 6000 FXS Analog Interface Module User device profiles: Cisco IP Phone 7900 Series and Cisco SoftPhone Define a format for the CSV data file. You can use the Cisco Unified CallManager BAT spreadsheet or a text editor to create the CSV data file. Validate the data input files with the Cisco Unified CallManager database. Cisco Unified CallManager runs a validation routine that checks the CSV file and the template for errors against the first-node database. Insert the devices into the Cisco Unified CallManager database.
Step 2 Step 3
Step 4
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Configuring BAT Templates

This subtopic describes how to configure BAT templates.
Configuring Cisco Unified CallManager BAT Templates
Enter the phone template name
Configure device parameters
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Cisco Unified CallManager BAT gives you a fast and efficient way to add, update, or delete large numbers of phones in batches, rather than performing individual updates through Cisco Unified CallManager Administration. You can use Cisco BAT to work with these types of phones: Cisco IP phones (all models) CTI ports H.323 clients Use Cisco Unified CallManager BAT phone templates to define the common phone attributes to add a group of new phones. Prior to creating the template, make sure phone settings such as device pool, location, calling search space, button template, and softkey templates have already been configured in Cisco Unified CallManager Administration. You cannot create new settings in Cisco Unified CallManager BAT. After you create a phone template, you can add lines, services, and speed dials. Use this procedure to create a phone template:
Choose Bulk Administration > Phones > Phone Template in the menu. The Find and List Phone Templates window displays. Click Add New button. The Add a New Phone Template window displays. From the Phone Type drop-down list box, choose the phone model for which you are creating the template. Click Next. Choose the device protocol from the Select the Device Protocol drop-down list box. Click Next. The Phone Template Configuration window displays with fields and default entries for the chosen device type.
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Step 5 Step 6
In the Template Name field, enter a name for the template. The name can contain up to 50 alphanumeric characters. Example: Sales_7960. In the Device Information area, enter the phone settings that this batch has in common. Some phone models and device types do not have all the attributes that the table lists. After you have entered all the settings for this Cisco Unified CallManager BAT phone template, click Save. When the status indicates that the transaction has completed, you can add line attributes. Find the Line Template you want to add lines to. In the Line Template Configuration window, click Line [1] Add a new DN in the Associated Information area. The Line Template Configuration window displays.
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Configuring Cisco Unified CallManager BAT Templates (Cont.)
Enter the line template name
Configure line parameters
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To add one or more lines to the Cisco Unified CallManager BAT template or to update existing lines, follow this procedure. The button template in use for this Cisco Unified CallManager BAT determines the number of lines that you can add or update. You can create a master phone template that has multiple lines. Then, you can use the master template to add phones with a single line or up to the number of lines in the master template. After you clicked Line [1] Add a new DN in the Associated Information area of the Line Template Configuration window, the Line Template Configuration window displays:
Step 1
Enter or choose the appropriate values for the line settings, such as Route Partition, Calling Search Space, Presence Group, and others. Keep in mind that all phones or user device profiles in this batch will use the settings that you choose for this line. Treat all fields as optional. Click Save. Cisco Unified CallManager BAT adds the line to the phone template configuration. Repeat the described procedure to add settings for any additional lines.
The maximum number of lines that display for a Cisco Unified CallManager BAT template depends on model and button template that you chose when you created the Cisco Unified CallManager BAT phone template.
Step 2 Step 3 Note
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Uploading Data Input Files

This subtopic describes how to upload a data input file to be ready to start the bulk additions.
Uploading Data Input Files
Locate the file to upload
Select the Target and Transaction Type
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To add devices or records to Cisco Unified CallManager using Cisco Unified CallManager BAT, you first must upload a data input text file in CSV format.
Note The procedure for creating the CSV file is described in detail later in this lesson.
The data input text file will later be validated and used for the device insertion procedure. Use this procedure to upload the CSV file to the Cisco Unified CallManager server:
Choose Bulk Administration > Upload/Download Files. The Find and List Files window displays. Click Add New. The File Upload Configuration window displays. In the File text box, enter the full path of the file you want to upload or click Browse and locate the file. From the Select the Target drop-down list box, choose the target you want to use the file for. From the Transaction Type drop-down list box, choose the transaction type the file defines. If you want to overwrite an existing file with the same name, check the Overwrite File if it exists check box. Click Save. The status displays that the upload is successful.
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Validating Data Input Files

This subtopic describes how to validate data input file.
Validating Data Input Files
Select the file to validate
Select the template to be used Start Validation
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When you choose Validate Phones, the system runs a validation routine to check that the CSV data file and Cisco Unified CallManager BAT phone template have all required fields populated, such as device pool and locations. The validation also checks for discrepancies with the first-node database. To validate your CSV data file phone records, use this procedure:
Step 1 Step 2
Choose Bulk Administration > Phones > Validate Phones. The Validate Phones Configuration window displays. Either select Validate Phones Specific Details radio button to validate phone records that use a customized file format, or select Validate Phones All Details radio button to validate phone records from an exported phones file that was generated by using the All Details option. In the File Name drop-down list box, choose the CSV data file that contains the unique details for the phones or other IP telephony devices. You have uploaded this file for this specific bulk transaction. For the Specific Details option, in the Phone Template Name drop-down list box, choose the Cisco Unified CallManager BAT phone template that you created for this type of bulk transaction. To verify the chosen CSV data file with the first-node database, click Submit. The job is submitted and executed immediately.
Step 3
Step 4
Step 5 Step 6
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Verifying Validation Results

This subtopic describes how to verify that the validation has completed successfully.
Verifying Validation Results
Verify that all records were processed successfully
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Follow this procedure to ensure that the validation procedure successfully processed all CSV file records:
Choose Bulk Administration > Job Scheduler, select Show Completed Jobs and click Find to search for all the completed jobs. Select your validation job from the list of completed jobs shown. Verify that all records are processed successfully. The Number of Records Processed column displays the number of records that have been successfully processed. The Number Of Records Failed column displays the number of records which could not be processed and could not be inserted. The Total Number Of Records column shows the total number of records found in the job, in this case the lines in the CSV. To view detailed information about the current job, click on the Log File Name link.
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Inserting IP Phones into Cisco Unified CallManager Database

Now the devices can be added to the Cisco Unified CallManager database.
Inserting IP Phones into Cisco Unified CallManager Database
Select the file and template
Select immediately
Start the insertion

To add phones, CTI ports, or H.323 clients, use this procedure:

Step 1 Step 2
Choose Bulk Administration > Phones > Insert Phones. The Insert Phones Configuration window displays. Either click Insert Phones Specific Details radio button to insert phone records that use a customized file format, or click Insert Phones All Details radio button to insert phone records from an exported phones file that was generated by using the All Details option. In the File Name drop-down list box, choose the CSV data file that you created for this specific bulk transaction. Check the Allow Update Phone with Custom File check box to allow updating the phone with the custom file you chose. Checking the Override the existing configuration check box overwrites the existing phone settings with the information that is contained in the file that you want to insert. For the Specific Details option, in the Phone Template Name drop-down list box, choose the BAT phone template that you created for this type of bulk transaction. If you did not enter individual MAC addresses in the CSV data file, you must check the Create Dummy MAC Address check box. If you are adding CTI ports, the dummy MAC address option provides a unique device name for each CTI port in the form of dummy MAC addresses. This field automatically generates dummy MAC addresses in the following format: XXXXXXXXXXXX, where X represents any hexadecimal (0-9 and A-F) number. If you do not know the MAC address of the phone that will be assigned to the user, choose this option. When the phone is plugged in, a MAC address registers for that device. If you supplied MAC addresses or device names in the data input file, do not choose this option.
Step 3
Step 4
Step 5
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Note
You can update the phones or devices later with the correct MAC address by manually entering this information into Cisco Unified CallManager Administration or by using TAPS.
In the Job Information area, enter the job description. Click the Run Immediately radio button to insert the phone records immediately, or click Run Later to insert the phone records at a later time. Click Submit to create a job for inserting the phone records.
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Verifying the Status of Insertion

This subtopic describes how to verify that the insertion has completed successfully.
Verifying the Status of Insertion
Verify a successful insertion
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In the job scheduler, verify that the phones have been inserted successfully.
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Creating CSV Files

This topic describes how you can create CSV files for Cisco Unified CallManager BAT.
Creating CSV Files

Cisco provides a template to create CSV files which have the mandatory format to work with Cisco Unified CallManager BAT:
The template is a Microsoft Excel spreadsheet that uses macros. The template can be personalized for specific needs. The file can also be created using a text editor, such as Microsoft Notepad: Use a separate line to enter data for each record. Separate each data field with a comma and include comma separators for blank fields. Do not enter blank lines, otherwise errors occur during the insert transaction.
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Cisco Unified CallManager BAT uses data that is entered in CSV file format to provide information for insert transactions to the Cisco Unified CallManager database on the first node server. By using the CSV data format, you can build a textual file that contains data records in a tabular format. You can create a CSV data file by using a text editor, such as Microsoft Notepad. You must use a separate line to enter data for each record. Separate each data field with a comma and include comma separators for blank fields. Enter data on every line in the data file, because an error occurs during the insert transaction if you enter a blank line in a CSV file. When you insert the data records to the Cisco Unified CallManager database, Cisco Unified CallManager BAT accesses a set of designated folders that reside on the server that is running the first-node database. For Cisco Unified CallManager BAT to access the appropriate CSV data file for the transaction, you must upload the CSV data file to the first-node database server of Cisco Unified CallManager.
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Creating CSV Files (Cont.)
Select bat.xlt
Download
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Follow this procedure to use the Cisco BAT template to create a CSV file for a Cisco Unified CallManager BAT transaction:
Choose Bulk Administration > Upload/Download Files. Select the bat.xlt from the list of available files. Click Download Selected to download the file to a specified destination on your PC.
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Using Excel template create the file format
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As soon as you have downloaded the template, double-click to create a Microsoft Excel .xls file based on that template and perform the next steps:
Step 4
Click Create File Format to define how the file should be formatted.
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Add/remove fields Order fields
Commit changes
Apply settings
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Step 5 Step 6 Note
From the newly opened window, add or remove fields so that the template includes all the fields that should be available within the CSV file. Sequence the fields into an order that meets your personal needs.
The order is only used to handle the structure of the file. It does not influence the Cisco Unified CallManager configuration.
Step 7 Step 8
Click Create to apply the settings. On the newly opened page, commit the changes to generate the new file format.
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Populate fields with values
CIPT1 v5.03-21
Step 9
As soon as the file format is configured, you can start entering the values into the Excel spreadsheet.
Export to CSV
Save changes
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Step 10 Step 11
After all data fields have been populated, click Export to BAT Format to create the new CSV file. The system prompts you for a file destination and suggests a path and name for it. Click OK to save the CSV file on your PC.
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Scheduling Cisco Unified CallManager BAT Jobs

This topic describes the steps to schedule Cisco Unified CallManager BAT jobs on Cisco Unified CallManager.
Scheduling Cisco Unified CallManager BAT Jobs

The following tasks need to be performed to schedule Cisco Unified CallManager BAT jobs on Cisco Unified CallManager:
1. Create a task. 2. Specify the time for starting the task.
CIPT1 v5.03-23
You can schedule bulk transactions and specify a time for starting these transactions. All jobs that are submitted through the Bulk Administration menu in the Cisco Unified CallManager Administration queue on the BPS. Depending on the start time specified for a job, the transaction starts running. If no start time is mentioned, the transactions execute in the order in which they are received.
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Scheduling Cisco Unified CallManager BAT Jobs: Create a Task
Specify to run the task later
Create a Job
To create a task in Cisco Unified CallManager, perform these steps:

Step 1 Step 2
Specify that a task will run later, not immediately, by clicking on the Run Later radio button. Click Submit to create the Cisco Unified CallManager BAT job.
Scheduling Cisco Unified CallManager BAT Jobs: Create a Task (Cont.)
Select the task scheduled
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Step 3
From the job scheduler, select the task from the list.
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Scheduling Cisco Unified CallManager BAT Jobs: Specify Time
Configure start time for scheduled task
Activate the job
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Step 4 Note Step 5
Configure the date and time when the task should be started.
Task date and time is based on the local date and time of the Cisco Unified CallManager.
Activate the job by clicking Activate Job button.
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Cisco TAPS Overview

This topic describes the purpose of the Cisco Unified CallManager Tool for Auto-Registered Phones Support (TAPS).
Introduction to TAPS
1 3 4
Call TAPS DN Dial own phone DN Phone Configuration Dummy MAC Address
Actual MAC Address
1. 2. 3. 4. 5. 6.
Cisco Unified CallManager BAT adds dummy MAC address into Cisco Unified CallManager Administration Administrator plugs phones into network User or administrator dials a TAPS directory number User or administrator dials own directory number Cisco Unified CallManager downloads configuration Phone updates Cisco Unified CallManager with correct MAC address
CIPT1 v5.03-27
The TAPS application enables you to call the TAPS directory number and download preconfigured phone settings for your directory number. TAPS works in conjunction with Cisco Unified CallManager Administration to provide these capabilities: Configure TAPS: Allows you to enable TAPS usage for all phones that use autoregistration, or to limit TAPS use to only phones that are added through BAT with dummy MAC addresses. Secure TAPS: Allows you to keep some directory numbers from being updated through TAPS. User locales for TAPS: Allows you to choose the languages for TAPS prompts. When new phones are added to Cisco Unified CallManager, TAPS works in conjunction with Cisco Unified CallManager BAT to update phones that were added to Cisco Unified CallManager BAT using dummy MAC addresses. After Cisco Unified CallManager BAT has been used to bulk-add telephones with dummy MAC addresses to Cisco Unified CallManager Administration, you can plug the telephones into the network. You can dial a TAPS directory number and then dial the directory number of the telephone, which causes the phone to download its configuration. At the same time, the telephone is updated in Cisco Unified CallManager Administration with the correct MAC address. You must make sure that autoregistration is enabled in Cisco Unified CallManager Administration for TAPS to function.
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Autoregistration Options for TAPS

This subtopic describes the available options for TAPS for autoregistration of phones.
Autoregistration Options for TAPS

Set one of the following options for using TAPS to update autoregistration of phones:
Using TAPS with new phones Updates MAC addresses and downloads a predefined configuration for new phones Using TAPS for reloading configurations Reloads the configuration for replacement phones
CIPT1 v5.03-28
You can set these options for using TAPS to update autoregistration of phones: Using TAPS with new phones: After Cisco Unified CallManager BAT has added the new phone configurations with dummy MAC addresses in Cisco Unified CallManager Administration, you can plug the phones into the network. You can dial a TAPS directory number to download the phone configuration. At the same time, the phone is updated in Cisco Unified CallManager Administration with the correct MAC address. You must ensure that autoregistration is enabled in Cisco Unified CallManager Administration for TAPS to function. Example: You have 100 new-hire employees starting on Monday. You must add these users and their new phones to Cisco Unified CallManager Administration. You can use Cisco Unified CallManager BAT to create a phone template for these 100 phones and a CSV data file for phones and users. By using the dummy MAC address option in the CSV data file, you do not need to add the individual MAC addresses for the new phones. With autoregistration enabled in Cisco Unified CallManager, you can plug the phones directly into the network. You or the new employee can load the configuration by dialing the TAPS directory number and following the voice-prompt instructions. Using TAPS for reloading configurations: When you must replace an existing phone that is not functioning, you can use TAPS to download the existing phone configuration to the new phone. After the user receives the new phone and plugs the phone into the network, the user dials the TAPS directory number to download the configuration of the previous phone. The user makes no configuration changes during this process. In Cisco Unified CallManager Administration, you must enable autoregistration. You must ensure that you configure TAPS usage for all phones to enable a user to download an existing phone configuration.
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Example: John's Cisco IP Phone 7940 is short-circuited during a lightning storm. He receives a new Cisco IP Phone 7940 and plugs it into the network. John can dial the TAPS directory number, and the new phone will download the configuration that was previously used for the damaged phone. TAPS automatically updates device information in Cisco Unified CallManager Administration.
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Using TAPS
This topic describes how to use TAPS.
TAPS Installation Prerequisites

With Cisco Unified CallManager BAT Release 5.0(1), TAPS is installed separately. TAPS installation prerequisites: Ensure that the first-node database for Cisco Unified CallManager is configured and running. Ensure that the Cisco CRS server is configured. Ensure that the latest Cisco Unified CallManager BAT is installed on the first-node database server for Cisco Unified CallManager.
CIPT1 v5.03-29
Before Cisco Unified CallManager BAT Release 5.0(1), TAPS installation and uninstallation for Cisco Unified CallManager was part of the BAT installation program. With Cisco Unified CallManager BAT Release 5.0(1) and above, TAPS is installed separately as an optional component of BAT. It is installed on both the Cisco Unified CallManager first-node server and the Cisco CRS applications server. These prerequisites apply to the TAPS installation for BAT: Ensure that the first-node database for Cisco Unified CallManager is configured and running. The first-node database can reside on its own server or on the same server as Cisco Unified CallManager. Before installing TAPS, ensure that the latest BAT release is installed on the first-node database server for Cisco Unified CallManager. You must have the IP addresses for the Cisco Unified CallManager first-node server. Ensure that the Cisco CRS server is configured. The Cisco CRS Version 4.5 application can reside on its own dedicated server or it can be located on the same server as Cisco Unified CallManager. Be sure to use the locale installer to create the country-specific TAPS prompts.
Note When you install TAPS in a network with a dedicated CRS server, you must run the TAPS installation program again on the CRS server. Use CRS online help for assistance with installation and configuration.
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TAPS Installation
This subtopic describes how to install TAPS.
TAPS Installation
Installation of TAPS on a CRS application server includes these steps:
Step 1: Log in to the first node with administrative privileges. Step 2: Using Applications > Plug-ins, list all the plug-ins. Step 3: Download Cisco TAPS for Windows to the server running CRS. Step 4: Install Cisco TAPS for Windows on the CRS server.
CIPT1 v5.03-30
Follow these steps to install TAPS on a CRS application server:

Log in with administrator privileges to the system that is running the Cisco Unified CallManager first-node database. Choose Applications > Plugins. The Find and List Plugins window displays. In the list of plug-ins, find Cisco TAPS for Windows. Click Download and save the plug-in to your Windows server where the CRS application server is installed. The Welcome window for the installation wizard opens. This installation program installs TAPS on the CRS applications server. Click Next.
When you install TAPS in a network with a dedicated CRS server, you must run the TAPS installation program again on the CRS server. Use CRS online help for assistance with installation and configuration.
Note
Enter the CRS application server IP address in the IP Address field, and click Next. The Installing TAPS on Apps Server window displays a progress bar that shows the status of the installation. Click Next to begin the installation. The Installation Completed window displays when the installation ends. Click Finish.
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TAPS Activation
This subtopic describes how to activate TAPS after it is installed.
TAPS Activation
1. Access Cisco Unified CallManager Serviceability tool 2. Choose Tools > Service Activation 3. Activate Cisco TAPS service
CIPT1 v5.03-31
You can activate and deactivate TAPS service by using the Cisco Unified CallManager Serviceability tool. Follow these steps to activate the service:
Access the Cisco Unified CallManager Serviceability tool. Choose Tools > Service Activation. The Service Activation window displays. Choose the appropriate server from the drop-down list box. Click Next. Choose Cisco TAPS Service from Database and Admin Services of the Cisco Unified CallManager Services list and click Save.
If the service is already activated, the Activation Status will display as Activated.
Note Step 5
The service is activated, and the Activation Status column displays the status as Activated.
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Configuring Autoregistration Options for TAPS

This subtopic describes how to configure autoregistration options for TAPS.
Configuring Autoregistration Options for TAPS

Use TAPS to update phones that autoregister:
Phones that were added with BAT and have a dummy MAC address Existing phones in Cisco Unified CallManager Administration
Choose System > Service Parameters to select the option that fits your requirements.
CIPT1 v5.03-32
The Configure TAPS option provides two ways to use TAPS to update phones that autoregister with the Cisco Unified CallManager database: Update phones that are added by using Cisco Unified CallManager BAT and have a dummy MAC address. Update existing phones in Cisco Unified CallManager Administration. The default setting limits the use of TAPS to phones that have a dummy MAC address with a device name that starts with the prefix BAT. You can set the Configure TAPS option to allow any phone to autoregister in the Cisco Unified CallManager system, including phones that have a standard MAC address.
Note Using TAPS leads to the depletion of the autoregistration range on the Cisco Unified CallManager. After updating through TAPS, the autoregistered directory number of the phone becomes an unassigned directory number. You should delete unassigned directory numbers using BAT periodically to free up the autoregistration range.
Follow these steps to set the Configure TAPS option:

In the Cisco Unified CallManager Administration window, choose System > Service Parameters. The Service Parameter Configuration window displays. From the Server drop-down list, choose the appropriate server. From the Service drop-down list, choose Cisco TAPS Service. Choose one of these two options from the Parameter Value drop-down list box:
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Allow Auto-Registered phones to reset with a profile with a dummy MAC address. TAPS updates autoregistered phones with a profile that has the dummy MAC address only. Allow Auto-Registered phones to reset with any profile. TAPS updates autoregistered phones with any profile.
Step 5 Step 6
Click Save. A status message indicates that the update is successful. To return to the TAPS Options window, click Back.
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Summary
Summary
Cisco Unified CallManager BAT automates the administration of a large number of similar phones, users, or ports. Cisco Unified CallManager BAT uses a template and CSV file to insert bulk transactions into the CallManager database. Cisco Unified CallManager BAT must be installed on the same server as the first node database for Cisco Unified CallManager. CSV files can be edited using the Cisco Unified CallManager BAT spreadsheet or text editor.
CIPT1 v5.03-33
Summary (Cont.)
Cisco Unified CallManager BAT validates the CSV file and template. Inserting Cisco IP phones, users, or devices is the last step in Cisco Unified CallManager BAT. To update phones, find them, configure values, and reset the phones. Bulk transactions can be scheduled as jobs at any specific time Cisco Unified CallManager BAT jobs queue up on the Bulk Provisioning Service. TAPS, an optional application that Cisco Unified CallManager BAT provides, retrieves the predefined configuration for autoregistered IP phones. TAPS must be installed manually on CRS application server.
CIPT1 v5.03-34
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References
For additional information, refer to these resources: Cisco Systems, Inc. Cisco Unified CallManager Bulk Administration Guide 5.0(1) http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_b ook09186a0080530796.html Cisco Systems, Inc. Cisco Unified CallManager Administration Guide, Release 5.0(2), Bulk Administration Tool http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_c hapter09186a00806333b3.html
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Module Summary
Module Summary
Cisco Catalyst switches provide three important functions in an IP telephony deployment: voice VLANs, CoS, and PoE. Cisco IP telephony endpoints support the G.711 and G.729 codecs and include the full range of desktop, PC-based, and conference-room IP phones. To register to a Cisco Unified CallManager, an IP phone obtains a VLAN ID and IP address and downloads configuration information from the TFTP server. Configuring Cisco Unified CallManager to automatically or manually add IP endpoints entails changing the Cisco Unified CallManager name to its IP address, configuring device pools, and assigning phone button templates.
CIPT1 v5.03-1
Module Summary (Cont.)

Cisco Unified CallManager Release 5.0 supports SIP phones using standard compliant SIP line side. Cisco SIP phones support either basic or advanced features and are fully managed by TFTP. Third-party SIP phones are also supported with very basic features. Cisco Unified CallManager BAT enables you to add large numbers of devices, users, and ports to the Cisco Unified CallManager database. Cisco Unified CallManager BAT can be used anytime but is typically launched during deployment or when making wholesale changes to phone extensions. Day-to-day changes to individual phones are normally done in Cisco Unified CallManager Administration.
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This module describes the LAN infrastructure that provides IP phones with electrical power and VLANs, and IP endpoints used with Cisco Unified CallManager Release 5.0. The module lists IP phones, describes their features and explains how to configure Cisco Unified CallManager to support them. Finally, the module describes how to automate changes of the Cisco Unified CallManager database with Cisco Unified CallManager BAT.
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Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and solutions are found in the Module Self-Check Answer Key. Q1) Which command enables inline power on port 3 of module 2 of a Cisco Catalyst 6000 Series switch that is running Cisco Catalyst software? (Source: Configuring Cisco Catalyst Switches) A) B) C) D) E) Q2) power inline in global configuration mode power inline auto in interface configuration mode set port inline power default set port inline power 3/2 802.3 set port inline power 2/3 auto
Cisco Catalyst switches can provide which three functions in an IP telephony deployment? (Choose three.) (Source: Configuring Cisco Catalyst Switches) A) B) C) D) E) convert SCCP signaling packets from the IP phone to MGCP power IP phones through the same Ethernet cable that carries data enable the classification and prioritization of voice packets at Layer 2 instruct the IP phone to change the CoS of an incoming data frame determine available bandwidth before placing a call over the WAN
Q3)
What are the two requirements for configuring dual VLANs on a single port of a Catalyst switch that is attached to an IP phone? (Choose two.) (Source: Configuring Cisco Catalyst Switches) A) B) C) D) E) Configure the interface as a trunk. Tag the voice packets with an identifier. Extend the CoS boundary to the IP phone. Ensure that the IP phone has an internal switch. Ensure that the IP phone is inline power-capable.
Q4)
What must you do if you do not want to trust the device that is attached to the IP phone? (Source: Configuring Cisco Catalyst Switches) A) B) C) D) Change the CoS value on the voice VLAN to a number lower than 5. Use the switchport priority extend untrust interface command. Do nothing; the IP phone will automatically change the CoS to 0 for the attached devices. Create a dual VLAN on the access port and place the data packets in the native VLAN.
Q5)
Which two protocols does Cisco Unified CallManager use to support Cisco IP phones? (Choose two.) (Source: Evaluating Cisco IP Telephony Endpoints) A) B) C) D) MGCP SIP H.323 SCCP
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Q6)
Which three of these statements correctly describe SIP feature levels supported by Cisco Unified CallManager? (Choose three.) (Source: Evaluating Cisco IP Telephony Endpoints) A) B) C) D) E) Very basic SIP phones support the minimum features for SIP compliance outlined in RFC 3216, the core SIP standard. This category also includes thirdparty SIP phones. Basic SIP phones support only the minimum features for SIP compliance outlined in RFC 3216, the core SIP standard. This category includes the older family of Cisco SIP phones and third-party SIP phones. Basic SIP phones support the core SIP standards plus several additional features. This category includes the older family of Cisco SIP phones. All Cisco phones are advanced SIP phones, while third-party SIP phones are basic SIP phones. Advanced SIP phones support all the features of the Basic SIP phones, but use SIP extensions to try to achieve feature parity with SCCP phones.
Q7)
Which of these process choices describes the Cisco IP phone (SCCP-controlled) startup process. (Source: Evaluating Cisco IP Telephony Endpoints) A) Process 1: Obtain power from the switch Load the stored phone image Obtain an IP address Configure the VLAN Request the configuration file and the profile file Register on Cisco Unified CallManager. Process 2: Load the stored phone image Obtain power from the switch Configure the VLAN Obtain an IP address Request the configuration file and the profile file Register on Cisco Unified CallManager. Process 3: Obtain power from the switch Load the stored phone image Configure the VLAN Obtain an IP address Request the configuration file and the profile file Register on Cisco Unified CallManager. Process 4: Obtain Power from the switch Load the stored phone image Request the configuration file and the profile file Configure the VLAN Obtain an IP address Register on Cisco Unified CallManager.
B)
C)
D)
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Q8)
Which five settings are required to configure a device pool? (Choose five.) (Source: Configuring Cisco Unified CallManager to Support IP Phones) A) B) C) D) E) F) G) H) I) Media Resource Group List Cisco Unified CallManager group date/time group region softkey template device pool name phone template phone load device defaults
Q9)
Which of these is a valid phone button template configuration for a Cisco IP Phone 7960? (Source: Configuring Cisco Unified CallManager to Support IP Phones) A) B) C) D) 0 lines, 6 speed dials 2 lines, 4 speed dials 6 lines, 1 speed dial 4 lines, 4 speed dials
Q10)
Which navigation path would you use to configure Cisco Unified CallManager to automatically register an IP phone? (Source: Configuring Cisco Unified CallManager to Support IP Phones) A) B) C) D) E) System > Server System > Cisco Unified CallManager Service > Service Parameters Device > Phone System > Device Pool
Q11)
How does Cisco Unified CallManager tie configuration information to the IP phones in the Informix database? (Source: Configuring Cisco Unified CallManager to Support IP Phones) A) B) C) D) E) IP address unique GUID host name MAC address device pool ID
Q12)
Which five of these Cisco IP phone models are supported by Cisco Unified CallManager with the SIP protocol? (Choose five.) (Source: Configuring SIP Endpoints) A) B) C) D) E) F) G) H) Cisco Unified IP Phone 7970/71 Cisco IP Communicator Cisco Unified IP Phone 7960/61 Cisco Unified IP Phone 7940/41 Cisco Unified IP Video Phone 7985G Cisco Unified IP Phone 7911 Cisco Unified IP Phone 7902G Cisco Unified IP Phone 7905/12
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Q13)
Which two mechanisms do Cisco SIP phones and Cisco Unified CallManager use for digit collection? (Choose two.) (Source: Configuring SIP Endpoints) A) B) C) D) DTMF for en-bloc collection (single SIP NOTIFY message transfers all digits together). DTMF for collection one digit at a time. KPML for faster collection of one digit at a time for faster call routing. KPML for either collecting one digit at a time or by using SIP dial rules that tell SIP phone how many digits to collect before considering it a valid pattern.
Q14)
How does a third-party SIP phone identify itself to Cisco Unified CallManager? (Source: Configuring SIP Endpoints) A) B) C) D) uses directory number sent in REGISTER message to Cisco Unified CallManager uses MAC address sent in REGISTER message to Cisco Unified CallManager uses digest authentication when it sends the REGISTER message to Cisco Unified CallManager with the username in the message uses the Authentication Name when it sends REGISTER message to Cisco Unified CallManager
Q15)
SIP profiles provide which specific SIP information for the phone? (Source: Configuring SIP Endpoints) A) B) C) D) range of allowed directory numbers and allowed SIP URI format default telephony event payload type, registration and keepalive timers, media ports, and dynamic DNS server addresses local dialing rules, allowed range of directory numbers, and button templates registration details and timers, and they are mechanisms used to drive communication between third-party SIP phones and Cisco Unified CallManager
Q16)
BAT enables you to perform which tasks? (Source: Managing Endpoints with Cisco Unified CallManager BAT) A) B) C) D) E) reload configuration for replacement phones add, update, and delete IP phones, users, and devices collect user data such as passwords and account identification update MAC addresses and download configurations for new phones save a detailed log of Cisco Unified CallManager events for troubleshooting
Q17)
On which server must you install BAT? (Source: Managing Endpoints with Cisco Unified CallManager BAT) A) B) C) D) first node subsequent node primary server secondary server
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Q18)
Before you can create a BAT template, you must make sure which four of these settings have already been configured in Cisco Unified CallManager Administration? (Choose four.) (Source: Managing Endpoints with Cisco Unified CallManager BAT) A) B) C) D) E) F) device pool directory number softkey template MAC address calling search space button template
Q19)
Which two items must be installed for TAPS to function properly? (Choose two.) (Source: Managing Endpoints with Cisco Unified CallManager BAT) A) B) C) D) CRS Cisco Unified CallManager BAT plug-ins Extension Mobility
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Q1) Q2) Q3) Q4) Q5) Q6) Q7) Q8) Q9) Q10) Q11) Q12) Q13) Q14) Q15) Q16) Q17) Q18) Q19) E B, C, D B, D C B, D A, C, E C B, C, D, E, F B B D A, C, D, F, H B, D C B B A A, C, E, F A, B
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CIPT1

Implementing Cisco Unified CallManager Part 1

Students, this letter describes important course evaluation access information!

Get Started with Cisco Unified CallManager

Introducing Cisco Unified CallManager Release 5.0

Evaluating Cisco Unified CallManager Release 5.0 Deployment Options

1-50 1-51 1-52

Installing Cisco Unified CallManager Release 5.0

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

2006 Cisco Systems, Inc.

Administration of Cisco Unified CallManager Release 5.0

Administering the Cisco IP Telephony Platform

Performing General Administration

2-55 2-56 2-57 2-59 2-62

Configuring Cisco Unified CallManager Release 5.0 Basic Settings

Implementing Disaster Recovery

Deployment of Cisco Unified CallManager Release 5.0 Endpoints

Configuring Cisco Catalyst Switches

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

2006 Cisco Systems, Inc.

Evaluating Cisco IP Telephony Endpoints

Configuring Cisco Unified CallManager to Support IP Phones

2006 Cisco Systems, Inc.

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

Configuring SIP Endpoints

Managing Endpoints with Cisco Unified CallManager BAT

2006 Cisco Systems, Inc.

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

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

2006 Cisco Systems, Inc.

Learner Skills and Knowledge

Learner Skills and Knowledge

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Learner Skills and Knowledge (Cont.)

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

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Course Goal and Objectives

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2006 Cisco Systems, Inc.

A Introduction to M Cisco Unified

Cisco Unified CallManager Features and Services

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

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Cisco Icons and Symbols

IP Phone Switch Router

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Cisco Glossary of Terms

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Your Training Curriculum

Cisco Career Certifications

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

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Cisco Career Certifications: CCVP

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

2006 Cisco Systems, Inc.

Get Started with Cisco Unified CallManager

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

2006 Cisco Systems, Inc.

Introducing Cisco Unified CallManager Release 5.0

Cisco Unified Communications Solution