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Documente Profesional
Documente Cultură
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Module Objectives
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Objectives
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Cisco Unified Communications Manager Overview Cisco Unified Communications Manager Signaling and Media Paths Cisco Unified Communications Manager Hardware, Software, and Clustering Cisco Unified Communications Manager Cluster Cisco Unified Communications Manager Hardware Requirements
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Cisco Unified Communications Operating System Cisco Unified Communications Operating SystemAccess Cisco Unified Communications Manager Database
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1-19 1-21 1-22 ''^
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Cisco Unified Communications Manager Licensing Model Overview CiscoUnified Communications Manager Licensing
License File Specifics License File Example License File Request Process (FlexLM) Obtaining Additional Licenses Licensing Functional Diagram Calculating License Units Generating License Unit Report
Uploading License File Summary
References 1-38 Understanding Cisco Unified Communications Manager Deployment and Redundancy
Options
Objectives
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1-41 1-42 1-45 1-49 1-52 1-56 1-57 1-59 1-61 1_62
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Cisco Unified Communications ManagerDeployment Options Overview Cisco Unified CommunicationsManager Single-Site Deployment Cisco Unified Communications ManagerMultisite Deployment with Centralized Call Processing Cisco Unified Communications Manager Multisite Deploymentwith Distributed Call Processing Cisco Unified Communications ManagerMultisite Deployment with Clustering over the WAN Cisco Unified Communications ManagerDeployment on Virtualized Servers Cisco Unified CommunicationsManager Call-Processing Redundancy 1:1 Redundancy Design 2:1 Redundancy Design Summary
References
Module Summary
References Module Self-Check
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Module Objectives
Cisco Unified Communications Manager Initial Configuration Overview Cisco Unified Communications Manager Network Configuration Options Overview
Network Components
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Enterprise Phone Configuration Changing Enterprise Phone Configuration Cisco Unified Communications Manager Service Parameters
Example of Service Parameters Changing Service Parameters Service Parameter Configuration Screenshot Cisco CallManager Service Parameters Screenshot Summary
References
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Managing User Accounts in Cisco Unified Communications Manager Objectives Cisco Unified Communications Manager User Accounts Overview Types of User Accounts in Cisco Unified CommunicationsManager
Data Associated with User Accounts
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User Privileges User Privilege Component Interaction Roles and User Groups Example User Management Options Lightweight Directory Access Protocol Cisco Unified Communications Manager End-User Data Location Managing User Accounts Using the Administration GUI Application User Configuration Page
Implementing Cisco Unified Communications Manager. Part 1 (CIPTt) v8.0
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Role Configuration Page User Groups User Group Configuration Page: User Assignment
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Cisco Unified Communications Manager BAT Components Bulk Provisioning Service Managing User Accounts Using Cisco Unified Communications Manager BAT Step 1: Configuring a Cisco Unified Communications Manager BAT User Template Step 2: Creating the CSV Data Input File Step 3: Uploading the CSV Data Input File Step 4: Starting Cisco Unified CommunicationsManager BAT Job to Add Users Step 5a: Job StatusList of Jobs Step 5b: Verifying Job StatusJob Details
LDAP Overview
LDAP Directory Integration with Cisco Unified Communications Manager LDAP Support in Cisco Unified Communications Manager
LDAP Integration. Synchronization
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3_2i 3-22 3_23 3-24 3-26 3_27 3_27
Implementing IP Phones
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Objectives
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3-33 3-35 3-37 3-39 3-40 3-43 3-44 3-45
Date/Time Group
Device Pool
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3-60 Step 1:Assigning the Default Autoregistration Protocol 3-61 Step 2: Cisco Unified Communications Manager Group Configuration 3-62 Step 3: Cisco Unified Communications Manager Configuration 3-63 Cisco Unified Communications ManagerBAT and Auto-Register Phone Too! 3-64 Cisco Unified CommunicationsManager Auto-Register Phone Tool 3-65 Cisco Unified Communications Manager Auto-Register Phone Tool Requirements Process ofAdding IP Phones Using the Cisco Unified Communications Manager Auto -Register
Using Cisco Unified Communications Manager BAT for Adding Phones toCisco Unified
Communications Manager
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Phone Tool
3-66
Step 1: Verify Bulk Provisioning Services Have been Activated Step 2: Configuring Cisco Unified Communications Manager Phone Template
Step 3: Creating the CSV Data InputFile Step 4: Uploading CSV Files Step 5: Validating Phones Configuration
Step 7: Verify Phone Insertion
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Step 1: Third-Party SIP Phone Configuration Steps 2 and 3: Third-Party SIP Phone Configuration Step 4: Third-Party SIP Phone Configuration
Directory Number Considerations Directory Number Line Appearance
Summary
References
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3-95 3-97 3-97
Module Summary
References
Module Self-Check
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2010Cisco Systems,
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MGCP Support in Cisco Unified CommunicationsManager Cisco Unified CommunicationsManager Configuration Server PRI Backhaul
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4-30 4-32 4-33 4_39 4_40 4_41
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Step 1:Configure Basic Cisco IOS H.323 Functionality Step 2: Configure Cisco IOS Call-Routing Information Step 3: Configure Cisco Unified Communications Manager Redundancy on H.323 Gateways H.323 GatewayCall Survivability
H.323 Gateway Call-Survivability Configuration
4.47 4-48
4.49 4.51 4-52 4.53 4-54 4,55 4-56 4.58 4-59 4-60 4_g1 4_g3 4_64 4_oc 4_gg
SIP Gateway Implementation Cisco Unified Communications Manager SIP Gateway Configuration Step 1: Add a SIP Trunk Step 2: Configure SIP Trunk Parameters Step 1: Configure Basic Cisco IOS SIP Functionality Step 2: Configure Cisco IOS Call Routing on SIP Gateways Step 3: Configure Cisco IOS SIP UserAgent Parameters SIP Considerations: DTMF Signaling SIP Considerations: MTP Allocation SIP Considerations: MTP Allocation Configuration Summary References
4-67
Dial Plan Components Dial Plan Components and Dial Plan Components and Dial Plan Components and Dial Plan Components and Dial Plan Components and Endpoint Addressing Endpoint Dialing Endpoint Dialing Example
Objectives
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Functions: Functions: Functions: Functions: Functions: Endpoint Addressing Call Routing and Path Selection Digit Manipulation Calling Privileges Call Coverage
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Cisco Unified Communications Manager Call-Routing Logic Cisco Unified Communications Manager Digit Analysis
Call-Routing Table Example Call-Routing Table Entries (Call-Routing Targets) Sources ofCall-Routing Requests (Entities Requiring Call-Routing Table Lookup) Route Pattern: Commonly Used Wildcards Route Pattern Examples
4"^
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Digit-Signaling Methods
User Input on SIP Phones User Input on Type-A SIP Phones: No SIP Dial Rules Configured on the Phone User Input on Type-A SIP Phones: SIP Dial Rules Configured on the Phone User Input on Type-B SIP Phones: No SIP Dial Rules Configured on the Phone User Input on Type-B SIP Phones: SIP Dial Rules Configured on the Phone
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Special Call-Routing Features Route Filters The! Wildcard Urgent Priority Blocked Patterns Call Classification Secondary Dial Tone Cisco Unified Communications Manager Path Selection
Path-Selection Example
Cisco Unified Communications Manager Path-Selection Configuration
Route-Group Functionality
Local Route Groups
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jl\
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Local Route-Group Functionality Route-Group Configuration Local Route-Group Configuration Route-List Configuration Route-Pattern Configuration
Summary
References
~l ^o
Using Partitions and CSSs to Implement Calling Privileges for On-Net Calls
Objectives Calling Privileges Overview
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Calling-Privileges Requirement Example Calling-Privileges Configuration Elements Partitions and CSSs Partition <None> and CSS <None> Analogy: Locks and Key Rings Basic Partitions and CSS Example CSS Partition-Order Relevance Partitions and CSS Example with Multiple Best Matches Phones That Have a Device CSS and Line CSS Example with IP Phone Line CSS and Device CSS CoS Sample Scenario Configuring Partitions and CSSs Creating Partitions Assigning Partitions Creating a CSS Assigning a CSS to an IP Phone
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Example of Partitions and CSSs Sample CoS Solution Partition and CSS Considerations Partition and CSS Considerations ImportantRules and Guidelines Summary References
CIPT1
Course Introduction
Overview
Implementing Cisco UnifiedCommunications Manager, Part I (ClI'Tl) v8.0 prepares you for implementing a Cisco Unified Communications Manager solution at a single-site environment. This course focuses primarily on Cisco Unified Communications Manager Version 8.0. which is the catl-routing and signaling component for the Cisco Unified Communications solution. You will perform postinstallation tasks, configure Cisco Unified Communications Manager, implement Media Gateway Control Protocol (MGCP) and H.323 gateways, and build dial plans to place on-net and off-net phone calls. You will also implement media resources, IP Phone Sen ices. Cisco Unified Communications Manager native presence, and Cisco Unified Mobility
Ability to configure Cisco IOS gateways with traditional and VoIP call legs
introducing Cisco Voice and Unified Communications Administration (ICOMM) v8,0 Implementing Cisco Voice Communications and QoS
(CVOICE)v8.0
"To provide learners with the necessary knowledge and skills to implement a single-site Cisco Unified
Communications solution that is based on Cisco
Upon completing this course, you will be able to meet these objectives: Describe Cisco Unified Communications Manager, including its functions, architecture, deployment and redundancy options, and how to install or upgrade it Perform Cisco Unified Communications Manager initial configuration and user
management
Configure Cisco Unified Communications Manager to support on-cluster calling Implement PSTN access in Cisco Unified Communications Manager and build a dial plan in a single-site Cisco Unified Communications Manager deployment
Implement Cisco Unified Communications Manager media resources
Course Inlroduclion
Course Flow
This topic presents the suggested flow of the course materials.
Course Flow
Media Resources
introductionto
CiscoUnified
Communications
Calling {Cont.)
Single-Site Off-Net
[ Calling (Cent.)
Feature and
Manager
Application
Implementation
Calling
Lund
Administering Cisco
Unified
Communications
Suigie-Site Oft-Net
Calling (Cont.)
Single-Site Off-Met
Calling (Cont. J
Media Resources
Application
Implementation
(Cont.)
The schedule reflects the recommended structure for this course. This structure allows enough
time for the instructor to presentthe course information and for you to work through the lab aeti\ities. Ihe exact timing of the subjectmaterials 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.
Manager
Gatekeeper
Cisco Unity
Connection
Cisco Unified
Border Element
Gateway
Cisco Untied
Messaging Gateway
Communicator
Communications
Manager Express
Cisco Unified Communications
Course Introduction
You arc encouraged tojoin the Cisco Certi fication Communit). a discussion forum open to
am one holding a\alid Cisco Career Ccrtificalion (such as Cisco CCII-\ CCNA\ CCDA".
CCNP". CCDP\ CCIP\ CCVP'". orCC'SP'"). It provides a gathering place forCisco certified
prolessionals to share questions, suggestions, and infonnation about Cisco Career Ccrlilication programs and other certification-related topics. For more information, visit
hup: \\\\n.cUco.com'go certifications.
Expert
Professional
impksmantmgCisco unm<i communlcaions Manager, Part 1 ImfDeimntns Cisco Unfiad canvmintcatom M&tagar.PmtZ
Associate
CemrmiiicaSaJs integmting Cisco unified Commailcetlafis Apfiteatkms www cisco com/go/cert ifications
Voice Networking
Course Introduction
Module 1
Communications Manager
Overview
Cisco Unified Communications Manager is the software-based, call-processing component of
the Cisco Unified Communications solution.
This module describes the characteristics ofCisco Unified Communications Manager and explores the available deployment models for using Cisco Unified Communications Manager in
a Cisco Unified Communications solution.
Module Objectives
Upon completing this module, you will beable to describe Cisco Unified Communications
Manager, including its functions, architecture, deployment and redundancy options, and how to
install or upgrade. This ability includes being able tomeet these objectives:
Lesson 1
Architecture
Overview
A Cisco Unified Communications deployment relies on Cisco Unified Communications
Manager for its call-processing andcall-routing functions. Understanding the rolethat Cisco Unified Communications Manager plays in a converged network from a system, software, and hardware perspective is necessary to successfully install and configure Cisco Unified
Communications Manager. This lesson introduces the Cisco Unified Communications solution and describes the role,
architecture, characteristics, hardware and software requirements, and the licensing model of
Cisco Unified Communications Manager.
Objectives
Upon completing this lesson, you will understand Cisco Unified Communications Manager
architecture. This ability includes being able to meet these objectives:
Describe the components of a Cisco Unified Communications solution and the functionality of each component
Describe the architecture and role ofCisco Unified Communications Manager Describe the hardware requirements for Cisco Unified Communications Manager Describe the characteristics ofthe Cisco Unified Communications Operating System Describe the characteristics of the Cisco Unified Communications Manager database and
how it provides redundancy
Describe the licensing model ofCisco Unified Communications Manager and describe how to calculate, verify, and add license units toCisco Unified Communications Manager
- Rich-media conferencing
Third-party applications
The Cisco I nitied Communications system fully integrates communications by enabling data. \ oice. and video to be transmitted overa single network infrastructure using standards-based
IP. Leveraging the framework that is provided by Cisco IP hardware and software products, the
Cisco Unified Communications system has the capability to address current and emerging communications needs in the enterprise environment. The Cisco Unified Communications
applications. The Cisco Unified Communications system provides and maintains ahigh level ot
availability, quality of service (QoS). and security for the network. The Cisco Unified Communications system incorporates and integrates the following
communications technologies:
IP telephony: IP telephony refers to technology that transmits voice communications over anetwork using IP standards. Cisco Unified Communications includes hardware and software products, such as call-processing agents. IP phones (both wired and wireless},
voice-messaging s\stems. \ ideo devices, and many special applications. Customer Contact Center: Cisco Unified Contact Center products area combined
strategv with architecture to enable efficient and effective customer communications across agloballv capable network. This strategy allows organizations to draw from abroader
range of resources to sen ice customers. They include access to alarge pool ofagents and multiple channels ofcommunication, as well as customer self-help tools.
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Video telephony: Cisco Unified Video Advantage products enable real-time video
communications andcollaboration using the same IP network andcall-processing agent as Cisco Unified Communications. With Cisco UnifiedVideo Advantage, making a video call
is just as easy as dialing a phone number. Dedicated video hardware products such as
video-enabled desktop phones are also available.
integrated set of IP-based tools for voice, video, and web conferencing.
Third-party applications: Cisco works with leading-edge companies to provide a broad selection of third-party IPcommunications applications and products. This collaboration helps businesses focus on critical needs such as messaging, customer care, and workforce
optimization.
1-5
Call processing
Signaling and device control Dial plan administration
Phone feature administration
Directory services Programming interface to external applications Includes a backup-andrestore tool (disaster
recovery system)
Cisco Unified Communications Manager extends enterprise telephony features and functions to packet telephony network devices. These packet telephony network devices include Cisco IP phones, media-processing de\ ices. VolP gateways, and multimedia applications. Additional data, voice, and \ ideo sen icessuch as converged messaging, multimedia conferencing, collaborative Contact Centers, and interactive multimedia response systemsinteract with the IP telephony solution through the Cisco Unified Communications Manager application programming interface (API).
Cisco Unified Communications Manager 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 de\ice control: Cisco Unified Communications Manager sets up all 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 Communications Manager uses to determine call routing. Cisco Unified Communications Manager provides the ability to create scalable dial plans for the users. Phone feature administration: Cisco Unified Communications Manager extends services such as hold, transfer, forward, conference, speed dial, last-number redial. Call Park, and other features to IP phones and gateways.
Directory services: Cisco Unified Communications Manager 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. Cisco Unified Communications Manager allows synchronization from the following
directories to the database:
OpenLDAP 2.4
Programming interface to external applications: Cisco Unified Communications Manager provides a programming interface to external applications such as Cisco IP Communicator. Cisco Unified IP Interactive Voice Response (IVR), Cisco Personal Assistant, and Cisco Unified Communications Manager Attendant Console. Backup and restore tools: Cisco Unified Communications Manager provides the Disaster Recovery System (DRS) tools to provide a means of backing up and restoring the Cisco Unified Communications Manager configuration database, as well as the Call Detail Records (CDR) and the Cisco Unified Communications Manager CDR Analysis and Reporting (CAR) database.
Signaling Protocol
iSCCP/SIPI
Signaling Protocol
SCCP/SIP)
Cisco Unified Communications Manager uses the Session Initiation Protocol (SIP) or the
Skinny Client Control Protocol (SCCP) to communicate with Cisco IP phones for call setup
and maintenance tasks.
Whenthe call is set up. mediaexchange occursdirectly between the Cisco IP phones using Real-TimeTransport Protocol (RTP) to carry the audio.
Example: Basic IP Telephony Call In the figure. User A on IP Phone A (left telephone) wants to make a call to IP Phone B (right telephone). User A picks up the handset and dials the number of User li. Inthis environment, dialed digits aresent to Cisco Unified Communications Manager, the call-processing engine. Cisco Unified Communications Manager finds the address and determines where to route the
call.
Using SCCP or SIP. Cisco L'nified Communications Manager signals the calling party over IP to initiate a ringback. and Party Ahears the ringback tone. Cisco Unified Communications Manager also signals the call to the destination phone, which starts ringing. When User B accepts the call, the RIP media pathopens between the two stations. User A or
User B may now initiate a conversation.
The Cisco li* phones require no further communication with Cisco Unified Communications Manager until either User A or User B invokes a feature, such as Call Transfer, call
conferencing, or call termination.
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- No customer access to operating system Only GUI and CLIaccess to appliance system Third-party access via documented APIs only
Supports clustersfor redundancyand load sharhg: Provides database redundancy by sharing a common database Provides ca-processing redundancy by Cisco Unified Communications
Manager groups
including TFTP, mediaresources, conferencing, and call processing - Maximum ofeight nodescan be used forcall processing (tunning trie Cisco Unified Communications Manager service)
Cisco Unified Communications Manager is a complete hardware and software solution that works asan appliance. The appliance isessentially a closed system that supports only applications and utilities that are authorized by Cisco. Key goals ofthe appliance model are to
simplify the installation and upgrade ofthe system and to hide the underlying operating system and its tools. An appliance-based model makes itpossible for an administrator toinstall. ' implement, and manage a Cisco Unified Communications Manager server without requiring
knowledge or having access tothe underlying operating system. The Cisco Unified Communications Manager appliance has these features:
Complete hardware and software solution:
Cisco Unified Communications Manager servers arepreinstalled with all software that is required to operate, maintain, secure, and manage a server or cluster of
servers(including Cisco Security Agent).
Can also be field-installed on supported Cisco Media Convergence Server (MCS) platforms or third-party server platforms that arc approved by Cisco.
Appliance operating system improves installation and upgrade and increases security and
reliability.
You can upgrade Cisco Unified Communications Manager servers while they
continue to process calls.
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Interfaces provide access to the system via either command-line interface (CUI)and GUI for administration purposes or through documented APIs for third-parly access. Outputs \ arious management parameters via a published interface to provide infonnation to approved management applications such as. butnotlimited to. NetlQ
Vi\inet Manager. HP OpenVicw. and Integrated Research PROGNOSIS.
Operates in a headless manner (without keyboard, mouse, or Video (iraphies Array [VGA]
monitor support) or. in the case of some of the hardware platforms, in a headed manner
(with keyboard, mouse, and monitor).
fhe Cisco Unified Communications Manager appliance supports clusters for redundancy and
load sharing. Database redundancy isprovided by sharing a common database, whereas callprocessing redundancy is pro\ ided by Cisco Unified Communications Manager groups: Acluster consists ofone publisher and a total maximum of20 servers (nodes) running
\ arious ser\ices, including TFTP. media resources, conferencing, and call processing. You can have up to a maximum of eight nodes torcall processing (running the Cisco
CallM,ina<ter service).
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Manager Cluster
5W Conferencing
IDS Datafca
*_*eMifia .
The Cisco Unified Communications Manager service provides call routing, signaling, and
media control for an IP telephony enterprise deployment.
A cluster is a setof networked services that work together to provide the Cisco Unified
Communications Manager service in addition todedicated servers providing database, application. TFIP.and media services such asconferencing and music onhold (MOH). These services can be provided by the subscribers and the publisher and can be shared by all servers.
Clustering provides several benefits. It allows the network to scale to several thousands of
endpoints. provides redundancy incase ofnetwork orserver failures, and provides a central
point of administration.
configuration settings for all devices. These settings arc stored in adatabase using IBM
Informix Dynamic Server (IDS). The database isthe repository for information such asservice
parameters, features, device configurations, and thedial plan.
Cisco Unified Communications ManagerClustering The database replicates nearly all information in astar topology (one publisher, many
subscribers). However. Cisco Unified Communications Manager nodes also use a second
communication method to replicate run-time data in a mesh topology (every node updates every other node). This type ofcommunication is used for dynamic information that changes
more frequently than database changes. The primary use ofthis replication is tocommunicate newly registered phones, gateways, and digital signal processor (DSP) resources, sothat optimum routing ofcalls between members ofthe cluster and the associated gateways occurs.
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The following are the minimum hardware requirements for Cisco Unified Communications
Manager:
2-GH/ processor
2-GB RAM
Minimum requirements remain the same as for Cisco Unified CallManager Version 5.0. but
onlv specific Cisco MCS models areapproved. Note Cisco Unified Communications Manager server support matrix and hardware specifications
can be found at the following URL
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Because voice networks should maintain a high uptime, Cisco Unified Communications Manager most be installed on a server that meetsCisco configuration standards. Forthis
reason. Cisco has collaborated with two server hardware manufacturers. Hewlett-Packard and
IBM. which designed these server hardware platforms specifically for Ciscovoice applications.
The following URI.s provide a listof the IBM and HP hardware ptatfonns thatareapproved by
Cisco:
Imp:-'www.cisco.eom'en/US/prod/colSatcral/voicesw/ps6790/ps574S/ps37R/product_soluti
on_ov erv iew()9186a0(>80107d79.html
InCisco Unified Communications Manager Versions 7.1(3) and 8.0, Cisco is officially supporting VMware installations on VMware ESXi 4.0 and allowing complete licensing. Cisco Unified Communications Manager can alsobe installed on anyother VMware
platform but will not be supported for production use.
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Appliance operating system (based on Red Hat Linux) Operating system updates provided by Cisco (along with application updates) Unnecessary accounts and services disabled IBM Informix Dynamic Server as the database
DHCP server
Cisco Security Agent The Cisco Unified Communications operating system is also used for these other Cisco Unified Communications applications:
'['he Cisco Unified Communications Operating System is an appliance operating system that is based on Red Hat Linux. Cisco provides operating system updates (along with application
updates) through patches that arc signed b\ Cisco. Unsupported software and applications (not
signed bv Cisco) cannot be uploaded or installed into the appliance.
Root access to the file sv stem is not permitted, andall unnecessary accounts and services have
been disabled in the appliance operating system.
IBM IDS is installed as the database for the Cisco Unified Communications applications.
Cisco Sccuritv Agent, a host intrusion-prevention system, isalso built into the appliance to provide protection against known and unknown attacks. A DHCP server is integrated into Cisco Unified Communications Manager toprovide DHCP
services to IP phones.
The Cisco Unified Communications Operating System is also used for these other Cisco
Unified Communications applications:
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The Cisco Unified Communications Operating System is a hardened operating system. The root and other common but unnecessary default accounts of the native operating system have been
disabled.
There is no possibility to access the native operating system directly or to install any unsupported applications or software. Access to the platform and upgrading of patches can only be done through the Cisco CLI and GUI.
There is also no access to native operating system debug interfaces; however, traces, alarms, and performance counters can be enabled and monitored through the Cisco CLI and GUI. There is no direct access to the file system; only some files and directories are accessible through the Cisco CLI and GUI for maintenance purposes.
To require support from Cisco, activate remote account support for a specific time for remote Cisco Technical Assistance Center (TAC) access.
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Pnvacy. DND
Hunt group login status
The data in the Cisco Unified Communications Manager database is divided into two types.
Static configuration data is createdor modified as part of the configuration of the Cisco Unified Communications Manager cluster. Read/write accessto this dala is provided for the publisher only. Subscribers will provideonlv read-only accessto this data. If the publisher is not available, this data cannot he modified. Replicationof the data is from the publisher to the
subscribers.
I)\ namic user-facing features data is created or modified when certain user features are
modified b> the useror b\ an application feature. Read/write access to thisdata is provided on
all servers. This data can be modified even if the publisher is unavailable. User-facing features
data canbe replicated from the server where the change wasinitialed to all otherservers within
the Cisco Unified Communications Manager cluster.
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publisher failure:
^^^H r.nmnoneni
CCMAdmh
CCMUser
BAT
Function
' When
Always
Always
^^H
Provisions everything
Provisions user settings
Provisions everythkig
Always
Always Always
Sometimes
TAPS
AXL
AXIS-SOAP CCM
Inserts phones
Autoregistration only
Always (local) Always (local)
LDAP Sync
License Audit
Sen ices thatuse the publisher will be affected in the event of a publisher failure. These sen ices mainly provide configuration changes to the Cisco Unified Communications Manager cluster. Thereplication of thisdatawillalways be initiated from the publisher to the subscribers. The figure showsthe list of servicesthat rely on the publisher.
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User-Facing Features
User-facing features arc independent of the publisher, because their data can be written to
subscribers.
User-Facing Features
' Call Forward All (CFA)
Cisco Extension Mobility login * Hunt-group logout * Device Mobility * CTI CAPF status for end users and application users
These features do not rely on the availability of the publisher, because necessary data can be written to
subscribers.
fhe user-facing features that are listed in the figure do not rely on the availability ol'the
publisher; the dv namic user-facing features data can be written to the subscribers to which the device is registered, "fhe data is then replicated to all other servers within the cluster. Bv allowing the data to be written to the subscriber, the user-facing features can continue to function in the event of a publisher lailure. This functionality has been introduced with Cisco Unified Communications Manager Version 6.0. In all earlier versions, these user-lacing
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publisher to subscribers).
Only user-facing features
data are writable on a subscriber and
bidirectionally replicated
between all servers.
Subscriber
Subscriber
Iuser-facing
features
(user-feeing
features
writable)
writable)
Replication is now fully meshed between all servers within a cluster. However, only userfacing features datafor example. Cisco Extension Mobility featuresare writcable on a subscriber andare replicated from an updated subscriber to all otherservers. All nonuser-facing
features data can be written only to the publisher database and will get replicated from the publisher to all subscribers. Therefore, most data (all nonuser-facing features data) is still replicated in hub-and-spokc style (publisher to subscribers), while user-facing features data is replicated bidirectionally between
all servers.
Database access between members of a cluster is protected1 By IP access control (dynamic firewall "iptables")
By security password
Special configuration procedue required to enable database access for subscribers. At publisher, using Cisco Unified Communications Manager Adminislration, add
subscribers list of servers before installation of subscriber
Durrg subscriber installation, enter same database security password that was configured dunng instalation of publisher.
Publisher Subscnber
The first method is IP access control using "iptables" (dynamic firewall), and the second method is the use of a database security password.
The procedure to allow new subscribers to access the database on tlie publisher is as follows: Add the subscriber to the publisher database using Cisco Unified Communications Manager Administration.
During installation of the subscriber, enter the same database security password that was
entered during installation of the publisher.
After this configuration, the following process occurs to replicate the database from the publisher to the new Iv added subscriber: 1he subscriber attempts to establish a connection to the publisher database using the database management channel.
The publisher verities the subscriber authenticitv and adds the subscriber IP address to its
dynamic firewall (iptables).
The subscriber is allowed to access the publisher database.
Note
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Licensing Models
* Two basic licensing models are available - Device-based licensing
device-based licensing model isexplained in more detail on the following pages. For more details regarding the user-based licensing model, refer tothe following documentation on
Cisco.com: Cisco Unified Workspace Licensing, http:''www.ciseo.eom/en/US/partner/produets/ps9I56/index.htm!.
1-21
Licenses are required per cluster and provided by license tiles License file is bound to MAC address of publisher (running the licensing
service)
Manager.' including third-party SIP phones, and compare that number with the number of
license units that have been purchased. License enforcement occurs atthe time ofphone
provisioning and Cisco Unified Communications Manager service activation. The publisher is the only licensing server. The licensing server is the logical component that keeps track of the licenses that are purchased and the licenses that are used. If the publisher
fails, no new phones can register, and no configuration changes will be allowed; however,
existing phones still operate.
Cisco L'nified Communications Manager tracks the license compliance for dev ices,
applications, and software.
Device license units (DLUs)
Application licenses
Hie Cisco Unified Communications Manager software will be lied tothe MAC
address of the publisher.
IfCisco Unified Communications Manager is installed ona VMware LSXi server, the license isnot tied tothe MAC address of the publisher. Instead, a hash of various
Implementing Cisco Unified Communications Manager, Part 1(CIPTl) v8 0
2010 Cisco Systems. Inc
1-22
Application licenses are required for every call-processing serverthat is. servers that are running the Cisco CallManager service.
I hesc application licenses are also called Node Licenses.
Software licenses
Software license is tied to the major version of the software. Software licenses will be required for upgrade to Cisco Unified Communications
Manager Version 8.0.
Licenses are createdand distributed in accordance with the Cisco FlexLM process.
Note IfCisco Unified Communications Manager is installed within a VMware virtual machine, a demo license with 150 DLUsand three call-processing servers is automatically installed
1-23
Amount of DLUs depends on device type and device capabilities: Cisco phone or third-party
phone
Number of lines
:!!:"!"
:,.,
Video capabilities Number of units required per device can be viewed from
Cisco Unified Communications
, ,-..1
Manager Administration.
::';:':
Device License Units
Third-partv DLUs
fhe Cisco units are I'or Cisco dev icesonly. The third-party units canbe converted to Cisco
units but not \ ice versa.
Cisco Unified Communications Manager tracks the number of units that each device requires, asshown in the figure, F.ach device type corresponds to a fixed number of units. The amount of DLUs depends on device type and device capabilities, such as:
Video capabilities
The number ofrequired units per dev ice can be viewed from Cisco Unified Communications Manager Administration. DLUs are perpetual and device-independent.
1-24
A software license has to be added (5.x only required node licenses and DLUs).
MAC address of the license server (publisher) Version (major release) of the Cisco Unified Communications Manager software
Number of node licenses
NumberofDLUs
License files are additive (multiple license files can be loaded). The Cisco FIcxl.M process is used to obtain licenses, and a digital signature ensures the integrity of license files.
1-25
1000 DLUs:
INCREMENT PHONE_UNIT ciSCO 8.0 permanent uncounted \
VENDOR_STRING=<Count>1000</Count><OrigMacId>OOOBCD4EE59D</OrigMacId><Li
cFIleVersion>l.0</LicFlleVerslon>
HOSTID=000bcd4ee5 9d
NOTICE=-cLicFileIDs2D0 50B26140539162</LicFileIDxLicLineID>2</LicLineID
> \
F359
D594
Unified Communications Manager Version 8.0. 'fhere is no expiration date for this license, as indicated by the keyword '"permanent."
Note If this license had been a Cisco Unified Communications Manager node license, the
1-26
**,,
""
Generate
license
fie
fl^^^ft*
As shown in the figure, the license file requestprocess includes these steps:
Step 1
Step 2
Step 3
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.
Step 4
Step5
fhe customer registers the product atCisco.com or public web page and provides
the MAC address of the publisher devicethat will becomethe licenseserver.
The license fulfillment infrastructure validates the PAK, and the license key
generator creates a license file.
Step 6
Step7
The license file is delivered viaemail to the customer. Theemail message also
contains instructions on how to install the license file.
1-27
New
License
2 additional
! 100 units
-. 100 units
Cisco
* hoc- units
Server
200 units
The process of obtaining additional DLUs and node licenses includes the following steps:
The customer places an order for the additional licenses for a license server (publisher
MAC address has to be specified).
When the order is received. Cisco.com generates a license file with the additional count and
sends it to the customer.
fhe new license file has to be uploaded to the license server and will be cumulative.
For example, if >ou have an existing license file that is uploaded to Cisco Unilied Communications Manager that contains 100 DLUs, and you purchase another 100 DLUs. die second license file that is generated will contain only 100 DLUs. When this license file is uploaded to Cisco Unified Communications Manager, the 100 DLUs from the first license tile are added to the dev ices of the second license file, resulting in a total of 200 DLUs.
1-28
The Lev licensing components of the Cisco Unified Communications Manager licensing are the
license server and the license manager.
License Server
Manager cluster. The publisher takes onthe functionality ofthe license server and is responsible for keeping track of the licenses that are purchased and the licenses that arc used. When you request a license file, the MAC address of the publisher isrequired to generate the license file. Once generated, the license file hasto be loaded to thepublisher, which has to have
the corresponding MAC address.
License Manager
Manager. This logical component acts asa broker between Cisco Unified Communications Manager applications that use licensing information and the license server. When the License Manager receives a request from the Cisco Unified Communications Manager application, it forwards the request to the license server and responds back to the application alterthe license
server has processed the request.
1-29
BS^^H
(forDenrt(5)
Unified CM'
An administration subsvsiem and alann subsystem complete the functional diagram. Details of
these two subsv stems are as follows:
keeps infonnation about the license units that are required for each phone 1} pe. Ihe customer can view this information using a GUI.
Supports a GUI tool that calculates the required numberof phone unit licenses, 'fhe customer inputs phone tvpes and the number of phones of each type that the customer wants to purchase. The output is the total number oflicenses that the customer would need lor the given configuration. Supports a GUI tool that displays the total license capacity and the number of licenses in use and the license file details. The tool can also report the numberof
available licenses,
fhe alarm subsv stem generates alarms that are routed to event logs or sent to a management station as Simple Network Management Protocol (SNMP) traps to notify the
administrator of these conditions:
occurs when more licenses are used than available, but the amount of exceeding licenses is in an acceptable range (5 percent overdraft is permitted).
License scner dov\n: Occurs when the license manager cannot reach the license
serv er,
Insufficient licenses: Occurs when the license server detects the fact that there are
not sufficient licenses to fulfil! the request and raises an alarm to notify the
administrator.
1-30
Issues wilh license file: Occurswhen there is a versionmismatch betweenthe license file and Cisco Unified Communications Manager(license file version
mismatch alarm), or when the number of licenses in the license file is less than the
number ofphones that are provisioned (license file insufficient licenses alann).
Another cause of this condition is an invalid MAC address (for instance, after a network interface card [NICJ change).
1-31
Calculating Lici
Cisco Unified Communications Manager Administration includes a license calculator that displays the amount of units consumed per device and calculates the total amount of required units for a given
number of devices.
OBwattfeBidwBf
bn'f hrtiM
lk<i this procedure to calculate the numberof phone licenses that are required when the number of phone tvpes 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 numberof license units that are consumed per device and the
current number of devices is displayed.
Step 2 In the Number of Devices column, enter the desired number of devices,
Step 3
Click Calculate. Ihe total number of Cisco Unified Communications Manager node license units and DLUs thai are required for specifiedconfiguration is displayed.
1-32
Usethis procedure to generate a license unitreport: Step 1 Choose System > License >License Unit Report.
Step 2
The License Unit Report window displays the number ofphone licenses and number
of node licenses, in these categories:
Units Authorized
Units Used
Units Remaining
Manager Administration.
Step 2
Step 3
Choose System> License >License File Upload. The License File Upload window
displays.
Click Upload License File. Click Browse to choose the license file from the local
directory.
Click Upload.
($
Step 4
Step 5
Step 6
Click Browse to choose the license file from the local directory.
Click Upload.
1-35
Upload Result
CCM20070822:16rjeill55,lic
11 Continue |
Step7
After the upload process is complcle. the Upload Result file displays. Click the Continue promptwhen it appears. The contentof the newly uploaded license tile
will be displaved.
1-36
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Cisco Unified Communications is a comprehensive
Summary (Cont.)
Access to the system is only allowed through the use of
Cisco CLI and GUI tools.
Cisco Unified Communications Manager uses an Informix Dynamic Server database, and configuration information in the database is replicated fromthe first node to all
subsequent nodes within a cluster.
References
for additional infonnation. refer to these resources:
Cisco Sv stems. Inc. Cisco t mfied Communications System Re/ease 8.xSRKD San Jose
California. April 2010.
hup: wuw.cisco.com en/l S-docwoice ip comnv'cucni/srnd/8\/iicSxsriKl.pdf Cisco S\stems. Inc. Cisco I'nified Workspace Licensing, California, February 2010.
http:' www.ciM.-o.com eiVUN.partner'pnxluets/ps9156.inde\.html
1-38
Lesson 2
and redundancy options ofCisco Unified Communications Manager and tofollow the
recommended design and deployment practices.
Objectives Upon completing this lesson, you will be able to understand the Cisco Unified Communications
Manager deployment and redundancy options. This ability includes being able tomeet these
objectives:
deployment with centralized call processing, and list the reasons for choosing this
deployment option Describe the characteristics of a Cisco Unified Communications Manager multisite
deployment with distributed call processing, and list the reasons for choosing this
deployment option
depkmnent with clustering over the WAN, and list the reasons for choosing this
deployment option
F.xplain how call-processing redundancy is provided in a Cisco Unilied Communications Manager cluster, and identifv the requirements for ditferent redundancy scenarios
1-40
Multisite WAN with centralized call processing Multisite WAN with distributed call processing Clustering over the IP WAN
Selection ofthe type ofdeployment model isbased onseveral factors, including tlie following: Size: Number ofIPphones. Cisco Unified Communications Manager servers, and other resources, such asgateways or media resources (conference bridges, music onhold [MOH]
servers, and so on)
Geographical distribution: Number and location of sites Network characteristics: Bandwidth anddelay of network links, and type of traffic that is
carried over ttie network
1-41
physical location. IP WAN (if one) is used for data traffic only: PSTN is used for all external
calls.
The single-site or Campus model for Cisco Unified Communications consists of a Cisco
Unified Communications Manager clusterthat is located at a single site, or campus, with no lelephonv sen ices provided over an IP WAN. All Cisco Unified Communications Manager
servers, applications, and digital signal processor (DSP) resources arc located in the same phvsical location.
An enterprise wouldtvpically deplov the single-site model over a UAN or metropolitan-area network (MAN), which carries the voicetraflic within the site. In this model, calls beyond the
I.AN or MAN use the public switched telephone network (PSTN).
In a single-site deplov ment model, all Cisco Unified Communications Manager servers,
applications, and DSP resources are located in the same physical location.
fach clustersupports a maximum of 30.000 IP phones. If there is a need to deploy more than 30.000 IP phones in a single-site configuration, multiple clusters that are inside a IAN or within a MAN can be implemented and interconnected through intercluster trunks.
(iatevvav trunks that connect dircctlv to the PS'IN manage external calls. If an II* WAN exists
between sites, it is used to carry data traffic onlv: no telephonv services are provided over the
WAN.
Design Guidelines
Single-site deployment requires that, for future scalability, best practices that are specific to the
distributed and centralized call-processing modelare recommended.
Current calling patterns within tlie enterprise must beunderstood. How and where are users
making calls? How many calls are intersite versus intrasitc? Ifcalling patterns dictate that most
calls areintrasite. using the single-site model will simplify dial plans and avoid having to provision additional dedicated bandwidth for voice across the IPWAN,
Because VoIP calls are within the UAN or campus network, it is assumed thatbandwidth is not
a concern. Using G.722 orG.711 coder-decoders (codecs) for all endpoints will eliminate the requirement ofDSP resources for transcoding, and those resources can be allocated to other
functions such as conferencing and MediaTermination Points(MTPs).
All off net 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 of Media Gateway Control Protocol (MGCP) gateways for the PSTN gateway is
recommended if11.32*3 functionality isnot required. When deploying multiple clusters, choose
a uniform gateway and centralize the gateway functions using H.323 gatekeepers rather than
using MGCP gateways.
Single-site deployment does not always equate toa single cluster. Ifthe site has more than 30.000 IP phones, install multiple clusters and configure intereluster trunks (ICTs) between the
clusters.
Ease of deployment
Benefits
A single infrastructure for a converged network solution provides significant cost benefits and enables Cisco Unified Communications to take advantage of the many IP-based applications in
the enterprise.
Single-site deplovment allows each site to be completely self-contained, ('alls between sites will be routed over the PS'fN, Additional provisioning of WAN bandwidth is not needed. Dial plans are also easier to provision. Ihere is no dependency for service in the event of an IP WAN failure or insufficient bandwidth, and there is no loss of call-processing serviceor
functionality.
The multisite WAN with centralized call-processing model consists of a centralized Cisco Unified Communications Managerclusterthat provides services for many sites and uses
The figure illustrates a typical centralized call-processing deployment, with a Cisco Unified Communications Manager cluster at thecentral site and an IP WAN with QoS thatis enabled to
connect all the sites. Theremote sitesrelyon the centralized Cisco Unified Communications Manager cluster to manage their call processing. Applications such asvoice mail and
interacts e voice response (1VR) systems are typically centralized as well toreduce the overall
costs of administration and maintenance.
The Cisco Unified Survivablc Remote SiteTelephony (SRST) feature that is available in Cisco
IOS gateways provides call-processing services toremote IP phones during a WAN outage. When the IP WAN is down, the IPphones at the remote branch office can register to the Cisco
Unified SRST router. The Cisco Unified SRST router can process calls between registered IP
To avoid oversubscribing theWAN links with voice traflic, causing deterioration of the quality
of established calls. Call Admission Control (CAC) is used to limit the number of calls between
the sites.
Centralized call-processing models can take advantage ofautomated alternate routing (AAR) features. AAR allows Cisco Unified Communications Manager todynamically reroute a call
over the PSTN if the call is denied because of CAC.
1-46
Maximum of2100H.323 devices (gateways, multipoint control units, trunks, and clients) or 1100MGCP gateways
Cisco Unified SRST on the branch router limits remote offices to a maximum of 1200 Cisco IP phoneswhen using a Cisco
3945 Integrated Services Router.
Design Guidelines
When implementing the multisite WAN model with centralized call processing, the following
guidelines should be considered:
Maximum of 2000 locations per Cisco Unified Communications Manager cluster. Maximum of2100 H.323 devices (gateways, multipoint control units, trunks, and clients) or 1100 MGCP gateways per Unified Cisco Unified Communications Manager cluster.
reduce voice cut-through delays.
Minimize delay between Cisco Unified Communications Manager and remote locations to
Use the locations mechanism inCisco Unified Communications Manager to provide CAC into and outof remote branches. Thelocations can support a maximum of 30,000 IP
phones per cluster when Cisco Unified Communications Manager runs on the largest supported server. Since Cisco Unified Call Manager Version 5.0, you can use Resource
Reservation Protocol (RSVP)-based CAC between locations.
There is no limit to the number of IPphones at each individual remote branch. However,
the capability that the Cisco Unified SRST feature provides in the branch router limits
remote branches to a maximum of 1200 Cisco IPphones on a Cisco 3945 Integrated
Senices Router during a WAN outage orfailover to SRST. Other platforms have different
limits.
Ifadistributed call-processing model ismore suitable for the business needs ofa customer, the
choices include installing a Cisco Unified Communications Manager cluster at the remote
branch or running Cisco Unified Communications Manager Express on the branch router.
1-47
Processing (Cont.)
Benefits
* Acommoninfrastructure for a converged solution. PSTN callcost savingswhen using the IP WAN forcalls
between sites.
dialed This practice is known as tail-end hop-off (TEHO). * Maximum utilization ofavailable bandwidth by allowing voice traffic to share the IPWAN with other types of traffic.
Use of Cisco Extension Mobility features between sites.
Use of AAR in case of insufficient bandwidth.
Centralized administration.
Benefits
Multisite WAN with centrali/cd call processing saves PSTN costs for intersite calls by using the IP WAN instead ofthe PSTN. IP WAN can also be used tobypass toll charges by routing calls through remote site gateways, closer to the PS'IN number that isdialed, fhis practice is
known as tail-end hop-off (TfHO). 1hiK) is disallowed in some countries, and local regulations should be verified before implementing ITT10.
"fhis deplov ment model maximizes the utilization of available bandwidth by allowing voice
traffic to share the IP WAN withother types of traffic. Deploying OoS and CAC ensures voice
quality, AAR reroutes calls over the PSTN ifCAC denies the calls because ofoversubscription. Cisco Extension Mobility can be used within the Cisco Unified Communications Manager
cluster, allowing roaming users to usetheirdirectory numbers at remote phones as iflhev were
at their home phones.
When using the multisite WAN with centralized call-processing deployment model. Cisco Unilied Communications Manager Administration is centralized, and therefore simpler, compared with a multisite with distributed call-processing model where multiple clusters have
to be separately administered.
This topic describes the characteristics of aCisco Unified Communications Manager multisite
Communications Manager
signaling.
The model for a multisite WAN deployment with distributed call processing consists of
multiple independent sites, each with its own Cisco Unified Communications Manager cluster
that is connected to an IP WAN that carriesvoice traffic between the distributed sites. The IP
WAN is used with the PSTN for intersite calls.
Cisco Unified Communications Manager, applications, and DSP resources may be located at each site. IP WAN carries only signaling traffic for intersite calls, butsignaling traffic for calls within a site remains local to the site. Thisway, the amount of signaling traffic between sites is
reduced compared with a centralized call-processing model. Bach site is completely selfsufficient and will continue to operate independently ifother sites fail or the IP WAN becomes
unavailable.
With the use ofgatekeepers, a distributed call-processing model can scale tohundreds ofsites. It also provides transparent use ofthe PSTN if the IPWAN isunavailable.
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. Follow tlie bestpractices guidelines lorsingle-site and multisite deployments in addition to the guidelines here,
which are specific to this deploy ment model.
When using gatekeepers to control the intereluster communication, this deploy ment model
scales to hundreds of sites. A gatekeeper is an 11.323 device that provides CAC and I-.164 dial plan resolution. Additional gatekeeper guidelines include the following:
Gatekeeper networks can scale to hundreds of sites. 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 thetopology.
Itis recommended to use gatekeeper redundancy support to provide a gatekeeper solution with high availability. It is also recommended to use multiple gatekeepers to provide spatial
redundancy within the network.
!t is recommended to use a single WAN codec, because the 11.323 specification does not
allow tor Layer 2. II'. User Datagram Protocol (UDP), or Real-Time Transport Protocol (RTP) header overhead in the Bandwidth Request. (Header overhead is allowed only in the
pay load or encoded voice part of the packet.). Using one type of codec on the WAN simplifies capacity planning by eliminating the need to overprovision the IP WAN to allow
lor the worst-case scenario.
Noloss offunctionality during IPWAN failure, because there is a call-processing agent at each site
Benefits
fhe multisite WAN withdistributed call-processing deployment model is a superset of both single-site and multisite WAN with centralized call processing. The multisite WAN with distributed call processing model provides the following benefits:
PSTN callcost savings when using the IP WAN for calls between sites Use of the IP WAN to bypass toll charges by routing callsthrough remote site gateways,
closer to the PSTN number dialedthat is, TEHO
rmq over
IPWAN
m*
QoS Enabled Bandwidth
Cisco suppons Cisco L'nified Communications Manager clusters over a WAN. Some of the characteristics include the following:
Applications and Cisco Unified Communications Manager of the same cluster that is
distributed over the IPWAN.
Two to four sites for local failover(two Cisco Unified Communications Manager
servers per site) [}p to eight sites for remote failover across the IP WAN (one Cisco Unified
The cluster design is useful for customers who require more functionality than the limited feature set that is offered by SRS'I. This network design also allows remote offices to support more IP phones than SRST if the connection to the primary Cisco Unified Communications
Manager is lost.
1-52
remote to the Publisher (Database and inter-server traffic) Upto eightsmall sites using the remotefailover deployment
model
Design Guidelines
round-trip delay of80ms between them. In comparison, high-quality voice guidelines dictate that one-way. end-to-end delay should notexceed 150 ms. Because of this strict
guideline, this design can be used only between closely connected, high-speed locations.
Foreven' 10.000 busy hourcallattempts (BHCAs) within the cluster, an additional 900
kb/s of WAN bandwidth for intracluster run-time communication must be supported. The
BHCA represents the number ofcall attempts that are made during the busiest hour of the
day.
Up toeight small sites are supported using the remote failover deployment model. Remote failover allows you to deploy oneserver perlocation (maximum of eight call-processing seners are supported in a cluster). If CiscoUnified Communications Manager fails. IP phones will register to another server over the WAN. Therefore, SRST isnot required in this deployment model (although it issupported). The remote failover design may require signi ficant additional bandwidth, depending onthe number oftelephones ateach location.
1-53
Note
In prior versions ofCisco Unified Communications Manager, subscriber serversin the cluster use the publisher database forread/write access, and they use theirlocal database for read-only access when the publisher databasecannot be reached. Starting with Cisco
Unified Communications Manager Version 6.x, subscriber servers in the cluster read their
local database. Even database modifications can occurinthe local database (for special applications such as user-facing features). IBM Informix Dynamic Server (IDS) database
replication is used to synchronize the databases on the various servers in the cluster
Therefore, when recovering from failure conditions such as the loss of WAN connectivity for
an extended period, the Cisco Unified Communications Manager databases must be synchronized with any changes that might have been made during the outage.
This process happens automatically when database connectivity is restored and can take longerover low-bandwidth links. In rare scenarios, manual reset or repair of the database replication between servers inthe cluster mightbe required,which is performed by using the
commands such as utils dbreplication repair all or utils dbreplication reset all at the command-line interface (CLI) Repair or reset of database replication using the CLIon remote subscribers over the WAN causes all Cisco Unified Communications Manager
databases in the cluster to be resynchronized, in which case additional bandwidth above
1 544 Mb/s might be required. Lower bandwidthscan take longerfor database replication
repair or reset to complete.
Maximum utilization of available bandwidth by allowing voice trafficto share the IP WAN with other types of traffic
Failover across WAN supported
Benefits
Clustering over tlie IP WAN provides a combination ofthe benefits ofthe two deployment
models to satisfy specific site requirements.
Although there arc stringent requirements, clustering over the IP WAN offers these advantages: Single point ofadministration for users for all sites within the cluster
Feature transparency
The clustering over IP WAN design isuseful for customers who want to combine these advantages with the benefits that are provided by a local call-processing agent ateach site (intrasitc signaling is kept local, independent ofWAN failures) and requires more functionality at the remote sites than thatprovided bySRST. This network design also allows remote offices tosupport more Cisco IP phones than SRST (1200 IP phones using Cisco 3945 routers) in the
event of WAN failure.
These features make clustering across the IP WAN ideal asa disaster-recovery' plan for business continuance sitesor as a single solution for up to eightsmall or medium sites.
1-55
Cisco Unified Communications products can run as virtual machines on a selected set of supported virtuaiization server technologies. The principal component of a virtual server is the Cisco Unified Computing System (UCS) Platform along with its hypervisor virtuali/ation
technology.
Unified Computing is an architecture that integrates computing resources (CPU. memory, and I/O). IP networking, network-based storage, and virtuali/ation. into a single highly available system. This level of integration provides economies of power andcooling, simplified server connectivity into the network, dynamic application instance repositioning between physical
hosts, and pooled disk storage capacity.
Thearchitecture uses a unified fabric that provides transport for LAN. storage, and highperformance computing traffic ov er a single infrastructure with the help of technologies such as Fiber Channel over Fthemet (FCoh). Cisco's unified fabric technology is builton a 10-Gb/s Fthemet foundation that eliminates the need for multiple sets of adapters, cables, and switches for LANs. Storage Area Networks (SANs), and high-performance computing networks.
1-56
- Prioritized list of call-processing servers (one or more). - Multiple Cisco Unified Communications Manager groups
can exist in the same cluster.
Managergroup assigned, which will determine the primary and backup server to which it can register.
single IP PBX system. With Cisco Unified Communications Manager Version 8.0. a cluster may contain up to 20servers, ofwhich a maximum of8 servers may run the Cisco
Cal'lManager service performing call processing in acluster. Other servers can be used as TFTP
servers or provide media resources such assoftware conference bridges or MOH.
Cisco Unified Communications Managercall-processing redundancy is implemented by
grouping servers that arerunning the Cisco CallManager service into Cisco Unified Communications Manager groups. A Cisco Unified Communications Manager group is a
prioritized list of one or more call-processing servers.
The following rulesapply for the Cisco Unified Communications Manager groups:
Multiple Cisco Unified Communications Manager groups can exist in the same cluster.
Fach call-processing server can be assigned to more than oneCiscoUnified
Communications Manager group.
liach device has to have a Cisco Unified Communications Manager group assigned, which will determine the primary and backupserversto which it can register.
1-57
Cisco IP phones register wilh their primary server. When idle, the IP phones and Cisco Unilied Communications Manager exchange the signaling application keepalives. In addition, Cisco IP
phones establish aTCP session with their secondary server and exchange TCP keepalives. When the connection to the primary server is lost (no keepalives received), the IP phone
registers to the secondarv server. The IPphone will continuously try to re-establish a
connection with the primary server: ifsuccessful, the IP phone will reregister with the primary
server.
1-58
1 Redundancy Design
High availability (upgrade)
Increased server count
Simplified configuration
1S.OOG IP ptwnss 30,000 IP phones
Cisco MCS 784S
%:*
' - *"
TFTP Server
{NorRequired
MOOOj
Cisco IPphone registrations will never overwhelm the backup servers, even if multiple primary servers fail concurrently. However, the 1:1 redundancy design has an increased server count compared with otherredundancy designs and may not be cost-effective.
The otherservices (dedicated database publisher, dedicated TFTP server, or MOH servers) and media-streaming applications (conference bridge or MTP) may also be enabled on a separate
server that registers with the cluster.
Fach cluster must also provide a TFTP service. The TFTP service is responsible for delivering IP phone configuration files to telephones, along with streamed media files, such as MOH and ring tiles. Therefore, the server that isrunning the TFTP service can experience a considerable network and processor load. Depending on thenumber of devices thata server is supporting, you can run the TFTP service ona dedicated server, on the database publisher server, oron any
other server in the cluster.
In thisexample, a Cisco Media Convergence Server (MCS) 7845 Series is used as the dedicated database publisher andTFTPserver. In addition, there aretwo call-processing servers supporting a maximum of 7500CiscoIP phones (on the CiscoMCS 7845 Series
platform). One of these two servers isthe primary server: the other one isa dedicated backup
server. Thefunction of thedatabase publisher and theTFTP server can be provided by the
primary orsecondary- call-processing server ina smaller IP telephony deployment (fewer than
1000 IP phones). In thiscase, only twoservers areneeded in total.
1-59
When you increase the number of IP phones, you must increase the number ofCisco Unified
Communications Manager servers that are required to support the telephones. Some network engineers ma; consider the 1:1 redundancy design excessive, because awell-designed network is unlikely to lose more than one primary server at atime. With the low possibility ofserver loss and the increased server cost, many network engineers choose to use a 2:1 redundancy
desien.
1-60
15.000 IP phones
30,000 IP phones
1to
7S00 rsoito
13,000
15.001to 22.500
22.5Glto
30.000
Although the 2:1 redundancy design offers some redundancy, there isthe risk ofoverwhelming the backup server if multiple primary servers fail. In addition, upgrading the Cisco Unified Communications Manager servers can cause a temporary lossof some services such asTFTP
or DIICP. because a reboot of the Cisco Unified Communications Manager servers is needed after the upgrade is complete.
Network engineers use this2:1 redundancy model in mostIP telephony deployments because
of the reduced server costs. If a Cisco MCS 7845 Series is used (shown in the figure), that
server is equipped with redundant, hot-swappable power supplies and hard drives. When these servers areproperly connected and configured, it is unlikely thatmultiple primary servers will
fail at the same lime, which makes the 2:1 redundancy model a viable option for most
businesses.
As shown in the first scenario, when using no more than 7500 IPphones, there are no savings in the 2:1 redundancy design compared with the 1:1 redundancy design, simply because there is
only a single primary' server.
In the scenario with up to 15.000 IP phones, there are two primary servers (each serving 7500 IPphones) and onesecondary server. As long asonly oneprimary server fails, the backup server can provide complete support. If both primary servers failed, thebackup server would
only be able to serve half of the IP phones.
The third scenario shows a deployment with 30,000 IP phones. Four primary servers are
required to facilitate this amount of IP phones. Foreach pair of primary servers, there is one backup server. As long as no more than two servers fail, the backup servers can provide
complete support, and all IP phoneswill operate normally.
Summary
This topic summarizes the key points that were discussed in this lesson.
In the single-site deployment model, the Cisco Unified Communications Manager, applications, and DSP resources are at the same physical location; all off-site calls are
processed by the PSTN.
The multisite with centralized call-processing deployment model has a single Cisco Unified Communicafions Manager
cluster. Applications and DSP resources can be centralized
Clustering over the WAN provides centralized administration, a unified dial plan, feature extension to all offices, and support for more remote phones during failover. But it also places strict delay and bandwidth requirements on the WAN.
1-62
References
For additional infonnation. refer to these resources:
Cisco S> stems. Inc. Cisco Unified Communications System Release 8.x SRND. San Jose.
California. April 2010.
liitp:.'wvtvv.cisco.coni/en/US/docs/voicc.jp..conim/cucm/srrid/8x/tic8\snid.pdr.
1-63
1-64
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Cisco Unified Communications Manager is the central
This module describes the main characteristics of Cisco Unified Communications Manager. The module describes the role that Cisco Unified Communications Manager plays in the overall Cisco Unified Communications solution, and the Cisco Unified Communications Manager
hardware and software requirements. Also, themodule describes the four call-processing deployment models and how Cisco Unified Communications Manager clusters provide
redundancv and failover.
References
For additional infonnation. refer to these resources:
Cisco Systems. Inc. Cisco Unified Communications System Release 8.xSRND. SanJose.
California. April 2010.
hup: .vvvvvv.cisco.com/cn/US/doesAoicc_ip_comm/cucm/smd/Sx/uc8Ksrud.pdf
Cisco Systems. Inc. Cisco Unified Workspace Licensing, California, February 2010. lit[p:'-'ww w-cisco.com/en/US/parliier/prodiicts/ps9i56/indcx.html
1-65
1-66
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.
Ql)
Which two ofthe following options are not part ofthe Cisco Unified Communications
architecture? (Choose two.) (Source: Understanding Cisco Unified Communications
Manager Architecture) A) B) C) D) E)
F)
Ci)
third-party applications
Q2)
Which three of the following options areCisco Unified Communications Manager functions? (Choose three.) (Source: Understanding Cisco Unified Communications
Manager Architecture) A) B) C) D) F) F) G) packet routing signaling and device control dial plan administration phone-feature administration storing voice mail providing call-center functionality filtering IP packets
Q3)
List the minimum hardware requirements for Cisco MCS platforms that are required by
Cisco UnifiedCommunications ManagerVersion8.0. (Source: Understanding Cisco
Unified Communications Manager Architecture)
Q4)
Which database is used by Cisco Unified Communications Manager Version 8.0? (Source: Understanding Cisco Unified Communications Manager Architecture)
A) B) C) D) E) IBM Informix Dynamic Server Microsoft SQL Server 7.0 Microsoft SQL Server 2000 Oracle Microsoft Jet
Q5)
Which two of the following featuresrely on the publisher? (Choosetwo.) (Source: Understanding Cisco Unified Communications ManagerArchitecture)
A) B) Call Forward All Message Waiting Indicator
C)
D) E) F)
1-67
Q6)
What is a licensing overdraft, and by what percentage is it allowed? (Source: Understanding Cisco Unified Communications Manager Architecture)
Q7)
Which ofthe following options is not a Cisco Unified Communications Manager licensing tool? (Source: Understanding Cisco Unified Communications Manager
Architecture)
A)
B)
C)
I))
Q8)
Q9)
Which codecs are recommended in a single-site Cisco Unified Communications Manager deployment? (Choose two.) (Source: Understanding Cisco Unified Communications Manager Deployment and Redundancy Options)
A) (i.721
B) C) D) F.)
QIO)
Which statement is true about a multisite WAN with centralized call-processing Cisco Unified Communications Manager deployment? (Source: Understanding Cisco Unified Communications Manager Deployment and Redundancy Options)
A) B) C) [)) fhe 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. fhe IP WAN carries no call-control signaling for intrasitc calls.
Ql 1)
Which Cisco Unified Communications Manager deployment model oilers the highest scalability ? (Source: UnderstandingCisco Unified Communications Manager
Deployment and Redundancy Options)
A) B) C) D)
Q12) Which two options are features ofCisco Unified Communications Manager clustering
over the WAN11 (Choose two.) (Source: Understanding Cisco Unified Communications
Manager Deployment and Redundancy Options)
A) B) C)
D)
E)
highestscalability
A) B) C) D)
18nodes.6 of them used for call processing 18nodes. 8 of them used for call processing 20 nodes. 6 of them usedfor callprocessing 20 nodes, 8 of them used for call processing
1-69
Q->
H. C. L)
Q3)
Q-l)
Q>>
06)
Ql i OS)
Q'>\ 0 I (1 I OH i
An o\erdiait is acondition where more deuces rcgislcr to Cisco Unified Communication-, Manager ihan there are license units purchased Cisco Unilied Communications Manager allows a 5 percent overdraft
B A.C.I:
B. I: C IS
012)
01.')
A. D
li
1-70
Module 2
This module describes the configuration ofinitial settings and explains how tomanage users in
Cisco Unified Communications Manager.
Module Objectives
Upon completing this module, you will be able to perform Cisco Unified Communications
Manager initial configuration and user management. This ability includes being able tomeet
these objectives:
Manage user accounts, including integrating Cisco Unified Communications Manager with
a corporate LDAP directory and enabling multiple levels of user privileges
2-2
Lesson 1
Objectives
Upon completing this lesson, you will be ableto activate required CiscoUnified Communications Manager Services and settings anddecide whether to use the Domain Name System (DNS), litis ability includes beingable to meettheseobjectives:
List elements that are used for general, initial configuration
Listnetwork configuration options of CiscoUnified Communications Manager Listthe reasons for using NTP servers andenabling DHCP services in CiscoUnified
Communications Manager
Describe the reliance on DNS by IP phones when servernames are used insteadof server
IP addresses
Describe the difference between network and feature services and explain how they can be managed using Cisco Unified Serviceability
Verifythe Fdfaving
Network settings
fcMlxf^ls"1**!^ "inpijj^j,
NTPservers, DHCP services, remove
DNS reliance.
After installing Cisco L'nified Communications Manager, some initial configuration has to be done before starting to deplov endpoints. This initial configuration includes the following:
Configure network settings: Basic network settings have already been configured during installation. However, some of them should be revisited -for example, use ol external Network Time Protocol (NTP)and DNS servers. Network settings that are not configurable
during installation--for example, enabling DHCP services on Cisco Unified Communications Manager have to be addressed before endpoint deployment.
\ erify network and feature services: Cisco Unified Communications Manager servers run network sen ices (automatically activated) and feature services (activated by the
administrator). Alter installation, network services should be checked, and desired feature
sen. ices have to be activated.
cluster-wide configuration settings called enterprise parameters. After installation, enterprise parameter default values should be verified and modified, if required. Configure service parameters: Cisco Unified Communications Manager Services have configurable parameters that can usually be set per Cisco Unified Communications Manager server. After installation and service activation, service parameter default values should be verified and modified, if required.
2-4
Master NTP
Reference Clock
-^
NTP
DHCP/TFTP
DNS
Cisco Unified Communications Manager network configuration options include the use of external NTPand DNS servers and the ability to provide DHCP and TFTP servicesto
endpoints.
2-5
Network Components
fhis section describes the function ofnetwork components that are used orprovided by Cisco
Unified Communications Manaaer.
DHCP server provides IP address configuration and TFTP server location to the IP phones. TFTP server provides device configuration files, ringer files, and firmware upgrades to the IP phones. A Cisco Unified Communications Manager server
(typically publisher) can provide both the DHCP and
TFTP services.
DNS server
fhe NTP is a protocol for synchronizing the clocks of computer systems over IP networks, ll has a hierarchical organization by the use of clock strata. Stratum 0 is an extremely precise clocksource, such as an atomic clock or radioclock. A stratum I server is directly connected to
a stratum 0 clock and can prov ide time infonnation to other (stratum 2) devices, which
themselves can serve stratum 3 devices.
Cisco Unilied Communications Manager uses NTP to obtain time information from a time server (Upically stratum 1). Onlv the publisher will send NTP requests to the external NTP server or servers; subscribers will synchronize their time with the publisher.
DHCP is a protocol that allows IP endpoints to obtain their IP settings from a server. The most important settings arc IP address, subnet mask, and default gateway. In addition, the DNS serveraddress and special functions, such as the TFTP serveraddress tiiat is used by Cisco IP phones, can be assigned to the client. Cisco Unified Communications Manager features a DHCP server, which is designed to serve Cisco IP phones on!v.
TFTP is a simple FTP and is used by Cisco Unified IP phones to obtain configuration files and their software. A Cisco Unified Communications Manager cluster has to run the TFTP service at least on one server to be able to support Cisco Unified IP phones.
DNS is a name resolution protocol that allows IP applications to refer to oilier svstems bv logical names instead of IP addresses. A Cisco Unified Communications Managet cluster can
be configured to use either DNS or IP addresses.
Subscriber
Manager. At least one external NTP server must be reachable and functioning when installing
the Cisco Unified Communications Manager publisher to complete the installation. Cisco is
recommending using a minimum of three external NTP servers ina production environment. It isextremely important that all network devices have accurate time information, because the system time ofCisco Unified Communications Manager isrelevant inthe following situations: Cisco IP phones display date and time information; this information isobtained from Cisco
Unified Communications Manager.
Call Detail Record (CDR) andCall Management Record (CMR), which are used for call reporting, analysis, and billing, include dale andtime information. Alarms andevents in log files, as wellas traceinformation in tracefiles, include time infonnation. Troubleshooting a problem requires correlation of infonnation that is created by different system components (Cisco Unified Communications Manager, Cisco IOS
gateway, and soon). This problem-solving isonly possible if all devices inthe network
have the same correct lime information.
therefore rely oncorrect date and lime. These features include timc-of-day routing and
certificate-based security features. Note Certificatesincludea validity period. Ifa system that receives a certificatehas an invalid (future) date, it may consider the received certificateto be invalid (expired).
Administering Cisco Unified Communications Manager
2-7
1o ensure that all network devices have correct date and time, it isrecommended that all network dev ices use NTP for time synchronization. The master reference clock should be a
stratum I NTP server.
2-8
Ho*tnani> or IP Addrtu
10.1.1.101
status
i AddNCW
Tomodify NTP configuration inCisco Unified Communications Manager, use Cisco Unified Operating System Administration web pages and goto Settings >NTP Servers. There, you
can add. delete, and modify NTP servers.
Note Though an NTP server must be reachable during installation of Cisco Unified Communications Manager, the NTPserver configuration can later be deleted from Cisco Unified OperatingSystem Administration web pages. This is not recommended.
2-9
Sufficientfor IP phone purposes Not designed to serve other network devices (PCs) Only for smaller deployments (up to 1000 IP phones)
Multiple DHCP services per Cisco Unified Communications Manager cluster:
'fhe Cisco Unified Communications Manager DHCP server is designed to serve IP phones in
small deplovmenls (maximum of 1000 devices). It provides a subset of Windows. Linux, or Cisco IOS DHCP server functionality that is sufficient for IP phones, but it should not be used for other network dev ices (such as PCs).
Note The DHCP server of Cisco Unified Communications Manager must not be used with deployments of more than 1000 registered devices. Even ifthere are fewer devices, the
CPU load of the services has to be watched closely, and ifhigh CPU ioad is experienced. the DHCP service should be provided by other devices (forexample, dedicated DHCP
server switch, router, and so on).
Multiple DHCP services can be configured per Cisco Unified Communications Manager cluster, f.ach Cisco Unified Communications Manager DHCP server can be configured with
multiple subnets. In nonattached subnets. DHCP relay must be enabled so that the DHCP requests that were sent out by the clients are forwarded to the Dl ICP server.
2-10
2 3.
Add and configure the DHCP server. Configure the DHCP subnets.
To activate andconfigure the Cisco Unified Communications Manager to provide the DHCP
service, the following steps need to be followed:
Step 1
Step 2 Step 3
2-11
iJi.dteit
l>dlTFV.dLc[l
M activated
C.sin F>ten3ed fl^clFO'i
Activate the DHCP Monitor Service from Cisco Unified Serviceability > Tools > Service
Activation.
2-12
||
tothe phones.
0 tim*.uK}'
r-
?*newBl[T 11 Time4ieOk
tngCIi) T.ms)"
DHCP server configuration includes the selection ofthe CiscoUnified Communications Manager cluster member thatshould run the DHCP service (drop-down list) and general
(default)parameters, such as DNS and TFTP server addresses.
2-13
' 5e"
Cisco Unified Communications Manager DHCP Subnet Information configuration includes the selection ofthe DHCP server, the network ID ofthe subnet, up to two continuous IP address ranges (to allow excluded ranges in between), subnet mask, default gateway, and .ill parameters for which the defaults have been set under Cisco Unified CM Administration > System >
DHCP Server Configuration.
DNS Considerations
This topic describes the advantages and disadvantages of using IP addresses versus DNS.
Cisco Unified Communications Manager can use DNS names (default) or IP addresses for system addressing.
Advantages of using IP addresses Does not require a DNS server Prevents the IP telephony network from faiing when the DNS server is unavailable Decreases the amount of time required when a device attempts to contact the Cisco Unified Communications Manager
server
Easier IP address changes because of name-based IP paths Serverto IP phone NAT possble Possibirtyto have redundant IP phone
services
Simplifies troubleshooting
Cisco Unified Communications Manager can either use IP addresses or names to refer to other IP devices in application settings. When names arc used, they need to be resolved to IP addresses by DNS.
Both methods have some advantages as follows: Using IP addresses: The system does not depend on a DNS server, which prevents loss of
serv ice when the DNS server cannot be reached. When a device initiates a connection for
the first time, the time that is required to establish the connection is shorter because no name resolution (DNS lookup sent to the DNS server, and DNS reply sent back from the server) is required. By eliminating the need for DNS, there is no danger of errors that are caused by DNS misconfiguration. Troubleshooting is simplified because there is no need to verify proper name resolution. I sing DNS: Management is simplified because logical names are simpler to manage than 32-bit addresses. If IP addresses change, there is no need to modify the application settings because they can still use the same names; only the DNS server configuration has to be modified in this case. IP addresses of Cisco Unified Communications Manager servers can be translated toward IP phones, because the IP phone configuration files include server names, not the original server IP address (which should appear differently to the IP phone). As long as these names are resolved to the correct (translated) address when IP phones send out DNS requests, the Network Address Translation (NAT) is no problem.
In general, due to the additional point of failure that is caused by configuration errors or because of unavailability ofthe service, the recommendation is not to use DNS with Cisco Unified Communications Manager.
2-15
Note
Most IP clients cache the IP address information that is received from the DNS servers to
SCCP Call Flow with DNS The figure illustrates acall between IP phones where DNS is used.
SCCP Call Flow with DNS
Before sending packets, IP phones will query the DNS server to
resolve the IP address ofthe Cisco Unffed Communications
Manager server.
IP Phone A
IP Phone B
1) DNS Query
and Response
1] DNSQuery
and Response
Before the IP phone can communicate with Cisco Unified Communications Manager, it has to
resolve the name ofthe server (obtained from the configuration file, which was downloaded from aTFTP server). Only then can signaling messages be exchanged between the IP phone
and Cisco Unified Communications Manager.
Note
SCCP = SkinnyClientControlProtocol
2-17
i! Signaling Proioco
1) Signaling Pralocol
IPPhoneA
PPhoneB
Manager servers, the need for the evlra step of DNS resolulion is eliminated. The signaling
session can be set up immediately and calls can be processed even ifthe DNS service is not
available. Therefore, the recommendation is to remove DNS reliance.
When IP addresses are used instead of DNS names for the Cisco Unified Communications
2-18
Unified Communications Manager to contact the Cisco Unified Communications Manager without resolving a DNS name:
In Cisco Unified
.i|ir,[Jr Cicce Unified CM Admmistratjcm
; MoMriWaa v cam < tmt
Communications Manager
Administration, choose
.,
DnM. Hn** njMter
By default- Cisco Unified Communications Manager propagates the machine name and not the
IP addresses of its active Cisco CallManager Services. (These hostnames are part of TFTP
configuration files for devices such as IP phones.) Removing DNS reliance refers to the requirement for IP phones to use DNS servers to resolve
hostnames of Cisco CallManager Services.
Step 3
Note
Change the server name to the IP address ofthe server and save the changes.
Repeat Steps 2 and 3 for each server in the cluster.
Note
By default, hostnames are also used in phone URLs. When removing DNS reliance, hostnames that are used in these phone URLs also have to be replaced by IP addresses.
Phone URLs are configured by so-called enterprise parameters. Enterprise parameters and their configuration are explained in a later topic of this lesson.
2-19
Network Services
Communications Manager application features; for example, TFTP, call processing, or serviceability reports.
Must be activated manually using Cisco
A Cisco Unified Communications Manager cluster can consist of up to 20 servers, F.ach server can fulfill different tasks, such as running a 1IIP or DHCP server, being the database publisher, processing calls, providing media resources, and so on.
Depending on the usage of a server, different services have to he activated on the system. "Ihere
are two tvpes of serv ices on Cisco Unified Communications Manager servers as follows: Network services: These services are automatically activated and are required for the operation ofthe sener. Network services cannot be activated or deactivated by the administrator, but they can be stopped, started, or restarted from Cisco Unified Serviceability > Control ("enter > Network Services. Examples for network services are Cisco CDP. Cisco DB Replicator, and Cisco CallManager Admin. Keaturc serv ices: These serv ices can be selectively activated or deactivated per server to assign specific tasks or functions (such as call processing. TFTP. and so on) to a certain
server, feature services can be activated and deactivated by the administrator using Cisco
Unified Serviceability > Service Activation. They can be started or restarted from Cisco l'nified Serviceability > Control Center > Feature Services. Kxamples for feature services include Cisco CallManager. Cisco TUT*, or Cisco DirSync.
2-20
Network Services The figure shows a list ofnetwork services that are categorized in groups.
Network Services
Platform Services: A Cisco DB, A Cisco DB Replicator, Cisco Tomcat, SNMP Master Agent, etc.
Note
Cisco Unified Communications Manager Real-Time Monitoring Tool (RTMT) can be installed on an administrator PC. The listed Cisco Unified Communications Manager RTMT services
are required for the client application that is running on the administrator PC to communicate
with Cisco Unified Communications Manager.
DRF stands for the Disaster Recovery Framework. It allows backup and restore tasks to be performed from the Disaster Recovery System (DRS).
2-21
* CM Services: Cisco CallManager Personal Directory, Cisco Extension MobilityApplication, Cisco CallManager Cisco fP Phone Services, Cisco Change Credential Application * CDR Services: Cisco CDR Agent * Admin Services: Cisco CallManager Admin
Note
To enable CDR records, the CDR Services need to be running and the CDR Records
CallManager service parameter needs to be enabled
Feature Services
The figure shows a listof feature services that are categorized in groups.
Feature Services
Performance and Monitoring Services: Cisco Serviceability Reporter, Cisco CallManager SNMP Service
CM Services: Cisco CallManager, Cisco TFTP, Cisco Messaging Interface, Cisco IP Voice Media Streaming App,
etc.
Security Services: Cisco CTL Provider Voice Quality Reporter Services: Cisco Extended Functions
All feature services are disabled by default after installing Cisco Unified Communications
availability of the feature by activating or deactivating the corresponding feature service. Cisco Unified Communications Manager automatically enables the required network services depending on the activated feature services.
2-23
Service Activation
feature services are activated from Cisco Unified Serviceability,
Service Activation
To enable Cisco Unified Communications Manager feature services, perform the following tasks:
Access Cisco Unified Serviceability.
Go to Tools > Service Activation.
To activate or deactivate feature services for a server, perform the following steps in Cisco
Unified Serv iceabilitv:
Step t Step 2
Go to Tools > Service Activation. Select the server where vou want to activate or deactivate a service.
Step 3
Set or remove the check box for each service that vou want to modify and save the changes.
Step 4
Verifv that the sen ice has been started by using the control center ('fools > Control
(enterFeature Services).
5 Go to Control
Center-Feature
server configuration.
eld Sarvr*
Services.
Deactivated Deactivated
DedCti.'str-cJ
activated.
? Deselect the
Desftrwated
rjeacKkated
Deaclrvrfled
Da(tsted
HdSvated
The Service Activation web page is used to selectively activateand deactivate feature services
per server in the cluster.
2-25
W^^
2>5^. :
Ssit Mi DdMbA&e ant jUil&i Servtui
J}
Start, stop, restart,
and refresh selected service
and Up Time
I Runlet J ii.Jj.Mi
Ki
Select service to
start, stop, or
restart.
I he control center for feature sen ices is used to start, stop, or restart and to verifv the status (started or not miming) and the activation status (activated or deactivated) of feature services per server in the cluster.
2-26
Cisco Unified Communications Manager Enterprise Parameters and Enterprise Phone Configuration
This topic describes the purpose of enterprise parameters, lists some of them, and shows how to
change them.
Enterprise Parameters
Used to define clusterwide system settings.
Apply to all devices and services in the same cluster. Afterinslallation, enterprise parameters are used to set initial
values of device defaults.
Enterprise parameters are used to define clusterwide system settings and apply to all devices and serv ices in the cluster. After installation, enterprise parameter default values should be
verified and modified if required before deploying endpoints. Some enterprise parameters will specify initial values of device defaults.
Note Change enterprise parameters only if you are completely aware of the impact of your
ixample of Enterprise
This parameter provides a
unique Identifier tot Ms cluster. Specifies ihe protocol with which auloregislered phones
St an dAioneC luster
Autoregistraion Phone
Protocol
should boot during hirealization Determines whether to display dependency records These parameters are used to
CCMUser Paramaers
Dependency records are a feature of Cisco Unified Communications Manager that allows an administrator to viev\ configuration database records that reference the currently displayed record. Dependencv records are useful when you want to delete a configuration entry (for
example, a device pool), but the deletion fails because the record is still referenced (lor example, bv an !P phone). Without dependency records, you would have to check each device, whether it uses the dev ice pool that you tried to delete.
2-28
Step 2
Note
2-29
.nterpnse Pa rami
tn
\ - T"-"
At the fnterprise Parameters Configuration web page, you will find enterprise parameters that are grouped into categories with the current configuration and the default value shown per
parameter.
2-30
it
j "-''-'
l3 &-"~ *i it-*'-*
I Change machine
names to IP address.
_]
Note
When removing DNSreliance, all hostnames within enterprise URL parameters have to be
changed to IP addresses.
2-31
configuration and Device Configuration settings. tf different parameters are set at multiple locations, the
setting that takes precedence is determined in the following order: Device Configuration
._ Common Phone Profile
Defined and activ ated hnterprise Phone Configuration parameters only have an effect on phone models that support the corresponding setting.
Parameters that vou set under Hnterprise Phone Configuration may also appear in the Common Phone Profile and the Device Configuration settings for various devices. If you set these same parameters in these other windows as well, the setting that takes precedence is dclennined in
the following order: 1. Device Configuration window settings
2. Common Phone Profile window settings 3. f.nterprise Phone Configuration window settings
page, choose System > Enterprise Phone Configuration. Only settings where the Override Common Settings checkbox is
checked take effect.
IrWUM
Modified parameter
v\
2-33
Service Parameters
Communications Manager allow you to configure parameters for different services. Examples of
service parameters ofthe Cisco Unified
* Defining Cisco Extension Mobility maximum login time * Defining voice media-streaming application codecs
Service parameters are used to definesettings for a specificservice for example, the callprocessing Cisco CallManager serv ice. They can be configured separately for each server in the cluster. After installation (or activation of feature services), service parameter default values should be verified and modified, if required, beforedeploying endpoints. The most important service parameters for the Cisco CallManager service are the following:
T302 timer: Specifies the interdig.it timer for variable-length numbers. Reducing the
default value will speed up dialing (shorter postdial delay).
CDR and (MR: CDRs and CMRs are the basis for call reporting, accounting, and billing,
fhe serv ice parameters are used to enable CDRs and CMRs.
Cisco Extension Mobility maximum login lime: After expiration of this timer, a user is logged out of Cisco l:\tension Mobility regardless ofthe idle time ofthe device.
Codecs of voice media-streaming applications
2-34
Description
Default Value
IM
False
30s
15s
False
(Cick Advanced
button first)
for PRI and enamel associated signaling (CAS) bterfacesin real thiefor
troubleshooting.
Bydefault, notall service parameters aredisplayed. To seethe complete listof service parameters, click the Advanced button. TheChange B-Channel Maintenance Status service parameter is anexample of a Cisco CallManager service parameter, which is notshown by
default.
From the Cisco Unified Communications Manager Administration page, choose System > Service
Parameters Select the server and choose the service.
Update the appropriate parameter settings. * Ifyou cannot find the parameter, click the Advanced button to display hidden parameters To save the changes, click Save.
To modify service parameters, perform the following steps in Cisco IJnitled Communications Manager Administration:
Step 1 CJo to System > Service Parameters.
Step 2
Step 3
Note
Select the server and the service for which you want to change service parameters.
Change the sen ice parameter values as desired and save the changes.
Ifyou cannot find the service parameter that you want to change, click Advanced to see the
complete list of available service parameters. By default, not all service parameters are
displayed
2-36
UHtlimM'*
*MWMBm '
5
Select server
_ 'tischcogJ - ^ _,
Irt Schemed -
3 T33l {Atfwe)
At the initial screen, you have to select theserver and the service forwhich you want to seeor
change the seniee parameters.
^TsaJe^nechanges. ^"3den_sen|ice_parameterS
-d
Service
parameter name
At the Sen ice Parameter Configuration web page, you will find service parameters that are grouped into categories with the current configuration and the defaultvalue that is shown per
parameter.
2-38
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Cisco Unified Communications Manager initial configuration includes network configuration, activation of feature services, and enterprise and service parameter configuration. Cisco Unified Communications Manager network configuration options include NTP configuration, DHCP server configuration, and using DNS versus IP addresses.
The Cisco Unified Communications Manager DHCP service is designed to serve IP phones.
Summary (Cont.)
Network services are automatically activated, while feature services are activated by the Cisco Unified Communications Manager Administrator. Enterprise parameters are used to define clusterwide system settings. Enterprise phone configuration is used to define global settings for phones supporting these parameters. Service parameters are used to configure parameters of specific services.
2-39
References
For additional information, refer to these resources:
2-40
Lesson 2
Manager
Overview
Cisco Unified Communications Manager includes several features that are related to user accounts, including end-userfeatures and administrative privileges. Cisco Unified Communications Manager user accountscan be managed using Cisco Unified Communications Managerconfiguration tools or by integrating Cisco UnifiedCommunications Managerwith a Lightweight Directory Access Protocol (LDAP)directory. This lessondescribes the typesof user accounts that arc used by Cisco Unified Communications Manager and how they can be
managed.
Objectives
Uponcompleting this lesson,you will be able to manageuser accounts including integrating
Cisco Unified Communications Manager with a corporate LDAP directory and enabling
multiple levels of user privileges. This ability includes being able to meet theseobjectives:
Identify the different user accounts in Cisco Unified Communications Manager and explain
how they are used
Describe how to add and delete users and how to assign authorization rights to them
Describe the purpose of Cisco Unified Communications ManagerBATand list its features
Describe how Cisco Unified Communications Manager BAT can be used to manage users Identify LDAP characteristics and list the types of LDAP support that are provided by
Cisco Unified Communications Manager
Cisco Unified Serviceability Cisco Unified Operating System Administration Disaster Recovery System CDR Analyzing and Reporting (CAR) Cisco Unified Communications Manager applications: Cisco Extension Mobility Cisco Unified Communications Manager Assistant
Directories
Several Cisco I nitied Communications Manager features require u.ser accounts to be able to
authenticate the user. These features include administrative and user web pages and applications that require the user to log in. such as: Cisco Intension Mobility Cisco Unified Communications Manager Assistant
Cisco IP phones can browse directories to find the directory number for a given useniame. To be able to provide this infonnation. Cisco Unified Communications Manager needs to know
users and their extensions.
When using Cisco CallManager Cisco IP Phone Services, the services can be configured to require a user login before providing access to the service. Users can authenticate with their useniame and a password (alphanutneric) or PIN (numeric), depending on the application. Cisco Unified Communications Manager sends authentication requests to an internal library, the Identity Management System (IMS) library, which is responsible for authenticating the credentials against the embedded database (by default).
2-42
Application Users
Associated with an individual person
administrator logins
Included in user directory
Cannot usei_DAp
The two types of useraccounts in Cisco Unified Communications Manager areas follows:
End users: All end users are associated with a physical person and an interactive login.
This category includes all IPtelephony users as well asCisco Unified Communications Manager administrators when using the usergroups and rolesconfigurations.
Application users: All application users areassociated with Cisco Unified Communications features or applications, such as Cisco Unified ContactCenter Express or Cisco L'nified Communications ManagerAssistant. These applications need to authenticate with Cisco Unified Communications Manager, but these internal "users" do not have an interactive login andserve purely for internal communications between applications.
CCMSysUser
IPMASecureSysUser, IPMASysUser
Manager Assistant
Cisco WebDialer Web Service
WDSecureSysUser, WDSysUser
2-43
The attributes that are associated with end users are separated into three categories and include the following information:
Personal and organizational settings:
User ID. First. Middle, and Last Name
Password
Application and feature parameters (for example. Cisco Lxlension Mobility profile.
Presence Group. Mobility. Certificate Authority Proxy Function (CAFF), and so on)
Note
Application users are associated with a subset of these attributes, which are the ones that
are shown in italics
2-44
User Privileges
Cisco Unified Communications Manager allows the assignment of user privileges to
application and end users.
User Privileges
Privilegesare assigned to applicationusers and end users.
Privileges include these accesses:
- Access to user web pages. - Access to administration web pages.
User privileges include these configuration elements' - User groups {a listof applicationand end users). - Roles (a collection of resources for an application).
Each role refers to one application.
Each application has one or more resources (static list). Per role, access privileges are configured per application
resource.
Access to application interfaces, suchas computer telephony integration (CTI) and Simple
Object Access Protocol (SOAP)
User priv ileges are configured using two configuration entities as follows:
Each role refers to one application, andeachapplication hasoneor moreresources (static list per application). Perrole, access privileges are configured per application resource. Roles arc
assigned to user groups.
2-45
r Privil
Users n " User Groups ' n Roles n 1 Applications 1 . 1 Privileges
App)scation2
Resource! < ifad
RBSOurce2'
<Kf
RBSOurC93
(none)
Resource*
'ea"-"P"*"1
The diagram shows four users (User I to User4) and two user groups (Group1 and Group2). Lserl and Lser2 arc assigned to Group!: User3 is assigned to both groups; and Uscr4 is
assigned to Group2.
Ihere are tliree roles (Rolel to Role3). Rolel is assigned to Group I: Role2 is assigned to both
groups: and Role3 is assigned to Group2.
Rolel and Rolc2 both refer to Application!. Application! has three application resources (Resource! to Resources). Rolel and Role2 have different privileges that are assigned to resources of Application!. Role? refers to AppHcation2 and has privileges that are assigned to the four application resources (Resource! to Resourcc4) of Applieatton2.
2-46
with read-only access to Cisco Unified Communications Manager Administration Solution: Two user groups and two roles
User Group
Role
* Application
Resource
Cisco Unified
Communications Administration Cisco Unified Communications
AAR Group
web pages
Cisco Unified
CM Group
web pages
DRF Show
only
User "Kim Lu" User "Tom Adams"
Standard
CCMADMIN
Manager
Administration
Read-Only
Status page
In the example, the goalis to have administrators who havecomplete access to all configuration pages of Cisco Unified Communications Manager Administration and
administrators who have read-only privileges to these configuration pages.
TheCisco CallManager Administration (thatis, CiscoUnified Communications Manager Administration) application has webpages that areassociated with a function, such as:
Call Park web pages (used to configure the Call Park feature)
Disaster Recovery Framework (DRF) Show Status page (used to check the status of
disaster recovery system backup or restore jobs)
These web pages are application resources ofthe Cisco CallManager Administration
application.
Cisco Unified Communications Manager has standard roles (that is, roles that exist bv default), which are associated with the Cisco CallManager Administration application, such as:
Role Standard CCMADMIN Administration
"The first role has all application privileges set to "update," while in the second role, all application privileges are set to "read."
2-47
Cisco Unified Communications Manager hasseveral standard usergroups, including user group "Standard CCM Super Users" and usergroup "Standard CCM Read-Only." User group.
Standard CCM Super Users, is associated with role Standard CCMADMIN Administration, and user group Standard CCM Read-Only is associated with role Standard CCMADMIN ReadOn I\.
Ba^ed on the prev iously mentioned default roles and usergroups, in order to assign complete access to all configuration pages of Cisco Unified Communications Manager Administration to an end user, the end user has to be assigned to the standard user groupStandard CCM Super Users. Lnd users uho should have read-only access to all configuration pages of Cisco Unified Communications Manager Administration have to be assigned to the standard user group Standard CCMADMIN Read-Onlv. No further configuration is required, because the appropriate application privileges are preconfigured in the default roles, and the default roles are preassigned to the corresponding default user groups.
Note Cisco Unified Communications Manager has numerous default user groups (24 in Cisco Unified Communications Manager Version 8 0), which cover the needs for the most typical requirements. Examples of these default user groups are the previously mentioned Standard CCM Super Users and Standard CCMADMIN Read-Only user groups as wet as the other
user groups, such as "Standard CAR Admin Users," "Standard CCM Server Maintenance."
"Standard CCM Server Monitoring," "Standard CCM Phone Administration," "Standard CCM End User," and "Standard CCM Gateway Administration "
- LDAP authentication
For user authentication
User management options in Cisco Unified Communications Manager include the following: Using Cisco Unified Communications Manager Administration, User Management menu items: Thisoption is suitable forconfiguring a fewusersor doingsingle updates to the configuration. It does not scale for mass deployment of users. Using CiscoUnified Communications Manager Bulk Administration Tool (BAT):
Cisco Unified Communications Manager BAT allows bulk administration of several
configuration elements, including users. Cisco Unified Communications Manager BAT is a good option for initial (mass) deployment when LDAP integration is not used. LDAP integration: This option is available onlyto end users. LDAP integration provides
two functions, which can be enabled independent of each other:
LDAP synchronization: Allows user provisioning where personal and organizational data aremanaged in an LDAP directory and replicated to the Cisco
Unified Communications Manager configuration database.
LDAP authentication: Allows user authentication against an LDAP directory.
When using LDAP authentication, passwords are managed in LDAP. (LDAP authentication requires LDAP synchronization.)
2-49
Examples.
Microsoft Active Directory, iPlanet. Sun ONE, OpenLDAP Cisco Unified Communications Manager supports two types of integration
- LDAP synchronization
LDAP synchronization and LDAP authentication
When using LDAP. some user data is no longer controlled via Cisco Unified Communications Manager Administration.
LDAP directories are services that store user infonnation in a specialized database, fhe
database is optimized for a high number of reads and searches, and occasional writes and
updates. Directories tvpieally store data that do not change often, such as employee infonnation. user privileges on the corporate network, and so on. The LDAP provides applications with a standard method for accessing and potentially modifying the information that is stored in the directory. This capability enables companies to centralize all user infonnation in a single repository that is available to several applications. This also results in a remarkable reduction in maintenance costs through the ease of adds, moves, and changes.
Lxamples for LDAP directories are Microsoft Active Directory, il'lanet or Sun ONI! LDAP Server, and OpenLDAP or Microsoft Active Directory Application Mode. Cisco Unified Communications Manager supports two tvpes of integration: LDAP synchronization and LDAP authentication. When using LDAP. some user data is not controlled by Cisco Unified Communications Manager Administration web pages.
Authentication
Local
LDAP (replicated
LDAP(replicated
to local)
to local)
Local
Locai
LDAP
Local
Local
Locat
As shown in the table, without LDAP integration, all end-user data is stored in the Cisco Unified Communications Manager database and configured via Cisco Unified Communications Manager Administration.
Note Application user data is always controlled by Cisco Unified Communications Manager
Administration and stored in the Cisco Unified Communications Manager database.
Whenusing LDAP synchronization, personal and organizational settingsare configured and stored in LDAP. With each synchronization, the data is replicated to the Cisco Unified Communications Manager database. However, as long as LDAP synchronization is enabled,
this data cannot be modified in Cisco Unified Communications Manager. User passwords and Cisco Unified Communications Manager configuration settings are still configured using Cisco Unified Communications Manager Administration and stored in the Cisco Unified Communications Manager database only.
When using LDAP authentication, personal and organizational settings are also controlled by LDAP. because LDAP synchronization is mandatory. User passwords, however, are configured and stored in LDAP only. The passwords arc not replicated to the Cisco Unified Communications Manager database. To store the password for a Cisco Unified Communications Manager user in LDAP (the user has to exist in the Cisco Unified Communications Manager database so that Cisco Unified Communications Manager settings can be configured for the user), the user has to exist in both databasesthat is, in LDAP and in
the Cisco Unified Communications Manager database.
2-51
Management
Requires sufficient privileges1
* Use master administrator
CCKAdministrafor
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Bui MmmtnSos -
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User Group
Cisco Unified Communications Manager user management is perlonned from Cisco Unified
CM Administration > I'scr Management. To be able to manage users, the administrator needs to use an account that has sufficient privileges. It can be the default administrator account, which is created during Cisco Unified Communications Manager installation, or any end-user account that has the user management privilege assigned.
The user management menu includes options to configure application users, end users, roles, and user groups.
2-52
The most important settings are the User ID and the Password.
^^^^^nSSWmmWmmmmmmmmMJSOm
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to user groups
View roles of
application user
At the bottom ofthe Application User Configuration page, the application user can be added to user groups. The roles that are assigned to the user groups, of whichthe application user is a
member, are displayed in the Roles list box.
2-53
X~ &mm.
E<M f rerfflntpfll
The Lnd User Configuration screen is like the Application User Configuration screen, fhe User ID. Password, and Group Membership (not shown in the screenshot) are the most important
settings.
2-54
Roles
Cisco Unified Communications Manager includes standard rolesas shown in the figure.
Roles
Standard (default) roles exist; standard roles cannot be
deleted.
Custom roles can be created by adding new roles or by copying and then modifying a standard role.
Standard roles cannot be deleted or modified. Custom roles can be created from the beginning
2-55
Configured
privilege per
=^
application
resource
As shown in the figure, an application lias to be selected on the Role Configuration page. After selecting an application, the application resources are displayed and read or update privilege
can be assigned to each application resource.
2-56
User Groups
Cisco Unified Communications Manager includes standard usergroups as shown in the figure.
User Groups
Standard (default) user groups exist; standard user groups
cannot be deleted.
ttfTHTJtUl ITf
*
IS 3
6
G B
Standard user groups cannot be deletedor modified. Customuser groups can be created from the beginning or by copying and then modifying a standard user group.
2-57
>
"
B
Scad A* Dup A.I I *0a 4*J*3d
As shown in the figure, application and end users can be assigned to the user group on the User
Group Configuration page.
2-58
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-">-*"
ita'**,-..*
..-.,..,..
'
To assign roles to a user group,choose the Assign Role to User Group item from the Related
Links list box at the User Group Configuration page. A new window, in which you can assign
or delete roles, will be displayed.
2-59
Credential Policies
This section describes the principles and configuration of credential policies.
A credential poliev defines password requirements and account lockouts for user accounts. Credential policies that arc assigned to user accounts control the authentication process in Cisco Unified Communications Manager. After a credential policy has been added, that new poliev can be assigned as the default poliev for a credential type or to an individual application or end
user.
PINs and end-user passwords. 1he poliev contains settings for failed login resets, lockout
durations, expiration periods, and credential requirements. The Credential Policy Configuration
window allows configuration of new credential policies for the system or site.
Passwords can contain am alphanumeric ASCII character and all ASCII special characters. A nontrivial password meets the following criteria:
Must contain three ofthe four allowable characteristics: uppercase character, lowercase character, number, and symbol Must not use a character or number more than three times consecutively Must not repeat or include the alias, useniame. or extension Cannot consist of consecutive characters or numbersfor example, passwords such as
654321 orAUCDFFG
PINs can contain digits (Olo 9) onlv. A nontrivial PIN meets the following criteria: Must not use the same number more than two times consecutively Must not repeat or include the user extension or mailbox or the reverse ofthe user
extension or mailbox
Must contain three ditferent numbers: for example, a PIN such as 121212 is trivial
2-60
Must not match the numeric representation (that is, dial by name) for the first or last name
ofthe user
Must not contain groups of repeated digits, suchas 408408. or patterns that arc dialed in a
straight line on a keypad, such as 2580, 159,or 753
2-61
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s**oai
:.:~ctwAi ;*QeisnStrata
iXl
-
Credential PoIkv
AddNeiv
SeledAU
Clear All
Delete Seleaed
Credential Policy
After installation of Cisco Unified Communication Manager, one default credential policv i; applied to all end users and application users.
Note
2-62
<='-
can be modified.
Even though the default credential policy cannot be deleted, it allows editing of its parameters.
2-63
4. Save policy
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\-j"
^BjIv'!- i'*fW*tp'^lPfc'^-
Display Name
Enter a number in the range from 1 to 100. To allow unlimited failed logins, enter 0 or check the No Limit for Failed Logons
check box Uncheck the check box to enter a value greater than
Specify the number of minutes before the counter is reset for failed login attempts. After the counter resets, the user can try logging in again Enter a number in the range from 1 to 120 The default setting specifies 30
Description
Credential Changes
Specify the number of minutes that are required before a user can change credentials again. Enter 0 to allow a user to change credentials at any time. Uncheck the check box to enter a value greater than 0. The default setting specifies 0.
option for low-security accounts or multiple user accounts, for example. The default setting specifies 180.
Minimum Credential Length
Specify the number of previous user credentials to store. This setting prevents a user from configuring a recently used
credential that is saved in the user list.
Enter a number in the range from 0 to 25. If no previous credentials should be stored, enter 0. The default setting specifies 12. Inactive Days Allowed
Specify the number of days that a password can remain inactive before the account gets locked.
Enter a number in the range from 0 to 5000. The default setting
specifies 0.
Enter a number in the range from 0 to 90 to specify the number of days before a user password expires to start warning
notifications. The default setting specifies 0.
Check this check box to require the system to disallow
credentials that are easily hacked, such as common words, repeated character patterns, etc. The default setting checks the check box.
2-65
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Custom credential policies can be assigned to three default credential policy groups as follows: Default Credential Policy for end users defining password credential rules Default Credential I'oliev for application users defining password credential rules Default Credential Policy for end users defining PIN credential rules
2-66
Q'
Assign Credential Policy
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Credential Poltsy Defi* Information
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Credential Type
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Credential Pokey'
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Select the configured
custom Credential Policy.
S*ve
Select a custom policy from the drop-down menu and save the changes.
2-67
To assign a custom policv to an end user orapplication user, open the i-nd I'ser or Application
user window and click the F.dit Credential button.
Selecta custom policv from the drop-dow n menu and save the changes.
Manager BAT
Cisco Unified Communication Manager BATallows
Cisco Unified Communications Manager BAT allows mass configuration ofCisco Unified Communications Manager configuration items, including users, phones, directory numbers,
gateways, and so on.
2-69
1Exports data (phones, users, gateways, etc.): Exported files can be modified and reimported. Exportand import of complete configuration (tar archive)
is possible.
Integrated with the Cisco Unified Communications Manager Administration pages and available by default {no plug-in
required) Supports localization
Note
The import and export function of Cisco Unified Communications Manager BAT can be used
to move data records from one Cisco Unified Communications Manager cluster to another.
Integrated with the Cisco Unified Communications Manager Administration pages and
available by default (no plug-in required)
Supports localization
The Cisco Unified Communications Manager Auto-Register Phone fool is also available from the Bulk Administration menu but requires additional products.
2-70
Cisco Unified Communications Manager BAT has its own main menu in Cisco Unified
Communications Manager Administration.
As shown in the figure. Cisco Unified Communications Manager BAT menu items include the
following options:
2-71
Cisco Unified Communications Manager BAT templates are used to define general settings that
fit all the dev ices that should be added. Comma-separated values (CSV) files arc used to define specific settings per device that should be bulk-configured. Adding, updating, and deleting devices and records is initiated from the Cisco Unified Communications Manager Administration BAT menu, based on Cisco Unified Communications Manager BA I
configuration requests that are referring to BAT templates and BAT CSV files, Cisco Unified Communications Manager BAT jobs can be executed immediately orscheduled for a later
time.
Cisco {inified Communications Manager BAT can he used to work with the following types of
devices and records:
Add. update, and delete IP phones including voice gateway phones. CTI ports, and H.323
clients,
Migrate phones from Skinnv Client Control Protocol (SCCP) to Session Initiation Protocol
(SfP),
Add. update, and delete Cisco Unified Communications Manager Assistant and niaiumer
associations.
Add. update, and delete ports on a Cisco Catalvst 6000 family fXS Analog Interface
Module.
Add ordelete Cisco VG200 and Cisco VC1224 analog gateways and ports.
Note
The Cisco Catalyst 6000/6500 WS-X6624 and Cisco VG200 products have reached end of
life(EOL).
Update or export Cisco Unified Presence or Cisco Unified Personal Communicator users.
Populate ordepopulate the Region Matrix.
Insert, delete, or export the access list.
2-73
Cisco Unified Communications Manager BAT utilizes adedicated feature service, the Bulk Provisioning Serv ice (BPS). for maintaining and administering submitted Cisco Unified
Bulk Provisioning
BPS administers and maintains all jobs that are submitted
BPS has to be activated onlyon the Cisco Unified Communications Manager publisher.
fhe BPS is activated from Cisco l'nified Seniceability> 'looks > Service Activation. It is
required for executing submitted Cisco Unified Communications Manager BAT jobs. The BPS
has to be activated on the Cisco Unified Communications Manager publisher server only.
2-74
Ti
Deactivated
DCtHJtKl
CMtCUVIMcl DMCBvated
Activate BPS.
JUB*ea
The figure shows the BPS being activated on the Service Activation page ofCisco Unified
Serviceability.
2-75
The Cisco Unified Communications Manager BAT configuration procedure includes these steps:
Step 1.Configure a Cisco Unified Communications Manager
BATuser template.
Step 2: Create the CSV data inputfile. Step 3' Upload the CSV data input fie.
Step 5: Verify the status of the Cisco Unified Communications Manager BATjob.
Stepl
Step 2
Create the CSV data input file. This file includes the users to be added to the
configuration database, for each user, ihere will be one record containing all settings
ofthe corresponding user.
Step 3
Step4
Upload the CSV data input file. 'IheCSV file needs to beuploaded to the Cisco
Unified Communications Manager publisher server.
Start the Cisco Unified Communications Manager BAT jobto add users.
Step 5
Verifv the status ofthe Cisco Unified Communications Manager BAT job.
2-76
The figure shows the Cisco Unified Communications Manager BAT User Template
Configuration page.
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ir luilm Caaftrtaa
fineirr*
template name.
~
Configure default
user parameters.
Aname for the phone template has to be configured, and the default user configuration
usemame in the data CSV file.
parameters have to be selected. These default values can be overwritten with specific values per
2-77
Cisco provides a template to create CSV files that have the mandatory format to work with Cisco Unified Communications Manager 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 texteditor, such as
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.
fhe CSV file has to beina special format and has to include specific values. Therefore, it is
recommended to createthe CSV file by using a Microsoft fxcel macrothat can be downloaded
from the Cisco Unified Communications Manager server. Use the Upload/Download Files
menu item in the Bulk Administration menu to download the file. The Hxeel macro will allow
vou to enter the configuration data in a spreadsheet and then save the data in the appropriate CSV format. Alteniativelv. you can create the CSV file on your own as long as you use the correct sequence ofconfiguration parameters (separated by a comma). Make sure that you
follow these rules when creating a CSV file on yourown:
(. ise a separate line to enter data for each record,
Separate each data field with a comma and include comma separators for blank fields. Donot enterblank lines: otherw ise. errors occur during the insert transaction.
To create the CSV file using the Microsoft Excel template, make sure that macros are
enabled within Excel
Note
2-78
You have to specify the local file, the configuration target {users, phones, gateways, and so on),
and the transaction type (add. delete, or update).
Note
At this time, you only uploaded the CSV file. The selected transaction type will not be
executed unless you proceed with the next step.
2-79
3. Select immediately or
queuejob and start later or configure start time
To start a Cisco Unified Communications Manager RA 1job for adding users, go to Cisco
Unified CM Administration > Bulk Administration >Users > Insert Users. Atthe Insert Users configuration page, perform the following actions:
Select the user template {which you created in Step I). Select the CSV file (which you created and uploaded in Steps 2 and 3).
Specify to either run the job immediately or to run fhe joblater.
It'you choose the option to run the job later, you will have to configure the start time using the
Job Scheduler.
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To access the Job Scheduler, go toCisco Unified CM Administration > Bulk Administration >Job Scheduler. Tlie Job Scheduler provides a list ofjobs, displays the status
ofthe jobs, and allows configuration ofthe start time for scheduled jobs.
2-81
When clicking ajob ID from the list of Cisco Unified Communications Manager BAT jobs thai
tep 5b:
*"lsw ***tSSSEESBSBEM
9:
2. Click to open
log file.
Ihe job details include infonnation about thejob result, the number of records that are
processed, and the number ofrecords that failed. Ifyou want to sec more details for example,
if yourjob had errors click the log filename.
2-82
LDAP Overview
This topic describes LDAP directory' services.
LDAP Characteristics
LDAP directories typically store data that do not change often,
such as employee information.
LDAP directories typically store data that docs not change often, such as employee
information, user privileges onthe corporate network, and soon. The information isstored in a database that isoptimized for a high number ofread and search requests and occasional write and update requests. LDAP directories store all user infonnation in a single, centralized repository that isavailable
to all applications. Applications can access the directory using the LDAP, providing astandard
method for reading and potentially modifying the infonnation that is stored in the directory.
2-83
Cisco Unified Communications Manager can integrate with LDAP directories to benefit from a
^|
User
"PF^bBqw
Provisioning
Authentication
IPTfetephony&KlUseiJ
Integration between voice applications and a corporate LDAP directory ts a common task for many enterprise IT organizations. 1lowever. the exact scope ofthe integration varies from
company to company, and it can translate to oneor more specific andindependent
requirements.
For example, one common requirement isto enable user lookups (sometimes called the "white
pages'" sen ice) from IP phones so that users can dial acontact directly after looking up its
number in the directory.
Another requirement istoprovision users automatically from the corporate direetorv into the
user database ofunified communications applications. This method avoids having to add.
remove, or modify core user information manually each time that a change occurs in the
corporate directory.
applications that are using the corporate directory credentials is also required. Ihis method enables the IT department to deliver single login functionality and reduces the number of passwords that each user needs tomaintain across different corporate applications.
ACisco Unilied Communications system can satisfy each ofthese requirements using different
mechanisms according tothe Cisco Unified Communications Manager version thai is used.
Cisco Unified IP phones that are equipped with adisplay screen can search auser directory when auser presses the Directories button on the phone. The phones use 11 IIP to send requests to a web server. The responses from the web server must contain some specific XML objects
that the phone can interpret and displav.
Bv default Cisco Unified IP phones are configured to perform user lookups against the
change this configuration so that the lookup is performed on acorporate 1 .DAI drrcctory. In
this case the phones send their HTTP requests to an external web server that operates as a
phones via IfffP.
proxv and translates these requests into LDAP queries against the corporate directory. Inc LDAP responses are then encapsulated in the appropriate XML objects and sent back to the
Communications Manager
Supported directories.
Microsoft Active Directory Application Mode 2003 Microsoft Lightweight Directory Services 2008
iPlanet Directory Server 5.1
Cisco Unified Communications Manager supports two types ofLIMP integration, which can
be enabled independent of each other:
LDAP synchronization: Allows user provisioning where personal and organizational data
are managed in an LDAP directory and replicated to the Cisco I Inified Communications
Manager configuration database.
against an LDAP directory. When using LDAP authentication, passwords are managed in
LDAP.
2-86
Full synchronization:
- Microsoft Active Directory
- OpenLDAP
All synchronization agreements must integrate with the same LDAP family (Active Directory, iPlanet orSun ONE, OpenLDAP or Active
DirectoryApplicationMode).
Application users are not affected by LDAP integration. They are always configured from
Cisco Unified Communications Manager configuration database.
Cisco Unified Communications Manager Administration, and their data is always stored in the
Depending on the directory server that is used, LDAP synchronization is performed in one of
the following ways:
Full synchronization: This method is used with Active Directory versions. Full
svnehronization means that all records are replicated from the LDAP directory to the Cisco Unified Communications Manager database. In large deployments, this method can cause considerable load: therefore, synchronization times and jobs have tobe carefully selected. Incremental synchronization: This method is used with all other supported directory servers. Because only changes are propagated to the Cisco Unified Communications
Manager database, this method requires fewer resources than the full synchronization
method.
synchronization agreements with Active Directory and any other LDAP server.
CiscoUnified Communications Manager uses LDAP version 3.
One LDAP user attribute (for example, sAMAccountName, uid, mail, ortelephoncN umber) has
to be mapped to the User ID field ofauser in Cisco Unified Communications Manager and
must be unique across all users.
2-87
Users are added ordeleted in ihe LDAP directory. " All personal and organizational user data are configured in LDAP Users and their personal and organizational data are replicated from LDAP to Cisco Unified Communications Manager. These
parameters are read-only in Cisco Unified Communications Manager Administration.
'I his process uses a service called directory synchronization (DirSync)on Cisco Unified Communications Manager to sy nchroniz.c anumber ofuser attributes (either upon request or
periodically ) from a corporate LDAP directory , When this feature is enabled, users are automatically provisioned from the corporate directory.
When using this feature, end userscannot be added or deleted from Cisco Unified
Communications Manager Administration. They are added and deleted in the LDAP directory,
and all personal ororganizational sellings that are associated with the users are configured in'
LDAP.
Users and their associated personal and organizational data are replicated from LDAP to Cisco Unified Communications Manager. These parameters are read-only in Cisco Unified
Communications Manager Administration. User passwords andCisco Unified Communications
Manager settings are still configured from Cisco Unified Communications Manager Administration and are stored only in the Cisco Unified Communications Manager database.
Therefore, these settings cannot be configured in LDAP.
LDAP
Authentication
B- LDAP (replicated
Local
to local)
LDAP
Local
Controlled Devices
When LDAP synchronization is enabled, the User ID, First, Middle and Last Name, Manager User ID. Department. Phone Number, and Mail ID values can only bedefined on the LDAP sener and appear read-only onthe Cisco Unified Communications Manager End User web
page.
configured from Cisco Unified Communications Manager Administration or Cisco Unified Communications Manager
user web pages.
Users and their persona! and organizational data are still stored in the Cisco Unified Communications Manager local
database'
With LDAP authentication. Cisco Unified Communications Manager authenticates user credentials against a corporate 1. DAPdirectory. When this feature is enabled, end-user
passwords arcnot stored inthe Cisco Unified Communications Manager database anymore (and arealso not replicated to that database) butareonly stored inthe LDAP directory.
Personal user data is also managed in LDAP and replicated into the Cisco Unified
Manager database (to assign Cisco Unified Communications Manager usersettings to the user) and inthe LDAP directory (toassign the password to the user). Toavoid separate management of user accounts in these two databases. LDAP synchronization is mandatory with LDAP
authentication.
2-90
integration
Personal and organizational
settings:
Synchronization'
Local
LOW3(replicated to local)
Password
Cisco Unified Communications
Local
Local
Manager Settings:
Local
Local
When LDAP authentication is enabled, onlythe following settings canbe managed locally:
PIN and Digest Credentials
Groups and Roles
Associated PC's
Controlled Devices
Extension Mobility Profile and CAPF Presence Group and Mobility All other settings must be managed at the LDAP server.
2-91
Summary
This topic summarizes the key points thatwere discussed in this lesson.
Summary
1Cisco Unified Communications Manager has application
users and end users
Application and end users can be configured one-by-one using Cisco Unified Communications Manager
Administration.
Cisco Unified CommunicationsManager BAT allows bulk configuration of users, phones, and other configuration
entities.
LDAP directories are centralized storage of user information. Cisco Unified Communications Managercan integrate with
LDAP for user provisioning and authentication.
References
For additional information, refer to these resources:
Cisco Systems. Inc. Cisco ( nified Communications Manager Administration Guide. Re/ease 8.Of I/. San Jose. California. February 2010. Imp:-www.cisco.com eir'l'S dot*.\nice ip_comm/eucm/drs/8 0 l/drsagXOI.himL
Cisco Sy stems. Inc. ('isco I. nified Communications Manager System Guide Release 8.0(1).
San Joe. California. February 2010. http:.''wwu.^iveo.C(im.en,l S docs \oice ip comnrcuciiMidniin/S I) l/eem-,\>'iii.Tm-K()lem.htnil.
Cisco Sy stems. Inc. (. isco !. nitied ( ommwiications System Re/ease 8.x SRMX San Jose.
California. April 2010.
2-92
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Initial configuration ofCisco Unified Communications Manager
includes service activation, general configuration of service,
This module describes CiscoUnified Communications Manager Services activation and initial
configuration parameters. Further, the module describes the user-management options that are
available in Cisco Unified Communications Manager.
References
For additional infonnation. refer to these resources:
http://w^vw.cisco.com/en/US/docs/voice_ip_c()mm/cticm/drs/8J)_l/drsag801.htnil.
Cisco Systems. Inc. Cisco Unified Serviceability Administration Guide, Release 8.0(1).
California. May 2009.
http:. www.ciseo.com/en/US/docsAoice ipconim/cucm/drs/8 0 1/drsagKOI .htriil. Cisco Systems. Inc. Cisco Unified Communications Manager System Guide Release 8.0(1).
San Jose. California. February 2010.
Cisco Systems. Inc. Cisco Unified Communications System Release 8.x SRf\D. San Jose,
California. April 2010.
Cisco Systems. Inc. Cisco Unified Communications Manager Bulk Administration Guide.
Release 8.0(1). San Jose, California, February 2010.
cm.htinl
) 2010 Cisco Systems, Inc
2-94
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found inthe Module Self-Check Answer Key.
Q1)
Which two options are not initial configuration steps? (Choose two.) (Source: Managing Services and Initial Configuration of Cisco Unified Communications
Manager)
A)
B)
C) D)
E)
Q2)
Which is not anetwork configuration option ofCisco Unified Communications Manager? (Source: Managing Services and Initial Configuration ofCisco Unified
Communications Manager)
A) B)
HSRP
NTP
DNS DHCP
C) D)
03)
Cisco Unified Communications Manager Version 8.0 provide IP phones with IP addresses by DHCP. (Source: Managing Services and Initial Configuration ofCisco
Unified Communications Manager)
A) B) C) D)
has to
cannot
can
subscribers
Q4)
What must be done toremove DNS reliance? (Source: Managing Services and Initial
Configuration of Cisco Unified Communications Manager)
A)
B)
C)
D)
05)
Which services cannot be activated ordeactivated by theadministrator? (Source: Managing Services and Initial Configuration ofCisco Unified Communications
Manager)
A)
B) C) D)
enterprise services
cluster-wide services network services feature services
2-95
06)
AI B| C) D)
1hey are used to define cluster-wide system settings. Areload is required after changing any of them. They apply to all devices and are configured per services. They allow the configuration of IP phone URLs.
F) 07}
C) D) OS)
Which two options are features that do not interact with Cisco Unified
Which two configuration elementsare used to assign privileges to users?(Choose two.) (Source: Managing User Accounts in Cisco Unified Communications Manager)
A)
B)
functional groups
roles
C) D) P.)
QIO)
Which function is not performed by the Cisco Unified Communications Manager Bulk
Administration 'fool? (Source: Managing User Accounts in Cisco Unified Communications Manager) A) B) C) D) F) F| adding or deleting a large number of similar records
exporting data records
updating a large number of similar records exporting the Cisco Unified Communications Manager configuration
importing data records converting SIP phones to SCCP
011)
Which option is not a step of adding users with the Cisco Unified Communications Manager Bulk Administration Tool? (Source: Managing User Accounts in Cisco Unified Communications Manager) A)
B) C)
verifying the status ofthe Cisco Unified Communications Manager BAT job
uploading a user template uploading a CSV data input file
D)
2-96
Ql 2) Which two choices are the supported LDAP integration options? (Choose two.) (Source: Managing User Accounts in Cisco Unified Communications Manager)
A) B) C) D) L) LDAP synchronization LDAP replication LDAP authentication LDAP authorization LDAP distribution
2-97
Q- 0-m
Q(>)
C A. D
\
li. 1Li. C V
li
012)
A. (.'
Module 3
This module describes theendpoints that are supported by Cisco Unified Communications
Manager, as well as their characteristics, protocol, and feature support. The module also
describes how Cisco Catalyst switches can provide power to endpoints and support VLAN
separation for voice and data traffic. Tlie module also explains how toimplement Cisco IP and
third-party phones using thedifferent protocols.
Module Objectives
Upon completing this module, you will be able toconfigure Cisco Unified Communications Manager tosupport on-cluster calling. This ability includes being able tomeet these objectives:
Describe thegeneral features and unique characteristics ofthe H.323. SCCP, and SIP endpoints thataresupported by Cisco Unified Communications Manager
Implement SCCP and SIP (Cisco and third-party) phones inCisco Unified
Communications Manager
3-2
Lesson 1
Understanding Endpoints in
Cisco Unified Communications
Manager
Overview
An important task in implementing and supporting a Cisco Unified Communications deployment is managing theend-user devices, orendpoints. It is important to beable to distinguish between various Cisco Unified Communications end-user devices thatyou may encounter during the course of deploying and administering a Cisco Unified Communications
network. In addition, understanding the boot and registration communication betweena Cisco
IPphone and Cisco Unified Communications Manager is important for understanding normal
voice networkoperations and for troubleshooting purposes. This lessondescribes the variousmodelsof Cisco IP phonesand how they work within a Cisco
IP telephony solution. The lesson also introduces the IP phone power-up and registration process and describes third-party Session Initiation Protocol (SIP) and H.323 endpoints.
Objectives
Upon completing this lesson, you will be able to describe thegeneral features and unique
characteristics ofthe H.323. SCCP,and SIP endpoints that inlerwork with Cisco Unified Communications Manager. This ability includes being able to meet these objectives:
Describe how Cisco Unified Communications Manager supports third-party SIP IP phones
in
Cisco SIPOnly
Phones
A varietv of endpointsCisco as well as third-party productscan be used with Cisco Unified Communication^ Manager. Endpoints include IP phones, analogstationgatewayswhich allow analog phones to interact with Cisco Unified Communications Managerand video
endpoints.
Cisco Unified Communications Manager supports three protocols to be used for endpoints:
Skinm Client Control Protocol (SCCP). SIP. and 11.323.
3-4
If no local voice VLAN ID is configured, the Cisco IP phone sends out aCisco
Discovery Protocol frame with a VoIP VLAN query.
If the Cisco Catalyst switch has a voice VLAN configured, it will send out a Cisco Discovery Protocol frame with the voice VLAN ID for the Cisco IP phone.
Cisco Unified Communications
DHCP
Manager
Cisco TFTP
When connecting to the VoIP network, the Cisco IP phone goes through astandard startup
these steps may not occur on your Cisco IP phone:
process consisting of several steps. Depending on your specific network configuration, some of
Step 1 Obtaining power from Ihe switch: The Cisco IP phone obtains power from the
switch Two methods ofproviding power via aswitch port are available: Cisco
prestandard Power over Ethernet (PoE) and IEEE 802.3af PoE. Initially, the switch provides the maximum power that is available depending on the used PoE method.
Alternatively, wall power or an in-line power injector can power the IP phone.
Step 2
memory in which the phone firmware image is stored. At startup, the phone runs a
bootloader that loads the phone image from flash memory. Using this image, the
phone initializes its software and hardware.
Loading the stored phone image: The Cisco IP phone has nonvolatile Hash
Step 3
Voice VLAN configuration (IP phone): Cisco IP phones can use IEEE 802. lQ VLAN tagging to differentiate voice traffic from data traffic of aPC that is attached to the PC port ofthe phone. The voice VLAN ID can be configured locally at the Cisco IP phone or at the Cisco Catalyst switch. If no voice VLAN is configured
Discovers Protocol message that includes a VoIP VLAN query, 'fhis Cisco
real power demand oftheCisco IPphones.
localh the Cisco IP phone is requesting the voice VLAN ID by sending out aCisco
Discovery Protocol message also includes the required power for the used phone
model, which allows the switch to possibly reduce the supplied power to match tlie
3-5
Step 4
\ o.ce \ LAN configuration (snitch): Ifavoice VLAN ID is configured on the switch. ,t will respond to the received Cisco Discovery Protocol message and inform
send out two more Cisco Discovery Protocol messages asking for the voice VI. AN ID before it will continue the boot process. This results in longer boot times ifno unee VLAN ,s configured on the switch. The switchport voice vlan untagged command will instruct the switch to respond with aCisco Discovery Protocol
message to speed up the phone boot process.
Protocol message. If no voice VLAN is configured on the switch, it will not respond uith aCisco Discovery Protocol message. In this ease, the IP phone will tvpicallv
die Cisco IP phone about the voice VLAN ID by also sending out aCisco Discovers
3-6
DHCP
Mana9er
Cisco TFTP
Cisco Certificate Trust List (CTL) file is used in voice security-enabled environments and is
not covered in this course
Step 5
Obtaining an IP address: If the Cisco IP phone uses 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 DHCP is not used in your network, a static IP address and TFTP server address must be assigned to each IP phone locally. If the DHCP server does not respond, the IP phone will make use ofthe last used configuration that is stored in NVRAM. Requesting the configuration file: The Cisco IP phone requests various files from
the TFTP server. The first file that it tries to download is the Certificate Trust List
Step 6
(CTL) (CTLSEP<M.4C>.tlv), which is only used if cryptographic features arc enabled in Cisco Unified Communications Manager.
The phone now requests its individual configuration file (SEP<il/.4C>.cnfxml). which is only present on the TFTP server if the phone is already configured as an SCCP device in Cisco Unified Communications Manager. If this file is not a\ ailable. it further tries to download the SIP-based configuration file
(SlP<A/-JO.cnf).
3-7
The individual and the default configuration file contains a prioritized list
of up to three Cisco Unified Communications Manager call-processing
nodes and the mod el-specific phone load version to use The Cisco IP phone compares its installed phone load version with the
load version defined within the received confi guration file. If the load
version is different, it requests the new load version from the TFTP server
and reboots Cisco Unified Communications
DHCP
Manager
CiscoTFTP
Step 7
Default configuration file: If the TF1 P server responds wilh a "File not found" error message on the previously requested configuration files, the phone requests the XMLDefault.cnf.xml file. Like the individual configuration file, this file contains a prioritized list of up to three call-proccssing nodes and the phone load version that is to be used for each phone model. Phone Load check: Once fhe phone has received either the individual or the default configuration file, it compares its local load version with the load version that is specified within the configuration file. If they are different, the phone downloads the
new load version from the TF'l P server and reboots.
StepB
3-8
Ifthe Cisco IP phone is already configured within Cisco Unified Communications Manager, it will successfully register and will be instructed by SCCP messages to set up the display layout (directory number, softkey buttons, speed dials, etc.). If localization or customer ringers are configured for the phone,
additional files will be downloaded.
Cisco Unified Communications DHCP
Manager
Cisco TFTP
Step9
Step 10
Registeringon CiscoUnified Communications Manager: The phone attempts to registerwith the highestpriority call-processing node on the list.
If the phone is already configured as an SCCPphone at Cisco Unified Communications Manager, it will successfully register and will be instructed by SCCPmessages to set up the display layout. The display layout includes attributes such as directory' numbers, softkey buttons, speed dials, and so on.
If localization or custom ringers are configured for this phone, additional files will
be downloaded from TFTP.
Step 11
3-9
Autoregistration disabled: Cisco Unified Communications Manager will not allow registration. The Cisco IP phone will display "Registration
Rejected "
Cisco Unifed Communications
DHCP
Manager
cisco TFTp
Step 12
If the Cisco IP phone is not \et configured and received the list of call-processing nodes, from the default configuration file, the following options are possible:
Autoregistration enabled: After the phone tried to register at the call-processing node. Cisco Unified Communications Manager dynamically creates an individual
configuration file for this phone and requests it to reboot. After reboot, the phone will suceessfullv register.
Autoregistration disabled: Cisco Unified Communications Manager will not allow registration. The Cisco IP phone will display a "Registration Rejected" message on the phone display.
3-10
The boot sequences that are used for SIP phones are like the boot sequences that are used for SCCP phones, except for the follow ing main differences: SIP phones get their entire configuration from a configuration file. Therefore, the SIP configuration file is much larger for SIP than for SCCP.
If local dial rules are configured for the SIP phone, these rules will also be downloaded.
Some SIP phone models also download a separate softkey configuration file.
3-12
Communications Manager
"|
3 Phoneloadfile(optional)
5 Phone registers
6 Localization files
'^
S Custom ringers
The first stepsarethe same as with SCCP phones andare not shown in the diagram. In the diagram and following steps, it is assumed thatthe SIP phone has obtained an IPaddress and
information about how to reach a TFTP server:
Step 1
"fhe SIP phone boots and tries to download a CTL file. The CTL file contains a set
of certificates and is only used when Cisco Unified Communications Manager
cluster security has been enabled.
Step2
The SIPphone requests its S1P<M/I0.cnf file from the Cisco TFTP server. If a SIP phone is new. this file will notbe found, because the phone is notcurrently configured in the Cisco Unilied Communications Manager database. Contrary to SCCP configuration files, the SIPconfiguration file alsocontains components such
as directory numbers, softkey configuration, and so on.
Step 3
TheSIP phone requests the .Loads file, if one wasspecified in the default configuration file, to see whatimage the phone should be running. If the .Loads file specifies an image thatis different from the image that is contained in the SIP phone, the SIP phone attempts to obtain the newimages from the CiscoTFTP server. If the image is downloaded andverified successfully, tlieSIP phone will
reboot to load the new image.
Step 4
Step 5
The nextstepis to register with the highest-priority CiscoUnified Communications Manager server. The SIPconfiguration file indicates whether the SIPphone should
connect using User Datagram Protocol (UDP) or TCP. Localization files will be downloaded.
Step 6
3-13
Step 7
Most current Cisco IP phones (Type-B) also support the download of softkey
configuration files. Type-A phones (Cisco Unified IP Phones 7940and 7960)do not
support custom softkey configuration files.
Step 8
Custom ringtones will alsobe downloaded using a separate custom ringers file.
3-14
* H.323 endpoints can be voice or video devices. * H.323 endpoints are normally H.323 terminal devices, especially video endpoints. H.323 phones do not register with Cisco Unified Communications Manager and only have to be known by IP
address.
* H.323 phones need to have their own dial plan and act as a peer to Cisco Unified Communications Manager.
The H.323 client consumes two device license units.
the H.323 protocol. H.323 phones support multiple lines and canbeeither video or audio endpoints (where video endpoints include audio capabilities). In H.323 terminology, the
endpoints are H.323 terminals.
11.323 phones do not register with Cisco Unified Communications Manager but are configured by IP address, which becomes a problem if dynamic IPaddresses areused. In such a case, an 11.323 gatekeeper can be used for dynamic endpoint registration.
Configuration must beperformed onboth Cisco Unified Communications Manager and onthe
phone itself. This includes dial plan configuration, because the H.323 phone routes calls
autonomously: however, all callscan be routed to Cisco Unified Communications Manager. Each H.323 phone consumes two device license units in Cisco Unified Communications
Manager.
H.323 Endpoints
"Ihe figure shows an example of an 11.323 endpoint.
ndpoints
Common1\ used 11.323 endpoints arc 11.323 video devices from different vendors. H.323
endpoints are often deplov ed withan 11,323 gatekeeper processing the registration of the
devices.
3-16
H.323 endpoints support only a few features compared with Cisco IP phones that are using
SCCP or SIP. The features thatare not supported include butarenotlimited to the following:
fhere is no support for phone button templates and softkey templates. The user interface
depends on the H.323 product that is used.
Telephony features and applications such as:
Cisco IP Phone Services
3-17
.7
Ihe high-level configurations for H.323 phone implementations include the following points:
The H.323 phone has to be addedto Cisco Unified Communications Manager wilh its IP
address and direetorv numbers specified. The 11.323 phone has to be configured wilh the IP address of Cisco Unified Communications Manager. Adial planmustbe configured on both devices. Typically, all callsfrom the H.323 phoneare
routed to Cisco Unified Communications Manager to take advantage of the centralized dial plan of Cisco Unified Communications Manager.
Note
3-18
Third-party SIP phones register with Cisco Unified Communications Manager but are not recognized by a device IDsuch as a MAC address. SIP Digest Authentication is used instead to identify the endpoint that is trying to register. Configuration is performed on Cisco Unified Communications Manager and on the phone itself.
butCisco IPphones that are using the SIP protocol have many more telephony features than
third-part; phones that are using the SIP protocol.
Two different types of third-party SIP phones canbeadded to Cisco Unified Communications
Manager:
m Basic phones: Support only a single line andconsume three device license units Ad\anced phones: Support up to eight lines and video andconsume six device license
units
In termsof telephony features, there is no difference in basic versus advanced third-party SIP
phones.
Third-partv SIP phones register with Cisco Unified Communications Manager butdo notuse a
MAC address-based device ID. Cisco Unified Communications Manager uses SIP digest authentication to identify a registering third-party SIP phone. Both Cisco Unified Communications Manager and the third-party SIP phone must be
configured.
fhe following audio and videostandards are supported for third-party SIP phones:
Audio
Video
3-20
Cisco Unified IP Phones 7940 and 7960 can be loaded with a standard SIP firmware. In this
case, the phone is configured as a third-party SIP phone rather than as a Cisco Unified IP Phone
7940 or 7960 in Cisco Unified Communications Manager.
Cisco is working with key third-party vendors who arepart ofthe Cisco Technology Development Partner Program and who are developing solutions that leverage the SIP capabilities ofthe new Cisco Unified Communications Manager and Cisco Unified Communications Manager Express. Vendors include Linksys (hardware phones), IPceterate (unified client for educational environment usage), Research in Motion (RIM) (BlackBerry 7270 wireless LAN handsets). IP blue(soflphone), andGrandstream (Grandstream GXP2000
IP phone).
Cisco is also participating inan independent third-party testing and interoperability verification
process that isbeing offered by tekVizion. This independent service was established to enable
third-party vendors to test and verify the interoperability oftheir endpoints with Cisco Unified
Communications Manager and Cisco Unified Communications ManagerExpress.
Note Cisco Unified IP Phones 7940 and 7960 are end of life (EOL) and can no longer be purchased.
3-21
Third-party SIP phones support only a few features compared with Cisco IP phones using SCCP or SIP The features that are not supported include but are not limited to the following:
MAC address registration
Phone button templates
The limitations of third-partv SIP endpoints are the same that apply to 11.323 endpoints. These limitations include but arc not limited to the following:
MAC addres>-based registration. SIP phones need to be configured by their IP address in Cisco Unified Communications Manager instead of a MAC address-based device ID. fhere is no support for phone button templates and soflkey templates, fhe user interface
depends on the SIP product that is used.
Telephonv features and applications such as the following are not supported:
Cisco IP Phone Services
3-22
Cisco Unified Communications Manager can be configured to check the key (i.e., digest credentials) of a username that is used by a third-party SIP device, or to ignore the key and
only search for the username.
SIP digest authentication is specified in RFC 3261 and RFC 2617. It is based on a client/server model, in which the server challenges and the clientresponds, andprovides authentication of SIP messages by a username and a keyedhash.
SIP digestauthentication allowsCisco UnifiedCommunications Manager to act as a serverto
challenge the identity of a SIP device when it sends a request to Cisco Unified Communications Manager. When digest authentication is enabled for a phone, Cisco Unified Communications Manager challenges all SIPphone requests except keepalive messages.
CiscoUnified Communications Manager doesnot support responding to challenges from SIP phones, butit can challenge SIPdevices that areconnecting through a SIP trunk and can respond to challenges that are receivedon its SIP trunk interface.
In Cisco Unified Communications Manager, SIP digest authentication is used to identify a
third-partv SIP phone, because these phones do notregister with a MAC address-based device
ID.
Cisco Unified Communications Managercan ignorethe keyed hash that is provided in a digest
authentication response and only check if theprovided username exists and is bound to a thirdparty SIP phone. This behavior is thedefault. Alternatively, Cisco Unified Communications Manager canbe configured to check that the keythatwas used at the third-party SIP phone to generate the keyed hash matches the locally configured key (called "digest credentials") at the
end-user configuration in Cisco Unified Communications Manager.
3-23
Communications
Manager
Configuration
Database
REGISTER 1001
Username - "3rdpsip"
Third-party SIP phones cannot be configured using the Cisco Unified Communications Manager TFTP server. Instead, they need to be configured using the native phone configuration mechanism, which is usually a v\cb page or a TFTP file. The device and line configuration in the Cisco Unified Communications Manager database must be synchronized with the native phone configuration manually (for example, extension 1002 on the phone and 1002 in Cisco Unified Communications Manager). Also, if the directory number of a line is changed, it must be changed in both Cisco Unified Communications Manager Administration and in the native
phone configuration mechanism.
Third-party SIP phones include their directory number in the registration message. Thev do not
send a MAC address: thev must identify themselves b\ using digest authentication. For this purpose, the SIP RtGISTKR message includes a header with a username and the keved hash, as shown in the example:
Authorization: Digest
username="3rdpsip",realm="ccmsipline",nonce-"GBauADss2qoWr6k9y
3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5, response="12 6c064 3a4923359ab59d4f53494552e"
When Cisco Unified Communications Manager receives the registration message, it searches for an end user that matches the provided useniame in the SIP message (in this ease, "jrdpsip"). If found. Cisco Unified Communications Manager will use the digest credentials that are configured for that user to verify the keyed hash ("response" in the example). If the
keved hash is acceptablethat is. Cisco Unified Communications Manager and the third-partv SIP phone share the same kev that is used for the hash- the user passes authentication.
Note
Cisco Unified Communications Manager must be explicitly configured to verifythe keyed hash Bydefault, Cisco Unified CommunicationsManager only searches for the end
username.
Cisco Unified Communications Manager then searches for a third-party SIPphone that is
associated with the end user, and verifies that the configured directory number matches the
number that is provided by the third-party SIP phone in its registration message. If the phone Is found and the directory numberis the same, the third-party SIP phone registered successfully
with Cisco Unified Communications Manager.
?
:.'. '-
Configure the third-party SIP phone and its directory numbers in Cisco Unified Communications Manager.
Select the configured end user as Digest User on the third-party SIP phone configuration window. Configure the third-party SIP phone with the IP address of Cisco Unified Communications Manager (proxy address), end-user ID, digest credentials (optional), and directory
numbers
Step 2
Note
Wrien configuring the third-party SIP phone in Cisco Unified Communications Manager, you must specify a dummy MAC address. The entered MAC address will not be used to identify
the device, but it is required, because inside the Cisco Unified Communications Manager
configuration database, phone records are uniquely identified by MAC addresses.
Step 3
Step 4
Associate the third-partv SIP phone with the end user configured in Step !.
Configure the third-part} SIP phone with the IP address of Cisco Unified Communications Manager (proxj address), end-user ID. digest credentials (optional), and directory numbers.
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Cisco Unified Communications Manager supports SIP, SCCP, and H.323 protocol for endpoints. Cisco IP phones follow a specific process during bootup,
allowing the IPphoneto learna voice VLAN ID, obtain IP configuration from a DHCP server, and download its
configuration from a TFTP server.
H.323 phones have to be configured on Cisco Unified CommunicationsManager and also manuallyon the phone.
References
For additional information, refer to these resources:
lutp'.';v\\vv\.cisco.coni/cn/US/docs/voice_ip_eomm/cucm/drs/X_0_l/drsagX()l.htiiil,
Cisco Systems. Inc. Cisco Unified Communications System Release 8.x SRND. San Jose.
California. April 2010.
3-27
3-28
Lesson 2
Implementing IP Phones
Overview
Adding, updating, and deleting phones arc important functions inthe day-to-day activities ofa
CiscoUnified Communications Manager Administrator. Cisco Unified Communications Manager provides varioustools to accomplish these tasks. This lesson describes howto implement Skinny Client Control Protocol (SCCP) and Session Initiation Protocol (SIP) phonesCisco and third-parly phonesin Cisco Unified Communications Manager (manually, using autoregistration, and with Cisco Unified
Communications Manager Bulk Administration Tool [BAT]).
Objectives
Upon completing this lesson, youwill beable to implement SCCP and SIP phones (Cisco and third-party phones) in Cisco Unified Communications Manager and harden theCiscoIP phones. This ability includes being able to meet these objectives:
Describe how to enable autoregistration for automatic insertion of new phones to the Cisco Unified Communications Manager configuration database
Describe how Cisco Unified Communications Manager BAT and Cisco Unified
Autoregistration
Devices automatically
added
Cisco Unified
Communications Manager
Bulk Administration Tool
Bulk add
MACaddresses required
in BAT files
Cisco Unified
Very scalable
MAC addresses not
required
Manual Configuration
Simple
There are basicallv four methods ol'adding IP phones to the Cisco Unified Communications
Manager:
Autoregistration allows the administrator to add Cisco IP phones to Cisco Unified Communications Manager without first compiling a list of MAC addresses of the endpoints. Without autoregistration. changes in the configuration mustbe done manually. Without using Cisco Unified Communications Manager HAT and Cisco Unified Communications Manager Auto-Register Phone fool, there is no easy way for the phone to be associated with the correct user. If the user has specific requirements, these will have to be updated manually after the
device has been registered.
Cisco Unified Communications Manager BAT allows bulk adds of phones, but MAC addresses
of IP phones must be known and included in the BAT files.
3-30
The Cisco Unified Communications Manager Auto-Register Phone Tool is more scalable, butit
requires a separate Cisco Customer Response Solutions (CRS) server; therefore, the
administrator must be familiar with the installation and configuration ofthe Cisco CRS server.
When using the Cisco Unified Communications Manager Auto-Register Phone Tool. MAC addresses are automatically added and associated with thecorrect phone configurations that have been added previously using Cisco Unified Communications Manager BAT (with dummy
MAC addresses only).
Adding phone devices manually isthe easiest way toadd IPphones to Cisco Unified
Communications Manager, but it hasthe disadvantage of being tedious and time-consuming. Theadministrator mustmanually compile a listofthe MAC addresses ofthe IP phones and ensure thatthey are correctly entered when creating device records for the phones.
Regardless ofthe configuration methods and tools that are used, the various endpoint-related
configuration elements remain the same.
3-31
Datemme Group
Device Pool Cisco Unified
Enterprise Phone Configuration Phone Security Profile Softkey Template Phone Button Template SIP Profile (SIP phones only)
Common Phone Profile
The figure shows some basic endpoint configuration elements. Some configuration elements can be assigned to the endpoint and some elements are assigned indirectly through a device pool.
Hvamples of elements that are a^sig[^cd through a device pool are as follows:
Configuration elements can be optional or mandatory. Some mandatory elements have predefined defaults, and the administrator can make use of these defaults in basic scenarios.
3-32
Phone NTP Reference This subtopic describes the configuration ofphone Network Time Protocol (NTP) reference.
If NTP servers do not respond, the Cisco IP SIP phone uses the date header in the 200 OK response.
'
Phone HTP Reference Inform*!!**!
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You can configure phone NTP references in Cisco Unified Communications Manager Administration to ensure that a Cisco IP SIP phone receives its date and time from an NTP server. If no NTP server is reachable, the Cisco IP SIP phone uses the date header in the 200
OK response to the REGISTER message fortliedate and time. CiscoUnified Communications Manager is using its internal NTP synchronized timewith the configured Date andTime Group
ofthe IP phones to provide time information for SCCP phones within SCCP messages.
After the phone NTP reference hasbeen added to CiscoUnified Communications Manager
Administration, it must be added to a date/time group. You can configure priorities ofthe phone NIP references In the date/time group.
The date/time group configuration is referenced from a device pool, and the device pool is assigned to a device at the device configuration page.
"fhe table describes the Phone NTP Reference Configuration fields.
3-33
Description
Enter the fP address of the NTP server that the Cisco IP SIP phone should use to receive its date and time. Cisco Unified
Description
From the drop-down list box, choose the mode for the phone
NTP reference. The values that are available are as follows:
the phone accesses date and time information from any NTP server, but gives the listed NTP servers (1st - primary, 2nd = secondary) priority For example, ifthe phone configuration contains NTP servers where A - primary NTP server and B secondary/backup NTP server, the phone uses the broadcast
packets (derives the dale and time) from NTP server A. If NTP server A is not broadcasting, the phone accesses date and
time information from NTP server B If neither NTP server is
broadcasting, the phone accesses date and time information from any other NTP server. If no other NTP server is broadcasting, the phone will derive the date and time from the
Note
Although selectable. Cisco Unified Communications Manager currently does not support the
multicast and anycast modes If either of these modes is selected, Cisco Unified
3-34
Date/Time Group
This subtopic describes the configuration ofdate/time groups.
Date/Time Group
Date/time groups define time zonesfor devices that are connected
to Cisco Unified Communications Manager.
Date/timegroup is assigned to the device pool. The device pool is assigned to the device.
Use date/time groups todefine time zones for devices that are connected toCisco Unified Communications Manager. Each device exists asa member ofonly one device pool, and each
device pool hasonly one assigned date/time group.
Installation of Cisco Unified Communications Managerautomatically configures a default
date/time group called CMUocal. CMUocal synchronizes tothe active date and time ofthe
operating system on the server where Cisco Unified Communications Manager is installed.
After installing Cisco Unified Communications Manager, you can change the settings for
CMI.ocal.
Note
CMLocal resets to the operating systemdate and timewheneverCisco Unified Communications Manager gets restartedor whenthe Cisco Unified Communications
Manager software is upgraded toa new release. Do not change thename ofCMLocal
3-35
1hetable describes the Date/Time Group Configuration fields. Date/Time Group Configuration Field Descriptions
Field
Description
Group Name
Time Zone
Separator
Date Format
Choose the date formatfor the date that displayson the Cisco
Unified IP phones.
Choose a 12- or 24-hour time format.
Time Format
To ensure that a SIP phone receives its date and time configuration from an NTP server, add the phone NTP
Find the phone NTP reference that needs to be added Only phone NTP references that exist in the Cisco Unified
3-36
Device Pool
This subtopic describes the configuration of device pools.
Device Pool
Device pools define sets of
common characteristics for
devices.
Device pools define setsof common characteristics fordevices, 'fhe device pool structure supports the separation of user and location information. Thedevice pool contains only deviceand location-related information. The Common Device Configuration window records all the user-oriented information, such as type of softkey template that is used, and localeinformation.
You should ensure that each device is associated with the correct device pool and common
Cisco Unified Survivable Remote Site Telephony (SRST) reference. The Cisco Unified SRST Reference field allows the administrator to specify the IP address ofthe Cisco Unified router. Cisco Unified SRST enables routers to provide call-handling support for
The dev ice pool combines all the individual configuration settingsthat have been created into a single entity, "fhis element canthen be assigned to individual devices, suchas IP phones. This process will configure these devices with mostofthe configuration elements thatthey need to operate efficiently in the IP telephony network.
Choose System > Device Pool. The find and List Device Poolswindow opens.
Click the Add New button to open the Device Pool Configuration window.
Choose, at a minimum, the Cisco I 'nified Communications Manager Group. Date<Time Group. Region, and a Softkey Template.
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
Assigns the correct time zone to the device.
Manager Group*
Date/Time Group* Region"
Softkey Template*
Defines the type and order of the softkeys that are displayed on the LCD of a Cisco IP phone Configures Cisco Unified SRST and chooses the gateway that
will support the device if the connection to Cisco Unified
SRST Reference"
Defines who an IP phone is able to call if it autoregisters with Cisco Unified Communications Manager.
Assigns media resource support to a device for functions such as conferencing, transcoding, or music on hold (MOH)
Chooses the audio that Cisco Unified Communications
Manager should play when you press the Transfer or Conference button on the Cisco IP phone.
User Hold MOH Audio Source Chooses the audio that Cisco Unified Communications
Manager should play when you press the Hold button on the
Cisco IP phone.
Network Locale
User Locale
Defines the language that the device uses. Defines the amount of time that the IP phone monitors its
connection to Cisco Unified Communications Manager before it unregisters from Cisco Unified SRST and reregisters to Cisco
Unified Communications Manager. This setting is to ensure that the registration is stable in case of a flapping link. The default for the enterprise parameter specifies 120 sec, which can be modified on a device-pool basis or left at the default value.
Note
Communications Manager
Communications Manager in
the list serves as the primary
Cisco Unified
(backup) CiscoUnified
Communications Manager
servers.
The first Cisco Unified Communications Manager in the list serves as the primary Cisco Unified Communications Manager for that group, and the other members of the group serve as secondary and tertiary (backup) Cisco Unified Communications Manager servers.
Each device pool has one Cisco Unified Communications Manager group that is assigned to it.
Communications Manager in the group that is assigned to its device pool. If the primary Cisco Unified Communications Manager is not available, the device tries to connect to the next Cisco
Unified Communications Manager that islisted in the group, and so on.
When adevice registers, it attempts to connect to the primary (first) C.sco Unified
Cisco Unified Communications Manager groups provide the following important features for
the unified communications system:
Redundancy: This feature allows the administrator to designate aprimary and backup
Cisco Unified Communications Manager foreachgroup.
. Call processing load balancing: This feature allows the administrator to distribute the
control ofdevices across multiple Cisco Unified Communications Manager servers.
For most s\stems, there is aneed for multiple groups, and asingle Cisco Unified Communications Manager can be assigned to multiple groups to achieve better load
distribution and redundancy.
Single-Site On-NetC3lling
3-39
Regions
Ihis subtopic describes the configuration ofregions.
Use regions tospecify thebandwidth that is used for anaudio or video call within a region and between regions by codec type The audio codec determines the type ofcompression and the
maximum amountofbandwidth that is used per audiocall.
Regions are used to specif} the maximum bandwidth that is used per audio or video call within
sounds best on loss} links and on links with low packet loss. Always choose aIink Loss l\pe
The configured maximum region bandwidth determines possible audio codecs that can be used lor acall within aregion or between aregion pair. The Region Configuration has an additional I.ink loss Upe parameter that Cisco Communications Manager uses to select the codec that
C ommunications Manager prefers the best-sounding codec (regardless ofthe codec bit rates)
that does notexceed the configured maximum bitrate.
(i.722 64 k is preferred over G.722.1. Cieneral codec references are as follows:
selection process takes place in Cisco Communications Manager: In general. Cisco Unified
If the chosen region bandwidth includes multiple selectable codecs, the following codec
G.722 at all bit rates (64. 56. and 48 kb/s) is preferred over G.71 1.
On low-loss links. G.722 is preferred over Internet Speech Audio Codec (iSAC).
On loss\ links. iSAC is preferred over G.722.
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Lossy
AAC-LD
L16 iSAC
AAC-LD
L16
32 32
56 24 48
G.722 64 k
64
32 56 24 48 64 64
G.722.1 32 k
G.722 56 k
G.722.1 24 k
G.722 48 k
64 64
56 56 16 16 13
G.711 a-law 64 k
G.711 mu-1aw56k
56 56
16 16
G.711 a-law 56 k
iLBC G.728
13
13 8 8 8
13 8
8 8 8 7 7
G.729AB
G.729 G729A
8 7
7
Note
The\ideo call bandwidtli comprises the sum of the audio and video bandwidth of the video
call.
TheG.722 and iLBC codecs can generally be enabled and disabled viaa Cisco CallManager
senice parameter (G.722 Codec Enabled/iLBC Codec Enabled option). If G.722 isenabled (default setting), its usage can befurther controlled ona pcr-device basis via the phone configuration page (Product Specific Configuration Layout >Advertise G.722 codec).
3-41
Note
The Advertise G722 Codec parameter indicates whether Cisco IP phones will advertise the G722 codec to Cisco Unified Communications Manager. Codec negotiation involves two
steps First, the phone mustadvertise the supported codecs to Cisco Unified
Communications Manager (not all endpoints support the same setof codecs}. Second,
when Cisco Unified Communications Manager receives thelist ofsupported codecs from all phones that are involved in thecall attempt, itchooses a commonly supported codec that is based on various factors, including the region pair setting. Valid values specify Use System Default (this phone will defer tothesetting that isspecified in theenterprise parameter,
Advertise G.722 Codec), Disabled (thisphone will notadvertise G.722 to Cisco Unified Communications Manager), or Enabled (this phone will advertise G 722 to Cisco Unified
Communications Manager).
Step 1
Step 2
Step 3
Choose S\stem >Region, fhe default region that was created during the Cisco
Unified Communications Manager installation appears.
(Tick Add New to configure the regions.
Gi\e the new region a unique name. Click Save.
Step 4
Note
Choose the codec and \ ideo bandw idth asappropriate between the regions.
Cisco Unified Communications Manager allows a maximum of2000 regions.
3-42
Locations
Locations
Use locations to implement CAC in a centralized
call-processing deployment.
CAC enables you to regulate audio qualityand video availability by limiting the amountof bandwidth that is
available for audio and video calls.
system. CAC enables the administrator to regulate audio quality and video availability by limiting the amount ofbandwidth that is available for audio and video calls that go in or out of
a location.
Note
If CAC is notusedto limit the audio and video bandwidth on IPWAN links, an unlimited
number of calls can be active on that link at the same time. This situationcan cause the
quality ofall audio and video calls todegrade as the link becomes oversubscribed.
cluster provides call processing for all locations on the IP telephony network. The Cisco
Unified Communications Manager cluster usually resides at themain (orcentral) location,
along with other devices such as phones and gateways, 'fhe remote locations contain additional
devices but no Cisco Unified Communications Manager. IP WAN linksconnectthe remote
locations to the main location.
Common Settings" box for each setting you wish to update. If you do not
check this box. the
In the Enterprise Phone Configuration window, you can configure parameters that will apply to
all phones that support these parameters.
Select the Override Common Settings box for each setting you wish to update. Ifyou do not
check this box. the corresponding parameter settingdoes not take effect. Note Note1 Parameters that you set in this window may alsoappear in theCommon Phone profile
window and the Device Configuration window for various devices. Ifyou set these same
parameters in these other windows too, the setting that takes precedence is determined in the following order. 1) Device Configuration window settings, 2) Common Phone Profile window settings. 3) Enterprise Phone Configuration window settings.
3-44
Profile Configuration window includes securityrelated settings such as device security mode, CAPF settings, digest authentication settings (for
~_ tOttn*, j L**UH
Manager Administration.
The Phone Security Profile Configuration window includes security-related settings such as device security mode. Certificate Authority Proxy Function (CAPF) settings, digest authentication settings (for SIP phones only), and encrypted configuration file settings. A security profile must be applied to all phones that are configured in Cisco Unified Communications Manager Administration. The administrator can use existing security profiles
that have security disabled.
3-45
Device Settings
This subtopic describes device settings.
Device settings contain default settings, profiles, templates, and common device configurations
that \ oil can assign to the de\ ice or device pool.
3-<J6
Device Defaults
phone model. Thedevice pool and phone button template for autoregistered phones will be defined
here.
aT
_^
Use device defaults to setthe default characteristics of each type ofdevice thatregisters with a Cisco Unified Communications Manager. Thedevice defaults for a device type apply to all
autorcgistered devices ofthat type within a Cisco Unified Communications Manager cluster.
You can setthe following device defaults for each device type to which they apply:
Device load: Lists the firmware loadthat is used witha particular typeof hardware device
Device pool: Allows the administrator tochoose the device pool that isassociated with
each typeof device in caseof autoregistration
Phone button template: Indicates the phone button template that each type ofdevice uses
in case of autoregistration
When a device autoregisters with Cisco Unified Communications Manager, it inherits the
default settings for its device type.
Step 1 Step 2
Step 3
In Cisco Unified Communications Manager Administration, choose Device > Device Settings > Device Defaults toopen the Device Defaults Configuration
window.
In the Device Defaults Configuration window, modify the appropriate settings for
the device.
Click Save tosave the changes inthe Cisco Unified Communications Manager
configuration database.
The following example shows a partofthe XMLDefault.cnf.xml file that is based on the
configured device defaults:
iprocessNodeName-ir.
i 'callMacager>
.v/member.e. 'menbe: ~ ->
1.1.1= piocessKodeNaine:-
tenters -
=< 'addresss
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-address
T.entsri .-
<:oad:rforT.aMcri3 0:;2
<loadlnf cr;".aficr.52C
<lcad!nforn-.at.iO:ii9-' nodel= "C-.sco 6961"--SCCP6 9xx . 8 - S 2 70 - 7</loadIntormation497 s <loadlnformatio:!?00 JS model- ,,Ci sco 797 0 " >SCCP7 0 . 9-0 2TH1 -7Sw'loadInformationSOO06> <loadlnf ci'-naricr.l 15
<.lcadIr.oiTat:on4SJ
tloadInfcrnatio:-.J0012
<loadIafcr7iaticr.4 3 6 -load Information^ 7 5
tloadlnfoi-t.a-ior.3 03
<load Information! jj
<loadInroir.:it:cnJCCIj
<loadInfcr^iatior.4 8:
<loadIafoi-na:ior.li
<-IoadIr,forTa-iOi'.4;2 <_oad Information 3 ib
mcdel= "Ci-ico ATA 166" -ATAG 302 04SCCPO902 02JV./loadlnf onnat ionl2>
rode 1= "Cisco J551" >IP5951 . 8 -1 -2SRW loadlnf ormat ion412 > mode 1 = "Cisco 7 921" .CP7 9 21G 1.3. 3 :/loadlnforn at J on3 6 5 -.
3-48
>S
Creating and using templates provides a fast way to assign a common button configuration to
many Cisco Unified IP phones.
Cisco Unified Communications Manager includes several default phone button templates.
When adding phones, one ofthese templates can be assigned tothe phones. Cisco
Communications Manager also allows defining thephone buttons without using a previously
created phone button template by clicking Modify Button Items onthe phone configuration page. In this case. Cisco Unified Communications Manager dynamically creates a phone button
template thatis associated with this device only. Make sure that all phones have at least one assigned line, which isnormally button 1. Additional lines toa phone depend onthe model ofCisco Unified IPphone. Phones generally have several features, such as speed dial and Call Forward, which arcassigned to the remaining
buttons.
Before adding an> IPphones tothe system that are supposed to use nonstandard phone button allocation, you should create custom phone button templates for these IPphone models.
3-49
Softkey Template
fhis subtopic describes the configuration ofsoftkey templates.
Softkey Template
Softkey template configuration allows the administrator to configure softkey layouts that are assigned to Cisco Unified IP phones.
Softkey template configuration allows the administrator to manage softkeys on Cisco IP phones. Cisco Unified Communications Manager supports two types ofsoftkey templates:
standard and nonstandard. Applications thatsupport softkeys can have oneor more standard softke\ templates that are associated with them; for example. Cisco Unified Communications Manager has the standard feature andfhe standard usersoftkey templates thatare associated
uith it. Standard softkey templates cannot be modified ordeleted. To create a new softkey
template, copy one ofthe templates, edit it. and save it with a new name, or create a new one from the beginning.
Choose De\ ice > De\ ice Settings >SoftkeyTemplates to access the Softkey Template Configuration window in Cisco Unified Communications Manager Administration. Tomodify softkeys for a chosen softkev template, choose Configure Softkey Layout from the
Related finks menu.
3-50
2 Assign softkeys
for the selected
phone slate
The screen shows the configuration ofsoftkeys based ondifferent phone states. Every phone
state has a set of possible softkeys to choose from.
3-51
SIP Profile
This subtopic describes the configuration of SIP profiles.
are associated with SIP trunks and SIP endpoints. SIP profiles include information such as name, description, timing, retry, Call Pickup URI, etc.
Sn>Pro(B*<jmfiaarati
uned in Pfioiif
A SIP profile comprises the set of SIP attributes that are associated with SIP trunks or SIP
endpoints. SIP profiles include information such as name,description, timing, retry. Call Pickup Uniform Resource Identifier {URI). and soon. The profiles contain some standard entries that cannot be deleted or changed.
Note A SIP URI consists of a call destination that is configured with a user@host format, such as xten3@CompB Cisco com or 2085017328@10.21.91.156:5060.
A default SIP profile, calledthe Standard SIP Profile, can be assigned to SIP phones on the SIP
phone configuration page, 'fhe Standard SIP Profile cannot be deleted or modified. To create a new SIP profile. cop> the default SIP profile, edit it. and save it with a new name, or create a new profile from the beginning.
3-52
Common phone profiles include phone configuration parameters and are assigned to IP phones.
Common Phone Prolitn t onfiqurfltioo
-Umnmri Phone Profile IimuilimiLiuh
Name*
Description Local Phone Unlock Password
Common phone profiles include phone configuration parameters such asthe phone password (for supported Cisco IPphones). Do Not Disturb (DND), and personalization settings, including end-user access to background images. After a common phone profile has been configured, use the Common Phone Profile Configuration window to associate anSCCP or SIP
phone with it.
The administrator can choose to use the default standard common phone profile, which is createdwhen Cisco Unified Communications Manageris installed, if no specificsettings are required.
3-53
Phone
Softkey Template
Date/Time
Group
ComtnM
..Phone &tlie'
Ihe arrows show the assignment of elements, for example, the N'fP Reference is applied as an element ofthe Date/Time Group, and the Dale/fimc Groupis applied as an elementofthe
Device Pool configuration. The Device Pool is one ofthe elements in the device record ofan IP
phone, allowing the IP phones to inherit or acquiresettings that have been defined in the
various elements.
In some cases,such as Locations, the elementcan be applied lo boththe Device Pool and the phone configuration, in which case the \alue that is applied to the phoneconfiguration will
have higher priorih.
Someof the elements apply onl\ to specificdevice fvpes. 1or example, the SIP Profile applies
only to a SIP phone.
3-54
IP Phone Autoregistration
This topic describes how autoregistration works.
Autoregistration
Supported by all Cisco IP phones. Existing endpoints are not affected.
An autoregistration directory number range is configured, and each phone thatis added byautoregistration is assigned with
the next available directory number ofthe configured range. * Cisco Unified Communications Manager BAT can be used to make bulk changes after autoregistration. The Cisco Unified Communications Manager Auto-Register Phone Tool can be used to associate phones with specific
directory numbers.
Autoregistration allows Cisco Unified Communications Manager to issue directory numbers to new IP phones, which issimilar tothe way in which the DHCP server issues IP addresses. With autoregistration configured and enabled, when a new IP phone boots and attempts to register with Cisco Unified Communications Manager for the first time. Cisco Unified Communications Manager issues a directory number from a configured range. After Cisco
Unified Communications Manager issues thedirectory number, it adds thephone to its
configuration database with the used device ID (MAC address) and the assigned extension. After the phone is added, the assigned directory number usually has tobe modified, because a
specific extension is intended tobeused for a given phone.
Therefore, autoregistration only slightly simplifies registration when you add a large number of IP phones. The MAC addresses ofthe phones are automatically added to the Cisco Unified
Communications Manager configuration database. The directory number per phone must still
be modified.
Some phone settings, such as device pools, need to be globally changed from their default
\ alucs. You can use Cisco Unified Communications Manager BAT after phones have been
autorcgistered.
For large deployments, you can use the Cisco Unified Communications Manager Auto-Register Phone Tool, which allows specific extensions tobe assigned toindividual phones based on user
input.
Autoregistration issupported by all Cisco IP phones and does not affect IP phones that are
alread} configured.
3-55
Autoregistration Process
This subtopic describes the autoregistration process.
Because the Cisco IP phone is not yel configured, the Cisco TFTPserver
returns TFTP Error code 1 ('File nolfound').
Manager
Cisco TFTP
server
Step 2 Step 3
Because the phone is not \et configured in Cisco Unified Communications Manager,
the IKIP server will return the TFTP Frrorcode I ("File not found").
Afler not receiving the individual configuration file, the IPphone requests the
general configuration file (XMLDefault.cnfxml).
3-56
The Cisco IP phone verifies its local phone load version and downloads a
new version if necessary.
configuration file and requests the Cisco IP phone toreboot. The Cisco IP phone reboots, downloads the new configuration file, and
Step 4 Step 5
Step 6 Step 7
The TFTP server provides the general configuration file that contains the If
version for each phone model.
addresses ofup to three call-processing nodes as well as the usable phone load
fhe Cisco IP phone verifies its local phone load version and downloads anew
version if necessary.
The Cisco IP phone tries to register at the specified call-processing node. After receiving the registration request from the Cisco IP phone. Cisco Unified ^
reboot.
the IP phone. Cisco Unified Communications Manager now requests the IP phone to
TFTP server and will successfully register atthe specified call-processing node.
Step 8
After the reboot, the phone will receive its individual configuration file from the
Single-Site On-NetCalling
3-57
Autoregistration is always enabled for only one Cisco Unified Communications Manager group butcan be activated
selectively on group members.
Administrators should carefully evaluate autoregistration before implementing it. because its use can pose asecurity risk to the network. Autoregistration allows anyone with phvsical access
to the voice network to connect an IP phone and use it. whether authorized ornot. For this
reason, many organizations, as part oftheir security policy, disable the use ofautoregistration
or use autoregistration in a secure staging environment for initial Cisco Unified Communications Manager configuration.
Arange of director) numbers must be configured on Cisco Unified Communications Manager directory number from that range. Only asingle directory number is assigned per IP phone, and
you cannot control which device will receive which directory number.
tor autoregistration: Cisco Unified Communications Manager assigns the next available
that support only one protocol will still be able to autoregister. even ifautoregistration protocol
The default protocol for autoregisiered IP phones is set globally within the cluster and can be set to either SIP or SCCP, lorendpoints that are SIP- and SCCP-capable. the endpoint firmware is automatically converted to match the default autoregistration protocols. Endpoints
3-58
Configuring Autoregistration
fhis topic describes how to enable autoregistration for automatic insertion of new phones to the
Cisco Unified Communications Managerconfiguration database.
For each Cisco Unified Communications Manager of the Cisco Unified Communications Manager group, enable or
disable autoregistration; ifenabled configure a range of directory numbers to be assigned. Manual reconfiguration or Cisco Unified Communications Manager BAT may be used to personalize a unregistered
devices.
"ITiere are four steps in configuring autoregistration; the fourth step is optional although commonly required: Step 1 Step 2 Step 3 Verify that the desired autoregistration default protocol is selected. Verify that the desired CM group is enabled for autoregistration. Configure Cisco Unified Communications Manager member servers of that group selectively to be used for autoregistration, and if enabled on a particular server, set
this server directory number range.
Step 4
Reconfigure the automatically added phones, applying the individually required configuration settings. This can be done using Cisco Unified Communications Manager BAT for groups of phones that share some settings, or manually for each phone.
3-59
Autoregistration
Phone Protocol
The default autoregistration protocol is an enterprise parameter, configured under System > tnterprise Parameters, litis parameter specifies the protocol that should be used on Cisco IP phones that support SCCP and SIP. 1he default autoregistration protocol is SCCP. Restart all services for the parameter change to
take effect.
3-60
-Ovia Untiled
Met |
J^wfyCorMg !
*6i !
First, go to System > Cisco Unified CM Group and choose the group that you want to configure. At the Cisco Unified Communications Manager group that should provide the autoregistration service, check the Auto-registration Cisco Unified Communications Manager Croup check box. You can only enable autoregistration on one Cisco Unified Communications Manager group. Activating autoregistration on one Cisco Unified Communications Manager group automatically disables the check box on the group that previously had autoregistration enabled (if applicable).
3-61
:il.i-d f..nni
.^
Sj
>-"
' OisaCle^ an th s
Complete these steps to enable autoregistration on a specific Cisco Unified Communications Manager sencr. (fhis ser\erhas to he a member ofthe Cisco Unified Communications Manager group that is configured for auloregistralion.) Step 1 Step 2 Step 3 from Cisco Unified Communications Manager Administration, choose System >
Cisco I nified Communications Manager.
Click Find and choose the server that should be configured for autoregistration. 1 nder the Auto-Registration Information section, enter the appropriate director} number range in the Starting Directory Number and finding Directory Number
fields.
Step 4
Step5
Note
1insure that the Auto-registration Disabled on this Cisco Unified Communications Manager check box is unchecked.
Click Save.
Specifying a valid range of directory numbers in the Starting Directory Number and Ending Directory Number fields automatically unchecks the Auto-registration Disabled check box.
3-62
Cisco Unified Communications Manager BAT allows for bulk updating, addition, ordeletion ofrecords, including the capability to add phone records tothe configuration database. When using Cisco Unified Communications Manager BAT toadd phones, you must specify the MAC addresses ofthe IPphones along with the respective directory numbers inthe BAT files.
Note The MAC address is printed intextand Universal Product Code (UPC) form on both the
shipping box oftheIP phone andontheIP phone itself, which allows you to use barcode
scanners rather than manually typing MAC addresses into BAT files.
Alternatively, you can use autoregistration first sothat Cisco Unified Communications Manager includes all phones with their MAC addresses and the directory numbers that were assigned by autoregistration. The administrator can then modify the directory numbers in the exported files by replacing the directory numbers that were assigned by autoregistration with those that areactually desired for tlie individual phones, 'fheseedited files can then be used by
Cisco Unified Communications Manager BATto update the phone records in the database.
However, both methods do not scale for large deployments.
3-63
Ihis subtopic describe?, how the Cisco Unified Communications Manager Auto-Register Phone
fool allows phone additions in largedeployments.
Desired phones and their directory numbers are addedwith dummy MAC addresses using Cisco Unified Communications Manager BAT.
* Autoregistration is enabled so that new phones can be used to call an IVR application, whichallows users to enter their directory
number
Auto registered phone will be deleted. * Scalesto large deployments: MAC addresses are automatically added
The Cisco Unified Communications Manager Auto-Register Phone Tool is a set of Cisco CRS scripts and a script application that has to be configured on a Cisco CRS server.
With the Cisco I nified Communications Manager Auto-Register Phone Tool, new phones and their director} numbers areadded with dummy MAC addresses (any arbitrary MAC addresses) so that you onl\ have to specif} those settings that cannot beautomated. Usually, Cisco Unified Communications Manager BAT Is used for that purpose. After you add these phone records to Cisco Unified Communications Manager with Cisco Unified Communications Manager BAT.
}ou mustapph the appropriate MAC address to each individual phonerecord.
The process is automated b\ enabling autoregistration to enable newly added IPphones to place a call loan interactive \oice response (IVR) application thai, is running on Cisco CRS. When a phone usercalls into that application, the useris prompted to enterthe desired director}
number.
TheS}slem knows which ofthe prepared phone records (with dummy addresses) the calling phone is supposed to usethe record that hastheentered directory number configured. At thisstage, the s_\stem knows all the required information: the MAC address of this phone as well as the phone configuration record to be applied to this phone, fhe Cisco Unified Communications Manager Auto-Register Phone Tool will now updatethe Cisco Unified Communications Manager configuration database by removing the phone record that was added b} autoregistration (to free up the MAC address in the configuration database) and b\ changing the dummy MAC address ofthe desired phone record to the oneofthe phone.
As a result. MAC addresses were learned automatically and were automatically associated with the correct phone record (based on user input).
3-64
Communications Manager.
The Cisco Unified Communications Manager Auto-Register Phone Tool CRS script has to be downloaded from the Cisco Unified Communications Manager plug-in page and installed
on a Cisco CRS Server.
The Cisco Bulk Provisioning and Cisco Unified Communications Manager Auto-Register Phone Toot must be activated and running. All the necessary CRS files can be downloaded from the Cisco Unified Communications Manager plug-in page and then be installed and configured on the Cisco CRS server. Installation prerequisites for the Cisco Unified Communications Manager Auto-Register
Phone Tool are as follows:
The Cisco Unified Communications Manager publisher is running, and integration with Cisco CRS is configured.
The Cisco CRS server is running, and integration with Cisco Unilied
Communications Manager is configured.
After installation ofthe Cisco Unified Communications Manager Auto-Register Phone Tool, vou can configure optional parameters in Cisco CRS.
Details for installation, configuration, and integration of the Cisco CRS server are not part of this course and are covered in the Deploying Cisco Unified Contact Center Express (UCCXD) course. For instance, an Administrative XML(AXL) account needs to be configured for Cisco CRS so that it can access and update the Cisco Unified Communications Manager database. Additional information can also be found at Cisco.com.
Note
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Process of Adding IP Phones Using the Cisco Unified Communications Manager Auto-Register Phone Tool
fhe figures illustrate the process of adding IP phones when using the Cisco Unified
Communications Manager Auto-Register Phone Tool.
Process of Addin
Cisco Unified Communications Manager routes call to the Cisco Unified Communications Manager Auto-Register Phone Tool application on Cisco CRS
Follow these steps to add an IP phone using the Cisco I Inilied Communications Manager AutoRegister Phone fool: Step 1 Use Cisco Unified Communications Manager BAf to preconfigure phone device
records with dumm\ MAC addresses.
Note
Make sure sufficient Device License Units (DLUs) are present when importing phones using
BAT The phone import job will only import phones as long as DLUs are available and report
errors in the import log file for phones that could not have been inserted due to a lack of
DLUs
Step 2
A new phone is plugged into the network. It autoregisters to Cisco Unified Communications Manager, which creates a new device record with a directory number from the autoregistration range.
Step 3
Step 4
The phone user dials the number ofthe Cisco Unified Communications Manager Auto-Register Phone Tool CRS application.
Cisco Unified Communications Manager routes the call to the Cisco Unified Communications Manager Auto-Register Phone fool application on Cisco CRS.
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6 7
Phone downloads new configuration from Cisco Unified Communications Manager TFTP and registers.
Step 5
Cisco CRS prompts the user to enter a directory number and looks up the number in the phone configuration records that were previously added using Cisco Unified Communications Manager BAT and have a dummy MAC address.
Cisco Unified Communications Manager sends a reboot request to the Cisco IP phone.
Step 6
Step 7
While the phone reboots, Cisco CRS deletes the autoregistered phone entry and updates the dummy MAC address ofthe found phone record with the actual MAC address ofthe phone in the Cisco Unified Communications Manager configuration
database.
Step 8
fhe IP phone downloads its newly created configuration file from Cisco Unified Communications Manager TFTP.
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Using Cisco Unified Communications Manager BAT for Adding Phones to Cisco Unified Communications Manager
This topic describes the procedure of using Cisco Unified Communications Manager BAT to
Manager BA"
The Cisco Unified Communications Manager BAT configuration process includes these steps:
2. Verify that the Bulk Provisioning Services have been
activated on the Publisher server.
'
-i ;;
0
"fhe following procedure for using Cisco Unified Communications Manager BAT to add
phones to Cisco Unified Communications Manager is similar to the procedure for using Cisco Unified Communications Manager BAT for adding users:
Step 1 Step 2 Step 3 Step 4 Step 5
Step 6
Verify that the Bulk Provisioning Services have been activated. Configure the Cisco I (nified Communications Manager BA f template. Create the comma-separated \ allies (CSV) data input file. I 'pload the CSV data input tile. Validate the data input file.
Insert the devices into the Cisco Unified Communications Manager database.
Step 7
A template name must be assigned and mandatory device parameters must be configured. Only the common parameters, shared by all phones, are configured through the templates. Individual
parameters are entered to the CSV data file.
Prior to creating the template, you should ensure that phone settings such as device pool, location, calling search space (CSS), button template, and softkey templates have already been configured in Cisco Unified Communications Manager Administration, fhese settings cannot be created by Cisco Unified Communications Manager BAT. Use the following procedure to create a phone template: Step 1 Step 2 Step 3 Step 4 Choose Bulk Administration > Phones > Phone Template in the menu. The Find
and List Phone Templates window displays.
Click the 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 the
template is to be created. Click Next.
Choose the device protocol from the Select tlie Device Protocol drop-down list box. Click Next. The Phone Template Configuration window displays with fields and default entries for the chosen device type.
Step 5
In the Template Name field, enter a name for the template. The name can contain up
to 50 alphanumeric characters (for example: HQ_CapcTo\vn).
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Step 6
In the De\ ice Information area, enter the phone settings that the phones to be added ha\e in common. Some phone models and device types do not use all the attributes,
which are shown.
Step 7 Step 8
Step 9
After all the settings tor this Cisco Unified Communications Manager BAT phone
template ha\c been entered, click Save.
When the status indicates that the changes are saved, you can add line attributes.
find the line template to add lines to.
Step 10
In the Phone lemplate Configuration window, click Line [1| Add a new DN in the Associated Information area, fhe Line'lemplate Configuration window displays.
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The phone button template that was selected in the previous step determines the number of
master phone template mat has multiple lines. Then, the administrator can use the master
window appears and must be configured in the following way:
lines that the administrator can configure in the line template. The administrator can create a
template to add phones with asingle line or up to the number of lines in the master template.
After the administrator clicks Line |1| Add a new DN, the Line Template Configuration
Step 11
Search Space Presence Group, and others. Keep in mind that all phones that are added by this Cisco Unified Communications Manager BAT job will use the settings
that are chosen for this line.
F.nter or choose the appropriate values for the line settings, such as Partition, Calling
Step 12 Click Save. Cisco Unified Communications Manager BAT adds the line to the
phone template configuration.
Step 13
Note
Repeat the described procedure to add settings for any additional lines.
The maximum number of lines that display for a Cisco Unified Communications Manager
BAT template depends on the model and button template that the administrator chose when
creating the Cisco Unified Communications Manager BAT phone template.
3-71
The figure shows an example of configuring lite bulk.xll template to support phone and
aCSV file to add phones. To add adirectory number column, follow these steps:
Step 1
Step 2
The bulk.xll template, b; default, does not show acolumn for direetorv numbers when ereatim
Click the Create File Format button at the bulk.xll template.
Anew configuration window will pop up.
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In the pop-up window, add the Directory Number entry to the Selected Line Fields.
Click the Create button to add the additional directory number column.
Define the MAC Address. Description, and Directory Number for each phone that should get inserted via Cisco Unified Communications Manager BAT.
Click the Export to BAT Format button to create the CSV file.
Step 6
3-73
This^subtopic describes how to upload adata input file containing the individual ph,
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Use the following procedure to upload the CSV file containing fhe device data to the Cisco
Choose Bulk Administration >( pload/Download Files. The Kind and I ist files
window displays.
Step 1
Step 2 Step 3
Step 4
Click Add New, The file Upload Configuration window displays. In the file text box. enter the complete path ofthe file to be uploaded, or click
HroMse and locate the tile.
|-rom the Select The Target drop-down list box. choose the targel that the file is to
Step 5
Note
from the Select fransaction lype drop-down list box, choose the transaction t\pe
Specific Details, for validating records that follow the Default or Custom file format
All Details for validating records from afile that was generated with the export utility by
using the All Details option
Step 6
Step 7
Ifthe file is to overwrite an existing fife with the same name, check the Overwrite
Hie if it exists check box.
Click Save and wait for updated status infonnation. The status should be Successful.
3-74
template to validate.
3^
(V"" Fil''
JVieiv File)
-Job Information-
Jotj Description
Start validation.
and Cisco Unified Communications Manager BAT phone template have populated all required fields such as device pool and locations. The validation also checks for discrepancies with the first node database (for instance, an already existing entry with the same MAC address).
To validate the CSV data file phone records, use the following procedure:
When performing this step, the system runs avalidation routine to check that the CSV data file
Step 1
Step 2
Click either the Validate Phones Specific Details radio button to validate phone
using the All Details option.
records that use acustomized file format or the Validate Phones All Details radio
button to validate phone records from an exported phones file that was generated by
Step 3
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. This is the file that was
uploaded previously.
Step 4
Step 5
For the Specific Details option, in the Phone Template Name drop-down list box, the
administrator can choose the Cisco Unified Communications Manager BAT phone template that was created for this type ofbulk transaction.
To start the verification, click Submit.
Step 6
Step 7
Check for the status ofthe verification. Proceed to the next step only ifthe
verification was successful.
Single-Site On-NetCalling
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^S
Step 8
Open the Job Scheduler and check if the validation returned errors.
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(i)j- i.t.
and template.
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Click Run
Immediately.
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To start the bulk add ofthe phones that are listed in the uploaded and verified data file, perform the following steps:
Step 1 Choose Bulk Administration > Phones > Insert Phones. The Phone Insert Configuration window displays.
Step 2
Click either the Insert Phones Specific Details radio button to insert phone records
that use a customized file format or the Insert Phones All Details radio button to
insert phone records from an exported phones file that was generated by using the All Details option.
Step 3 In fhe File Name drop-down list box, the administrator can choose the CSV data file that was 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 that
the administrator chose.
Step 4
Checking the Override Configuration Settings check box overwrites the existing phone settings with the information that is contained in the file that is to be inserted. For the Specific Details option, in the Phone Template Name drop-down list box. choose the Cisco Unified Communications Manager BAT phone template that was
created for this type of bulk transaction. If an individual MAC address is not entered
in the CSV data file, you must check the Create Dummy MAC Address check box. This is used when the Cisco Unified Communications Manager Auto-Register
Phone Tool is used.
Step 5
>2010 Cisco Systems. Inc.
Click the Run Immediately radio button to insert the phone records immediately, or click Run Later to schedule the job for a later time. Click Submit to submit the job for inserting the phone records. Check for ihe status of the job: this can be done at any lime by browsing lo Bulk Administration > Job Scheduler and clicking the appropriate Cisco Unified Communications Manager BAT job.
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M3El!llOBFOT"M)"
E u w g: :i igei:!<<
Open the Job Scheduler, choose the phone insert job, and open the associated log file. It should not show any failed insert attempts.
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Sin.
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Manager BAT
The new phones will also show up at the Find and List Phones page.
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3-79
Manually adding new IP phones to the network is oflen tedious, but it can constitute a large part of da\-to-da\ \oice network management. Provisioning a Cisco SIP phone is just like provisioning an SCCP phone.
The configuration procedure consists of these high-level steps: Stepl
Step 2
Step 3
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QjStatus: Ready
Select the type of phone you wouM like to createProduct Type:
"iscg 7975
To manually add an IP phone to Cisco Unified Communications Manager, go to Device > Add Phone and choose the phone type. (In the example, a Cisco Unified IP Phone 7960 was selected.) Then, choose the protocol that should be used with the Cisco Unified IP phone (SCCP or SIP), and click Next to go to the Phone Configuration page.
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(Device Pool)
f.ach phone in the Cisco Unified Comnuiniealions Manager configuration database is uniquel}
identified by a de\ice ID. which is built from its MAC address. The MAC address of a Cisco IP
phone is printed on a label on the back ofthe IP phone and can be viewed at the phone itself b\
pressing the Settings button.
In addition to the MAC address, the following mandator}' parameters have lo be sel:
MAC Address Device Pool
Built-in Bridge
Pri\ ac\
Note
Itis recommended to define a meaningful descriptionfor each configured IP phone, because the standard description will only be the MAC address plus the SEP prefix. It is helpful to contain directory number and associated user informationin the description to help locate phones in Cisco Unified Communications Manager.
Not all these mandatory parameters have to be configured; some of them have default values. Only those parameters that do not have default values set must be configured before the phone can be actually added into the configuration database.
3-83
* Busy Trigger
follow this procedure to configure a director} number for the manually added IP phone:
Step 1 At the Phone Configuration window in the left Associated Information column, click the Line |\] -Add a new 1)\ link to configure the first line with a directory
number.
Step 2
When the Director}' Number Configuration window appears, enter the director}
number ofthe IP phone in the appropriate field.
Note
Leave parameter Route Partition selection as <None> (default partition]. Route partitions will
be covered later in this course.
Step 3
Note
Click Save.
Use the same procedure to configure additional lines if the phone has more than one line
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Note
Verify that the correct directory numbers are assigned to the IP phone lines.
The easiest way to verify the directory numbers of a phone is to check at the phone Itself or view the phone configuration in Cisco Unified Communications Manager.
Note
3-85
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1he figure shows an example of a phone listing (after performing a find and List Phones procedure from Device > Phone). Successful phone configuration can be verified by checking
the following items: Look at the Status column and veril\ that the phone is registered.
If it is shown as unregistered, it means that the phone has previously registered but is no longer registered. If a phone has been reset, it may be shown as unregistered during the short time until it reregisters with Cisco Unified Communications Manager. If it is shown as unknown, it means that the phone has never successfully registered to Cisco Unified Communications Manager If the phone is registered, its IP address will be shown in the
Status column
Note
Look at the IP Address column to verify that the IP phone is registered to the intended Cisco Unified Communications Manager server.
If all Cisco Unified Communications Manager servers are operating, the IP phone should
Note
register with the primary server of the IP phone Cisco Unified Communications Manager group The Cisco Unified Communications Manager server that the phone registered with is
shown by its IP address
Tip
By clicking the device name of a specific phone of the list, the phone configuration page of the corresponding phone is shown You can then verify line configuration (directory numbers) and other parameters that are not shown on the Find and List Phones result page.
The high-level steps for adding a third-party SIP phone are as follows: Step 1
Step 2
Step 3
Step 4
Configure the third-party SIP phone to register with Cisco Unified Communications
Manager.
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Q"
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Once the SIP phone is registering. this password will be used for digest authenlicalion. Digesl
Aulhenlication needs lo be enabled
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The steps that are performed in Cisco Unified Communications Manager when you add thirdpart} SIP phones are as follows: Step 1
Note
third-party SIP phone configuration is using a Device Security Profile with enabled digest
authentication.
Some third-party SIP phones do not have a separate User IDand Auth ID. In this case, the user ID has to be set to the directory number at the third-party SIP phone. On the Cisco Unified Communications Manager side, the end username has to be identical with the
directory number of the IP phone
Step 2
When adding a third-party SIP phone, you must specify the type ofthe phonebasic or advanced.
Note
Note
Step 3
In the Protocol Specific Infonnation pane ofthe Phone Configuration window. choose the end user that was configured in Step 1 from the Digest User drop-down
list.
Cisco Unified Communications Manager. The User Name has to match the directory number
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The final step to add a third-part} phone lakes place on fhe third-party phone itself. Therefore, the configuration depends on the product that is used. The example shows fhe configuration of the X-fiie third-party SIP softphone.
In the prox\ address field ofthe third-party phone, specify the IP address or fully qualified domain name of Cisco Unified Communications Manager. The User Name has to be set to the director} number that is assigned to the IP phone in Cisco Unified Communications Manager. The Authorization username lias to match the Digest User that was assigned to the phone. The password only needs lo be set if the Digest Credentials have been configured for the end user and if a phone security profile wilh checked "f.nable Digest Authentication" has been assigned.
Note If the Enable Digest Authentication check box is not activated in the phone security profile, only the username of the digest authentication is verified, but the password (Digest
A directory' number is an independent component within the Cisco Unified Communications Manager database. Once a directory number is associated with a device (IP phone), the device record in the Cisco Communications Manager database references to this directory number. A single director}' number can be associated with zero or more devices. If a director}' number is used with multiple devices, it is called a shared line. The director} number configuration consists ofthe following two types of parameters:
Director} number parameters that are shared with all associated devices. Examples of these parameters are Route Partition, Calling Search Space, Call Forward settings, and so on. Directory number parameters that are associated with a specific device. Examples of these parameters are Line Text Label, External Phone Number Mask, and so on.
- Phone configuration page (by clicking the line at phone configuration page) Directory Number configuration page (by selecting a device and clicking Edit Line Appearance] Shows shared settings and device-specific settings that apply only
to selected device
Changes to device-specific settings apply only to selected device (update of Line Appearance of selected device)
Different \\a_\s exist to view and edit the director} number con figuration: 1o \ iew and edit onh shared director} number parameters, open up the ("all Routing > Directory Number menu. Any changes that are made to these parameters apply to all
associated devices
To \ iew and edit both the shared and the device associated parameters, either choose the line configuration link at the phone configuration page or click the Kdit Line Appearance button from the Director} Number configuration page. Depending on the parameter that is being
changed, it might apply lo all associated devices or is specific to just one associated device.
The figure shows how to view or editall shared parameters as well as how to access devicespecific parameters by selecting one associated device andopening up the Line Appearance
configuration.
3-93
The same screen can be reached by clicking the directory number link at the Phone configuration page.
When opening up the LineAppearance configuration for a directory number, the configuration page alwa}S shows the associated device and allows you to go directly to the associated device configuration b\ selecting the related link.
3-94
Summary
This topic summarizes the key points that werediscussed in this lesson.
Summary
Some IP configuration settings are applied directly to the device, while other settings are applied by referencing configuration elements such as a device pool.
IP phone autoregistration automatically adds new Cisco IP phones to the configuration database and assigns one
directory number to the IP phone. Autoregistration configuration includes the configuration of a directory number range and activation of the feature on some
servers of a Cisco Unilied Communications Manager group.
The Cisco Unified Communications Manager Auto-Register Phone Tool requires a Cisco CRS server on the network.
* Cisco Unified Communications Manager BAT can be used to add and delete IP phones or to change their configuration. Manually adding IP phones is time-consuming.
References
for additional information, refer to these resources:
http:.;;w\vw.cisco.coni/cn/US/docs/voice_ip_comm/cucm/drs/8_0_ IAlrsag801.html.
Cisco Svstems. Inc. Cisco UnifiedCommunications System Release 8.x SRND. San Jose. California. April 2010. http:'www. cisco.eoni/en/US/docs/voiee_ip_comm/cucm/snid/8\/uc8\snid.pdf. Cisco Systems. Inc. Cisco UnifiedCommunications Manager BulkAdministration Guide, Release 8.0(1). San Jose, California, February 2010. http:'www\ cisco. coiii/cn/l.'S/partner'docs/voicc_ip__cc>mm/cucni/bat/8_0_l/bat-801cm.html
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3-96
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Cisco Unified Communications Manager supports a variety of endpoints. including Cisco IP phones and third-party phones.
SCCP, SIP, and H.323 can be used as signaling protocols to these endpoints.
Endpoints are configureddifferently based on protocol and vendor type (Cisco IP phones vs. third-party endpoints). Mass endpoint implementation can be simplified using the
Cisco Unified Communications Manager BATor Cisco Unified Communications Manager Auto-Register Phone Tool.
This module describes the endpoints that are supported by Cisco UnifiedCommunications
Manager. Itexplains the differences in thevarious Cisco IPphone models and third-party phones, fhe module further describes howto implement different endpoints in Cisco Unified Communications Manager manually, using the Cisco Unified Communications Manager Bulk Administration Tool (BAf) or the Cisco Unified Communications ManagerAuto-Register
Phone Tool.
References
For additional information, refer to these resources:
lHtp:.',ww\\.cisc{i.eiitn/en/lJS/d()cs/voice_ip_eomm/cucm/drs/8_0_l/drsag80l .html.
Cisco Systems. Inc. Cisco Unified Communications System Release 8.x SRND. San Jose.
California. April 2010.
litlp:''www.cKco.com/en/US/docs/voice ip comm/euem/srnd/8\/uc8\srml.pdr. Cisco Systems. Inc. Cisco Unified Communications Manager Bulk Administration Guide.
Release 8.0(1). San Jose. California. February 2010.
http:.'wv\vv.ci^co.com/en/US/partner/docs/voice ip_comm/cuem/bal/8_0_l/hat-80lcm.himl
3-97
3-98
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. Ql) Which endpoint is not supported by Cisco Unified Communications Manager? (Source: Understanding Endpoints in Cisco Unified Communications Manager)
A)
B) C) D)
Q2)
H.323 phones third-party SIP phones SCCP phones Skype software client
Which list of tasks best describes the boot process of a Cisco IP phone? (Source:
C)
D)
third-party SIP
H.323
Q4)
Which two items of infonnation are provided by a third-party SIP phone during registration with Cisco Unified Communications Manager? (Choose two.) (Source: Understanding Endpoints in Cisco Unified Communications Manager) A)
B) C) D)
directory number
MAC address username X.509 certificate
Q5)
Which two settings are not configurable at a device pool? (Choose two.) (Source: Implementing IP Phones) A) B) C) D) F) F) Softkey Template Media Resource Group List Cisco Unified Communications Manager Group DatcATimc Group Phone Button Template Region
Q6)
Which two statements do not apply to the autoregistration feature? (Choose two.) (Source: Implementing IP Phones)
A) B) Each autoregistered phone is added twice: once with SIP and once with SCCP. Only one directory number can be assigned per phone.
C)
D)
3-99
Q7|
. (Source:
Q8)
Which two components and features are not used by fhe Cisco Unified
Communications Manager Auto-Register Phone fool? (Choose two.) (Source:
Implementing IP Phones)
A) CRS
application plug-ins autoregistration Cisco Unified Communications Manager BAT Cisco Unified Communications Manager Extension Mobility
Which ofthe following is not a step in adding phones with the Cisco Unified
Communications Manager Bulk Administration Tool? (Source: Implementing IP
Phones)
A) B) C) D)
Upload a phone template. Start a Cisco 1(nitied Communications Manager BAT job Lo add phones. Configure a phone template. 1upload a CSV data input file.
QIO)
Which three of the following must be specified when adding a phone manually?
(Choose three.) (Source: Implemenling IP Phones) Al B) C)
i
F) l:] G)
3-100
C D A.C A. H
A.D
C B, L
A
A, B, D
3-101
3-102
Module 4
that provides endpoint addressing, path selection, calling privileges, digit manipulation, and call
Module Objectives
Upon completing this module, you will be able to implement PSTN access in Cisco Unified
Communications Manager and tobuild adial plan in asingle-site Cisco Unilied Communications Manager deployment. This ability includes being able to meet these
objectives:
Describe the implementation ofPSTN gateways in Cisco Unified Communication Manager Describe and configure Cisco Unified Communications Manager numbering plans, director} numbers, route groups, route lists, route patterns, digit analysis, and urgent
priorit} foron- andoff-net calls
Explain the need and uses for calling privileges and how to implement them in Cisco
Unified Communications Manager
Explain the need and uses for gateway selection and PSTN-access features and how to
implement them in Cisco Unified Communications Manager Describe call coverage and how to implement itin Cisco Unified Communications
Manager
4-2
Lesson 11
using different protocols such as Media Gateway Control Protocol (MGCP), H.323, or Session
Initiation Protocol (SIP) for signaling on VoIP call legs.
The purpose of this lesson is to describe the role and implementation of MGCP, H.323. and SIP gateways to provide PSTN access to aCisco Unified Communications Manager environment.
Objectives
Upon completing this lesson, you will be able to describe the implementation of PSTN gateways in Cisco Unified Communications Manager. This ability includes being able to meet
these objectives:
Describe the types of gateways that can interact with Cisco Unified Communications
Manager, anddescribe theirdifferences
Describe how to integrate MGCP gateways with Cisco Unified Communications Manager Describe how to integrate H.323 gateways with Cisco Unified Communications Manager Describe how to integrate SIP gateways with Cisco Unified Communications Manager
Gateway Review
teleph lephoin infrastructure.
his topic describes the importance of Cisco access gateways in the overall design ofthe IP
Gateway in an IP T
Headquarters
Branch Office
WAN-
PSTN
Agateway is adev ice [hat can translate between different types ofsignaling and media. One
type of gateway is the voice gateway. Avoice gateway isa router orswitch that converts IP
dev ices such as an analog phone or fax.
Note
voice packets to analog or digital signals that trunks or stations understand. Voice gateways are used msc\ eral situations: for example, connecting to a PSTN or PBX, or connecting indiv idual
This lesson provides anoverview ofthe voice gateways that you can usewith the Cisco
Unified Communications Manager system and describes their basic configuration. For more
information about configuring voice gateways, refer tothe implementing Cisco Voice
Communications andQoS (CVOICE) course.
ip
Telephony
Service
Provider
Analog Trunk
PSTN
DigitalTrunk
Access analog stationgateways: Access analog station gateways connect Cisco Unified Communications Manager to plain old telephone service (POTS) analog telephones, interactive voice response (IVR) systems, fax machines, and voice-mail systems. Station gateways provide Foreign Exchange Station (FXS) ports for connecting to analog devices suchas telephones and faxes.
Access analog trunk gateways: Access analog trunk gateways connect Cisco Unified Communications Managerto PSTN central office (CO) or PBXtrunks. Trunk gateways provide Foreign Exchange Office (FXO) ports for PSTN or PBX access and E&M ports for analog trunk connection to a legacy PBX. (E&M ports are known by various names, primarily recEive and transMit, earand mouth, or F.arth and Magneto.) Analogdirect inwarddialing(DID) is also available for PSTN
connectivity.
Cisco access digital trunk gateways: A Ciscoaccess digital trunk gateway connects Cisco Unified Communications Managerto the PSTN or to a PBXvia digital trunks, such as PRI common channel signaling (CCS). BR1,T1 channel associated signaling(CAS),or El. Digital T1 PRI trunks canalsoconnect to certain legacy voice-mail systems.
- DTMF relay: Signaling method that uses specific pairs of frequencies within the voice band for signals Supplementary services: Services that provide user functions, such as hold, transfer, and conferencing
Cisco Unified Communications Manager redundancy: Secondary Cisco Unified Communications Manager system that picks up control of all gateways initially managed by the primary Cisco Unified Communications Manager system
Call survivability: Preservation of a voice conversation between two IP endpoints when the Cisco Unified Communications Manager system to which an endpoint is registered is no longer
reachable
Dual tone multifrequency (DTMF) relay capabilities: DTMF signaling tones must be
processed, (iatewavs must separate DTMF digits from the voice stream and then send the call signaling in VoIP signaling protocols such as H.323. Cisco IOS Software MGCP, or
SIP.
Supplementary services support: fhese services are typically basic telephony functions,
such as hold, transfer, and conferencing.
Communications Manager clusters provide for Cisco Unified Communications Manager redundance The gateways must support the ability to re-home to a secondary Cisco Unified Communications Manager system if a primary Cisco Unified Communications Managers} stem fails, 'fhis t_vpe of support differs from call survivability during a Cisco
Unified Communications Manager or network failure.
Call survivability in Cisco Unified Communications Manager: Ihe voice gatewav preserv cs the Real-Time Transport Protocol (RTP) bearer stream (Ihe voice conversation) between two IP endpoints when the Cisco Unified Communications Manager system to which an endpoint is registered is no longer reachable.
An.v IP telephony gatewav1 that you select for an enterprise deployment should support these core requirements. Additionall}. even IP telephony implementation has sitc-speciflc feature requirements, such as analog or digital access. DID. and capacity.
4-6
MGCP: Client/server; simplified configuration, inwhich Cisco Unified Communications Manager maintains the dial plan
and route pattern
SCCP: Client/server; simplified configuration, in which Cisco Unified Communications Managermaintains the dial plan
and route pattern
H.323: H.323 uses apeer-to-peer model. You perform most ofthe configuration through
Cisco IOS Software on the voice gateway device. With the peer-to-peer model. Cisco Unified Communications Manager has no control over the gateway; this lack ofcontrol limits the Cisco Unified Communications Manager feature support on H.323 gateways. For
example, only devices that support H.323 version 2(H.323v2) can take advantage of Cisco
Unified Communications Manager supplementary services such as hold, transfer, and
conference features. However. H.323 gateways support additional Cisco IOS features
outside ofCisco Unified Communications Managerthat the other gateways do not. such as Call Admission Control (CAC) and Cisco Unified Survivablc Remote Site Telephony
(SRST). Examples ofCisco gateway devices that support H.323 include the Cisco Voice Gatcwav 224 (VG224) Analog Phone Gateway (FXS only) as well as Cisco 2600 Series
Multiservice Platforms: Cisco 2800.2900, 3800, and 3900 Series Integrated Services
Routers: and Cisco 3700 Multiservice Access Routers.
MGCP: MGCP uses a client/server model, with voice-routing intelligence that resides ina
call agent (the Cisco Unified Communications Manager). Because of its centralized
architecture. MGCP simplifies the configuration ofvoice gateways (the gateway requires
fallback). Ifthe MGCP gateway loses contact with Cisco Unified Communications
no dial peer configuration) and supports multiple (redundant) call agents in anetwork. MGCP gateways provide call survivability (the gateway maintains calls during failover and
Manager, the gateway falls back to using 11.323 control to support basic call handling of
FXS. FXO. Tl/El CAS. and Tl/EI PRI interfaces. Examples ofCisco gateway devices
that support MGCP are the VG224 Analog Phone Gateway (FXS only) as well as Cisco
2600 Series Multiservice Platforms and Cisco 2800, 2900, 3700. 3800 and 3900 Series
routers.
SIP: Ihe Internet Engineering Task Force (IETF) developed the SIP standard for
multimedia calls over IP. ASCII-based SIP works in client/server relationships as well as in
peer-to-peer relationships. SIP uses requests and responses to establish, maintain, and terminate calls (or sessions) between two ormore endpoints. Cisco Unified Communications Manager supports both SIP trunk and SIP line sides with voice media.
implementation. As with other protocols. SIP components lit under the device layer ofthe Cisco Unified Communications Manager architecture. As is true for the 11.323 protocol,
multiple logical SIP interfaces can be configured in a Cisco Unified Communications Manager database and can be associated with dial-plan elements.
SIP gatewav sare supported in Cisco Unified Communications Manager via SIP trunk
Skinny Client Control Protocol (SCCP, known also as Skinny): SCCP is aclient/server
protocol that uses Cisco proprietary messages tocommunicate between IP devices and
from Cisco 1inilied Communications Manager. Afler the phone registers, it is notified of new incoming calls and can make outgoing calls. SCCP is used for VoIP call signaling and
for enhanced features such as message waiting indication.
client. During registration, aCisco IP phone receives its line and all other configurations
Cisco Unified Communications Manager. The Cisco IP phone is an example ofadevice that registers and communicates with Cisco Unified Communications Manager as an SCCP
depends on site-specific requirements and your installed base ofequipment. You might prefer
MGCP lo 11.323 because ofthe simpler configuration ofMGCP. Or, you might prefer H.323 to
MGCP because ofthe interface robustness of 11.323 or the ability to use it with CAC or Cisco
Unified SRST.
Most gatewav devices support multiple gatewav protocols. Selecting the protocol to use
4-8
MGCP
Dumb
H.323
SIP '
Intelligent
implement
TCPandUDP
ASCII
No Yes*
implement
TCP
Signaling protocol
Code basis
Binary (ASN.1)
Yes
Yes
Call survivability
FXO caller ID
Call applications
usable
No
Yes
Yes
"fhe following table provides an overview ofthe features and functions that each signaling
protocol provides.
H.323
SIP
Intelligent
Not supported
Supported More effort to implement TCP and User Datagram
Supported
Not supported Easy to implement
TCP or UDP
Q Signaling (Q SIG)
Fractional T1/E1
Signaling protocol
Code basis
Protocol (UDP)
ASCII
ASCII
Call survivability
FXO caller ID
No
Yes* No
Yes
4-9
Note
MGCP FXO caller ID support was introduced with Cisco Unified Communications Manager
Version 8 0
Protocol Comparison The figure compares the advantages and disadvantages of H.323, MGCP, and SIP gateways
Protocols Compares
Centraiized dial plan
configuration
Centralized gateway
configuration
per gateway
Regional requirements can be
met
Stmple gateway
Pros
per gateway
configuration
Third-party telephony
system support
Easy
implementation
support
Complex configuration
f.ach ofthe three gateway protocols has adv anlages and disadvantages when compared tothe others. 1here isno best gatew av protocol. The most appropriate protocol depends onthe individual needs and demands hi a Cisco Unified Communications Manager environment.
Note
The Implementing Cisco Voice Communications and QoS (CVOICE) course provides detailed information about functions and features ofthe H.323, MGCP, and SIPprotocols.
4-10
MGCP Gateways
is defined under RFC 2705and is a master-slave protocol Allows a call-control device (such as Cisco Unified Communications Manager) totakecontrol ofa specific porton a gateway Provides centralized gateway administration and highly scalable gateway
solutions:
Use ofplain-text commands between Cisco Unified Communications Manager and the gateway, overUDP port2427 * Requires gateway tobesupported by Cisco Unified Communications
Manager
MGCP is aplain-text protocol that call-control devices use to manage IP telephony gateways. MGCP (defined under RFC 2705) is amaster-slave protocol that allows acall-control device,
such asCisco Unified Communications Manager, to take control of a specific gateway port,
litis approach has the advantage ofcentralized gateway administration and is used for largely
scalable IP telephony solutions. With this protocol, the Cisco Unified Communications
Manager knows and'controls the state ofeach port on the gateway. MGCP allows complete
control ofthe dial plan from Cisco Unified Communications Manager. MGCP also gives Cisco
Unified Communications Manager per-port control ofconnections tothe PSTN, legacy PBX. voice-mail svstems. POTS phones, and so on. MGCP isimplemented by a series ofplain-text commands that are sent over UDP port 2427 between the Cisco Unified Communications
Manager and the gateway.
For an MGCP interaction to take place with Cisco Unified Communications Manager, the
gateway must have Cisco Unified Communications Manager support. Use the Cisco Software
Adv iso'r tool to make sure that the platform and version ofCisco IOS Software or Cisco
CataKst operating system are compatible with Cisco Unified Communications Manager for
MGCP.
4-11
Endpoint Identifiers
Ihis subtopic describes how identifiers are associated with an endpoint.
FXS VWIC
2/1/1
4x
-*
When interacting with agateway, the call agent directs commands lo the gateway to manage an
endpoint ora group ofendpoints. As its name suggests, an endpoint identifier identifies
endpoints.
Endpoint identifiers consist of two parts: the local name ofthe endpoint (in the context ofthe
gatewav) and the domain name ofthe gatewav. fhe (a] separates the two parts. Ifthe local pail represents a hierarchv. then a slash (/) separates the subparts ofthe hierarchy. In the figure, the local ID might represent a particular gatewav/circuit number; the circuit number might in turn
represent a circuit ID/channel number.
4-12
MGCP and SCCP Interaction litis subtopic describes the interactions between Cisco Unified Communications Manager.
SCCP phones, and an MGCP gateway.
Communications
Manager
Manager.
Cisco Unified Communications
PSTN
Gateway
Manager is the master server for both protocols. The interactions arc as follows: IP phones communicate directly with Cisco Unified Communications Manager for all callsetup signaling.
MGCP gateways communicate directly with Cisco Unified Communications Manager for
all call-setup signaling.
Actual \oice traffic Hows directly between the IP phone and the MGCP gateway through
RTP over UDP.
4-13
MGCP support in Cisco Unified Communications Manager includes awide range ofanalog
and digital interfaces that can be used on several Cisco IOS router platforms. Cisco Unified Communications Manager allows fhe Cisco IOS MGCP gateway topull its
MGCP-related configuration from the CiscoTFTP server, fhis feature eliminates the need for manual MGCP gatewav configuration.
Cisco Unified Communications Manager also supports PRI backh.auling, which is supported on
ISDN PRI. With PRI backhauling. the MGCP call agent (Cisco Unified Communications Manager) takescontrol ofthe ISDN data channel (D channel).
4-14
Manager
Administrator configures MGCP gateway
in Cisco Unified Communications
I Manager.
Csco Unified
PSTN
apples MGCP
configuration
When using the Cisco Unified Communications Manager configuration server feature. Cisco
Cisco IOS gatewav downloads from the Cisco Unified Communications Manager TFTP server.
configuration commands. These commands take the form ofan XML configuration file that the This approach is recommended to integrate Cisco IOS MGCP gateways with Cisco Unified
Communications Manager.
stored in the centralized TFTP directory. Atailored XML file can be created and downloaded
from the TFTP server to a designated MGCP gateway.
Each MGCP gateway in the network has an associated, gateway-specific configuration that is
When changes are made to the configuration in the Cisco Unified Communications Manager
database. Cisco Unified Communications Manager sends a message to the affected MGCP
gateway This message instructs the gateway devices to download the updated XML configuration file. F.ach device has an XML parser that interprets the XML file according to device-specific requirements. Cisco MGCP gateways, for example, translate the content ofthe
XML file into specific Cisco IOS commands for local execution.
4-15
PRI Backhaul
This subtopic explains PRI backhaul, which is an important concept in implementing ISDN
PRI on an MGCP gateway.
PRI Backhaul
1 D-channel signaling is carried
in raw form back to the Cisco Unifiec Communications
Cisco Unified
Communications
Manager
Gateway
Manager to be processed. Gateway terminates data link layerand passes Layer 3 signaling (Q 931) to Cisco
Unified Communications
Communications Manager
APR) backhaul is an internal interface between Cisco Unified Communications Manager and Cisco MGCP gateways. In other words, a PRI backhaul is aseparate channel for backhauling
signaling infonnation. This channel forwards Layer 3 PRI (0.93 3) backhaulcd over a TCP connection. I.aver 3 infonnation is forwarded independent ofthe native protocol that is used on
the PSTN time-division multiplexing (TDM) interface.
A PRI isdistinguished from other interfaces in thai the data that ihe PS'fN receives on the D
channel must be carried in its raw form back lo the Cisco Unified Communications Manager, to be processed. The gateway docs not process orchange this signaling data, but simply passes the data on to the Cisco 1'nified Communications Manager through TCP port 2428. The gateway is
still responsible for the termination ofthe I.aver 2data. All the 0-921 data link layer connection protocols are terminated on the gatewav. However, everything above that (Q.931 network layer data and beyond) is passed onto the Cisco Unified Communications Manager.
Also, the gateway doesnot bring up the D channel unless it cancummunicale with Cisco
Unified Communications Manager to backhaul the 0.931 messages that are contained in the D
channel. The figure illustrates these relationships.
4-16
2
3
4.
MGCP gateway implementation includes configuration steps on both Cisco Unified Communications Manager (the MGCP call agent) and the MGCP gateway that will be
controlled.
The steps to configure an MGCP gateway differ depending on the type ofMGCP gateway
platform that is selected.
The high-level Cisco Unified Communications Manager configuration steps for implementing
an MGCP gateway are as follows:
Step 1 Step 2
Add the MGCP gateway toCisco Unified Communications Manager. Configure the MGCP gateway in Cisco Unified Communications Manager.
Step 3
Step 4 Step 5
Add one or more voice modules to the slots ofthe MGCP gateway in Cisco Unified
Communications Manager.
Add voice interface cards (VICs) tothe configured modules. Configure the MGCP endpoints (one ormore perVIC).
4-17
Manager Administration,
choose Device >
Gateway
Click the Add New
button. The Add a
Follow these configuration steps lo add an MGCP galevvav to Cisco IInilied Communicalions
Manager:
Step 1
Step 2 Step 3
Step 4
Click the Add New button. The Add a New Gateway window appears. From the Gateway Type drop-down list, choose the appropriate MGCP gatewav.
Click Next.
4-18
~m^ZZZHIII
! Chsnfi*Gateway,f*L
- Not Bdertei_--.
- -- Not Selected -
Step 5
Ifa Protocol drop-down list appears, choose MGCP, and then clickNext. Some gateways support SCCP and MGCP. The Protocol list appears only when you add
such a gateway.
Note
4-19
1 hostname (case
>Oi sensitive)
' Ti 3
.< SV
-. a-;
I "-=""' r..a-r,-
l%\ 1
l-C . Heijc , . ,
1.j.
,.=-
j1-"-'
2 Choose Cisco
Unified
Urrfe, L'
6 UarJ(e. C,
"""-"" 'i"-"J
"p
u;,
Conlcaured you
Communications
Manager Group
Cisco 2811 Integrated Services Router isbeing used. To configure an MGCP gatewav. follow these steps:
Fhe configuration ofan MGCP gatewav depends on the selected platform. In this example, a
Fntei- the name ofthe gateway in the Domain Name field. Fhe name must match the
hostname ofthe Cisco IOS router.
Choose a Cisco Unified Communications Manager group. Choose the installed network module type from the appropriate Module in Slot
drop-down list.
Step 4
Note
different value per ISDN interface The global ISDN switch type is part ofthe gateway configuration The interface-specific switch type can be configured at the MGCP endpoint
Step 5 Click Save when vou finish the gatewav configuration.
Tip
To display help for the configuration parameters, click the question mark (?) symbol.
Note
Sutunrt 1 ^ None *
Sutunrt ', c Wane >
Subuml 1 VW1C2-1MFT-T1E1-E1 Subunit 2 Subunrt 3 Hod.ils n Slot 1 ^ narw s < None > < None j
Sutunil J
configuration.
r-TlEl-Tl
'/wi;i-iFT-T[Ei-ei
VlC2-lMfT-TlEI-Tl
Vir.2-1HTT-TIEI-El
Use the show inventory command on Ihe routerlo find the VIC name.
To add endpoints. you select voice modules and VICs at the Gateway Configuration page. To
addendpoints to a gateway, follow these steps:
Step 1
Step 2
Locate the Configured Slots, VICs, and Endpoints section, in which the available
slots are listed for the displayed gateway. Choose the installed VIC from the drop
down list. Click Save to continuewith the endpointconfiguration.
The endpoint identifier (O/l/O) for the selected VIC appears. Click the endpoint icon to start the endpoint-specific configuration. Also, you should verify that the endpoint
identifier in Cisco Unified Communications Manager is identical to thecontroller or
interface ID on the Cisco IOS gateway.
Tip
Use the Cisco IOS show diag or show inventory command at the gateway, to display the
modules and interface cards with which the gatewayis equipped. Cisco Unified Communications Manager lists modules and interface cards according tothe product
number (or field-replaceable unit [FRU]), which ispart of the output of the show diag
command.
Note
If you use VICs that support T1 and E1 protocols, the implemented interface type must be configured on the Cisco IOS gateway before you configure the MGCP settings on Cisco
Unified Communications Manager. Use the card type command in global configuration mode tospecify whether E1 orT1 should be used.
4-21
1.W
lI;--"K,r
ri cla.. v i . t '
.-1.11
iBitt:;ji
?7fls-
-,....,.,.j^',,r'
'"
1 Enler enapoint
description and select device pool
...
"\
2 Verify ISDN PRI interface
. , - ,....,.
..>
After vou add voice modules and VICs in the Configured Slots. VICs and lindpoinls section of the gatewav configuration, the endpoints ofthe VICs are displayed. To configure an MGCP endpoint. follow these steps: Step 1 Step 2
Note
[inter a description and choose the appropriate device pool for the endpoint. In the Interface Information section, verily orchange the interface-configuration
parameters, as neccssarv.
Tip
3
Ei>abl
EchnCanceflaiicr.Ccve'aga <ma(' 6a
4. Verifyproduct-specific
Step 3
Note
Verify or change the configuration parameters in the PRI Protocol Type Specific
Information section.
Step 4
Tip
To display help for the product-specific configuration parameters, click the question mark (?)
symbol.
Note
4-23
Ouliiounri Calls
Ctfau:
Crirjma-"r Cls;3 - aN,
\\
5 Venfycall routingspecific configuration
anrj change if required
, j,or ,
-. i -..- ,
Step 5
Step 6
Note
Verifv or change the con11goralion settings inthe Call Routing Infonnation sections.
Click Save when vou finish the endpoini configuration.
Repeat Steps 1 through 6 for each endpoint
4-24
Configuration Methods
Different ways exist to configure MGCP on the Cisco
IOS gateway:
* Configuration server
Manual configuration
After vou add the MGCP gateway in Cisco Unified Communications Manager Administration, vouneed toconfigure the Cisco IOS MGCP gateway to register to Cisco Unified Communications Manager. There are three methods ofconfiguring a Cisco IOS Software
If more than one Cisco Unified Communicalions Manager TFTP server is deployed in die CiscoUnified Communications Manager cluster, configure the gateway with all Cisco Unified Communications ManagerTFTP server IP addresses.
Enable the configuration server feature.
Specify the IP address ofthe MGCP call agent (the Cisco Unified Communications
Manager server).
processing (that is. for running the Cisco CallManager service), configure the gateway with aprimary and redundant call agent. To do so. specify the IP addresses
of two Cisco Unified Communications Manager call-processing servers.
IfMGCP is to control FXS orFXO interfaces, use the service mpcpapp command
to enable MGCP on the corresponding POTS dial peers.
finable MGCP.
Note
I st' both a configuration server and manual configuration: Use a configuration server to configure the MGCP gateway.
Disable the con figuration server by using the ccm-manager config command.
Note
Manual!) remove the configuration that is received from the configuration server, or
add additional confi curat ion.
As long as the configuration server isactive onthe Cisco IOS gateway, the Cisco IOS
configuration will be rewritten everytimethe MGCP endpoint is reset from Cisco Unified Communications Manager. Also, as long as the configuration server is enabled, the MGCP configuration will be rewritten each time the MGCP gateway reloads.
Step 2
Step 3 Slep4
Step 5
Specify the IP address of the MGCP call agent (Cisco Unified Communieaficm Manager).
Enable MGCP.
oontroilei:
Steps Step 7
Step 8
Using the configuration server isthe fastest way toconfigure the Cisco IOS MGCP gateway.
However, additional configuration, which canbe applied by disabling the configuration server
and manually adding the necessary commands, might be required. Manual configuration offers the greatest flexibility but requires deeper knowledge ofthe Cisco IOS MGCP-relatcd
commands.
Gateway hostname must match name specified in Cisco Unified Communications Manager gateway configuration.
router(config)#ccm-manager config server <CM TFTP TP>
router(config)#can-manager config
Two commands arcrequired for a Cisco IOS MGCP gateway lo pull its MGCP configuration from a configuration server (a Cisco Unified Communications Manager TITP server).
The ccm-manager config server {IF address j list oj IF addresses) command specifics the IP address ofthe TFTP configuration server (the Cisco Unified Communications Manager TFTP server). If more than one Cisco Unified Communications Manager TFTP server is deployed in the cluster, vou can specify a listof IP addresses (with a space between each IP address). The
Cisco IOS MGCP gatewav will in the IP addresses in ihe specified order.
The ccm-manager config command enables the configuration server feature. Unless this command is entered, ihe ccm-manager config server command is ignored. For the configuration feature to work, the following prerequisites musl be met: IP connectivity between the MGCP gateway and the Cisco Unified Communications
Manager TFTP sener or servers
Ifall these conditions aremet and the gateway is configured with the ccm-manager config and the ccm-manager config server commands, the gatevvaj can download its XML configuration
file from the TFTP serv er.
Note
The name of the configuration file is n.cnf.xml, where nis the hostname of the Cisco IOS MGCP gateway; for example, HQ-1 cnf.xmt for agateway with the hostname HQ-1
The gatewav then parses the XML file, converts the information to appropriate Cisco IOS
configuration commands, and configures itself for MGCP operation.
The gateway then uses the MGCP protocol to register with Cisco Unified Communications
Manager.
After asuccessful configuration download, the MGCP gateway saves the running configuration to NVRAM which updates the startup configuration. Any manually added, previously unsaved configuration parameters are also saved to NVRAM. Manually added configuration parameters
are updates to the configuration that you made by using the command-line interface (CLI).
4-29
The example shows one Cisco (. nified Communicalions Manager server (providing call processing and 1FTP sen ices) with IP address 10.1.1.1. ACisco IOS MGCP gateway connects to the PSTN by using an fc 1interface (port 0/1/0). "1 hegateway and itsF1 PRI endpoint are added to Cisco Unified Communications Manager. At die gateway, the ecm-manager config
server 10 1.1.1 and ccm-manager config server commands are entered. No MGCP
configuration commands are entered manual!} because the configuration server feature automatical!} downloads and applies the MGCP configuration.
Afterthe gatewav downloaded its cnf,\ml configuration file from the Cisco Unified Communications Manager TTIP server, the following MGCP commands are addedand saved
to NVRAM:
controller El 0 :1 >'Q
framing crc4
linecode hdb3
ccm-manager music-on-hoid
mgcp
4-30
4-31
Communications Manager.
Configure the MGCP gateway router manuallyor remove the ccm-manager config command and modifythe downloaded
configuration.
In some situations, not all time slots of a Tl or F.I connection are lobe used. This type of PR!
Is called fractional Tl or fractional F I.
You can specifv the numberof usable li channels in Cisco Unilied Communications Manager,
bysetting the Cisco CallManager service parameter Change B-Channel Maintenance Status 2-5 . As mam as live PRI endpoints can beconfigured to have B channels in maintenance status. However, this setting has noeffect on the XMI. configuration file that is received through the MGCP configuration server. Therefore, the PKl group on the Cisco IOS MGCP
gatewav will alwavs allocate the maximum numberof R channels that are available for a
specific controller tvpe.
Io configure fractional I1or L1onthe Cisco IOS gateway, use manual configuration or
disable the configuration server and manually reconfigure the PRI group on the corresponding
Tl or Fl controller.
Note
The maximum number of PRI group Bchannels depends onthe number ofinstalled digital
signal processors (DSPs) on the Cisco IOS gateway Fractional T1 or E1 might also be used toshare DSP resources with other functions, such as transcoding orconferencing, rather
than allocating the resources solelyto PRIgroups.
4-32
gateway
interface ID.
r Interface lirfortiwtwn-
To put specific time slots ofan MGCP Tl orEl PRI into maintenance state, you need to
retrieve the MGCP endpoint ID (for example, SO/SUl/DSl-0@HQ-l) and check the Enable
Status Poll check box in the Interface Information configuration section.
Note
Use your operating system Copy and Paste functions toavoid spelling errors in the endpoint
ID.
Single-Site Off-NetCalling
4-33
Use"1" todisablea
B channel and "0"
to keep it enabled
The service-parameter
value starts with the
MGCPT1/E1 interface ID
From Cisco 1inilied Communications Manager Administration, choose System > Service
Parameter, choose Cisco CallManager Sen ice, and click fhe Advanced button to view the
This parameter allows Cisco l'nified Communications Manager to changeindividual B-channel maintenance status for PR! andCAS interfaces in real time (for use In troubleshooting).
The input format ofthe parameter is device name ~ B-channelmaintenance status.
The device name, as specified in the Gatewav Configuration page in Cisco Unified Communications Manager Administration, must match the specified gateway name. To avoid
an\ manual-Input mistake, copv the device name from Cisco Unified Communications
Manager Administration and paste It into this service parameter. 'Fheequals sign (--) is
mandator} and unique and distinguishes the device name from the R-channcl maintenance
status.
The B-channel maintenance status takes the form "x\\x xxxx xxxx xxx\ xxxx xxxx x\\\ \\\\. where \ can be one ofthe three digits:
0: In serv ice 1: Graceful out of serv ice 2: Forceful out of service
Graceful out of sen ice changes channel status until the active call ends, ifan active call exists
on that channel. Forceful out of service tears down the activecall first, then changes channel
status immediately, ifan active call exists on that channel.
The svstein ignores any values other than 0. I. or 2 because such values are invalid. Make sure
that the total number of digits is either 24 (for Tl) or 32 (lor hi). Any other length or a
mismatch (such as 24 digits for 1 1) is treated as an error, and no action is taken tor that dev ice.
4-34
The spaces between the device name and the equals sign, between the equals sign and B-
channel maintenance status, and between sets ofdigits within the B-channel maintenance status are all optional. You cannot use any spaces within the device name.
0000 0001." This entry has 24 bits (for Tl). The channel number begins with bit I and goes to bit 24. from left to right. In the case ofPRI, the last bit specifics the Dchannel, which is
unaffected. In this example, the fifth through eighth Bchannels are marked as out of service.
Another example would be 4"S1/DS1-0@DLS2-CM136-VG2002SLI .SllJSCMLAB.CISCO.COM - 0000 0000 0000 0000 0000 0000 11111111."' This
For example, consider the entry "S0/DSI-0@SDA123456789ABC =0000 1111 0000 0000
example shows 32 bits (for El). The Bchannel number begins with bit 1and goes to bit 32, from left to right. The 16th bit specifies the Dchannel. The last bit does not affect any channel.
In this example, the 21 st through 31 st channels arc marked as out ofservice. To use this feature, check the Enable Status Poll check box in the PRI Gateway Con figuration window. Then, click Update, and reset the gateway so that the service-parameter change will
take effect.
4-35
To change the PRI-group configuration to fractional Tl or hi. complete the following steps: Step 1 Ifthe configuration server was enabled, use fhe no ccm-manager config command
to disable it before changing any MGCP-related configuration.
Step 2
A PRI group canbe altered only if the corresponding voice port is placed in
shutdown state. To do so. use fhe shutdown command onthe corresponding voice
port.
Step 3
Step 4
Move to the corresponding T1 orE1 controller and put it into the administratively
down state, byusing the shutdown command.
Single-Site Off-NetCalling
4-37
Create and enable a new PRI group for the fractional T1/E1.
controller El 0/1/0
Step 5
Step 6
Create anew PRI group for the fractional Tl/HI. by using Ihe pri-group timeslots
1-4,16 sen ice mgcp command: use the no shutdown command lo enable the PRI
group.
Step 7
Re-cnable backhauling on the corresponding serial interface, by using the isdn bind13 ccm-manager command.
be "MULTIPLE FRAME
ESTABLISHED."
1. SAPI -
0,
State
-1MULTIPLE
FRAME ESTABLISHED |
An easy way to check the operation ofan MGCP-controlled Tl/El interlace isby using the
show isdn-status command andchecking the Layer 1 and Layer 2 status. If the interface is
4-39
Manager.
If the connection between Cisco Unified
All active calls are dropped (no call survivability}. Complete local dial plan configuration must be
configured.
Because of the client. ser\er architecture of MGCP, a constant connection must bepresent between the Cisco IOS MGCP gatewav and Cisco Unified Communicalions Manager. If the connection between Cisco Unified Communications Manager and the MGCP gateway is
unavailable, the following can happen:
If the gatewav was configured for failover (by using the ccm-manager fallback-mgcp
command), the MGCP gateway can fail over to local call control. All active calls are
dropped (there is no call survivability), anda eomplete local dial plan must be present. If no failover configuration is present, all callsare dropped, and the PRI interface goes
down.
4-40
The voice stream (RTP) occurs directly between the H.323 gateway and Cisco IP phone.
Cisco Unified Communications
Manager
H323
T1/E1 VWIC
-PSTN-
This figure demonstrates how Cisco Unified Communications Manager and H.323 gateway configurations relate to each other. In the figure, the voice-enabled router isthe H.323 gateway
that connects Cisco Unified Communications Manager to the PSTN by usinga digital (Tl/El)
voice WAN interface card (VWIC).
When calls aremade from the IPphone to the PSTN, thedial plan on Cisco Unified
Communications Manager must direct those calls to the H.323gateway.
1'hc following two sets of steps are required to configure an 11.323 gateway:
lo conligure an H.323 gatewav inCisco Unified Communications Manager, follow these steps:
Step 1 Add an H.323 gatewav.
Step 2
To configure H.323 gatewav functionality at the Cisco IOS router, follow these steps:
Step 1 Configure basic Cisco IOS II.323 functionalitv.
Step 2
Step 3
4-42
Communications Manager
Administration, choose
!
. , .
Cam 3B45
Cites 392 S
Cnco 3?<5
Cisco 68 i
Follow these steps toadd anew H.323 gateway loCisco Unified Communicafions Manager:
Step 1 Choose Device>Gateway from the menu.
Step 2
Step 3
Step 4
Click Add New toadd anew H.323 gateway toCisco Unified Communications
Manager.
4-43
Sottinqs
name of the H 323 gateway The device name must be unique for each configured H 323 gateway
Step 5
Inter the H.323 gatewav IP address or name (which must be resolvable bv using DNS) to the Device Name field. Fnter a descriptive name in the Description field
(this name is optional).
Step 6
Step 7
from the Device Pool drop-down list, choose thedevice pool to which thisgatewav
should belong.
Click Save.
4-44
5T.VH t>a<s"
Cak-*; 5F=J'5tce
A
m >
*<K*>
X
-
Verifyparameters thai
relate to inbound calls.
-cj#ltovtt>f iJrfDmutKH
MMtW
nannr Orf.J i
CflfcnflPffT. ^01^'
untac-f"
CB1 <:**-??'
CcMar-iper
r
CriHanarjT'
Verifv the necessary configuration settings for the added H.323 gateway.
Note Settings thatare shown in the figure will be explained inlaterlessons.
4-45
iSh<
After communication between the II.32.3 gateway and Cisco Unified Comnuiniealions Manager takes place, the 11.323 gateways source IP address is displayed on the Cialewav Configuration
page. Be aware that an 11,323 gatewav will never register with Cisco Unified Communications
Manager, so the registration status will always be Unknown.
It red commands are not configured, the IP address of the outgoing interface is
Cisco routers will always use the IPaddress oftheoutgoing interface as the source IPaddress
for IP packets that the router generates. To control which IP address touse for H.323 signaling,
add the commands that the figure shows. Add these commands atthe interface that should he
used as the source for H.323 traffic.
Note
The twocommands that are in red are optionalifthe outgoinginterface is simultaneously the
source interface for H.323 traffic
Note
By default, Cisco Unified Communications Manager accepts H.323 messages only from IP addresses that are specified as H.323 devicenames at the H.323 gatewayconfiguration. The advanced Cisco CallManagerservice parameter Allow TCP KeepAlives for H323can
control this behavior
4-47
ng
To routecalls from the 11.323 gatewav to Cisco Unified Communications Manager, you must
configure a single dial peer.
In thisexample, all calls with a called-party number that starts with 2 and that is four digits
long will be routed to the Cisco Unified Communications Manager that has the IP address 10.1.1.1. The defaultsignaling protocol for Cisco IOS dial peers is 11.323, so no further configuration is needed lo implement simplecall-routing ftinctionalitv.
H323
Calls from the H.323 Gateway to Cisco Unified Communications Manager Cluster By configuring dial peer hunting on the Cisco IOS H.323 gateway, you can configure prioritized dial peers to reach alternate call-processing nodes. Ifthe preferred Cisco Unified
with a different Cisco Unified Communications Managerwithin the cluster.
Communications Manager becomes unreachable, the Cisco IOS gateway tries to set up the call
As itis for IP phones, the H.323 gateway is associated with adevice pool in Cisco Unified
Communications Manager. The device pool ofthe H.323 gateway specifics a Cisco Unified Communications Manager group that contains an ordered list ofCisco CallManager services.
This Cisco Unified Communications Manager group defines which Cisco CallManager service
should be used for signaling calls towards the H.323 gateway. Ifthe IP phone that places the
call to the H.323 gateway is registered with aCisco CallManager service that is also amember
ofthe Cisco Unified Communications Manager group that the H.323 gateway uses, this service
gets priority over other Cisco CallManager services within this Cisco Unified Communications
Manager group. The service will be used for signaling towards the H.323 gateway.
4-49
Step 3; Configure
Communications I
1
To reduce the failover detection time, create an H.323 voice class
Apply the configured H 323 voice dass to the VoIP dial peer
destination-pattern 2...
voice-class h323 1
To configure dial peer hunting, create a second dial peer that has the same destination pattern but a different call-proccssing node IPaddress and an inferior preference. To speed upthe fai lover-detection time for dial peerhunling. configure an 11.323 voice class
with lower h225tcp establish and h225setup timeout parameter values, and bind thisvoice
class to the dial peers.
h225 timeout tcp establish sec: If the 11.323 galewav cannot establish a TCPconnection
to the Cisco 1. nified Communications Manager within the specified lime, the next dial peer
with inferior preference will be used.
h225 timeout setup sec: An H.225 setup message will be sent to Cisco Unified Communications Manager onlv afler the TCP connection is established. If Cisco Unified
Communications Manager does not respond within ihe specified time, the next dial peer
with inferior preference will be used.
II the H.323 voice class is used onlv to applv the two h225 timeout parameters, lite class does notneed to be applied to the dial peer that has the worst preference (the last dial peer ofthe hunt configuration). In this example, the voice-class command isoptional for dial peer 2.
Communications
Directory
Manager
Number
2001
h yz
H.323 call survivability describes the behavior ofactive calls ifcommunication between Cisco
Unified Communications Manager and the H.323 gateway is lost.
For active calls between the H.323 gateway and IPphones, a signaling session between the
H.323 gateway and Cisco Unified Communications Manager must also be maintained. Itthe
H.323 gateway can no longer communicate with Cisco Unified Communications Manager
while calls are active, these calls are torn down.
Note
The H.323 gateway detectsa connection failure with Cisco Unified Communications
Manager throughthe use of H.225 keepalivetimeouts.
Single-Site Off-NetCalling
4-51
__ keepalivecommand is configured
timeout keepalive-
no h225
to h323
To avoid the dropping of active calls during a communication failure between the 11.323 gatewav and Cisco L'nified Communications Manager, configure the global 11.323 no h225 timeout keepalive parameter.
H.323 Calls to PSTN Via VoIP Dial Peers (Cisco Unified Border Element
Configuration)
The global 11.323 no h225 timeout keepalive command has no effect on IP-to-IP calls. To
configure call sun iv ability for Cisco L'nified liorder Element configurations, create a H.323
voice class, set the call preserve parameter, and bind the voice class to the used VoIP dial
peers.
4-52
Primary
In Cisco Unified Communications Manager. SIP gateways arc implemented by using SIP trunks. The figure shows the SIP gateway-configuration scenario.
In the figure, the voice-enabled router is the SIP gateway that connects Cisco Unified Communications Manager to the PSTN by using adigital (Tl/El) VWIC. Cisco Unified Communications Manager establishes a SIP trunk to this gateway IP address. To route calls
from the cluster towards the PSTN network, the SIP trunk must beassociated with dial plan
information. In this scenario, the IP phone receives only PSTN calls but does not initiate calls,
so a dial plan is required only at the SIP gateway.
4-53
'/.
To configure a SIP gatewav in Cisco Unified Communications Manager, lollovv these steps:
Stepl Add a SIP trunk.
Step 2
To configure SIP gatewav funclionalitv onthe Cisco IOS router, follow these steps:
Step 1 Step 2
Step 3
Conligure basicCisco IOS SIP functionality. Configure the necessary Cisco IOS call-routing information.
Configure the SIP user agent parameters.
4-54
!U
Follow these steps to add a new SIP trunk to Cisco Unified Communications Manager:
Step 1 Choose Device > Trunk from the menu.
Step 2
Choose SIPTrunk from the Trunk Type drop-down list. SIP from the Device Protocol drop-down list, and None (Default) from the Trunk Service Type drop
down list.
Step3
Note
4-55
Traalf to*R*URtUoo
L35"'
*,. ^.
1 1 Define a unique device
harei.-erjj'''
l-.-p-run._oi
JDtljjhi
Li? svaem jcfajlr
^-,Ct 5!>,.
IWi'i!-ll'''"L:1
Step 1 Step 2
Filtera unique SIP trunk name intothe Device Name field, and (optionally) enter a
descriptive name in the Description field.
To associate the SIP trunk wilh the appropriate device pool, choosethe pool from
the Device Pool drop-down list.
SIP Information
configuration
Destination Address
Deinat>on address IPv6
|lP.l.l.I01
Destination Port*
5060
I 5. Select IheSIP
profile.
Step 3
Step 4 Step 5
Step 6
Choose a profile from the SIPTrunk Security Profile drop-down list. Choose a profile from the SIP Profile drop-down list.
Click Save.
4-57
SIP agnalng and VoIP media traffic will use the specified interface IP
address as the source voice service voip
sip
Unlike the source interface for H.323 traffic, the source interface (IP address) for SIP traffic and RIP media or other parameters (such as the transport protocol that is to be used) are configured as global SIP parameters. SIP supports signaling overTCPand UDP. so conligure the l.a\er 4 protocol according to the SIP trunk configuration on Cisco Unified
Communications Manager.
Note Although a physical interface is shown in the example, it is recommended to bind applications running on a Cisco IOS router to loopback interfaces so that the application does not rely on one specific interface to be up
4-58
The only mandatory configuration for SIP call-routing support on the Cisco IOS gateway isto change the
VoIPdial-peer protocol to SIP.
dtmf-relay h245-alphanumeric
The default signaling protocol on VoIP dial peers is H.323. You need to change the protocol to
SIP. to configure basic call routing towards Cisco Unified Communications Manager.
Use the session protocol sipv2 command to change the signaling protocol ofthe VoIP dial peer
to SIP.
Single-Site Off-NetCalling
4-59
Many parameters can be defined for the Cisco IOS SIP user agent.
Use ihe sip-ua command to
authentication, and so on
sip-server ipv4:lC
.1.1.1
dtmf-relay h245-alphanuinerio
codec g711ulaw
Manj parameters, such as im ite. response timers, or authentication and server settings can heconfigured \ ia the SIP user agent configuration onthe Cisco IOS galewa) .
Note The Implementing Cisco Voice Communications and QoS (CVOICE) course describes how
to configure Cisco IOS SIP gateways.
4-60
SIP Considerations: DTMF Signaling This subtopic describes considerations when implementing SIP gateways or trunks in
emironments thathave different DTMF signaling methods.
Signaling
Protocol
DTMF. Method
SCCP
SCCP
OOB signaling
OOB signaling
RFC 2833"
SIP
SCCP
TypeB TypeB
SIP trunk
RFC2833*or OOB signaling RFC 2833* OT OOB signaling RFC 2833* or OOB signaling
SIP
SIP
Depending on phone models and the used signaling protocol, different methods are available to
signal DTMF digits. In general, you can differentiate between in-band and out-of-band (OOB) DTMF signaling. OOB digit signaling on SIP phones Is implemented via Keypad Markup Language (KPMI.); SCCP phones generally use OOB SCCP messages for digit signaling.
This table shows the supported DTMF signaling methods that are based onphone type.
DTMF Signaling Methods
Device Classification
Used Signaling
Protocol
DTMF Method
SCCP-only IP phone
Type A Cisco phone
SCCP
SCCP SIP SCCP SIP SIP
OOB (SCCP)
OOB (SCCP)
Single-Site Off-NetCalling
4-61
Toprovide proper operation of DTMF. the called and calling dc\ices need to have a common
method for DTMF signaling. Depending on the phone models and signaling protocols that arcused ina Cisco I 'nitied Communications Manager infrastructure., this requirement can become
a problem. Cisco Citified Communications Manager checks for a common DTMF signaling
method andautomatical!) allocates a Media Termination Point (MTP) if no common DTMF
method is available. In that case. Cisco (Inilied Communications Manager also autoncgotiaies
the audio codec between the end devices and the MTP.
Note
Although an MTP acts like an intermediate device between the two communicating end
devices, one common audio codec still must be used. Ifno commoncodec exists, a
transcoding device is required. Atranscoder acts like an MTP, but an MTP is not necessarily
a transcoder
Dev ices such as SIP-based IP phones or SIP trunks can be forced loalways use an MTP. lo do so. cheek the Media Termination Point Required check box on the device configuration page. Ifan MTP is enabled statically, a preferred audio codec must be defined manually. In this
case, the SIP INV1TF message also contains Session Description Protocol (SDP) infonnation (known as Far!) Offer media capability exchange option).
Note
After a device is configured to always use an MTP, that MTP will be used even ifit is technically unnecessary because both devices might have a common DTMF method.
4-62
SIP Considerations: MTP Allocation This figure shows the DTMF signaling path, as well as the RTP path (in case an MTP is
allocated).
SIP Considerations:
SIP DTMF on Cisco Unified Communications Manager:
Type ACisco IP phones require an MTP onCisco Unified Communications Manager, for properoperation ofSIP DTMF towards a SIP trunk.
TypeA SCCP Phone
OOB
~&Vi
SIP Trunk
OOB
RFC 2833
_f 1 t>
->-, _^~
iRTPj..
v/
In this example, the SIP gateway expects DTMF signaling viaRFC 2833, whereas the Type A SCCP phone supports only OOB SCCP messages. Cisco Unified Communications Manager
needs to allocate an MTP to convert the OOBmessages from the phone to in-band RFC 2833
signaling.
After an MTP is allocated, two separate RTPstreams are present. The first RTP stream, without
in-band DTMF information, is active between the IP phone and the MTP. The second RTP stream,with in-band DTMF (RFC 2833).is presentbetweenthe MTPand the SIP gateway.
4-63
-.*- ![_tr
<
Point Required is
enabled.
To statieallv allocate an MIP for all calls on a SIP trunk, check the Media Termination Point
Required check box on the SIPTrunk Configuration page. After the MTP is statically enabled for all calls, also choose a codec from the MTP Preferred Originating Codec drop-down list.
4-64
Summary
fhis topic summarizes the key points that were discussed in this lesson.
Summary
Gateways are essential components in an IP telephony environment and provide functions such as conversion from TDM voice to VoIP,
DTMF relay, and so on.
Summary (Cont.)
H.323 gateways provide an easy and flexible way to connect VoIP
calls to the PSTN. Call-routing configuration needs to be applied
SIP gateways are implemented in Cisco Unified Communications Manager by using SIP trunk configuration. Call-routing
configuration needs to be applied on the gateway as well as on
Cisco Unified Communications Manager.
4-65
References
For additional infonnation. refer to these resources:
Cisco Systems. Inc. Cisco I nified Communications System Release 8 v SRND San Jose
California. April 2010.
hup: wwu.ci-.cn.coni en.'l Sdocswtice ip coiiim/cuein/srfid/H\.''iicK\sriid.pdf. Cisco S> stems. Inc. Cisco ( nified Communications Manager Administration Guide,
Release 8.0(11. San Jose. California. February 2010.
4-66
Lesson 2
Call-Routing Components
Overview
The dial plan is one ofthe key elements ofan IP telephony system. The dial plan is at the core
ofthe user experience because itdefines the rules that govern how a user reaches any
destination.
Endpoint addressing and path selection are the most important components ofadial plan. This
lesson describes endpoint addressing, digit analysis, and path selection in a Cisco Unified
Communications Manager deployment.
Objectives Upon completing this lesson, you will be able to describe and configure Cisco Unified
Communications Manager numbering plans, directory numbers, route groups, route lists, route
patterns, digit analysis, and urgent priority for on- and off-net calls. This ability includes being
able to meet these objectives:
Describe the concept ofendpoint addressing, including on-net versus orf-net dialing and
dialing-string length inuniform on-net dialing Describe the concept ofcall routing inCisco Unified Communications Manager
Describe how Cisco Unified Communications Manageranalyzes digits
Describe features that relate to call routing
* Calling privileges: Different groups ofdevices can be assigned to different classes ofservice, by granting or denying access to
certain destinations or resources.
Call coverage: Special groups of devices can be created to process incoming calls for a certain serviceaccording to differentrules, avoiding dropped calls.
Although most people are not acquainted with dial plans b> name, they use dial plans dailv. Adial plan is a numbering plan for a voice-enabled network. The dial plan is the way in which >ou assign individual orblocks oftelephone numbers (li.164 addresses) lophysical lines or circuits. The North American telephone network isbased on a 10-digit dial plan that consists of
3-digit area codes and 7-digit telephone numbers. Fortelephone numbers within an area code, a se\en-digit dial plan is used for the public switched telephone nelwork (PSTN). Features within
a telephone switch (such as Centres) support a custom five-digit dial plan for specific customers that subscribe to that senice. PBXs also support variable-length dial plans that
contain from 3 to II digits.
Dial plans inthe 11,323 network contain specific dialing patterns so that users can reach a particular telephone number. Access codes, area codes, specialized codes, and combinations of dialed digits are all a part of any particular dial plan. Dial plans that areused with voice-
capable routers essential l\ describe the process ofdetermining which and how many digits to store in each configuration. Ifthe dialed digits match the configured number and patterns, the
call Is processed for forwarding.
Designing dial plans requires knowledge ofthenelwork topology, current telephone-number dialing patterns, proposed router and gateway locations, and traffic-routing requirements. No
standard protocol is defined for the d\ namic routing of E. 164 telephony addresses. H.323 VoIP
dial plans are configured statical!} and are managed ongateway and gatekeeper platforms.
A dial plan consists of these components:
Kndpoint addressing (numbering plan): Assigning directory numbers to all endpoints (such as IP phones. fu\ machines, and analog phones) and applications (such as voice-mail
svstems. auto attendants, and conferencing systems) enablesyou to access internal and
external destinations.
4-68
Call routing and path selection: Depending on the calling device, you can select different paths to reach the same destination. You can also use asecondary path when the primary path is unav ailable. For example, acall can be transparently rerouted over the PSTN dunng
an IP WAN failure.
acall: for example, when acall that was originally dialed by using the on-net access code is
Digit manipulation: In some cases, you need to manipulate the dialed siring before routing
expanded to an extension. This necessity can occur before or after arouting decision has
been made.
rerouted over the PSTN, or when an abbreviated code (such as0 for the operator) is
Calling privileges: You can assign different groups ofdevices to different classes of service, by granting or denying access to certain destinations. For example, you might allow lobby phones to reach only internal and local PSTN destinations but give executive
phonesunrestricted PSTN access.
Call coverage: You can create special groups ofdevices to process incoming calls for a
certain service, according to different rules (top-down, circular hunt, longest idle, or broadcast). Doing so also ensures that calls are not dropped without being answered.
4-69
Cisco fOSGateway
i .. Uumtrmr . i^H
Endpoini addressing
Directory number
Dig* manipulation
Both Cisco I:nified Communications Manager and Cisco IOS gateways, including Cisco
Unilied Communications Manager Express and Cisco Unified Survivablc Remote Site Telephony {SRST). supportall dial plan components.
The figure compares the methods that Cisco Unified Communicalions Manager and Cisco IOS
galewav s use to implement dial plans.
4-70
4001
4002]
The diagram shows the essence ofascalable endpoint-addressing scheme that logically
includes geographical information as part ofthe endpoint directory number. In this example, the first digit ofevery endpoint also represents its location. (The digit 2
represents Headquarters. 3represents Site I,and 4represents Site 2.) All endpoints use the
complex.
same extension length offour digits. Variable extension lengths and overlapping endpoint addresses can make call routing, path selection, orgeneral dial plan implementation much more
Ihis figure shows an example ofgeneral call routing and path selection in a multisite
environment.
3001 "-30Q2 Multiple paths are available toestablish a call l--'/^ between HQ and Site 1. One path will be f
2001 2002 2003
An important part oteverv dial plan implementation is call routing and path selection. Munv
factors can be considered when deciding which path to take toconned two endpoints via WAN
or PS IN.
In this example, the WAN connection has priority when establishing calls between Headquarters and Site I. If the WAN is unav ailable or its bandwidth is exhausted, calls will be routed via the first PSTN gatewav (cheap PSTN). IfThe cheap PSTN connection is also unav ailable. a third option (with anexpensive PSTN gateway connection) will be used.
Ideallv. the end user does not realizewhich path was taken to establish the call. A core function to prov ide this transparency is digit manipulation.
4-72
4002
Example2:Acallissent
from the PSTN to 5125553001. The dialed
number is transformed to
extension 3001.
3001
'3002
Many situations require manipulation ofcalled- or calling-parly numbers. In the first example, ause'r on phone 2003 dials 4002. to reach auser in Site 2. Headquarters and Site 2are
connected via PSTN only, so the dialed number 4002 needs tobe expanded toa complete
PSTN numberso that the PSTN can successfully route the call.
The second example shows ascenario in which the complete PSTN ealled-party number ofan incoming call atSite 1needs to be trimmed to the extension length offour digits.
In most voice infrastructures, some type of calling privileges is implemented within a location,
between locations, and for calls lothe PSTN. Calling privileges are typically implemented according to the called and calling numbers. In this example, the user onphone 200I isallowed
to establish a call to Site 2 via the PS'fN. whereas the useron phone 2002 does not have
sufficient privileges to establish calls via the PSTN.
4-74
4001
4002
In general, call coverage provides functions to process calls that would otherwise be
unanswered orprovides service numbers through which calls can bedistributed among a group
of people.
In Example 1. the hunt-pilot number 2222 was created atHeadquarters. Calls to this pilot
number will be distributed among all members, based on a defined hunting algorithm.
In Example 2. a call to3001 isunanswered, and so the call isforwarded to extension 3002. Cisco IP phones can be configured to forward calls todifferent numbers, depending on the reason for not being able to process the call (Busy, No Answer, and so on) and on the origin of
the call (on- or off-net).
4-75
Endpoint Addressing
This topic describes how different endpoints can beaddressed in a Cisco Unified
Communications Manager dial plan.
Directory numbers are assigned to endpoints (phones, fax machines, and so on) and applications (voice-mail systems, auto-attendant, and so on).
directory-number digits.
numbers
The numberof required extensionsgenerally determines the length of DID numbers for inbound PSTN calls are mapped to internal directory
Cisco Umfi Communications
Manager
Cisco
Unity
Phone Numbers
Assigned to Endpoints
fhe number ot'dialuble extensions determines the quantity of digits that are needed to dial extensions. 1'or example, a four-digit abbreviated dial plan cannot accommodate more than 10.000 extensions (from 0000 to 9999). If0 and 9 arereserved as the operator code and off-net access code, respectively, then the number range is further reduced lo 8000 (1000 to 8999).
Ifdirect inward dialing (DID) isenabled for PS IN calls, ihen the DID numbers arc mapped to
internal direetorv numbers.
4-76
Endpoint Dialing
This subtopic describes ihe three types ofendpoint dialing calls.
Endpoint Dialing
On-net dialing: Calls that
408 555-4001
On-net dialing: These are all the calls that remain within one telephony system, such as an
internal call from one IP phone to another IP phone.
Off-net dialing: These are calls that are placed from one telephony system to another telephony system, such as a call from an IP phone tothe PSTN. Abbreviated dialing: This type ofdialing occurs when an off-net destination is dialed by
an internal number; for example, when a caller dials a four-digit extension toreach a
4-77
Site 1
In the figure, the IP phone with extension 2003. which isat Headquarters, dials 3001 toreach an IP phone that is at Site I o\erthe IP WAN. Because both devices are part ofthe same VoIP system (Cisco Unified Communications Manager) and the call isplaced over the IP WAN. the
call is an on-net call.
The IP phone with extension 2002 dials 95552001, and the call is routed to a PSTN destination
through a PS'IN gateway. The call is an off-net call.
The IP phone with extension 2001 dials 4001. which is an IP phone at Site 2. At that site. Cisco Unified Communications Manager Express is used for call processing. However, in contrast to
the first call. Site 2cannot be reached over the IP WAN: it can be reached only through the
PSTN. From an endpoint-dialing perspective, a four-digit extension can be dialed. Cisco Unified Communications Manager then changes the dialed extension to the PSTN number 40X
555-4001 before the call is sent out through a PSTN gateway. This scenario is an example of
abbreviated dialing.
4-78
Use
DID Ranges
Ranaes
ilBI
2XXX
3XXX
N/A
3P-9TXX
N/A
N/A
4{0-41XX
4?5-9]XX
5XXX 6XXX
613 5554[0-4PC<
450 5554[5-9JXX
416 555 5XXX
Site A extensions
Site F extensions
Future Future
N/A
69XX
514555 6[0-8]XX
7XXX 8XXX
9XXX
Adial plan can be designed so that all extensions within the system are reached in auniform origination point. Uniform dialing is desirable because of its simplicity. Auser does not need to
wa\ *that is a fixed quantity ofdigits is used to reach agiven extension from any on-net
remember different ways to dial anumber when calling from various on-net locations. The figure shows an example ofa four-digit uniform on-net dial plan. In the table that is in the figure, the various sites are assigned numbers in the following ways: Site A the company headquarters, requires more than 1000 extensions, so two entire ranges ofnumbers (1XXX*and 5XXX) are retained. (X is awildcard digit.) The corresponding DID ranges must also be retained from the local exchange carrier ofthe site. Site Bis assigned an entire range (2XXX). allowing for as many as 1000 extensions. Site Cis also assigned an entire range, but that range is split between 100 DID extensions (415 555 30XX) and as many as 900 non-DID extensions. Ifgrowth requires more
extensions for DID. any unassigned numbers from the non-DID range can be used. Sites Dand F. are each assigned 500 numbers from the 4XXX range. The corresponding
DID ranges must match each ofthe respective portions ofthe 4XXX range. Because the DID ranges are for different sites (probably from different PS'IN service providers), more
becomes increasingly difficult (or Impossible).
coordination etTort is required lo split the ranges between sites. As the quantity ofsites that are assigned within agiven range increases, making complete use ofan entire range
The Site Frange is split between 900 DID numbers (6[0-8]XX) and 100 non-DID numbers
(69XX).
The ranges 7XXX and 8XXX are reserved for future use.
4-79
When an enterprise consists of few sites, such an approach can be used with leu complications Ihe larger the enterprise (in terms ofthe number ofextensions and sites), the more challenges
The number ofextensions might exceed the range that the quantity of digits affords the dial plan. For instance, ifmore than 8000 extensions are required (considering the exclusions of the OXXX and 9XXX ranges), the system might require an abbreviated dial plan to use
more than four digits.
Matching on-net abbreviated extensions to DID numbers means that, when a new DID
exists in asystem that uses afour-digit uni form abbreviated dial plan, and DID range 650 556 1XXX is also being considered, it might be necessary to increase the quantity ofdigits
for on-net dialing to five. In this example, the five-digit on-net ranges 5IXXX and 61XXX
would not overlap.
existing on-net abbreviated dial ranges. For example, ifthe DID range of415 555 IXXX
range is obtained from alocal exchange carrier, the range cannot conflict with the pre
Most sy stems require the exclusion ofcertain ranges because ofoff-net access codes and
operator dialing. In such a sy stem, in which 9and 0are reserved codes, no dial plan (uniform ornot) can accommodate on-net extension dialing that begins with 9or0.
Ihere tore. DID ranges that would force the use of 9or 0as the first digit in the dial plan
cannot be used. For instance, in a live-digit abbreviated dial plan, the DID range 415 559 XXXX (or any subset thereof) cannot be used. In this example, alternatives include increasing the length ofthe abbreviated dialing lo six or more digits or avoiding any DID
range in which the last five digits start wilh 9.
After agiven quantity ofdigits has been selected and the requisite ranges (for example, ranges
beginning with 9or0)have been excluded, the remaining dialing space must be divided
between all sites. Most systems require that two ranges be excluded, leaving eight possibilities for the leading digit ofthe dial range. The table in the figure gives an example ofthe distribution ofdialing space for a typical four-digit uniform dial plan.
E.164 Overview
This topic provides an overview of E. 164 support in Cisco Unified Communications Manager.
E.164 Overview E.164 as an ITU-T recommendation:
Evolved from E.163 in 1997
numbering plan that is used in PSTN and some other data networks. E. 163 was the former
ITU-T recommendation for describing telephone numbers for a PSTN; these numbers were called directory numbers inthe United States. E.I63 was withdrawn, and some
recommendations were incorporated into Revision 1 of E.164 in 1997.
E.164 also defines the format oftelephone numbers. E.164 numbers can have a maximum of 15
digits, and international phone numbers are usually written with aplus sign (+) before the phone number, to represent the international call prefix. To actually dial such numbers from a
normal fixed-line phone, the appropriate international call prefix must be used. Every countryhas both a country calling code, which isused todial into the country, and an international call prefix (or international access code), which is used to dial out ofthe country. The ITU
approved 00 as the general prefix standard some time ago, and many (though not all) countries have implemented this standard. For countries that use aprefix other than 00, simply substitute that prefix for 00. For example, from North American Numbering Plan (NANP) countries
(including the United States and Canada), dial 011 49405055512.
4-81
This subtopic describes the new h. 164 support in Cisco Unified Communications Manager for
Cisco Unified Communications Manager can route calls that have been placed to E.164
numbers using a plus sign (+)as a prefix. Support for +dialing is implemented by recognizing the plus sign as dialable pattern that can be part ofcall-routing entries such as route patterns or
translation patterns.
Cisco Unified IP phones can place calls to PSTN destinations by using destination numbers in
F..164 format with a prefix. However, the plus sign cannot be dialed from the phone kevpad.
The user cannot manually enter the plus sign at the IP phone.
- dialing from Cisco Unilied IP phones issupported from call lists, directories, speed dials, and
applications (such as click to dial).
Routing Type
Intra site
Calls need to be routed and interconnected according tothe dialed number. Like IP routing, call
routing is destination-based routing. The figure shows the three major areas ofcall routing:
Intrasite routing: Call routing within a single site Intersite routing: Call routing between multiple sites
PSTN routing: Call routing between a site and the PSTN Cisco Unified Communications Manager can automatically route calls to internal destinations within the same cluster because Cisco Unified Communications Manager is configured with the direetorv numbers of itsassociated devices. This scenario can be compared to directly connected networks at a router, in IP routing. Forexternal destinations such as PSTN
destinations (including off-net intersite calls, which effectively are PSTN destinations because
they are addressed by their PSTN numbers) or other VoIP domains such as an Internet telephony service provider (ITSP) or another Cisco Unified Communications Manager cluster, an explicit routecalled aroute patternmust be configured. This route pattern is equivalent
to static routes in an IP router. Insummary, the Cisco Unified Communications Manager call-
routing table is built of connected devices. The table consists ofdirectory numbers ofregistered IP phones and ofstatically entered route patterns that point to external destinations.
Single-Site Off-NetCalling
4-83
408 555-1053
002 2003
In the example scenario in the figure. Cisco 1inilied Communications Manager has a basic
routing table that consists ofthe following entries:
2001. 2002. and 2003 are directory numbers of phones that are configured in Cisco Unified
Communications Manager at Headquarters.
Asecond site. Site I. has Cisco Unified Communications Manager Express and phones that use extensions inthe range of3000 to 3999. To route calls to this external system. Cisco Unified Communicalions Manager at Headquarters requires an entry in its routing table for destination 3XXX. (X isa wildcard digit in route patterns.) This entry' refers to Cisco
Unified Communications Manager hxpress at Site I. which is available viaa trunk.
Headquarters has a PSTN gateway. To route calls out tothe PS'IN. the route patient 9.! is configured in Cisco Unified Communications Manager, to point to the Headquarters PS'IN
gatewav. (The ! wildcard stands for oneor more digits, and theperiod |.| terminates the
access code 9.)
2001 to 2002: This call is internal. 1he dialed number 2002 is looked up in the call-routing
table, and the call is sentto the appropriate IPphone. 2002 to 914085551053: This call is sentto the PS'fN because it matches the 9.! route
pattern. Cisco Unified Communications Manager isconfigured to strip offthe PSTN access code 9before sending the call out lothe PSTN through lite Headquarters gateway. 2003 to3001: The dialed number 3001 matches the entry that refers to a trunk that points
to Cisco Unified Communications Manager Impress at Site 1. Cisco Unified Communications Manager sends a call-setup message to Cisco Unified Communications
Manager I-'xpress.
Description
pattern
numbers
Used to route calls to hunt-group members, based on a distribution algorithm (longest-idle, circular, and so on) Alows a call on hold to be sent to a number and retrieved from another phone by dialing the number Allows a conference callinitiator to set up a conference call and allowsattendees to jointhe conference by dialingthe
conference number
In the previous example, the call-routing table ofCisco Unified Communications Manager is composed ofdirectory numbers and route patterns. Additional routing components can be configured and are added to the call-routing table as possible call-routing targets. The table in
the figure shows a listof possible call-routing table entries. These entries are all possible call-routing targets. Ifa dialed number matches one of these entries, the call is routed to theappropriate entity. That entity can bea phone line, a trunk, a
gateway.a feature, or an application.
Single-Site Off-NetCalling
4-85
IP phones
Trunks
Gateways
Translation
After a translation pattern is best matched (as a targetof a callrouting table lookup), the transformed number is looked up
again mthe call-routingtable. The entity that generates this lookup is the translation pattern.
patterns
Voice-mai) ports
Avoice-mail systemcan be configured to allow calling other extensions or PSTN numbers {e.g.,ihemobie phone of an employee). Inthese cases, the call-routing request is received
from the voice-mail port of Cisco Unified Communications
Manager.
Call-routing requests that require a routing-table lookup include the simplest examplean IP phone placing a call -as well ascalls that are received from outside, through gatewavs or trunks. In addition, these sources ofcall-routing requests require a routing-table lookup: Translation patterns: Atranslation pattern is like a route pattern. A translation pattern
includes a pattern (theentry to the call-routing table): if thedialed number matches the
pattern, another number (the translated number that Is configured at the translation pattern)
is looked up in the routing table. A translation pattern, therefore, combines both roles in
one entity : fhe pattern is both a call-routing table target (it is matched by a dialed number)
and the cause of a second lookup for fhe translated number.
\ oice-mail ports: When a call has been sent to a voice-mail system, that system can request that the call betransferred to another directory number, to a PSTN destination (such asthe cell phone ofa user), orto an assistant. In all these scenarios, the voice-mail port is
the entity that requests the call that Cisco Unified Communications Manager is routing.
The distinction between call-routing sources and call-routing targets is important when
implementing features such as calling privileges, call classification, and others
Note
Description
Singledigit(0-9, *.#)
NANP
Route patterns can include wildcards, so one route pattern can represent multiple numbers, 'fhis abilitv helps to keep the call-routing table short and easy to interpret, like route aggregation in
IP routing.
The table inthe figure lists and describes the wildcards that can be used with route patterns.
Regarding the #wildcard, the implementation ofthe interdigit timeout termination is different from the implementation in Cisco IOS dial peers. In Cisco IOS dial peers, a dialed Uinstructs
the router not towait for additional digits. Only the digits that have been entered before the # are considered to be part ofthe dialed number. Therefore, the # isnot included in dial peer
patterns and can be used. The #symbol is not seen as part ofthe dialed number (and therefore
is notsearched for in a matching pattern). Rather, the symbol is recognized asan instruction to
stop waiting for additional digits. In Cisco Unified Communications Manager, the #is not only
the instruction to stop digit collection but isprimarily part ofthe dialed number. Therefore, if
users areto choose whether to use the # to prevent waiting fortheexpiration ofthe interdigit
timeout, all route patterns must be configured twiceonce with the # and once without.
Single-Site Off-NetCalling
4-87
1234
nx 12xx
Matches 1234
13[25-8]6 13["3-9]6
13!#
Matches any phone number that starts with + and is followed by one or more digits,as used by E.164numbers
The table in the figure shows examples ofroute patterns and the dialed strings that each pattern
matches.
Note that the asterisk in I*Ix isnot a wildcard, but rather a dialed digit.
Because the # symbol in pattern 13!# is used in the router pattern, it mustbe dialed. Otherwise,
the pattern is not matched. Therefore, ifusers should also be able to dial 13 followed by one or more digits, without pressing the # at the end. and simply wait for the interdigit timeout to
expire, an additional route pattern 13! is required.
Call-Routing LogicExample
User A
dials 1200
I
I.
Gateways 12XX
User B
Use'C
dials 1234
In practice, when multiple potentially matching patterns are present, the destination pattern is
chosen based on the following criteria:
Among all the potentially matching patterns, the pattern matches tlie fewest strings other
table includes the patterns IXXX. 12XX, 121X, and 1234.
than the dialed string. For example, the figure shows an example in which the call-routing
When User Adials the string 1200. Cisco Unified Communications Manager compares itto the
patterns in its call-routing table. In this case, there are two potentially matching patterns: IXXX
strings (from 1000 to 1999). whereas 12XX matches only 100 strings (from 1200 to 1299).
Therefore. 12XX is selected as the destination of this call.
and I2XX. Both ofthese patterns match the dialed string, but IXXX matches atotal of1000
When User Bdials the string 1212. there are three potentially matching patterns: IXXX, 12XX. and 121X As mentioned previously, IXXX matches 1000 strings and 12XX matches 100
strings. However. 121X matches only 10 strings. Therefore, 121X is selected as the destination
ofthe call.
When User Cdials the string 1234. there are three potentially matching patterns: IXXX. I2XX.
and 1234. As mentioned earlier, IXXX matches 1000 strings and 12XX matches 100 strings. However. 1234 matches only one string (the dialed string); therefore, 1234 isselected as the
destination of this call.
4-89
|tt-&y-un
1000
Dialed Digits
<none>
I
List Potential Matches List Potential Matches
List Current Match
Route Patterns
1XXX toxx
0 0
1001
Call Setup
Ifan endpoint sends dialed digits one by one. Cisco Unified Communications Manager starts
digit analysis immediately upon receiving the first digit. In fact, digit analysis starts even one
stepearlier, when a phone indicates an off-hook stateto Cisco Unified Communications
Manager. At that point. Cisco Unified Communications Manager looks up anull string dialed
number that matches all available call-routing tables.
Bv each additional digit that is received. Cisco Unilied Communications Manager can reduce ihe list ofpotential matches (that is. the call-routing table entries lhat match the digits that have
been received so far). After asingle entry, such as the directory number K10I in the example, is
Cisco Unified Communications Manager does not always receive dialed digits one by one. Skinny Client Control Protocol (SCCP) phones always send digit by digit. Session Initiation Protocol (SIP) phones can use en bloc dialing tosend the whole dialed string atonce, orcan use Keypad Markup Language (KPML) tosenddigit by digit. If digits are received en bloc, the whole received dial string ischecked at once against thecall-routing table.
matched, the so-called current match is used and the call is sent lo the corresponding device.
Note
Userdial string:
1211
1111
IISESr
i[23]xx J^:Ma^;:j^:;;;=;i;h*;
"J31 rp^8:SS^;Maii5|i;:
13[0-4]X _|':Oo^;^W|^i:|
Matches ^CC-ci-gst stings
13i
-beisiNWji
The figure shows an example ofdigit collection in Cisco Unified Communications Manager. Digit collection is stopped as soon as an entry in the call-routing table is matched in its complete length and no other potential matches exist. In the example, auser dials 1211. Cisco
Unified Communications Manager interprets the number digit by digit. After the first two digits have been anaK/ed. onlv two potential matches (the second and third entries) remain. All other entries in the call-routing table require adifferent digit than "2" at the second position. Cisco Unified Communications Manager continues collecting digits until itreceives four digits
(1211). Now. the second and third entries are matches. Because the second entry matches only
call.
10 numbers whereas the third entry matches 200 numbers, the second entry is used to route the
4-91
Ihe figure show san example in which the interdigit timeout must expire before Cisco Unified
1111
121X
r~ I
1[23JXX
131
13[0-4]X
Matrh H&HMl^^H
Match
Matches digitstnngs
r\
13!
In this example, a user dials 1311. Cisco Unified Communications Manager has three potential matches: 13(0- 4].\. I[23]XX. and 13!. Because the last entry is avariable-length pattern. Cisco Unified Communications Manager must wait for the user lo provide additional digits. Alter the
interdigit timer expires, all three patterns clearly match, and Cisco Unified Communications
Manager must route the call, according lo best-match logic. The result isthe use ofpattern
13J0-4]X. which stands for 10 possible numbers; 1[23]XX matches 200 numbers, and 13!
stands for unlimited possible numbers.
Ifthe user dials 131. Cisco Unified Communications Manager can match the last four route patterns. After receiving these three digits and after the interdigit timer expires, only two patterns131 and 13!remain to match. Again, the more specific pattern (131) is used to
route the call.
4-92
Addressing Method
Digit-by-digit En bloc (Type-B phones only)
En bloc
IP phone
SIP
Gateway
MGCP/SIP/H.323
Trunk
SIP, H.323
The table shows the addressing methods that Cisco Unified Communications Manager supports
for different devices.
in asingle SIP INVITE message. KPML allows digits to be sent one by one. SIP dial rules are
reorder tone, without sending any signaling messages toCisco Unified Communications
In SIP. en bloc dialing or KPML can be used. In en bloc dialing, the whole dialed string is sent
processed inside the SIP phone. Therefore, aSIP phone can detect invalid numbers and play a
Manager. Ifdialed digits match an entry ofaSIP dial rule, the dialed string is sent in asingle
SIP INVITE message to Cisco Unified Communicalions Manager. IfCisco Unified Communications Manager requires more digits. KPML can be used to send the remaining digits one by one. from the SIP phone to Cisco Unified Communications Manager. Trunks and ISDN PRIs can be configured for overlap sending and receiving, allowing digits to
be sent or received one by one over an ISDN PRI.
4-93
User Input on SCCP Phones This subtopic describes how Cisco Unified Communications Manager processes user input on
SCCP phones.
Can be controlled via fhe product-specific Enbloc Dialing configuration parameter (enabled by default). " ifthe numberis entered while the phone is on hook
and the Dial softkey is pressed to start the call,
en bloc signaling takes place.
- If fhe phone is placed off hook first and then digits are
dialed, digit-by-digit signaling is used.
Whether anumber is signaled digit-bv-digit or en bloc, depends not only on the configured
signaling protocol but also on the phone model (Type AorType B) that isused and on how the
phone number is dialed.
For Cisco SCCP IP phones, the following rules apply: Tvpe-A IP phones onlv support digil-bv-digil signaling.
Lit bloc dialing, which is enabled by default, can be disabled via the product-specific Lnbloc Dialing configuration parameter from the Phone Configuration page. Digit-bv-digit dialing is used whenever the number is dialed after the phone is put
off hook.
Note
The dialing behaviormay vary based on the phone load version that is used
4-94
Digit-Signaling Methods
The table shows the different digit-signaling methods, according to signaling protocol, phone
type, and dialing method.
Digit-Signaling Methods
mLmMMBmammm
Manual
Dial button
Digit-by-digit
Digit-by-digit
Digit-by-digit
En bloc
En bloc
En bloc
En bloc
En bloc
Speeddial
Call list
Digit-by-digit
Digit-by-digit
En bloc
En bloc
En bloc
En bloc
Digit-by-digit
En bloc
The table shows the behavior ofdifferent phones with different protocols for the following call
tvpes: manual dialing (that is. going ofi-hook first and then entering digit by digit), dial button
(that is. first entering the digits to be dialed and then going ofi-hook), speed dial (i.e. pressing a speed dial), and call list (that is, dialing adirectory or call list entry).
Note The dialing behavior also depends on the Cisco Unified Communications Manager version and on the phone firmware that is used. Therefore, you should not rely that IP phones in your environment behave asshown above. However, you should be aware, that the addressing method (digit-by-digit or en-bloc) varies based on used software, phone model,
protocol, and call type. __
4-95
Cisco Unified Communications Manager analyzes phone input, digit by digit, against configured dial plan and responds with feedback (dial tones, ringback. reorder tone, and so on).
Nodial plan information is at the IP phone.
SCCP message is sent DiatPlan
f
Any Dno.ie Model
ft
Signaling
IP phones that use SCCP immediately report every user input eveiil to Cisco Unified
Communications Manager. Lor instance, as soon as the user goes off hook, the phone sends a signaling message to the Cisco Unified Communications Manager server with which the phone is registered. The phone functions like aterminal, and the configured dial plan ofthe Cisco L'nified Communications Manager server makes all decisions that result from user input.
As the phone detects other user events, they are relayed individually toCisco Unified
Communications Manager. Auser who goes off hook and then dials 1000 would trigger five individual signaling ev ents from the phone to Cisco Unified Communications Manager. All the resulting feedback (such as screen messages, playing dial tone, secondary dial lone, ringback.
and reorder tone) is prov ided to the user as commands thai Cisco Unified Communications
Manager issues to the phone in response tothe dial plan configuration. Configuring dial plan infonnation on IP phones that run SCCP is neither required nor possible. All dial plan funclionalitv. including the recognition ofdialing patterns as user input is collected, is contained in the Cisco Unified Communications Manager cluster. Ifthe user dials a pattern that Cisco Unified Communications Manager denies, a reorder tone is
plaved lo the useras soon as that pattern becomes the bestmatch in Cisco Unified Communications Manager digit analysis. For instance, ifall calls to 91976 aredenied, a reorder
tone is sent to the user phone as soon as the user dials 91976.
4-96
User Input on SIP Phones Cisco Unified IP phones that run SIP have different capabilities, depending on the IP phone
model.
Modern Cisco Unified IP phonessuch as Cisco Unified IP Phone 7911, 7941, 7945, 7961, 7965, 7970, 7971
- Support KPML SIP dial rules can be configured on both phone types
Tvpe-A phones (Cisco Unified IP Phone 7905, 7912, 7940, and 7960) do not support KPML.
These phones do support SIP dial rules, which are configured in Cisco Unified
Communications Manager and downloaded tothe IP phone atboot time.
Type-B phones (Cisco Unified IP Phone 7911, 7941, 7961, 7970, and 7971) support KPMI.
and SIP dial rules.
4-97
Cisco Unified Communications Manageranalyzes the full dialed digits against configured dial plan.
PA'AM|y) \
j
Phone j^V
sumas Cisco ""^2
Unified IP Pnoies
7940 7SSC
j
/-t
Signaling
In this mode ofoperation, the phone accumulates all user-input events until the user presses
either the # kev or the Dial softkev. The function of these keys is like the Send button that is
used on many mobile phones,
lor example, a user who makes a call to extension 1000 would need lopress 1. 0. 0.and 0.
followed by the Dial softkey or the # kev. The phone would then send a SIP INVl'fL message
to Cisco Unified Communications Manager, to indicate that a call to extension 1000 is
requested. As the call reaches Cisco Unified Communicalions Manager, thecall is subjected to thedial planconfiguration for the phone. Thatconfiguration includes all the class of service (CoS)and call-routing logicthat is implemented in the Cisco Unified Communicalions
Manager dial plan.
4-98
The figure illustrates how user input is processed on Type-A SIP phones when SIP dial rules
are configured.
(Digit Analysis) ^^
Dial Plan
Cal in progress,callconnected,calldenied,andsoon
Dialing Actions
lOOOOial
Pattern 1 Timeout 0 .
such a5 Cisco
Unified P phones
7940.7960
Signaling
SIP dial rules enable the phone to recognize patterns that users dial. Alter the recognition has occurred, the sending ofthe SIP INVITE message toCisco Unified Communications Manager is automated. The user does not need topress the Dial softkey orwait for the interdigit timeout.
For example, ifabranch location ofthe enterprise requires that calls between phones within the
same branch be dialed as four-digit extensions, the phone can be configured torecognize the
four-digit patterns. As the figure shows, the user is not required to press the Dial softkey or wait
for the interdigit timeout.
In the diagram in the figure, the phone is configured to recognize all four-digit patterns that begin with l;the phone has an associated timeout value of0. All user-input actions that match the pattern trigger the immediate sending ofthe SIP INVITE message to Cisco Unified
Communications Manager, without requiring the user topress the Dial soUkey. Type-A phones that use SIP dial rules offer a way todial patterns that are not explicitly configured on the
phone. Ifadialed pattern does not match aSIP dial rule, the user can press the Dial softkey or
wait for interdigit timeout.
indication that the system has rejected the call. For instance, ifa SIP dial rule is configured on
end of dialing (afterpressing the final 4 key).
Communications Manager dial plan blocks such calls, the user will receive areorder tone at the
the phone to recognize calls that arc dialed in the form 919765551234 but ifthe Cisco Unified
4-99
f NOTIFY messages
SIP-Enhanced Phone such as Cisco
Unified >P
^reportedinSIP
(Digit Analysis.) ^
r _
DialPla.i
Prione797i
Dialing Actions
1 OOOD.al
Signaling
Toreport user activities. Type-B IP phones offer functionality that is based on KPML. Each
one ofthe user-input events generates its own KPML-based message to Cisco Unified Communication:, Manager. From the standpoint ofimmediately relaying each user action to Cisco I'nified Communications Manager, this mode ofoperation is like that ofphones that run
SCCP.
Every user kev press triggers a SIP Notify message to Cisco Unified Communications Manager, to report a KPMI event that corresponds to the key that the user pressed. This
messaging enables Cisco Unified Communications Manager digit analysis to reeogni/e partial
patterns as the user composes them. Digit analysis can also provide appropriate feedback: for example, an immediate reorder tone when an invalid number is being dialed.
In contrast to Tvpe-A IP phones that run SIP without dial rules. lype-BSIP phones have no
Dial softkey to indicate the end ofuser input. In the diagram, a user who dials 1000 is provided
call-progress indication (either a ringback or reorder tone) after dialing the lust 0 and without pressing the Dial softkev. This behavior is consistent with the user interface on phones that run
SCCP.
4-100
The figure illustrates how user input is processed on Type-B SIP phones when SIP dial rules
are configured.
- If additional digits are required, KPML is used. - Additional digits aresent one by one, using KPML.
f is sent when pattern
is recognized
SIP-Enhanced
Phone.
(Digit Analysis) ^
Dial Plan
such as Cisco
Unified P Phone
7971
Signaling
Timeout 0
Type-B IP phones can be configured with SIP dial rules so that the phone recognizes dialed patterns. In the figure, the phone is configured to recognize all four-digit patterns that begin
Manager.
with 1 and the phone has an associated timeout value of0. All user-input actions that match these criteria trigger the sending ofaSIP INVITE message to Cisco Unified Communications
As soon as SIP dial rules are implemented on Type-B IP phones, KPML-based dialing is used
ifadial string that matches adial rule and is passed on to Cisco Unified Communications Manager in the SIP INVITE message requires more digits at Cisco Unified Communications
dial string.)
Manager. (This requirement can occur because potential matches are longer than the provided
configured on the phone. Ifadialed pattern does not match aSIP dial rule, the user must wait for interdigit timeout before tlie SIP Notify message is sent to Cisco Unified Communications
Dial softkey at any time to trigger the sending ofall dialed digits to Cisco Unified
Communications Manager.
Tvpe-B phones that use SIP dial rules offer only one way to dial patterns that are not explicitly Manager. Unlike Tvpe-A IP phones, Type-B IP phones do not have aDial softkey to indicate the end of dialing, except when on-hook dialing is used. In the latter case, the user can press the
Ifaparticular pattern is recognized by the phone but is blocked by Cisco Unified
Communications Manager, the user must dial the entire dial string before receiving an indication that the svstem has rejected the call. For instance, ifa SIP dial rule is configured on Communications Manager dial plan blocks such calls, the user receives a reorder tone at the
end of dialing (after pressingthe 4 key).
) 2010 Cisco Systems, Inc
the phone to recognize calls that are dialed in the form 919765551234, but the Cisco Unified
4-101
digits .
receWd?
If KPML is supported and SIP dial rules are configured, digits are sent en bloc in a SIP INVITE message, after matching a dial rule. IfCisco Unified Communications Manager requires additional digits for the call-routing decision. KPML is used to transfer the additional digits. If no additional digits are provided. Cisco Unified Communications Manager stops digit
collection after e\piration of Ihe interdigit timer and rejects the call.
4-102
Cisco Unified Communications Manager collects digits and immediately passes them, one by one, tothe PSTN, as they
are dialed.
Useful forsimplifying variable-length PSTN dial patterns (need only oneroute pattern for all PSTN calls). Configured through route-pattern configuration. Overlap receiving Cisco Unified Communications Manager receives the dialed digits, one by one, from a PRI PSTN gateway.
Incountries whose national numbering plan is noteasily defined with static route patterns.
Cisco Unified Communications Manager can be configured for overlap sending and overlap
recei\ ing. Overlap sending means that Cisco Unified Communications Manager collects digits and passes them to the PSTN as end users dial them. To enable overlap sending, check the
man; F.uropean countries).
Allow Overlap Sending check box on the Route Pattern Configuration page. The route pattern needs to include only the PSTN access code (for example. "9." in North America or "0/" in
Overlap receiving means that Cisco Unified Communications Manager receives the dialed digits one by one from aPRI PSTN gateway and waits for completion ofthe dialed string before attempting to route the call to an internal destination. To enable overlap receiving, set
the OverlapReceivingFlagForPRI service parameter to True.
Single-Site Off-NetCalling
4-103
>Macro function that expands into a series ofroute patterns - Represents Ihe entire national numbering plan for a certain
country
Can modify and use @for other country numbering plans Can beusedwith route filters to block certain components of
the number
Ihe a wildcard is a special macro function that expands into aseries ofpatterns that represent the entire national numbering plan for a certain country, for example, configuring asingle untiltered route pattern, such as 9:a with the NANP adds 166 individual route patterns to the
Cisco L'nified Communications Manager interna! dial plan database.
You can configure Cisco Unified Communications Manager to accept other national numbering
plans, "fhe it wildcard can then be used for different numbering plans, even within the same Cisco {inilied Communications Manager cluster, depending on the value that isselected inthe
Numbering Plan field on the Route Pattern Configuration page.
The a wildcard can be practical in small and medium deployments but can become more difficult tomanage and troubleshoot in large deployments. Certain components ofthe
numbering plan can be matched bv using route filters.
Route Filters
This subtopic describes how route filters work together with numbering plans in Cisco Unified
Communications Manager.
Route Filters
Used only with @route pattern, to match certain patterns (e.g., all
1-900 calls) defined by clauses
- For example match all NANP dialed numbers that include the selection of a long-distance earner (e.g., 9.101044414085551234)
Route pattern: 9 @ - Route filter: IF TRANSIT-NETWORKEXISTS
numbers of anumbering plan. Aroute filter that is applied to apattern that does not contain the
'7i; wildcardis ignored.
Route filters can be used only with the @route pattern, to match certain elements or special
The logical expression that is entered with the route filler can be as many as 1024 characters,
excluding the NOT-SELECTED fields.
For large-scale deployments, use explicit route patterns, rather than using the @wildcard and
Route Pattern Configuration page.
route filters, 'fhis practice also facilitates management and troubleshooting because all patterns that are configured in Cisco Unified Communications Manager are easily visible from the
fags serve as the core component of aroute filter. Atag applies aname to asubset ofthe dialed-digit string. For example, the NANP number 972 555-1234 comprises the LOCAL-
AREA-CODE (972). OFFICE-CODE (555), and SUBSCRIBER (1234) route-filter tags. The following table shows a complete list oftags that are available for the NANP.
4-105
Tag
AREA-CODE
Description
COUNTRY CODE
END-OF-DIALING
INTERNATIONAL-DIRECT-DIAL
INTERNATIONAL-OPERATOR
LOCAL-AREA-CODE
This three-digit local area code in the form [2-9JXX identifies the
local area code for 10-digit local calls
LOCAL-DIRECT-DIAL
LOCAL-OPERATOR
LONG-DISTANCE-DIRECT-DIAL
LONG-DISTANCE-OPERATOR
NATIONAL-NUMBER
This tag specifies the nation-specific part ofthe digit string for an
international call.
OFFICE-CODE
SATELLITE-SERVICE
SERVICE
SUBSCRIBER
TRANSIT-NETWORK
This four-digit value identifies a long-distancecarrier. Do not includethe leading 101 carrier access code prefix inthe
TRANSIT-NETWORKvalue. Refer to TRANSIT-NETWORKESCAPE for more information.
TRANSIT-NETWORK-ESCAPE
The value for this field specifies 101. Do not include thefour-digit
Carrier Identification Code (CIC) in the TRANSIT-NETWORKESCAPE value. Refer lo TRANS IT-NETWORK for more
information
4-106
. Example 1: Aroute filter that uses AREA-CODE and the operator DOES-NOT-EXIST
selects all dialed-digit strings that do not include an area code; for example, seven-digit
calls.
Example 2: Aroute filter that uses AREA-CODE, the operator == and the entry 515
515XXXXXXXorl5I5XXXXXXX).
selects all dialed-digit strings that include the 515 area code (equivalent to aroute pattern
. Fxample 3: Aroute filter that uses AREA-CODE, the operator = and the entry 5J2-91X
selects all dialed-digit strings that include area codes in the range of 520 through 599.
0444 selects all dialed-digit strings that have the carrier access code 1010444.
. Example 4: Aroute filter that uses TRANSIT-NETWORK, the operator = and the entry
Sin9le-Sile Callin
4"17
The ! Wildcard
This subtopic describes the !wildcard that can be used in mule patterns.
T302 timer can be configured (typically reduced): - Service Parameter >Call Manager >Clusterwide parameters
(Device -General)
Different behavior, comparedto Cisco IOS dial peers In Cisco Unified Communications Manager, # is seen as partofdialed string (so if# is used, stnng does notmatch route pattern without #)
International destinations are usually configured by using the !wildcard, which represents any quantity ofdigits, for example, in North America, the route pattern 9.(II11 is typically configured for international calls. In most European countries, the same result is accomplished
by using the 0,00! route pattern,
Ihe! wildcard isalso used for deployments incountries inwhich the dialed numbers can be of vary ing lengths. In such cases. Cisco Unified Communications Manager does not know when
dialing is complete and will wait for 15 seconds (by default) before sending the call. This delay
can bereduced in oneofthe following wavs:
Reduce the I302 timer (Serv ice Parameter TimerT302_msec) to indicate the end ofdialing.
1lowev er. do not set this timer lo less than 4seconds, toprevent premature transmission of
the call before the user finishes dialing.
Configure a second route pattern, followed by the # wildcard (for example, 9.011 !# Ibr
North America or 0.00!# for Europe), and Instruct users to dial # to indicate the endof
dialing. This action is analogous topressing the Send button on a cell phone.
Note
Manager, the# is not only theinstruction tostop digit collection but isalso part ofthedialed number. Therefore, if users are tochoose whether to usethe# to prevent waiting for the expiration ofthe interdigit timeout, all route patterns must be configured twice (once with the
tt and once without)
4-108
Urgent Priority
This section describes urgent priority, where it can be configured, and how it works.
Urgent Priority
Configured under Route Pattern orTranslation Pattern configuration
- Used to force immediate routing as soon as match isdetected, even if other, longer route patterns are potential matches
Used with emergency-numberroute patterns
The Urgent Priority check box is often used to force immediate routing ofcertain calls as soon
asa match isdetected, without waiting for the T302 timer to expire when additional longer
potential matches exist. For example, in North America, ifthe patterns 9.911 and 9.[2-
9]XXXXXX are configured and auser dials 9911, Cisco Unified Communications Manager usually must wait for the T302 timer before routing the call, because further digits might cause the 9.[2-9]XXXXXX to match. However, when urgent priority is enabled for the 9.911 route
has finished dialing 99! 1. without waiting for the T302 timer. Effectively, enabling urgent priority excludes the specified route pattern from other, longer route patterns. Ifen bloc dialing isused and the provided number islonger than the urgent pattern, the urgent
pattern is not considered.
pattern. Cisco Unified Communications Manager makes its routing decision as soon as the user
Translation patterns always have urgent priority enabled; this urgent priority cannot be
disabled.
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Blocked Patterns
The subtopic describes the Mock This Pattern parameter.
A route pattern can be configured to either Allow or Block. Block patterns prevent calls to the pattern,clusterwide.
Route patterns and translation patterns can be configured loblock the pattern. Patterns that a
blocked prevent calls to the blocked pattern, throughout the cluster.
Note
If certain destinations should beblocked according tothe railing device oruser, calling
privileges must be configured With calling privileges, individual classes of service can be
configured percalling device Blocked patterns, however, generally do not allow calls tothe
matched number
Call Classification
Call Classification
Classifies a call as on-net or off-net
- Blocking off-net tooff-net transfers (toll-fraud prevention) - Drop conference when no on-net party remains
- Call Forward external versus Call Forward internal
The configuration at the route pattern is used for outgoing calls, whereas asetting at the device
is used for incoming calls.
At the route pattern, the Allow Device Override parameter can be activated to change the
default classification method for outgoing calls. When this parameter is activated, the
classification ofthe outgoing device, rather than the route-pattern classification, is used. This
parameter is useful when the route pattern refers to aroute list that has multiple options for path
selection. Assume that the first path isan intereluster trunk, which should be considered as an on-net call because ituses the IP network; the second path uses a PSTN gateway and should be considered as an off-net call. This distinction isimpossible ifthe route-pattern classification is
used.
Call Forward settings: Call Forward can be configured differently for internal (on-net)
and external (off-net) calls.
Block off-net to off-net transfers: This toll-fraud prevention feature ensures that the
Drop conference when no on-net party remains: This toll-fraud prevention feature
ensures that a conference isdropped when only external parties remain in the conference. If
the setting is not enabled, an internal facilitator can try to connect two external parties by setting up aconference and then dropping out, leaving the two external parties alone in the
conference.
4-111
Note
Call Forward is enabled at the phone. The other two features are Cisco Unified
Communicafions Manager service parameters.
4-112
Route Pattern*
Route Partition < None
Description
Numbering flan
Route Filter
MLPP Precedence*
Resource Pnontv Narr esoBce Network Doman
Default
<- None >
Route Class*
Default
Gateway'Route List*
Route Option
HQ_RL
Call Classification*
JQnNet
7^
Call classification can besetonthe route patterns and for gateways or trunks. Ifthe Allow Device Override check box ischecked, the call classification that isdefined on the outgoing
gateway or trunk will be used, and the route-pattern call classification will be ignored. Using the Allow Device Override function is important when aroute pattern points to aroute
list that contains gateways (associated with route groups) that have different configured call
classifications. Consider a WAN connection that has on-net classification asthe primary path
and a PSTN gateway with off-net classification as the secondary path. When Allow Device
Override isenabled, the classification always depends onthe used trunk orgateway rather than
on the route-pattern setting.
4-113
Dial tone does not necessarily change afterthe first dialeddigit Dial tone changes only ifall potentially matching routepatterns or
translation patterns have secondary dial tone enabled
550815
The secondarv dial tone typically indicates a call toihe PSTN afler the country-specific access
code (for example. 0 or 9) has been dialed.
The secondarv dial lone function can be enabled on route patterns and translation patterns. The
dial tone will change onlv ifall possibly matching route or translation patterns have the
secondarv dial tone enabled.
Pattern
Pattern:
<.
[2-9JXXXXXX
After a user dials1), the dial lone will immediately change because both matching route patterns
have the secondarv dial tone enabled. Orconsider another example:
Route Pattern: 9.t [Secondary Dialtone enabled]
After a user dials lK the dial lone does not change because only one matching route pattern has the secondary dial tone enabled. However, ifthe user then dials a 1, the dial tone will change
because only the route pattern that has the secondary dial tone enabled remains as apossibfe
match.
4-114
Numbems Plan'
Dour* Filter K.Pp Precedence RKnurc* Priority Ma network Domain < Hone
Def*uHj
Qattway/Route Ust1
fluufp OfHiQ"
aH ClflMm-caticn'
The secondary dial tone is configured by checking the Provide Outside Dial Tone check box on
the Route Pattern Configuration orTranslation Pattern Configuration page.
Single-Site Off-NetCalling
4-115
* Path selection is an essential dial plan element. After call-routing decision is done, where should the call be
sent?
Path selection is an essential dial plan element. After matching an entry in the call-routing
table. Cisco Unified Communications Manager must select how and where to route the call.
The route might be a VoIP path over an IP network, using a Cisco Unified Communications
Manager trunk, or the route might be a path, using the PS'fN. Cisco Unified Communications
Manager allows you to configure multiple paths for aroute pattern so that you can specify a
primarv path and oneor more backup paths.
4-116
Path-Selection Example
The figure shows two possible paths for adialed route pattern.
Path-Selection Example
For off-net calls, a route pattern mustbe configured on Cisco
Unified Communications Manager.
Forexample,to reach 408 526-4000, use: - IPWAN through an ICT as priority path If WAN is unavailable, try the second paththrough PSTN
Gatekeeper
San Jose
Routers/Gateways
408 526-4000
In the figure, ifauser dials along-distance PSTN number such as 91408 526-4000, the call is
sent over the 1P WAN. such as over a gatekeeper-controlled H.323 intereluster trunk. Ifthis
call uses the local PSTN gateway asa backup and issent through the PSTN. For such off-net calls, route patterns must beconfigured in Cisco Unified Communications
path does not work (because of network failure, no response from the other side, or so on) the
Manager. Assuming that the office in San Jose has a DID range of4000 to 4999. the route
pattern would be 9.14085264XXX. In this example, the 9is used as aPSTN access code, the I
is used to indicate a long-distance call. 408 isthe area code, 526 is the office code, and 4XXX
stands for station codes 4000 to 4999.
Note
Usually, digit manipulation must be performed depending onthe selected path. In the example, for the PSTN call, the access code 9 must beremoved and the calling-party number should be changed from the internal extension 1001 toa complete PSTN number. If
the call is sent overthe IPWAN, a different dial string thanthe PSTN number might be used
to address the destination.
4-117
Path-Selection Configuration
Cisco Ui
Matches dialed number for external calls
Second level of path selection Points to the actual device(s) Circular or Top Down distribution
Route patterns are strings ofdigits and wildcards, such as 9.4085264XXX, that are configured in Cisco l'nified Communications Manager and that are part ofthe call-routing table. If matched bv the call-routing logic, the route pattern can point directly toa device (such asa trunk ora gateway )ora route list. Route lists provide the first level ofpath selection, if multiple paths exist to reach the called number that matches the route pattern. Route lists include a prioritized list ofroute groups and allow digit manipulation lo be configured per route
group. A route group is the second level of path selection and pomts to devices that areselected according to a distribution algorithm (circular or top-down).
Cisco strong]} recommends using the eomplete route pattern, route list, and route group construct. This construct prov ides the greatest flexibility for call routing, digit manipulation,
route redundancy. and future dial plan growth. Ifroute patterns point directly todevices, the configuration might need to be changed when additional devices are added. Asingle device cannot be referenced both asa member ofa route group and directly from a route pattern.
4-118
To implement path selection in Cisco Unified Communications Manager, follow these highlevel steps:
Add devices (gateways and trunks). Build route groups from the available devices. Build route lists from the available route groups.
Step 4
Single-Site Off-NetCalling
4-119
Route-Group Functionality
1his subtopic prov ides more information about route-group configurat*
Aroute group isa list ofdevices that share the same requirements for digit manipulation (for example, multiple PSTN gateways).
PSTN
Aroute group is a list ofdev ices (gateways and trunks). Put such devices into the route group that has identical digit-manipulation requirements. Digit manipulation can then be configured
once per route group, during route-list configuration.
Note
Aroute group can beconfigured for circular distribution (round robin) orfor top-down
distribution (first entry ofthe list has the highestpriority). The circular distribution is used for
4-120
This topic describes the characteristics of local route groups in Cisco Unified Communications
Manager.
New entry in list of route groups that can beadded to route lists
- Standard Local Route Group is shown in addition to
configured route groups
- Can be added one time per route list (not mandatory) * New setting in device pools
Local route groups decouple the selection of a PSTN gateway or trunk for off-net dialing from
the route patterns that are used to access the gateway. This action can greatly reduce the complexity and size ofdial plans in Cisco Unified Communications Manager. Two new settings are available in Cisco Unified Communications Manager Version 8: Standard Local Route Group isanew entry in the list ofroute groups that can be added to a
route list. Aroute list can include this entry one time, but adding this entry to the list of
route groups in a routelist is not mandatory.
Local Route Group is anew drop-down list atthe device pool. The list includes all
value), or one route group can be chosen.
configured route groups. The Local Route Group parameter can set to <None> (the default
By using these settings, you can associate device pools with a local route group. Route patterns that use the local route group offer aunique characteristic: They allow for dynamic selection of the egress gatewav. depending on the device that originates the call. By contrast, calls that are
routed by route patterns that use static route groups route the call tothe same gateway, no
matter which device originates the call.
Single-Site Off-NetCalling
4-121
PSTN
ic-ttt O-oup
Gateway A
In the example, the route pattern 9.555XXXX points toa route list that contains only the
standard local route group. Gateway' A is associated with Route Group A, which acts us the
local route group to Device Pool A. Gateway Bis associated with Route Group B. which acts
as the local route group to Device Pool B. If a phone (hat is associated with Device Pool A
places a call to 95550815. the standard local route group for this device pool will be used to
place the call to the PSTN. The same applies to phones that arc associated with Device Pool B.
4-122
Route-Group Configuration
The figure shows an example ofroute-group configuration.
Route-Group Configuration
(relevant if TopDowndistribution
algorithm is selected).
In the example, two devices (Gateway 1and Gateway 2) have been put into the route group. The distribution algorithm is circular, so the order ofthe gateways isunimportant.
4-123
Loc
To use a local route group within a route list, device pools must
To use a local route group within a route list, device pools need tobe assigned a local route
group.
4-124
Route-List Configuration
This subtopic provides more information about route-list configuration.
Route-List Configuration
A route list is a prioritized list of route groups.
IP
PSTN
Aroute list isa list ofprioritized route groups. When configuring a route list, you can set up digit manipulation perroute group within the route list.
Single-Site Off-NetCalling
4-125
iwiletlsl&mOpmit)
In the example, two route groups have been added tothe route list. Route group WAN is listed first and therelbie has highest priority. Ifcalls cannot be set up by using any device ofthe
WAN route group, the next route group (PSTN) is used. Again. Cisco Unified Communications Manager tries all dev ices in that route group, according lothe route-group distribution
algorithm (circularor top-down). A route list can be disabled, which means that it remains in
the configuration database but is not used.
At the bottom ofthe Route List Configuration page, in the Roule List Details field, you can configure route-list details per route group. You can configure digit manipulation for each route
group that is a member ofthe route list.
4-126
Route-Pattern Configuration This figure shows how to configure aroute pattern in Cisco Unified Communications Manager.
Route-Pattern Configuration
Q'
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ED
Evil Jit DefftdK Hq_ftL
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h-jfrbenng '
RSuTt Filter
MLPO Pi-SMderte*
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jule C-ffiXtr
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u
i classification.
To configure aroute pattern, open Call Routing >Route/Hunt >Route Pattern and click
Add New.
Numbering Plan
Gateway/Route List
Call Classification
4-127
Route
Pattern
PSTN
I 5264XXX
Philadelphia
fP-WAN--
215 555-1xxx
Five-DigitInternal Dialing
Strip 52 and deliver
lii this example, there are two sites: San Jose and Philadelphia, liach phone has acorresponding
PSTN DID number. Isers dial five-digit extensions within a site (the last digit ofthe PS'fN office code plus the DID subscriber number), for intersite calls, users dial seven digits (the
PSTN office code that is used at each site plus the DID subscriber number).
At the Cisco L'nified Communications Manager in Philadelphia, a route pattern 5264XXX is configured for intersite calls toward San Jose, 'flic route pattern points to a route list that has two route groups One route group refers lo an intereluster trunk (configured as the primary path), and the other route group rciers to a group of PS'fN gateways (as a backup path). Depending on the chosen path, the following digit-manipulation requirements apply for a call
that is placed from Philadelphia to 526-4000:
Calls routed over the interelustertrunk: The first two digits (52)ofthe called number
(526-4000) must be stripped so thai Ihe receiving Cisco Unified Communications Manager
in San Jose finds the five-digit number asa configured directory number ononeofits IP phones. In addition, the calling-parly number must be changed from a five-digit extension
to a seven-digit intersite route pattern (by prefixing 55).
Calls routed over the PSTN: 'fhecalled number must be extended to a complete PS'fN number bv prepending 1408 to the dialed seven-digit number. At the receiving side, incoming calls from the PS'fN must be changed lo five-digit internal directory numbers,
fhe callingnumbermust be changed to a complete PSTN number.
More information about digit-manipulation configuration is provided in another lesson of this
module
Note
4-128
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
- CiscoUnified Communications Managerconfigurationincludesspecial for variable length numbers, blocked patterns, patterns with urgent
pnority, and classification of calls.
References
For additional information, refer to these resources:
Cisco Systems. Inc. Cisco Unified Communications System Release 8.x SRND. San Jose. liUp:.Vyvw\v.cisco.com/en/llS/docs/voicejp_comiii/cucm/smd/8x/uc8xsrnd.pdf. Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide,
California. April 2010.
http://www .eisco.com/en/US/does/voiec_ip_comin/cucm/drs/8_0_ l/drsag801.html. Cisco Systems. Inc. Cisco Unified Communications Manager Documentation Guidefor
Release 8.0(1). SanJose. California. January 2010.
mi
htip:.-\\\\v\.cisco.eoiiv;en/lJS/partner/docs/voice_.ip_comm/cuein/docgiiide/8._(l_l/dg80I.tit
4-129
4-130
Lesson 3
Objectives
Upon completing this lesson, you will be ableto explain the needand uses for calling privileges and howto implement them in Cisco Unified Communicalions Manager. This ability includes
being able to meet these objectives: Describe the tools Cisco Unified Communications Manager supports for calling privilege implementation
Explain the need and uses for callingprivileges and how to implement them in Cisco
Unified Communications Manager
Describe the implementation of partitions and CSSs for an on-net CoS example Describe important rules and considerations when implementing CoS by using partitions
and CSSs
Calling Privileges
Calling privileges (also called CoS) define the entries of a call-routing table that can be accessed by an endpoint that performs a call-routing request.
* Used to control telephony charges
Block costly service numbers
Restrict international calls
Route calls with the same number differently per user (different gateway per site for PSTN calls) Route calls to the same number differently per time of day
Calling privileges are configured to control which entries ofthe call-routing table are accessible from a particular endpoint (such as a phone, gateway, or trunk), 'fhe primary application of calling privileges is the implementation of CoS. CoS typically is used to control lelephonv charges b\ blocking eostlv service numbers and international calls for some users. CoS is also used to protect the privacy of some users: for example, to disallow direct calls lo managers except through their assistants.
Calling priv ileges can also be used to implement special applications, such as routing calls that have been placed to the same number in a different manner because of different calling devices.
For example, in a selective public switched telephone network (PSTN) breakout in a multisite environment with PS I N gateways at each site. PSTN route patterns should always be routed to the local PSTN galewav. Therefore, the same route patients must exist multiple times (once per site, in this example). Only the site-specific route patterns should be accessible by ihe devices
at this site.
Another application is time-of-dav routing, in which calls should take different paths depending
on when the call Is placed.
4-132
Allowed Destinations
Emergency
Internal
Local
Emergency
Local PSTN Internal
Lang Distance
Emergency
Local PSTN
Emergency
International
Local PSTN
The figure shows the calling classes in a typical CoS implementation and theirallowed
destinations. These calling classes can then be assigned to devices or users.
In the example in the figure, classInternal allows only internal andemergency calls.Class Local adds permission for local PSTN calls. Class Long Distance alsoallows long-distance
PSTN calls, and class International enables international PSTN calls.
4-133
Calling-Privileges Configui
Cat! Privileges Element
Group of numbers (directory numbers, route patterns,
Partitions
CSSs
Time Schedules and Time Periods
Used to allow certain partitions to be reachable only during a certain lime of the day
Used to track calls to certain numbers; code must be
Restricts outgoing calls to certain numbers; code must be entered by user to reach a certain number
Partitions and calling search spaces (CSSs) are explained in the following topics of this lesson.
The other three elements are discussed in the "Implemenling Gateway Selection and PS'fN
Access features" lesson.
4-134
- A device can call only those numbers that are in the partitions that are part of its CSS. - A CSS is assigned to any entity that can generate a callrouting request, including phones, phone lines, gateways,
and applications.
A partition is a group of dialable patterns with similar accessibility. A CSS defines which partitions are accessible to a particular device. A device can call only those call-routing table entries that are in partitions that are part ofthe CSS ofthe device.
Partitions are assigned to call-routing targetsthat is. any entry ofthe call-routing table, including voice-mail ports, directory numbers, route patterns, translation patterns, and Meet-Mc
conference numbers.
CSSs are assigned to the sources of call-routing requests, such as phone lines, gateways, trunks, voice-mail ports, and applications.
4-135
<None>
Before partitions and CSSs are configured, all entities lhat can have a partition (i.e., called entities such as directory numbers and route patterns) reside in partition <None>. and all entitles thai can have a CSS (calling entities such as phones or trunks) are assigned CSS <None>. Entities that are in partition <None> are always accessible (whether or not the calling entity has a CSS).
Bv default, all entities thai can be configured with a partition arc in partition <None>. and all entities that can he configured wilh a CSS are assigned CSS <None>. The source of a call-routing request can alwavs access the members of parlilion ---None> (also called Ihe null partition), regardless ofthe CSS of" that call-routing source. Entities that do not have an assigned CSS (in other words, entities lhat use CSS <None>) can access onlv the call-routing targets thai arc in partition <None>.
By default, no partitions or CSSs are assigned, and all entitiesare associated with the null partition andCSS <None>. Therefore, all calls are possible for all calling sources by default.
4-136
Analogy: Locks and Key Rings The figure shows the interaction of configured partitions and CSSs, the null partition, and CSS
<None>.
Phones
Phone5 No CSS
CSSs:
Assigned
Effective Permissions:
Phonel: Phone2, Phone3, Phone4, Phone5 Phone2: Phonel, Phone3, Phone4, Phone5
Phone3: Phone2, Phone4 Phone4: Phonel,(Phone4)
Phone5: Phone4
The example in the figure uses an analogy of locks and key rings. The locks represent partitions
that the administrator applies: the key rings represent the CSSs that the administrator
configures.
In the example Phonel is configured as amember ofthe blue partition, Phonc2 is in the red partition and Phone3 and Phone5 are in the gold partition. Phone4 has not been assigned to a partition. Following the analogy of locks and keys, there are three types of locks (blue, red. and gold). Two ofthese locks are assigned to one phone each, and one ofthe locks is assigned to
two phones. Phone4 isnot secured by a lock.
Tlie CSSs are represented as key rings: Phonel has akey ring with red and gold keys. Phone2 has akey ring with blue and gold keys. Phone3 has akey ring with only ared key. Phone4 has
a kev ring with only a blue key. and Phone5 has no keys.
As aresult of this implementation of locks and keys, the following effective permissions apply:
Phonel: Like all other phones. Phone I has access to all devices that do not have alock (Phone4. in this example). In addition. Phonel can unlock the red and gold locks because it
has the appropriate keys. Therefore, Phonel can access Phone2, Phone3, Phone4. and
Phone5.
Phone2: Like all other phones. Phone2 has access to all devices that do not have alock
(Phonc4) In addition. Phone2 can unlock the blue and gold locks because it has the appropriate keys. Therefore, Phone2 can access Phonel, Phone3, Phone4, and Phone5.
Therefore. Phone3 can access Phone2 and Phone4.
Phone3: Like all other phones. Phone3 has access to all devices that do not have alock
(Phone4). In addition. Phone3 can unlock the red lock because it has the appropriate key.
4-137
Phone4: Like all other phones. Phone4 has access to all devices that do not have alock
importance because Phonc4 usually does not place a call to itself.
appropriate kev. Iherefore. Phone4 can access Phone Iand itself, which is of no practical
(itself in this example). In addition, it can unlock the blue lock because it has the
Phone5: Like all other phones. Phone5 has access to all devices lhat do not have alock (Phone4). Phone5 cannot unlock any locks because it does not have anv kevs. Therefore
Phonc5 can access onlv Phone4.
1osummarize the analogy that is used here: Partitions are like identical locks, which can be
(partition) is applied to adevice, then all other devices can access that dcv'ice. If no keys are
present (no CSS is configured), then onlv devices that do not have a lock can be accessed
Note Calling-privilege implementation in Cisco IOS technologies is called class of restriction
(COR) The concept issimilar to calling privileges implementation in Cisco Unified Communications Manager. However, if there is no incoming COR list (equivalent lo a CSS
unlocked by the same kev: CSSs are like kev rings thai include certain kevs. Ifno lock
in Cisco Unified Communications Manager), all outgoing COR lists (equivalent to partitions)
can be unlocked From the perspective of the presented analogy, if no key ring is applied, all locks can be accessed COR is discussed in more detail in the Implementing Cisco Voice
Communications and QoS (CVOICE) course.
4-138
two partitions.
Phone 2-1
directory number
Phone 3-1
directory number
4001 is in partition
Atlanta and is not
included in routing
decision.
In the example, a phone has a CSS, which contains two partitions, Chicago and San Jose. A third partition. Atlanta, exists but is not included in the CSS ofthe phone. These phone directory numbers are assigned to the partitions: Directory number 3001 (Phone 2-1) is assigned to the Chicago partition.
Directory number 2001 (Phone 1-1) is assigned to the San Jose partition.
Directory- number 4001 (Phone 3-1) is assigned to the Atlanta partition.
The user dials 3001. which is the directory number of Phone 2-1. Cisco Unified Communications Manager takes the number 3001 and performs a callrouting lookup through the partitions that are configured in the CSS ofthe calling phone: Chicago and San Jose.
Cisco Unified Communications Manager finds a match in the Chicago partition, because the directory number 3001 of Phone 2-1 is assigned to this parlilion. Because no other matches exist, routing is complete, and Phone 2-1 rings.
Cisco Unified Communications Manager does not consider the Atlanta partition during the routing decision because that partition is not included in the CSS.
Note
4-139
Multiple identical entities can exist in the call-routing table bet must
be in different partitions
If no single best match exists, the call-routing table entry whose partition is listed first in the CSS of the calling device is used.
Resulting route-selection priorities1
Best match
Order of partition
A CSS is an ordered list of partitions: fhe partition that is listed first has higherpriority than a partition that is listed later. When Cisco Unified Communications Manager performs a callrouting lookup, all accessible entities (that is. all targets that reside in a parlilion that is listed in the CSS ofthe calling phone andall targets that do not have an applied partition) areconsidered
bv best-match logic.
Multiple identical entities can existin the call-rouling table, but they must be in different partitions. One exception lo thisrule is phone directory numbers. When twoor more devices
share the same directory number within the same partition, the directory' number is called a
shared line.
Note
More information about shared lines is provided in the module "Features and Applications
If no single best match is found, then Cisco Unified Communicalions Manager uses the entrv of the call-routing table whose partition is listed first in the CSS ofthe callingdevice. In summary, the entrv ofthe call-rouling table is chosen basedon the following order:
1, fhe best match is chosen.
2. If multiple equally qualified matches exist (there is no single best match), the order ofthe partition in the CSS ofthe calling device (thai is. the match that is found in theearlier listed
partition) is the tie-breaker.
contains two
partitions.
Phone 2-1
Chicago.
Phone 1-1
directory number
3001 is in partition
Atlanta and is not
included in routing
decision.
Inthe example, a userdials 3001 from a phone that liststhe Chicago partition first in its CSS,
followed by the San Jose partition.
Phone 1-1. Phone 2-1, and Phone3-1 are all configured with directory number3001. Phone 1-1 is in the SanJosepartition. Phone 2-1 is in the Chicago partition, and Phone 3-1 is in the
Atlanta partition.
In thisexample. Phone 3-1 is not considered for call routing because its partition is not accessible to the callinguser; the partition is not listed in the CSS ofthe callingphone. From the accessible directory numbers, an equal (complete) match is found for two entries: Phone 1-1 and Phone 2-1. Because Phone2-1 is in the Chicagopartition, which is listed first in the CSS of
the calling phone, tlie call is sent to Phone 2-1. If the partitions were listed in reverse order, the
call would be sent to Phone 1-1.
4-141
tfb^
*^!^
Device
On most sources of a call-routing request, such as a trunk, gateway, or translation pattern, onlv one CSS can be configured. On IP phones, however, a CSS can be applied per lineand once al
the device level.
If both line and device CSSs areconfigured, the CSS ofthe line from which the call is placed is
considered first. In other words, the CSS lhat is used is composed ofthe partitions that are
listed in the line CSS. followed by the partitions ofthe device CSS. Note On computer telephony integration (CTI) ports, the line CSS and the deviceCSS are placed
in reverse order. The partitions of the device CSS are placed before the partitions of the line
CSS
4-142
In the example in the figure, the line CSS ofthe calling phone includes the San Jose and Chicago partitions, and the device CSS ofthe calling phone includes the Atlanta partition. Route pattern 300X is in the San Jose partition, directory number 3001 isused atPhone 2-1 in theChicagopartition. and the same directory number (3001) isused atPhone 3-1 in the Atlanta
partition.
Ifthe calling phone dials 3001, Cisco Unified Communications Manager interprets the dialed digits and searches for the closest match. Because the two directory-number entries in the callrouting table are more specific (a complete match) than the route pattern (which represents 10
numbers), the route pattern is not a candidate for the final routing decision. Out ofthe two
equally matched directory numbers, the number ofPhone 2-1 isused to extend the call because
it is in'the partition that is listed first in the line CSS lhat is effectively used.
This example illustrates that the line CSS has higher priority than the device CSS. Ifthe line
CSS and device CSS were reversed, thecal) would be sent to Phone 3-1.
Note
Although route pattern 300X matches thedialed number and is listed in thefirst partition, itis
not used to routethe callinthis example. The first priority forthe call-routing decision is the best match; the order of partitions is important onlyifmultiple best matchesexist. A common misunderstanding is that the first matching pattern that is found {regardless of
partitions are immediately considered for best-match logic. The partition order is relevant
only ifmultiple best matchesexist.
4-143
calling privilege.
The example shows the use of partitions and CSS to implement the following four classes of
service:
Long Distance: Allows inlemal calls, local PSTN calls, and long-distance PSTN calls
International: Allows internal calls, local PSTN calls, long-distance PSIN calls, and
international PSTN calls
Local-PSTN: Applies to route pattern 9.[2-9|XXXXXX LD-PSTN: Applies to route pattern 9.I|2-9JXX[2-9|XX XXXX Intl-PSTY Applies to route patient 9.01 V.U
The follow ing CSSs are configured, each implementing the corresponding CoS:
("SS-Internal: Contains the Phones partition
When you applv the appropriate CSS to a phone, the phone is granted the permissions ofthe
respective CoS.
2.
Step2
Step2
Step 3
Assign CSSs to entities thatcan request lookups to the call-routing table to route a call. Examples of such entities arephones and phone lines, trunks, gateways, and
translation patterns.
Note
A translation pattern is used in both roles. This pattern is a dialable pattern in the call-routing
table (thatis, the target of a call-routing request). Ifmatched, the pattern invokes a newcallrouting request for the translated pattern. The partition at the translation pattern specifies
who can match the pattern (Ihe partition is required in the CSS of the calling device). The CSS at the translation pattern specifies the entries ofthe call-routing table that the
translation pattern can see foritscall-routing request, whentrying to find the translated
partem in the call-routing table.
4-145
Creating Partitions
The figure shows how partitions arecreated inCisco Unified Communications Manager.
Creating Partitions
Purtttk* ConHgantloa
Oi
<
Ittir} T*# parTff.nr i^ms Mm*o[ e^r^a *f?
P"Cr*i JntC"3 ^il^^Ti. ,:,-:& =S~"J LK< P&H Ci"i[* &3 .^i-S'N i^L'JIjna^lfKS1-!
3rr^='H J^erTi-.or.j1 BST!, (*z:m
11 rf^3 'il
arable Eenfl*
When vou add partitions thatshould be created. Cisco Unified Communicalions Manager allows vou to list all the partitions in one input window. To do so, specify one partition name
and description, separated bv a comma, per line.
4-146
Assigning Partitions
The figure shows how to assign partitions to directory numbers and route patterns.
Assigning Partitions
\d*~
[-Prttem Drfmttwi
yj Stttid- Raiv
HMI BvliLnr.
tfS-flJifflCS-lJ'WX*
j-p*nt
5pi"
!**, *..
>>-,* ; r ^ ' .
HLrt fc-**"*
I>(ltartt
,Nfl"
&*te*iyft4 UB*
llwiM"'
V aorX
*!,*-(..&
Note. Assign partitions todirectory numbers, route patterns, translation patlerns. and soon.
patterns, or any other call routing target. The figure shows examples ofassigning partitions to
directory numbers and route patterns.
Partitions can be assigned to phone lines (directory numbers), route patterns, translation
Single-Site Off-NetCalling
4-147
Creating a CSS
fhe figure shows how to create andconligure a CSS.
highlighted partition
to or from CSS.
When vou add a CSS. vou must configure the name, a description (oplional). and the ordered
list of partitions.
Note
The cder of partitions within the CSS is important when two equally qualified matches are found In such a case, theentry ofthe partition that islisted first is used for the call-routing
decision
4-148
SCCHS.*.5-J51S
0 QVFCI [S *>-<
KJ1CU551E6
Ocst'iot-or*
Auto !CC5
a<. pool*
WQ_Pf
^ Hone *
SCPOD!4CJ4S51fi-SCCP-ind'v>du*l TeropWe
<IH >
Assign CSS to
phone.
Corwi>o"':phwiF'^fi*ffB
C*ll^s set-cf' Si="
Note Assign CSSs to devices (phones/lines), gateways, translation patterns, and soon.
CSSs can be assigned to phones (as shown in the figure), phone lines, gateways, translation
patterns, orany other source ofa call-routing request.
4-149
Manage^
Manager
Yes
Yes
In the example, there are three priv ilege classes wilh partly asymmetric calling privileges. (Asymmetric means that although the manager is allowed to call employee phones, employees
4-150
The following steps are needed to implement asimple CoS scenario using partitions and CSSs:
Step 1
Step2 Step3 Step4 Step5
4-151
Assislante.CS5.,
Employees PT
2001 2002
2003
Employees PT
[| H
II
||
2001 2002
2003
Phone 1-1 |
Phone 1-2 H
Phone 1-1 li
Phone 1-3
H
H
Manager PT
2201
Assistants PT
2101
2102
2101
Phone 3-1
Manager PT
2201
Phone 3-1 M
The figure shows the required partition and CSS configuration to implement the following
calling privileges:
From
To Employees
Yes
To Assistants
Yes No Yes
To Manager
No Yes
Employees
Assistants
Yes Yes
Manager
Yes
4-152
Call forwarding
Communications Manager, many other features rely on CSS and partition configuration:
Implementation ofCoS is not the only application for CSSs and partitions. In Cisco Unified
Gateway selection
Automatic alternate routing (AAR)
Call forwarding
Note
Single-Site Off-NetCalling
4-153
Partition and CSS Considerations This subtopic describes important considerations when implementing and working with CSSs
and partitions.
Hven in smaller Cisco Unified Communications Manager deployments, CSS and partition configuration can become complex. One difficulty ofusing CSSs and partitions lo implement
CoS relates to the main other features and functions in Cisco Unified Communications
Manager that also rely on proper CSS and partition configuration. Changing the configuration offeatures such as gatewav selection orCisco Unified Communications Manager native
presence might affect CoS configuration, and vice versa.
4-154
regular configured CSS and partition with the name None. (<None>
The following rules and guidelines can further help you to understand the principles ofCSSs
lock-and-key concepts.
and partitions. Compared to the lock-and-key analogy, these rules follow adifferent approach in explaining the <None> partition and the <None> CSS. Still, they do not contradict any ofthe
Think ofthe <None> CSS and the<None> partition as a regularly configured CSS and
partition with the name None. (<None> does not mean nonexisting or not configured.)
The <None> CSS has only the <None> partition as a member.
The <None> partition isimplicitly the last member ofany other configured CSS.
Any dev ice can reach every directory number, route pattern, or other element that is
assigned the <Nonc> partition.
Asingle dialable number (directory' number, roule pattern, and others) can only exist once
within a partition.
The partition order within a CSS isatie-breaker only ifthe current-match number exists in multiple partitions. Longest match rule always has priority over partition order.
4-155
Summary
This topic summarizes the kev points that werediscussed in this lesst
Summary
Calling privileges are implemented to implement class of service or special applications lhat require calls to be treated differently
depending on the caller.
It is absolutely essential to be familiar with the principles and functionsof calling search spaces and partitions in order to
implement and troubleshoot a variety of features of Cisco Unified Communications Manager.
References
For additional infonnation. refer to these resources:
Cisco Svstems. Inc. Cisco Unified Communications Manager Administration Guide. Release 8.0(11. SanJose. California. February 2010.
Cisco Sv stems. Inc. Cisco Unified ('ommunications System Release 8.x SRXD. San Jose.
California. April 2010.
4-156
Table of Contents
Volume 2
4-157
Objectives
Cisco Unified Communications Manager Digit Manipulation Overview
4-157
4-159
Digit-Manipulation Requirements
Cisco Unified Communications ManagerDigit-Manipulation Flow
4-160
4-161
4-162
4-163 4-164 4-165 4-166 4-167 4-168 4-170 4-172 4-173 4-174 4-175 4-176 4-177 4-178 4-179 4-180 4-181 4-182
4-183 4-184 4-186 4-187 4-188
4-189
4-190
Incoming Calling- and Called-PartySettings Incoming Calling-Party PrefixExample: Globalization of Calling Number Incoming Calling Party Settings Configuration at Gateway Incoming Calling Party Settings Digit-Manipulation Order: Examples Incoming Calling Party and Incoming Called Party Settings in the Device Pool Calling-Party Transformation Order Called-Party Transformation Order Digit-Manipulation Considerations Summary
References
Implementing Gatewav Selection and PSTN Access Features Objectives Calling-Privileges Applications Overview Calling-Privileges Application Examples Implementing Time Schedules and Time Periods Time-of-Day Routing Applications
Time Periods and Time Schedules
4-209
Time-of-Day Routing-Configuration Procedure Creating Time Periods Creating Time Schedules Assigning Time Schedules to a Partition Implementing Gateway Selection and CoS Gateway-Selection Example: Configuration Gateway-Selection Example: Partitions and CSSs Gateway-Selection Example: Operation
4-218 4-219
4-220 4-221
Traditional-Approach Example: Single Site 4-222 4-223 Traditional-Approach Example Multiple Sites Line/Device Approach: Improves Scalability 4-224 Line/Device Approach: Concept 4-225 Line/Device Approach Example: Multiple Sites 4-226 Implementing 911 and Vanity Numbers 4-227 Vanity Numbers 4-228 Implementing Emergency and Vanity Numbers in Cisco Unified Communications Manager 4-229 Vanity-Number Example 4-230 Implementing Carrier Selection Based on Time of Day 4-231 Time-of-Day-Based Carrier-Selection Example 4-232
CMC and FAC CMC Call Successful Call CMC Call. Call Failure FAC Call: Successful Call FAC Call: Call Failure 4-233 4-234 4-235 4-236 4-237
4-245
4-245 4-246
4-247
Hunt Lists
4-253
Line Groups Line-Group Members Call-Hunting Flow Call-Hunting Scenarios Example 1. Internal and External Forwarding (No Hunting) Example 2. Internal and External Forwarding with Hunting Example 3: Internal and External Forwarding with Hunting Example 4: Internal and External Forwarding with Hunting Example 5: Using the Maximum Hunt Timer While Hunting Call-Hunting Configuration Step 1: Configuring Line Groups Step 2: Configuring Hunt Lists Step 3: Configuring Hunt Pilots Step 4: Configuring CFNC at Directory Numbers Summary
References
4-254 4-255
4-256 4-259 4-259 4-260 4-261 4-262 4-263 4-264 4-265 4-267 4-269 4-271 4-273
4-274
Module Summary
References Module Self-Check
4-275
4-276 4-277
4-284
Media Resources
Overview
zl
5-1
5-1 54
Objectives
Media Resources Overview
Media Resource Functions
5-3
5-4
5-5
5-7
Media Resource Signalingand AudioStreams Voice-Termination Signaling and Audio Streams Audio-Conferencing Signalingand Audio Streams Transcoder Signaling and Audio Streams MTP Signaling and Audio Streams MTP Types MTP Functions and Requirements AnnunciatorSignaling and Audio Streams MOH Signaling and Audio Streams Conference Bridge Overview Software Audio Conference Bridge Hardware Audio Conference Bridge Conferences per Resource Built-in Conference Bridge Resource Characteristics Meet-Me and Ad Hoc Conferencing Characteristics Conference Bridge Media Resource Configuration Step 1a: Activate Cisco IP Voice Media Streaming App Service Step 1b: Configure Cisco IP Voice Media Streaming App Service Parameters Step 1c: VerifySoftware Conference Bridge Media Resource Step 2a: Configure Cisco IOS Enhanced Conference Bridge Step 2b and 2c: Configure and Verify Cisco IOS Enhanced Conference Bridge Step 3: Configure Cisco CallManager Service Parameters Relating to Conferencing Meet-Me Conference Configuration Configure a Meet-Me Number or Pattern
MOH Overview
MOH Sources
UnicastMOH Multicast MOH
5-8 5-9 5-10 5-11 5-12 5-13 5-15 5-16 5-17 5-18 5-19 5-20 5-21 5-23 5-24 5-25 5-26 5-27 5-28 5-29 5-31 5-35 5-37 5-38
5-40
5-42
5-43 5-44
5-45
MOH Configuration Step 1: Plan Server Capacity Step 2a: Manage MOH Audio Files Step 2b: Configure MOH Audio Sources Step 2b: Configure Fixed MOH Audio Source Step 3: Configure MOH Server Step 4: Verify MOH Service Parameters Step 5a: Configure Multicast MOH Audio Sources Step 5b: Configure Multicast MOH Server Step 5c: Configure a Multicast Enabled Media Resource Group Annunciator Overview and Configuration Annunciator Features and Capacities
Annunciator Performance
5-46 5-47 5-49 5-51 5-52 5-53 5-54 5-55 5-56 5-58 5-59 5-60
5-61
5-62
5-63 5-64
Media Resource Design Media Resources Access-Control Example Intelligent Bridge Selection Intelligent Bridge Configuration Media Resource Access-Control Configuration
2010 Cisco Systems, Inc. Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) v8.Q
Step 1: Configure MRGs Step 2: Configure MRGLs Step 3: Configure Phones with MRGLs Summary
References
Module Summary
References Module Self-Check
5-77
5-77 5-79
5-81
6-1
6-1
Module Objectives
Configuring Cisco IP Phone Services
6-1
6-3
Objectives
Cisco IP Phone Services Overview
6-3
6-4
6-6 6-7
6-10 6-11
Default Cisco IP Phone Services Example: Corporate Directory Cisco IP Phone Services Redundancy
Cisco IOS SLB
6-12 6-14
6-15
Use of DNS to Provide Cisco IP Phone Services Redundancy Cisco IP Phone Services Configuration Step 1 Verify or Change Enterprise Parameters Step 2 Add a New Cisco IP Phone Service Step 3 Configure Cisco IP Phone Services Parameters Cisco IP Phone Services Subscriptions
Subscribe Cisco IP Phone Services: Administrator Subscribe Cisco IP Phone Services: End User
Summary
References
6-30
6-30
6-31
6-31 6-32 6-33 6-34 6-35 6-36 6-37 6-38 6-39
6-40
6-41
Subscribe CSS and Partition Considerations: Sample Scenario Presence Policy Example: Subscribe CSSs Presence Groups Presence Policy Example: Presence Groups Interaction of Presence Groups, Partitions, and Subscribe CSSs Cisco Unified Communications Manager Native Presence Implementation Step 1: Customizing Phone Button Templates Step 2: Applying the Phone Button Template to IP Phones Step 3: Configuring Presence-Enabled Speed-Dial Buttons Enabling Presence-Enabled Call Lists Enabling Presence on SIP Trunks Cisco Unified Communications Manager Presence Policies Configuration Step 3: Assigning Subscribe CSSs to Phones and SIP Trunks Step 1: Configuring Presence Groups
Implementing Cisco Unified Communications Manager, Part 1 (CIPT1) v8 0
6-42 6-43 6-44 6-45 6-46 6-47 6-48 6-49 6-50 6-51 6-52 6-53 6-54 6-55
2010 Cisco Systems. Inc
Step 2: Setting the Default Interpresence Group Policy Step 3a: Assigning Presence Groups to Lines and Phones Step 3b: Assigning a Presence Group to a SIP Trunk Summary
References
6-61
6-61 6-62
6-63
6-64
6-66
6-67 6-69
6-70 6-71
6-73
Relationship of Cisco Unified MobilityConfiguration Elements Cisco Unified MobilityConsiderations MVACall Flow with MGCP or SCCP PSTN Gateway Access CSS Handling in Mobile Connect CSS Handling in MVA Cisco Unified MobilityAccess-List Functions Operation of Time-of-Day Access Control MobilityPhone-Number Matching Cisco Unified MobilityConfiguration Step 1: Configure Softkey Template Step 2: Configure End User Step 3: Configure IP Phone Step 4: Configure Remote Destination Profile Step 5: Add Remote Destinations to Remote Destination Profile Step 6: Configure Service Parameters Step 7a: Configure Access List Step 7b: Apply Access List to Remote Destination Cisco Unified Mobility: MVA Configuration Procedure Step 1: Activate Cisco Unified Mobile Voice Access Service Step 2: Configure Service Parameters Step 3: Enable MVA per End User Step 4: Configure MVA Media Resource Step 5: Configure MVA on Cisco IOS Gateway Summary
References
6-74 6-75 6-76 6-77 6-78 6-79 6-80 6-81 6-82 6-83 6-84 6-85 6-86 6-87 6-89 6-90 6-91 6-92 6-93 6-94 6-95 6-96 6-97 6-99
6-100
Module Summary
References
Module Self-Check
6-101
6-101
6-103
6-106
ui
Lesson 4
Digit Manipulation
Overview Users of aphone svstem often need to reach various destinations, such as extensions within the
same site, different sites (sometimes with different dialing plans) within the same company, and
other companies within the same country or in different countries. Because these calls can take different paths, such as the IP WAN or apreferred public switched telephone network (PSTN) carrier, completing the calls often requires dialing various access codes, numbers ofdigits, or prefixes. In addition, restricting certain destinations, such as 900 numbers, is often prudent. To require users to understand the specific dialing patterns that are necessary to reach these
various destinations is impractical and inconvenient. Digit manipulation, orthe ability ofCisco Unified Communicalions Manager to add orsubtract digits to comply wilh a specific internal dial plan or national numbering plan, is important to providing transparent dialing and to
creating a unilied dialing planfor end users.
This lesson describes the digit-manipulation tools that allow a Cisco Unified Communications
external phone number masks, digit prefix and stripping, transformation masks, translation
and significant digits.
Manager Administrator to implement flexible and transparent dial plans. The lesson describes patterns, incoming called- and calling-party settings, called and calling transformation patterns,
Objectives Upon completing this lesson, you will be able to describe digit-manipulation elements in Cisco
Unified Communications Manager and describe how to implement them. This ability includes
beingable to meet theseobjectives:
Describe how to use external phone number masks in Cisco Unified Communications
Manager
Describe how to use translation patterns in Cisco Unified Communicalions Manager Describe how to use transformation masks in Cisco Unified Communications Manager
Describe how to use digit stripping and digil prefixes in Cisco Unilied Communications
Manager
Describe how to use significant digits in Cisco Unified Communicalions Manager Describe how muse global transformations in Cisco Unilied Communications Manager Describe how to use incoming number prefixes in Cisco Unified Communicalions Manager
4-158
Cisco IP Phones
]p Phone
How to manipulate
Off-Net Calls
Calling
Called
1002 9.1408-555-1111
706 555-1002
1408-555-1111
In some cases, manipulation ofthe calling and called (dialed) string is required before acall can be routed. For example, when acall to aPSTN number comes from an internal network, the access code 9must be stripped from the called number before the call is sent out to the PSTN.
PSTN number.
The calling-party number must also be changed from afour-digit extension to acomplete E. 164 In the example in the figure, an IP phone with extension 1002 calls PSTN phone 408 555-1 111. by dialing the PSTN access code 9followed by the PSTN number. Stripping 9from the called expanded to acomplete PSTN number so that when tlie PSTN phone rings, it sees the call
menu.
number before sending the call to PSTN is important. Otherwise, the PSTN switch will be unable to route the call to the correct destination. In addition, the calling-party number must be
coming from PSTN number 706 555-1002 rather than from extension 1002. This process allows the PSTN phone to call back the number conveniently from its Received/Missed Calls
Note In some countries, the calling-party number must besettothecorrect PSTN number ofthe
PSTN subscriber line or trunk.
Single-Site Off-NetCalling
4-159
Digit-Manipulation Requirements
1his topic describes some examples ofdigit-manipulation requirements.
lanipulation Requirements
Internal to PSTN
Internal lo PSTN
Internal to internal
PSTN to internal
outgoing PSTN calls, use either the external phone number mask orthe calling-party
transformation mask ofthecalling party in the PSTN route pattern or roule list.
To strip a PSTN access code before sending the call to the PSTN, use digit stripping
discard digits instruction (I>DI>in the PSTN route pattern or route list. To expand or modify an abbreviated number lo reach the actual destination (such as when
the access code 0 must be converted lo the actual internal extension ofthe operator), create
a translation pattern and use the ealled-party transformation maskto convertthe number.
This action is also applicable for calls to on-net sites, for which you must use the private IP
WAN as much as possible, even though the user calls the PS'fN numberto reach those
sites.
Io convert direct inward dialings (DIDs) lo internal directory numbers on incoming PSTN calls, use called-part) transformation masks in a translation pattern, orlimit the significant
digits on the appropriate gateway.
4-160
Description
Dialed number matches 9.!Route pattern configured with tliefollowing: - Called-party transformations > Discard digits: PreDot
- Calling-parly transformations: 40855530XX
- Route to the gateway
Cisco Unified Communications Manager strips off (discards) digit 9from thedialed
number andsends 13035556007 to PSTN via the Gateway, after modifying the calling-party number from 1005to406555-3005.
PSTN phone 303 555-6007 rings and sees408 555-3005 asthe calling number.
Tlie figure shows an example ofan internal caller dialing aPSTN number by using the PSTN
access code 9 followed by the PSTN number. In this example, the following digit
manipulations occur:
Cisco Unified Communications Manager discards the digit 9 before sending the call out lo
the PSTN.
The internal extension calling-party number isexpanded to the complete PSTN number.
Simple called- and calling-party transformations are used in the PSTN route pattern, to achieve
these two objectives.
4-161
PSTN phone dials 1-408-555-3010: PSTN switch routes the call tothe gateway and Cisco
Unrfied Communications Manager
incoming call dialed number matches 40855530XX translation pattern, configured as follows
Called-party transformation > Calledpartytransform mask 10XX (Optional) Calling-party transformation > Prefix digit. 91 CiscoUnified Communications Manager translates408555-3010 to 1010.
directory number.
Cisco Unified Communications Manager looks up1010 andfinds a registered phone with lhat
Step 2) prefixes the calling number with 91 toallow the internal usertocallbackthe PSTN
caller from IP phone via the Directory button.
Cisco Unified Communications Manager presents the call to extension 1010. It optionally (see
fhe PS IN phone calls the complete k.164 number ofthe destination. The PSTN gateway receives the call with 10 digits and passes it on to Cisco Unified Communications Manager. flic complete PSTN number is converted to the internal number by using the called-party
transformation mask ofa translation pattern that covers the complete DID range. The
resulting number matches an internal extension, andCisco Unified Communications Manager forwards the call to the IP phone that is registered with that extension. The IP phone receives the call, which is listed in the Received Calls menu, lb make it
easier for the IP phone user tocall back the number, you can use a calling-parly
transformation mask in the same translation pattern to add 91 lo the caller number, fhis
step is optional because the IP phone user can edit the number and manually add access
code 9 and long-distance code 1before calling back the PSTN number.
4-162
Characteristics
Significant digts
Caled- and caling-paity transformation pattern
Manager forincoming cals from a PSTNgateway or from a trunk />ppiescasing- and caIIed-number transformations forinbound or
outbound caBs, can be appSed to various coniguratton elements such as device pools, gateways, trunks, and so on Modifes the calling number of incoming PSTN calls, based on
The figure shows the main elements ofdigit-manipulation configuration, and their
characteristics. These elements will be explained in detail in subsequent topics.
4-163
This figure shows the relation ofcalling- and callcd-party digit-manipulation methods for calls lhat come from an IP phone or a gateway, respectively.
Cisco Unified Communications
it-Manioulation Mel
Caling-parly
transformation method
Caiied-party
transformation methods,
Incoming calling-party
seltmgs digit stripping,
transformation CSS
in order of operation:
Called-party settings,
transformation CSS
Calling-party
transformation
Caled-paity
ffa reformation methods, in order of
Calling-party transformation
methods, in order of
Calling-party transformation
methods, in order of
method
operaton-
operation1
Cal ling-parly
transform a ion
CSS
operation.
Discs a digits,
transformation mask,
External phone number mash, transformation mask, digit prefix Numbenng plan,
numoertype
Calling-party transformation
CSS, caller ID directory
number
While Cisco 1inilied Communicalions Manger processes a call, calling- and ealled-party
number can be modified at manydifferenl locations and by using various methods.
The figure shows when, where, and how digitmanipulation can lake place.
4-164
The external phone number mask instructs the call-routing component to use the external (PSTN) phone number ofa calling IPphone, rather than its internal directory number, for the
callerIDinformation. Theexternal phone number maskis set on a line-by-line basis on the Director. Number Configuration page of a device. The use ofthe external phone number mask
is enabled globally per PSTN route pattern.
The external phone number mask configuration can beapplied to many different call-routing
components and functions:
Roule pattern
Translation pattern
4-165
0&r^oi>NiBiibiCflJiBmj1wfl
<fv
Device n configuration
Type external PSTN number in
the External Phone Number Mask field
& PiUtmtiul&maiHB
i s. x *% m tH &
Ihe figure illustrates how to configure the external phone numbermaskon a line and how to
enable lis use in route patterns.
4-166
replacescalling-party number to 408 555-3005, strips9 from the dialed digits, and
sends call to PSTN gateway.
PSTN phone303555-6007 rings and sees 408 555-3005 as the calling number.
The figure shows a step-by-step example and description ofthe use ofanexternal phone
number mask.
4-167
Use this powerful tool to manipulate dialed digits and catlingparty number for any type of call.
When the digits match the translation pattern, Cisco Unified Communications Manager does not route the call to an outside entity (for example, a gateway); instead, it performs the translation first and then routes the call (to another translation pattern or to a route pattern).
Cisco Unified Communicalions Manager uses translation patterns lo manipulate dialed digits
before routing a call or to manipulate the calling-party number. In some cases,the dialed number is not the number that the system uses. In other cases, the dialed number is not a number that the PSTN recognizes. The translation pattern can also he used to block certain
patterns.
thesesituations, a unifonn dialing plan can be created and translation patterns can be applied to accommodate the unique office codes at each location. Thefollowing are additional examples
of situations in which translation patterns can be used:
Security desks and operator desks
Hotlines with a need for pri\ate line, automatic ringdown (PLAR) functionaliU Intension mapping from the public to a private nelwork
4-168
Routa Pattern
Translation patterns use theresults ofealled-party transformations as a setof digits for a new analysis attempt, 'fhesecond analysis attempt might match a translation pattern. In this case,
CiscoUnified Communications Manager applies the calling- andealled-party transformations ofthe matching translation pattern and uses theresults as the input for another analysis attempt.
To pre\ ent routing loops, Cisco Unified Communications Manager breaks chains of translation
patterns after 10 iterations.
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qurim
Go to Call Routing > Translation Pattern > Add New. Enter the Translation Pattern, including numbers and wildcards
(do not use spaces).
Specify the Calling PartyTransformations, Connected Party Transformations, and Called Party Transformations settings
(applicable only if Route the Pattern is selected).
Configuration of a translation pattern is like configuration of a roule pattern. Fiach pattern has
calling- and called-part\ iransfonnations sellings. The difference is that when Cisco Unified Communications Manager appliesthe translation pattern, it starts the digit analysis process
o\er. lo perform another call-routing process for the modified number.
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fc:
Route Option
Calling-, Connected-,
and Called-Party Transformation Settings
To configure atranslation pattern, choose Call Routing menu, and then choose Translation
Pattern. You can define the route pattern lomatch the calling- orcalled-party transformation
settings lhat shouldbe applied.
Ifyou click the Block This Pattern radio button, you must choose the reason for the translation
pattern to block calls. Choose one ofthese values from the drop-down list:
No Error
Unallocated Number
The transformation settings are not applicable when the Block This Pattern radio button is
selected.
Ifthe translation pattern contains the (: sign, you can select a Numbering Plan and Route Filter
to match certain number patterns ofthe selected numbering plan.
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San JOSe
PSTN
PSTN DID Range
Employee
Phones
Mtendani
14111)
408 555-1XXX
(attendant).
When the DID range from the centraloffice (CO) does not match the internal direetorv' number
range, a translation pattern can he used to map dialed DIDs to internal directory numbers.
Inthe figure, a San Jose. California, company has a PSTN DID range of 408 555-lXXX.
However, all the internal four-digit extensions begin with 4XXX. When the company receives an incoming call, the company can use a translation pattern that matches the assigned PSTN
DID range (408 555-1.XXX) and that has a transformation mask 4XXX. This mask converts the
dialed 408 555-I XXX PS'IN numbers to a 4XXX internal range, conserving the last three
digits. After Cisco l'nified Communicalions Manager applies the transformation mask, it
perfomis a new call-routing lookup for the translated tour-digit number, finds the directory
number in its call-routing table, and routes the call to the corresponding IP phone. In addition, there is a translation patient XXXX with a ealled-party transformation mask of 4111, This pattern routes callsthat are placed to unassigned directory numbers to 4111 (that Is. the directory numberofthe attendant). Assume that director; number 4333 does notexist, and an internal user dials 4333. Because no directory number 4333 exists, the translation pattern XXXX is the bestmatch, and the call Isrerouted to 411 I. Thesame happens for outside callers who dial 408 555-1333. Such a call first matches the translation pattern 408 555-IXXX and therefore is translated to 4333. After the translation, the call is processed likean internal call that is placed to 4333; Cisco l'nified Communications Manager does not find a directory
number entry and therefore matches XXXX again. The call is rerouted to 41 11.
Transformation Masks
Modify either the calling
number or called number (dialed digits)
An X in 3 mask
Birough.
Mask. 80S236XXX
806236000
Blanks block
number digits.
Dialing transformations allow the call-routing component to modify either the calling number orthe dialed digits ofa call. Transformations that modify the calling number are calling-party
transformations: transformations that modify the dialed digits areealled-party transformations. Transformation masks use mask operations to allow the suppression or insertion of leading digits or the changing of some, butnotall, digits.
Amask operation requires two items ofinformation: the number tomask and the mask itselt. In the mask operation. Cisco Unified Communications Manager overlays and aligns the number with the mask. That way. the last character ofthemask aligns with the last digit ofthe number. Cisco Unified Communications Manager uses the corresponding digit ofthe number wherever
the mask contains an X. If the numberis longerthan the mask, the mask obscuresthe extra
digits. Note Cisco Unified Communications Manager also allowsthe configuration of called translation
patterns, which translate dialed numbers by using transformation masks There are two main differences betweena route pattern with ealled-party transformation and a translation
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settings.
Transformation masks
! ID Pr-esp i(a
configured at route-list level have priority over those configured at routepattern level.
Ce[^N|,
20*
Os. Cri.
a^FlMrtimy
t C allManao.
Transformation masks can be configured inroute patterns, translation patterns, and per route
group {in route lists).
Ihe calling-parn and called-part} transformation settings that are assigned lo route groups in a route list override the corresponding transformation settings that are assigned to a route pattern
that is associated with that route list.
Usually, transformation masks areapplied at the route-list level. In this way. a different
transformation mask can be assigned to each roule group in the roule list.
l-'or example, a network administrator has created two route groups: the PSTN route group and
the IP WAN route group. Both of these route groups contain muliiple gateways that connect to theirrespective networks. When Cisco l'nified Communications Manager forwards a call to a gatewav in the PSTN route group, the network administrator applies a mask thattransforms the
number into an h. 164-compIiant phone number. However, when Cisco Unified
Communications Manager uses a gatewav from the IP WAN routegroup, Cisco Unified Communications Manager leaves the numberas a four-digit extension.
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Digit Prefix
Calling Party Tro information* -
Default
Party Transformations
and Called Party
Transformations
settings
Default
;oxx
Cisco CallManager
The digit prefix feature prepends digits toa number. Any phone keypad digits from
wellas * and#. can be prepended to the calling andcalled numbers.
The digit prefix feature can be applied toa calling- orealled-party number and confii under the corresponding transformation setting inthe route-pattern or translation-palt
configuration.
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Digit Stripping
The digit stripping feature is used tostrip digits from a dialed (ealled-party) pattern.
Digit Strippinj
Used to strip digits from a pattern
Is part of Called Party Transformations settings (Discard Digitsfield) DDl removes a portion ofthe dialed digitstring before passing the
number on
Ifthe @ sign (numbenng plan) is used in the pattern, not all DDIsare
supported
A DDl removes part of the dialed digit siring: for example, when an access code is needed to route the call to the PS'IN but the I'S I N switch does not expect thai access code. The DDl then passes the number on to the adjacent svstem.
Digit stripping Isconfigured under the ealled-party iransfonnations, by selecting a DDl. Digit stripping can be configured in route patterns and in routegroups of a route list.
For North American Numbering Plan (NANP)patlerns (a]), the entire rangeof DDIs is
supported. With non- a patterns, onlv a subset of fhe DDIs can be used.
For the PreDot DDl to work, the route pattern must include a dot (.). which is not dialed but
which Cisco Unified Communications Manager uses lo determine how manv digits to strip(all
digits before the dot).
PreAt
951010321011 33 1234*
The table inthe figure lists the most important examples of DDIs that are supported inCisco
Unified Communications Manager andexplains howtheywork. Note By default, Cisco Unified Communications Manager automatically removes a trailing #.This
behaviorcan be controlled via the Service Parameter > Call Manager > Clusterwide
parameter, which can besettoTrue (default) orFalse. If this parameter issetto False, the Trailing-* DDl will beapplied only if thecalled number includes a # as thelastdigit.
Note
The name 10-10-Dialing is used onlyfor historical reasons. In the past, 1010was used as carrieraccess code, followed bya three-digit Carrier Identification Code (CIC). Today, only the first three digits (101) indicate the carrier access code, followed byfour digits forthe
CIC.
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Cisco Unified
Communications
Manager
PBX
The PreDot and NoDigits DDIs are the onlv DDIs that can be used if the pattern does not
contain the a sign.
In the example in the figure. Ci>eo I inificd Communicalions Manager applies the PreDot DDl to the 9.SXXX route pattern. Cisco Unified Communications Manager then strips the 9 from
the dialed digits and sends only the 8123 lo the PRX.
InCisco Unified Communications Manager Administration, the Discard Digits menu, which the figure shows, is available via Call Routing >Translation Pattern or ( all Routing >
Route Pattern.
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Manager
consists of:
Carrier access code*: 101
-This is still called10-10dalirig altiojgh 101is now tie carrier access code and the CIC is tour
In this example. Cisco Unified Communications Manager applies the PreDot 10-10-Dialing DDl to the9. a, route pattern. This compound DDl performs two functions. First, the DDl strips
the access code 9 from the dialed number (9-1010-288-1-214-555-1212), then it removes the
carrier selection (the carrier access code 101 followed bythe four-digit CIC 0288) and sends
onlv 1 214 555-1212 to the gateway device.
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inificant Diqits
Instruct Cisco Unified Communications Manager to pay
attention to only the least-significant n digits of the called number, for incoming calls from PSTN or from another Cisco Unified Communications Manager cluster Are part of gateway and trunk configuration Affect all incoming calls received by the gateway or trunk; are not recommended for variable-length extension numbers
Ihe Significant Digits feature instructs Cisco Unified Communications Manager to pav attention to onlv the least-significant n digits ofthe called number for incoming calls that are received bv a galewav or a trunk. For example, setting Significant Digits to 5 on a PSTN gatewav causes Cisco Unified Communications Manager to ignoreall but the last fivedigitsof the called number for incoming PSTN calls. Using this feature is the easiest approach lo converting incoming PSTN called numbers lo theirinternal extensions. However, thisapproach
affects all calls that are received from the gateway or trunk. Iherefore. this approach is not
recommended when variable-length extension numbers are used.
I CtfmjPwtr PtifnUW*
Ofi-jiiMl&r
CrKB CallHanoflcr
Cmg CillHBoagpr Ciasi C*m*srh"s*f
Civn CallM*n5flB-
The Significant Digits feature is configured on the Gateway Configuration or Trunk Configuration pages and affects all incoming calls that the gateway or trunk receives.
Go to Gateway Configuration orTrunk Configuration >Call Routing Information Inbound Calls. In the Significant Digits field, specify the last n digits ofthe called number that Cisco Unified Communications Manager should process for incoming calls that are received by
the PSTN gateway or trunk.
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Description
In the example in the figure, the PSTN gateway receives an incoming call with the destination number 408 555-1010. If'Significant Digits = 4 isconfigured on the Gateway Configuration
page in Cisco Unified Communications Manager, then Cisco Unified Communications Manager strips off all digits except the last four digits (10I0). Cisco Unified Communicalions
Manager then looks up this number (1010) in itscall-rouling table and forwards the call to the
IP phone that is configured with that director; number.
Note
In contrast to using translation patterns to mapdialed E.164 numbers to internal directory numbers on incoming PSTN calls, this solution performs only one call-routing table lookup However, significant digits can be used onlyifallthe significant digits are the same (thatis, the complete directory number isalso used inthe DID range). If the PSTN DID range (for example. 1XXX) is different from the directory numbers that are used forthe phones (4XXX),
then translation patterns are required and significant digits cannot be used
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. Identical transformation patterns with different transformation settings can exit if they are
Egress devices (including phones) can be configured with called- and calling-party transformation CSSs. (At phones, only calling-party transformat.on is supported.)
The configured transformation CSS determines which transformation patterns are visible to
the device.
Called- and calling-party transformations arc applicable only to calls from Cisco Unified Communications Manager to the corresponding dev.ces, not for calls that are rece.ved Irom devices. The supported devices are gateways, trunks, and phones. To say that he transformations apply to outgoing calls is incorrect outgo.ng calls are usually defined as calls
that exit the cluster. As mentioned, transformation patlerns apply to calls that are sen to
call: itcan be seen as an outgoing call leg ofan internal or incoming call.
gatevvavs trunks, and phones. Acall to aphone is usually not considered to be an outgo.ng
device vou can configure the devices to use the corresponding device pool settings. If ne.ther the device nor the device pool are configured with atransformation CSS. no transformat.on is
performed.
Communications Manager receives from devices (call ingress or incoming call legs).
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rent
+1703XXXXXXX->XXXXXXX. suO
PreDot DDl
HC_G'V
Area Code 7D3
BranchJSW
Area CorJe 303
Called Party
Transformations CSS A
49691 -14085551234
Called Party
Transformations CSS B.
-17035551234 + 13035551234
transformations applv to outgoing call legs onlv. the incoming call leg is not considered in the figure. Only the localization ofthe called number at the selected outgoing gateway (after call
In the example, global call routing is enabled. Because called- and calling-party
1here are four callcd-party transformation patterns in three partitions. Partition Ais specific to
specific to Branch (area code 303), The IIQ gateway is configured wilh aealled-party
a ealled-party transtonnation CSS. which includes partitions Band C.
1iy (area code 703). Partition Bincludes generic transformation patterns, and Partition Cis
transformation CSS. which includes partitions Aand B. The Branch gatewav is configured with
pattern mpartition Cprov ides the same function for Branch numbers. The common partition B includes two transtonnation patterns: V+IXXXXXXXXXX represents all (other) 11 S area codes and \+.\ represents all other numbers (that is. international destinations)
Note
The (onlv) transformation pattern in partition Amodifies all 1\() globalized numbers to a seven-digit subscriber number (and sets the number type, accordingly). The transformation
As in call routing, a more specific pattern has priority over a less specific pattern.
As aresult ofthis configuration, culls to the following four destinations are transformed differently depending on the gateway towhich they are routed:
If the ISDN provider does not support number types, then a prefix of 011, instead of number
type international must be used
+17035551234 is sent out as 5551234, with type subscriber, at the HQ gateway and as 7035551234, with type national, at the Branch gateway.
Ifthe ISDN provider does not support number types, then a prefix of 1, instead of number
type national, must be used.
Note
+13035551234 is sent out as 3035551234, with type national, at the HQ gateway and as 5551234, with type subscriber, at tlie Branch gateway.
Ifthe ISDN provider does not support number types, then a prefix of 1, instead of number
type national, must be used.
Note
The different transformation CSS causes the different handling ofthe last two calls.
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MQ_GW
and HO ^hones Area Cooe 703 B
Calling Party
Transformations CSS A
Calling Party
Transformations CSS1 B. C
In the example, global call routing is enabled. All culling numbers (gateways and phones) are
already globalized (performed at call ingress). Only the localization ofthe calling number at the
selected outgoing gatewav (after call routing and path selection) is considered in this example.
There are three calling-party transformation patterns in three partitions. Partition A is specific
to HQ (area code 703). Partition B includes a generic transformation pattern for U.S. caller IDs. and Partition C is specific to Branch (area code 303). fhe HQ gatewav and HQ phones are configured with a calling-parly transformation CSS that
includes partitions A and B. The Branch gateway and Branch phones are configured with a calling-party transformation CSS that includes partitions B and C. fhe (onlv) transformation pattern in partition A modifies all HQ globalized numbers to a seven-digit subscriber number (and sets the number type, accordingly). The transformation pattern in partition C prov ides the same function for Branch numbers. The common partition II includes transtonnation pattern \+lXXXXXXXXXX and represents all (other) U.S. area codes.
Note
As a result of this configuration for the three call types that are shown, the calling-party number
will be transformed as follows:
For a call front an HQ phone to the PSTN via the HQ gateway, the (globalized) caller ID of
the phone. +17035551002. is transformed to 5551002. with type subscriber. For a call from a Branch phone to the PS'IN via the HQ gateway, the caller ID ofthe phone. +13035551001. is transformed to 3035551001. with tvpe national.
- -
fipFftjII
From the Cisco Unified Communication Manager menu, choose Call Routing > Transformation > Transformation Pattern > Calling Party Transformation Pattern and click Add New, to create calling-party transformation patterns. In the Pattern Deilnition section, define a matching pattern and assign a partition to this
pattern. In the Calling PartyTransformations section,you can specify calling-number transformation settings like those that are found in a route-pattern or translation-pattern
configuration.
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From the Cisco L'nified Communication Manager Administration menu, choose Call Routing > Transformation > Transformation Pattern > Called Party Transformation Pattern and click Add New, to create ealled-party transformation patterns. In the Pattern Definition section, define a matching pattern and assign a partition to this
pattern. In the Called Part; Transformations section,you can specifycalled-number transformation settings like those that are found in a route-pattern or translation-pattern
configuration.
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Transformation CSS
This figure shows how to configure CSSs to apply transfonnation patternsthat arc associated
with partitions.
Transformation CSS
CaSifig .s#3reti u<e tnnflourciitftn
- [aai*g 5wrcti Skc Infvnnttlan ;
J
_
--
be applied to many configuration elements, such as phones, gateways, device pools, and so on.
The CSS configuration to apply transformation patterns that are associated with partitions is identical to the CSS configuration to configure calling privileges. During the Cisco Unified Communications Manger digit-analyzing process, transformation
patterns within partitions are treated like any other dialable number. Therefore, creating
completely independent CSSs for calling privilege implementation as well as for called- and calling-transformation patterns is highly recommended.
4-1f
Not applicable to calls received from phones Use external phone number mask in E.164 format for globalization of phone numbers
I'hev allow the configuration of prefixes, digit stripping, and transformations to be applied to calling- andcallcd-party numbers on incoming calls (that is, callsthatare received through a gatewav or a trunk). Different settings can be configured per number type
(unknown, subscriber, national, and international).
Incoming calling- andealled-party settings can be configured for the device or device pool,
and as serv ice parameters ofthe Cisco CallManager service.
Incoming calling- andcalled-partv settings are not applicable to callsthat are received from phones. To globalize the calling-party number of a cal! that is received from a phone,
configure the external phone number mask in li. ]64 formal.
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Number
Manipulation
Gateway or
Trunk Yes
MGCP
Gateway
Yes No
Yes
No
No
type
Not every trunk or gateway type supports incoming called- and calling-party settings.
Although H.323 trunks and gateways support incoming called- and calling-party settings based onnumber type. Media Gateway Control Protocol (MGCP) gateways support only incoming calling-party settings based on number type. Because Session Initiation Protocol (SIP) docs not support number types. SIP trunks support incoming calling-parly settings only for unknown
number types.
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A', 69 3056412
0044 1234 567B90
Hamburg
PSTN
Gateway
V
Frankfurt Area (69) Hamburg Area (40)
Calling-party number of calls received through Hamburg gateway are normalized (globalized
to E 164 format)
v
Germany
+ for calls that are received with numbertvpc international, with additional two-digit
stripping
As a result of this configuration, the ealling-partv number of all calls that are received through the gatewav are normalized (globalized) to K.164 format, as shown in the figure.
4-192
To configure incoming calling-party settings, navigate to the Incoming Calling Party Settings
section on a Gateway Configuration or Trunk Configuration page.
The prefix field not only allows you to define anumber prefix but also accepts the following
entries:
<Blank>: If the Prefix field is empty, no prefix is added to thecalling number. Digit
strippingconfiguration still applies.
Default: Ifthe keyword Default isentered inthe Prefix field, the gateway or trunk configuration for this Number Type isignored. Instead, the incoming ealled-party settings on the device pool are applied. In this case, the Strip Digit setting isalso disabled.
If the Use Device PoolCSS check box is checked,the locally configured transfonnation CSS is
ignored. Instead, the transformation CSS that isdefined on the device pool is applied. In this
case, the Prefix and Digit Strip settings lhatareconfigured locally areused.
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3.
Incoming calling-part v sellings follow a specific order when applying digit manipulations
1. Digit stripping configuration is applied.
2. The configured prefix is added.
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This figure shows incoming calling- and ealled-party settings in the device pool configuration.
incoming Calling Party and Incoming Called Party Settings at the Device Pool
On the Device Pool Configuration page, as well as for H.323 gateway?
and trunks, Incoming Calling Party Settings and Incoming Called Party
Settings are available.
The configuration of incoming calling- and ealled-party settings inthe device pool isalmost
identical to the configuration of these settingson gateways or trunks.
The only differences are the following:
No Use Device Pool CSS check box is present.
If the Default keyword is used in any Prefix field, thecorresponding incoming calling- or ealled-party settings, asdefined inthe Cisco CallManager service parameter, are used.
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alhiii
Three Ivpes of calling-partv transformations can be configured on route patlerns and per roule group in a route li>t. in the following order:
4. The external phone number mask instrtiets the call-rouling component to use the external phone numberof a calling station, rather than its director)' number or the caller ID infonnation. The external phonenumber maskcan be appliedon a line-by -line basis through the Director. Number Configuration screen on the device.
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Discard Digits
10-lOrDialing
9 1808S551221
Prefix Digits
Called Number
880SS551221
DDIs allow the discarding of subsections of a numbering plan, such as the NANP. DDIs
can also be used to discard PSTN access codes, such as 9.
The callcd-party transformation mask allows thesuppression or insertion of leading digits or changes to existing digits whileleaving others unmodified.
Prefix digits allow the prepending of oneor more digitsto the called number.
Ifmultiple transformations are configured, Cisco Unified Communications Manager applies the
transformations in the following order, as presented in the example:
1. DDl first discards the digits from the dialed number. 2. Transformation then continues; the ealled-party transformation mask adds or removes
additional digits.
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Digit-Manipulation Considerations
Tins subtopic describes manipulation considerations.
igit-fVlanipulation Consideration:
Called-party transformation priority for incoming
PSTN calls:
Signfieant Die**
The final called-party number used for the dialed-number matching process is a combination of all three-digit manipulations.
On a gatewav or trunk, three-digit manipulation options arc available for the incoming calledparty number, before Cisco Unified Communications Manager starts digit analysis.
Ihese three settings are applied in the following order: 1. Significant digits
2. Pre 11 \ directory number
Example 1
Routs Qroup
" ,'"".-,1ii:'|
Manipulal
Define* -.^F.
Ma rpulaions
Example 2
Roule Patarn
Route List
j*MDtflOBd
on
Rous Group
Example 3
Foroutgoing calls, three levels of digit-manipulation options are available: Digit manipulation that is configured on the route pattern
These three levels ofdigit manipulation are not cumulative. Only the digit-manipulation configuration ofone level will be applied. The hierarchy for these digit manipulations is as
follows:
1. Digit manipulation settings onthe roule pattern take effect only when neither the route list
nor route group has defined digit manipulations and if no transformation CSS on the gateway or trunk matches any transformation patterns.
2. If no configured transformation CSS at the gateway or trunk matches buttheroute listor route group has configured digitmanipulations, those manipulations are used. Possible
route-pattem digit manipulations are ignored.
3. If any manipulation matches via a gateway or trunk transformation CSS, then route-list,
route-group, and route-pattern configurations are ignored.
Note
Called- andcalling-party digit manipulations are independent ofeach other. Therefore, called-party modification can beapplied via a route pattern, whereas thecalling party can be manipulated based on a transformation CSS on the used gatewayor trunk.
4-199
Summary
This topic summari/es the key points lhat were discussed inthis lesson.
bummary
Digit manipulation is an essential dial plan function Itis mandatoryto
Summary (Cont.)
Cisco Unified Communications Manager transformation masks are an
Cisco Unified Communications Manager digit stripping provides an easy way to apply DDl to route patterns or translation patterns Cisco Unified Communications Manager significant digits functionality
Cisco Unified Communications Manager global transformations provide a flexible and scalable way to implementglobalization and normalization
forfunotions such as globalized call routing. Cisco Unified Communications Manager incoming number prefixes are
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v8.0
References
For additional information, refer to these resources:
Cisco Sv stems. Inc. Cisco Unified Communications System ReleaseS.x SRND. San Jose.
California. April 2010.
Imp:'''www.cisco.coin'en/fiS/does/voice ip comni/eucm/snid/8\/uc8\srnd.pdi. Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide,
Release 8.0(1). San Jose. California. February 2010.
hup: v\uv\.cisco.com/en/US/docs/voicc_ip_comm.''cucm/drs/8_{l_t/drsagJ101.html.
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4-202
Lesson 5
Implementing Gateway
Selection and PSTN Access
Features
Overview
Gatewav selection and calling privileges are important dial plan components. Calling privileges are used to implement class of service (CoS). Based on the calling device or line, some destinations are permitted to access call-routing table entries; others are not. Implementation
applications include time-of-day routing, vanity numbers, Client Matter Codes (CMC), and
Forced Authorization Codes (FAC). and discusses different usage scenarios.
This lesson describes the configuration tools lhat can be used to implement calling privileges
Objectives
Upon completing this lesson, you will be able to explain the need and uses for calling privileges and how to implement Ihem in Cisco Unified Communications Manager. Th.s ability includes
being able to meetthese objectives:
List applications for calling-privileges configuration elements Describe how lime schedules and time periods work and how they are configured Describe the functions ofgateway selection and CoS and how toconfigure them Describe how toimplement 911 emergency calls and vanity numbers Describe how to implement carrier selection based ontime ofday
Describe how CMC and FAC work and how they areconfigured
Describe how to implement CMC and FAC
Calling-privileges configuration elements are used primarily lo implement class ofservice (CoS) when vou must pemiit or den> access to certain destinations, depending on the caller, fxainples include classes of service (international, long-distance, or local) for public switched telephone network (PS'IN) access, direct access to managers versus being transferred bv
assistants, pennis.ions that are based ontime ofday or thai depend on authorization codes, and
so on.
However, the same configuration tools can be used to implement other applications. Typicallv. calls are not permitted or denied in these applications but are routed in adifferent way" depending on who is placing the call. Samples include vanity numbers and emergen'cv dialing, time ofdav-based carrier selection lor PSIN calls, or private line, automatic ringdown'(PFAR).
Note
PLAR causes a phone to dial a specific, preconfigured number assoon asthe phone goes
off-hook.
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Description
CoS imits access tocertain destinations and specifies tracStional caling privileges: who isallowed tocall where
or whom.
Haeh application that is listed in the table has different requirements regarding the configuration
tools orelements that are needed toimplement the application.
CoS- CoS limits access to certain destinations. Partitions and CSSs, time schedules and
time periods, blocked patterns, CMC. and FAC (listed in descending popularity) are
implement vanity numbers and emergency dialing.
911 emergency calls and vanity numbers: Partitions and CSSs are typically used to
Time of day-based carrier selection: Partitions and CSSs, and time schedules and time
periods are typically used for this application.
Mandator call accounting: CMC are typically used to extend calls only ifthey are
marked with accounting information.
PLAR: Translation patterns, partitions, and CSSs are typically used to implement PLAR.
4-205
Identical route pattern is put into multiple partitions. At least one partition hastime information applied. If this partition is listed first in CSSs, it takes precedence
over other partition during the time applied to it. If timedoes not match, second partition of CSS is used
You can implement time-of-dav routing in Cisco Unified Communications Manager bv using time schedules and time periods to apply time and dale attributes to partitions. Time periods^
define time ranges ordales and are grouped into time schedules. Time schedules are then
assigned to partitions.
ACSS that includes a partition that isassocialed with a lime schedule has access to the partition onlv when the current date and time match the dale and time information that is specified in the time schedule. Ifthe configured lime schedule does not match the current date
and time, the partition is logically removed from the CSS.
Time-of-dav routing can be used to route calls differently based ontime: Identical route patterns are created and put into different partitions. At least one of these partitions has anapplied time schedule.
Ifthe partition with the time schedule is listed first in CSSs. that partition lakes precedence does not match the configured lime schedule, lite partition lhat has Ihe assigned lime schedule is ignored, and the next partition becomes the partition with the highest prioritv.
over other partitions during the time that isassociated with that partition. Ifthe current time
The following are some examples ofwhen time-of-day routing can be used:
Using time-of-day routing to control the call-routing path that is based on the current time,
Multiple providers for international calls might be available. Some ofthese
hours).
providers might have different prices, depending on the hours ofthe day (with calls typically being more expensive during business hours and less expensive during offWith time-of-day routing, international calls tocertain countries can use the
cheapest available provider, based on the current time. Therefore, make use ofthe
cheapest offer for any given time, instead ofusing the same provider for all calls to
certain countries.
Single-Site Off-NetCalling
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Repetition
M-F
weekdayhrs_TP
weekendhrs TP
Sat-Sun
newyears_TP noofficehours_TP
Time Schedule
January 1
Sat-Sun
Time Periods
SB* .lif
Time schedule
Group of time periods
*'i-.iiir^rF
Partition
3coAuaiir_!9 [
Regbnplo/oa*.TP
Atime period specifies atime range that isdefined by astart and end time and a repetition interval (davs ofihe week or a specific calendar date). One or more time periods are assigned to
a time schedule. The same time period can be assigned tomultiple time schedules.
Alime schedule is a group oftime periods. Time schedules are applied topartitions and make
the partition inactive in a CSS when the applied schedule does not match the current dale or
time.
In the example. CSS> that include the partition CiscoAuslinJ'T can access Ihe partition onlv on
Mondav lo Fridav. from 8 a.m. to 5 p.m. (0800 lo 1700).
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The figure shows an example ofhow international calls can be blocked during weekends and
on January I.
li 1 Cut-rain-m:: 10:>*t
RoutePattem:9.011!
Partition: Standard Route to PSTN
To implement this restriction, first create aroute pattern that allows international calls. Put the
route pattern into the standard partition, which has no time schedule applied.
Create a second, identical route pattern and putit into the Weekend partition.
Configure atime period for Saturday to Sunday, 0000 to 2400 hours. Configure another time period with aspecified date: January 1. Put these two lime periods into atime schedule, and
assign Ihe time schedule to the Weekend partition.
Assign to phones aCSS that contains the Weekend partition first, followed by the standard
partition.
So far. phone users can dial international calls at any time. During weekends and on January I.
thev are allowed to dial international numbers, according to the Weekend partition. When that
partition is inactive (all weekdays except January 1), users are allowed to dial international numbers, according to the standard partition. The task now istoconfigure the route pattern in
the Weekend partition to be blocked.
Note
Route patterns andtranslation patterns can be configured with the Block This Pattern
parameter, to deny the call if the call-routing logic (best-match, earlier-listed partition) selects
the pattern.
When the route pattern in the Weekend partition isconfigured toblock matching calls, international calls are impossible when the Weekend partition (as listed before the standard partition in Ihe CSS ofthe phones) isactive: on weekends and on January 1.
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Time-of-Day Routing-Configuratii
Procedure
Create time periods
: Create time schedules.
Step2
Step 3
Assign time schedules to partitions that should beactive only during the time that is
specified in the time schedule.
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IoThursday9:00to
18:00.
U35
Hi
H\nteekdays2_TPis E3
I activeon Friday
9:00 to 12:00.
"I
L-turQrpBd .
Two time periods are created in this example. The first is active Monday through Thursday, a.m. to 6p.m. (0800 to 1800). The second is active on Friday, 9:00 a.m. to noon (0900 to
1200).
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- 51fltu
CD'
-Tune ^thedule Infartnatlon
When creating a time schedule, vou must configure a name and a list of time periods.
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Partition InfwnwlK
to partition.
Set fixed time zone or usetime zone of calt-
originating device.
Partition is active in CSS only whencall is placed during time specified in timeschedule. International calls are possible only during business hours.
schedule. CompanyWeekdays_TS, is applied to the Intl-PSTN partition. As aresult, the Intltime schedule.
Time schedules can be assigned topartitions. In the example, the earlier contigured time
PSTN partition is active in CSSs only ifacall is placed during the time that is specified in the
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Gateway selection is an integral function implemented by CSSs and partitions: - Depending on the originating device, different gateways are
used for outbound calls.
Headquarters
Branch
Calls to the PSTN should be routed via the gateway that is associated with the geographical location ofthe willing IP phone. This recommendation is crucial especially (but not only) in
multisite environments.
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BR phones should see only the route pattern that points to the
BR gateway.
Step 1
Step 2
Use CSSs and partitions to make sure that Headquarters phones will reach only the route pattern that points to the Headquarters gateway.
Step3
Also configure CSSs and partitions to make surethat Branch phones will reach only
the route pattern that points to the Branch gateway.
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HQ-Phones PT
HO Phonel ||
HO Phone 2 H
H |f
M
2001
HQ Phone 1 H HQ Phone 2 H
2002
M 2002
BR-Phones_PT
H 3001
BR Phone 1 H BR Phone 2 11
H 3001
BR-Phones_PT
||
BRPhone 1 N
H [1
m
302
9'
3002
BR Phone 2 jj
HQ PSTN PT HQ-RL
H j
ij
9.i
BR_PSTN_PT BR-RL
H H
Bv creating Ui) CSS. which contains the HQ_PSTN_lT partition, and BR_ CSS, which contains the BR_PST\_PT partition, vou have separated the two 9.! route patterns.
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BR CSS HQ CSS
Alkr you apply the appropriate CSS to theheadquarters (HQ) and branch (BR) IPphones, calls
to the PSTN will use the HQ gateway for HQ IP phones and the BR gateway for BR IP phones.
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HQ-RL
Second Choice
BR-RL
-I
HQ-RG
HQ CSS
I Firsl Choice
BR-RG BR CSS
408 555-9999
Bv adding the branch route group (BR-RG) as second choice to the headquarters route list (HQRL) and adding the headquarters route group (HQ-RG) as second choice to the branch route list (BR-RL). you can implement gatewav redundancy.
\IQ phones will normal 1\ alwavs use the HQ gateway for PS'fN calls. However. HQ-RL will also send calls to the I'S IN via the BR gatewav. in ease the HQ gateway becomes unavailable.
Note Configuration of gateway redundancy typically also requires additional digit manipulations of
the calling-party numbers, in case an HQ phone uses the BR gateway and vice versa.
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HQ Device Pool
Common-I'
.:"'
_,
This example clearly shows the simplification ofthe dial plan by using local route groups and
thus reducing the number of required route patterns and route lists. The HQ device pool uses HQ-RG asits local route group; the BR device pool uses BR-RG as its local route group. Only one common route pattern isused. This route pattern points to a route listthat contains only the standard local route group, so proper gateway selection is
maintained.
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CoS for Off-Net Calls This topic describes how to implement CoS in Cisco Unified Communications Manager.
Lobby
Employee
Executive
Emergency
Local
Long Distance
International
PSTN
Emplo/ee
CoS is the collection of calling pennissions that are assigned to individual users. CoS canbe
implemented in different way?,.
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applied to the respective phones. No CSSs are applied to lines, so the phone CSSs apply to all
lines.
The traditional approach ofCoS implementation in Cisco Unified Communications Manager involves placing external route patterns into partitions. CSSs are configured per CoS and are
Avoiding the use ofaseparate line CSS might sound reasonable. Aphone typically should have
the same privileges on all its fines.
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Traditional-Appro
Route
Route
Groups
Devices
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Traditional-Approach Example:
Multiple Sites
Partitions Route Lists/Route Groups
DeviceCSS
(4 tor Site 1)
Trflilg
Gateways
The problem inmultisite environments islhat partitions and CSSs must provide two functions:
The selective PSTN breakout is achieved by creating all PSTN route patterns once per
gatewayalways ina different, site-specific partition. Inaddition, you must duplicate all these route patterns, varying thepartition by a CoS-specific tag. Then, Headquarters users with a
different CoS can have a CSS that includes a partition that providesaccessto a certainCoS and
their local gateway.
The figure shows thissolution. Look at thepartitions that include PSTN targets. The Site IEmergency partition, SitelNational partition, and Sitellnternational partition provide
three classes of service to Sitel users. The same three CoS partitions must exist for each additional site (SiteNEmergency. SiteNNational, and SiteNInternational). To calculatethe
number of required partitions, multiply the number of required classes of service bythenumber
of sites, fhis solution does not scale to large deployments.
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When applied to large, multisite deployments with centralized call processing, the traditional approach to implementing CoS in a multisite environment can result in many partitions and CSSs. This configuration is required because the device CSS is used to determine both the path selection {that is. which PSTN gatewav to use for external calls) and the CoS.
You can signiticantlv decrease the total numberof partitions and CSSs lhat are needed. I'o do
so. divide these two functions between the line CSS and the device CSS. This solution is called
the line/device approach. Based on the wav in which the line CSS and the device CSS for each IP phone are combined, follow these rules to implement the line/dev ice approach: Use the device CSS lo prov ide call-routing information: for example, which gatewav to
select lor all PSTN calls.
Use the line CSS to block route patterns that are not allowed by certain CoS (independent
ofthe used PS'fN tiatevvav).
Line/Device Approach: Concept The figure shows how the line/device approach works in Cisco Unified Communications
Manager.
Fa eachPSTN gateway, route patterns exist onceina different partition. Line CSScontains a partition with a singte route pattern thatmatches
international numbers and has been configured as a blocked pattern.
Una CSS
Mtecfivaly Modes
un***#dtoutM
Route/Translation Patterns
(weiJWSns to CoS).
OevfctGSS
altars accetl to
at external route*. Routed Route Patterns
Create an unrestricted CSS foreach site and assign it to the phone device CSS. This CSS
should contain apartition that features route patterns that route the calls to the appropriate local
gateway for each site.
Create CSSs that contain partitions with blocked route patterns for those types ofcalls that are
not permitted by the CoS ofauser. Assign these CSSs to the lines ofthe user phone. For
instance, ifa user has access to all types ofcalls except international, configure that user line
(or lines) with a CSS whose first partition includes aroute pattern that blocks calls to 9.0111.
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lultiple Sites
Route Route
In the example in the figure, the linedevice approach is used, resultingin a significantiv
simpler configuration.
One partition is used per CoS to block undesired destinations, fhese partitions are included in the lineCSS ofthe device, to alwavs block accesslo destinations that are not permitted for the corresponding CoS. regardless ofthe PSTN gateway that the devices in a location use.
In addition, all possible PSTN route patlerns are created onceper I'SIN gateway. The patterns are put into a partition that is included in thedevice CSS. This process allows the local galewav
to be used for all PS'fN calls that the line CSS does not block.
Ihis approach has a significant advantage: Only a single, site-specific partition (and device
CSS) is required for each site, to allow local gateway selection. Also, only one partition per
CoS (independent ofthe site) is required.
Rather than requiring numerous partitions that are calculated by multiplying classes of service and sites, the number of partitions is determined by adding (he required sites and classes of
serv ice.
for example, using the traditional approach with four sites and four classes of service, lb
partitions are required: using the line/device approach, the numberof required partitions drops
to 8,
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911 calls must always be sent to the local PSAP Calls to the same number must be routed differentlyper phone
(location).
9!1is a single number to call for medical, fire, and police emergencies in the United States and
Canada. Calls to 911 are routed to a public safety answering point (PSAP). The PSAP isthe first-tier triage call center for emergency calls. PSAP operators dispatch orconference with
medical, fire, and police resources, as necessary.
Emergency calls must be sent toa local PSAP through the local gateway. In a multisite environment, this requirement means that emergency calls that arc placed tothe same number must berouted differently, depending onthe physical location ofthe calling phone. The same method is usually applied toall PS'fN destinations, lokeep voice traffic offthe IP WAN and to
keep local gateways free during any PSTN outbreak.
Note Emergency calling in theUnited States andCanada includes additional aspects that are not
covered in this course.
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Vanity Numbers
fhis subtopic describes the characteristics of vanitv numbers.
anity Numbers
Vanity numbers provide a certain local service.
Dial 7999 at any site to reach local IT support (on-net). Dial 7998 at any site to reach local travel agency (off-net). Number can be a route pattern, directory number, or hunt pilot. 911 emergency dialing has the same basic concept: Dial 911 at any site to reach local emergency services
(off-net. emergency call)
Vanity numbers prov ideaccess to a ccilain local serv icevv ithin anenterprise. Users should be able to dial the same numberto access the appropriate locally provided service, no matter
where the users are located. For example. 7999 might always connect users to local ITsupport.
Vanitv numbers are not limited to internal services (such as the IT-support example). These numbers canalsobe configured to reach external local services, such as taxi or travel agencies,
by using abbreviated dialing (forexample. 7998).
The vanitv numbercan be a directory number, a route pattern, a hunt pilot, or a translalion
pattern.
Note
4-228
This subtopic describes how to implement vanity numbers in Cisco Unified Communications
Manager.
Implementing vanity numbers is like configuring selective PSTN outbreaks (always using the
local gateway for PSTN or emergency calls): Step1 Create onesite-specific partition per site.
Step 2
For each service, configure the same vanity number (route pattern, directory number, hunt pilot, ortranslation pattern) once per site. Put that number into the sitespecific partition thatyou created earlier.
Step 3
Note
Put the appropriate site-specific partition into the CSS ofthe phones ateach site.
If abbreviated dialing is used to reach external local services (such as using 7998 to reach a local travel agency), then a translation pattern is used for thevanity number.
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Vanity-Number Example
The figure shows anexample of vanitv-number implementation in Cisco Unified
Communications Manager.
CSS:
Standard
IT Helpdesk
San Jose
San Jose user is connected to San Jose helpdesk phone New York user is connected to New York helpdesk phone
CSS:
In the example in the tigure. the vanitv numberfor IT supportis 7999. fhere are two sites: New
York and San Jose. The direetorv number 7999. in San Jose, isput into the San Jose partition.
The same directory number, in New York, is put intothe New York partition. Phones that are
in New York have the New York partition listed first in their CSS. Phones that are in San Jose have the San Jose partition listed first in their CSS.
If a San Jose userdials 7999. the call is routed to the IT helpdesk phone, fhe San Joseuser cannot accessthe New Yorkdirectory number7999 because the San Jose phoneCSS does not include the New York partition. The reverse applies to users in New York.
Additional Example
Ifthe desired serv ice is provided extentallv. a translation pattern can be configured to translate the appropriate vanitv number (forexample. 7998 to access a local travel agency) to thesitespecific PSIN number. Creating the vanity number once persiteand putting it into a sitespecific partition canensure that users always match the vanity number translation pattern for their respective sites. This match is achieved by including the site-specific partition in the
phone CSS.
In the example, onlv the following changeswould be necessary: Usetwo 7998 translalion patterns instead of two 7999 directory numbers. As wilh the directory numbers, put these translation patterns into site-specific partitions (San Jose and New York). Configure the San Josetranslation pattern with the PSTN number ofthe San Josetravel agency. Configure the New York translation pattern wilh the PS'IN numberof the New York travel agency. Make
sure that the translation patterns are assigned CSSs that allow the patients to use the local
PSTN gatewav to route calls out to the translated PS'IN numbers.
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- Configure required route patterns (international, long distance, local,and so on) once per carrier and pointto appropriate carriergateway; put routepatterns intoa carrier-specific
partition.
Single PSTN access with carrier access code (101) and four-digit
CIC
Depending onthe way that the long-distance orinternational carrier isselected, you can use oneofthe following two approaches to implement time of day-based carrier selection.
Dedicated Gateway Per Carrier
Step 1
Configure the required route patterns (patterns that should berouted differently based on the time of day) once percarrier. Point each route pattern to the appropriate
carrier gateway.
Step2
Step 3
Step 4
Identification Code
Single PSTN Access with Carrier Access Code (101) and Four-Digit Carrier
Follow these steps to create single PSTN access:
Step 1
Configure the required route patterns (patterns that should be routed differently based on the time of day) once percarrier. Transform thedialed number to include the appropriate carrier access code andCarrier Identification Code(CIC).
Put the route patterninto a carrier-specific partition. Applytime-of-day attributes to each partition.
Include all the carrier-specific partitions in the phone CSSs.
Single-Site Oft-Net Calling 4-231
Step 2 Step 3
Step 4
) 2010 Cisco Systems, Inc.
In the example in the figure, two route patterns are configured for international calls. One route pattern is in partition CI 1I (indicating that this partition is using the carrier with CIC 111):the other route pattern is in the standard partition.
'fhe carrierwith CIC I I1 is lessexpensive for international calls during business hours. In all other situations (other PS'IN destinations or international calls that are placed outside business
hours), the standard PSTN provider should be used.
Ihe route pattern in partition CI 11 is set up as 9011.1 to allow the 9011 lo be stripped off by PreDot discard digit instruction (DDl). In addition, a prefix 1010111011 is configured. As a
result, stripping off 9011 and adding 1010111011 to the number results in a call with carrier access enabled (1010) using the carrier with the ID 111 followed by the international number (Oil) {country code 43. area code, and subscriber number 69918900009).
The other route pattern (9.0! 1! >is configured with PreDot DDL which strips off only the access code 9. resulting in a standard international call number (011 followed by the international
number).
When digit manipulation is set up correctly for both situationsmatching the 901 I.! roule pattern, which is in the Oil partition, and matching the 9.011! route pattern, which is in no partitiontheonlv problem is that the pattern in the CI 11 partition is preferred over the other pattern during business hours, fhis issue can be fixed easily by applying a corresponding time schedule to the partition and including the partition in the phone CSS.
4-232
Ifa route pattern that has CMC applied ismatched, the user isprompted toenler a client matter
code to extend the call. This client matter code is added to the Call Detail Records (CDRs) to allow the accounting and billingof calls,based on their client matter.
Ifa route pattern is matched to a pattern that has FAC applied, the user isprompted toenter an
authorization code to extend the call. The idea of FAC is to prevent calls from unauthorized
users. Inother words, the goal is to allow only FAC-protected patterns from those users who areauthorized to use thepattern (by knowing a corresponding authorization code).
Valid client matter codes and authorization codes are added to Cisco Unified Communications
Manager and an authorization level isassigned to FAC. If FAC isapplied to a route pattern, the minimum required authorization level must bespecified for the route pattern. To extend calls, users must enter any valid client matter code to pass CMC prompts and must enter a valid
authorization code with an authorization level equal to or greaterthan the level that is configured for the FAC-enabled router pattern.
4-233
Voice Gateway
.
User A dials a number that matches a route pattern for which the Require Client Matter Code parameter is enabled. Cisco Unified Communications Manager plays a tone to indicate to the user that a client matter code must be entered, "fhe user must enter any valid client matter code to extend the call. In the example. CMC 1234. 1244. and 34X9 are configured: the user enters 1234.The call is successful, and the entered code is included in the generated CDR.
4-234
3489
User A
Voice Gateway
The configuration is thesame thatwas used intheprevious example. However, thistime. User A enters5555 at the CMC prompt. This codeis invalid; therefore, the callis denied. A CDR is
generated and logs the attempted call.
4-235
CDR isgenerared.
User A
Voice Gateway
User A dials a number that matches a route pattern for which the Require Forced Authorization
Code parameter is enabled and the Authorization Level is set to 3. Cisco Unified Communications Manager play s a lone to indicate to the user that an authorization code must
be entered. To extend the call, the user must enter a valid authorization code with an
authorization level of 3 or above. In the example. FAC 1234 is configured with a level of 1. FAC 1244 is configured with a level of 2. and FAC 1888 is configured with a level of 7. At the prompt, the user enters 1888. The call is successful, and the name ofthe entered authorization code is included in the generated CDR.
4-236
tone.
User enters authorization code. Code is unknown or its authorization level is lower than
UserA
Voice Gateway
The configuration is the same as the one that is used in the previous example. However, this time. User A enters 1234 at the FAC prompt. Although this authorization code is valid, the call
is denied because the authorization level ofthe entered code (level 1) is lower than the required
level that is configured for the route pattern. A CDR is generated and logs the attempted call.
4-237
Communications Manager
CAR tool
When PS'fN calls arc established. CMC and FAC usage statistics can be written to Cisco Unified Communications Manger CDRs. CDRs are disabled by delaull. You must enable CDRs via the Cisco CallManager service CDR Enabled Flag parameter.
After CDR statistics are gathered, vou can analyze them by using the Cisco Unified
Communications Manager CDR Analysis and Reporting (CAR) tool.
4-238
Configuring CMC
'['his figure shows how to configure CMC.
Configuring CMC
Select Call Routing >Client MatterCodes and click Add New.
Client Matter CoAe* Configuration
meaningful description.
To configure CMC. choose Call Routing>Client MatterCodes and click Add New. liach configured client matter code must beunique within the Cisco Unified Communications
Manger cluster.
4-239
After \ ou add client matter codes. \ on can configure route patterns to require CMC to establish
a call.
4-240
Configuring FAC
This figure shows how to configure FAC.
Configuring FAC
Select Call Routing > Forced Authorization Codes
and click Add New.
I orrwt Authorijslion Code ConlMjUf-rt'
Q-X1
-Status
(QAdd SUCCCK
Forced AuthoriioHon Codr In(ormtlDn
AjithofizationLevel1
25
To configure KAC. choose Call Routing > Forced Authorization Codes and click Add New.
An authorization code can have an authorization level between 0 and 255. Each configured
authorization code must be unique within the Cisco Unified Communications Manger cluster.
*-^j
i.
:H^rr*
.^ if.
::nT"'
w.,-^
i^
y
Single-Site Off-Net Calling 4-241
After \ou add authorization codes, you can configure route patterns to require FAC with a specific minimum authorization level to establish a call.
i 2010 Cisco Systems, Inc.
Summary
Ihis topic summarizes the ke\ points that were discussed in this lesson.
Summary
Time schedules and time periods are used to activate or deactivate partitions within a CSS, depending on time or date information.
CMC is used to track calls to certain clients by requesting the client matter code to be entered and adding it into CDR. FAC is used to
allow access to route patterns only ifan authorization code with a high-enough level is entered when requested.
.urrmiary (Cont.;
Complexity of CoS implementation at IP phones can be reduced by using the line/device approach, which allows the effective CSS to be composed of a line and device CSS (in this order). Vanity numbers provide access to local services by dialing the same number from any physical location.
Time schedules and time periods can be used to route calls via
different gateways or carriers, depending on the time ofthe day or date, to take advantage ofthe cheapest rate at any time.
Usage CMC and FAC calls are written into CDRs and can be
4-242
References
For additional infonnation. refer to these resources:
http:.';www.cisco.com.;cn/liS/docs/voice ip comm/cucui/drs/8 0 I/drsagKOI.html. Cisco Systems. Inc. Cisco Unified Communications Manager System Guide Release 8.0(1).
San Jose. California. February 2010.
Cisco Systems. Inc. Cisco Unified Communications System Release 8.x SRND. San Jose, California. April 2010. hltp::^vww-.cisco.C(>m/cn/IJS/docs/voice_ip_comm/cuen5/snid/8x/ucXxsrnd.pdf.
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4-244
Lesson 6
method to make the lines work together so that ifone representative is busy or unavailable, the call will rotate to other members ofthe group until it is answered or forwarded to an autoattendant or voice mail. Hunt groups are the mechanisms lhat help these businesses manage
inbound calls from customers. These businesses typically need several phone lines and a
inbound calls. Ahunt group is agroup oftelephone lines that are associated with acommon number When acall comes in lo the number that is associated with the hunt group, the call
cycles through the group until an available line is found. This process is known as hunting.
features such asCall Forward, shared lines, and Call Pickup.
This lesson describes how to implement hunt groups and how to enable other call-coverage
Objectives
Upon completing this lesson, you will be able to describe call coverage and how to implement
it in Cisco Unified Communications Manager. This ability includes being able to meet these
objecti\es:
Describe how call hunting works in Cisco Unified Communications Manager Describe call-hunting scenarios in Cisco Unified Communications Manager
Coverage Support
Cisco Unified Communications Manager CallThis topic describes call-coverage features in Cisco Unilied Communications Manager.
Ring multiple phones atthe same time {shared number) Pick up a call ringing on other phone (Call Pickup/Group
Pickup)
Call coverage is part ofthe dial plan and ensures lhat all incoming calls are answered. The
following call-coverage features are typically implemented for individuals:
Call Kornard: Ifthe called phone does not answer the call, the call should be forwarded lo
another phone or \ nice mail.
Shared lines: Ashared line is adirector) number that is assigned lo more than one dev ice
allowing the call to be accepted on more than one phone.
Call Pickup: Call Pickup allows acall lhat is ringing on aphone to be picked up at another
phone.
selection of group members to abroadcast option that rings all members of aline group.
based on apilot number that, ifcalled directly or used as aCall forward target allows hunlino through multiple line groups. Several hunting algorithms e.xist. ranging from around-robin *
Call hunting is another complex and flexible feature lhat provides call coverage Call hunting is
4-246
Shared Lines
This subtopic describes the shared line feature in Cisco Unified Communications Manager.
Shared Lines
Same directory number is used on multiple phones.
All phones ring at the same time when directory number is called.
When a user answers the call fromone ofthe phones, all phones
stop ringing.
2000
Ashared line is implemented by assigning the same directory number to multiple phones. Ifthe number is called, all phones that are configured with this shared-line number ring. The first user that accepts the call isconnected to the caller, and all other phones stop ringing.
Single-Site Off-NetCalling
4-247
Shared-Line Configuration
Ihis figure shows how to conligure ashared line in Cisco Unified Communications Manger,
Shared-Line Confiqurati
Direetorv Number Cwifigurattoa
Direetorv I Number InFarmation Directory
. ,
~~-
Afctir-j Uni
fiK0C1=ted Ceic
SEPO0JD94C32E6F
SEP002*C4-51E6
Diiiooate Rev
As soon as a single direetorv number isconfigured for more than one phone, the Director;
Number Configuration page shows all the devices that are associated with that number.
4-248
Call Hunting
This topic describes the call hunting feature in Cisco Unified Communications Manager.
Call-Hunting Components
1-800-555-0111
Hurt List
iFirsiChoice
Secaid Clinic'
specHnmnurAofXbnanddiatftbufiGrt
agoriBim
tons to actual extensions
Una Group 1
Urn Group 2
1000 I
1001 |
1003 |
iao4|
Phone directory numbers or voice-mail ports: Numbers or ports are assigned to line
groups.
Line groups: Line groups are assigned to hunt lists. Ahunt list can have one or more line groups. At the line group, hunt options and distribution algorithms can be specified to
define howcall hunting should be performed forthe members of a linegroup. Hunt lists: Hunt listsareassigned to huntpilots. A hunt list is an ordered listof line
groups.
Hunt pilots: Hunt pilots arethe numbers thataredialed to invoke a hunting process. A hunt pilot can be called directly; forexample, to provide a certain service to customers. Also an IF phone canbe configured to forward callsthat it receives to the huntpilot, to
provide call coverage.
While hunting, the forwarding configuration of line-group members is notused. If the hunting algorithm rings a phone and thecall is notanswered, the Call Forward No Answer (CFNA) setting of that phone isignored. The hunting algorithm goes ontothe next line-group member.
4-249
Call-Hunting Operation
This subtopic describes call-hunting operation.
Line group
distributes
call to agents
2001
J^
2002
If no agenl answers,
hunt list sends call to
Line group
distributes call to
operators
in the examplein the figure, two line groups are configured: Agents (containing direetorv numbers 2001 and 2002) and Operators (containing directory numbers 2101 and 2102).
fhe line groups are assigned to the Helpdesk hunt list.
A hunt pilot, also named Helpdesk. with the pattern 2222 is configured to use the Helpdesk
hunt list for call coverage.
The following high-level steps describe how this hunt pilot processes calls: 1. A user dials 2222. matching the hunt-pilot number. The hunt pilot sends the call lo the
Helpdesk hunt li^t.
2. The hunt list picks the first line group. Agents. 3. The line group distributes the call to the assigned agent directory numbers. 4. If no agent answers, the hunt list sends the call to the second line group. Operators.
5.
fhe Operators line group distributes the call to the operator director} numbers.
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Hunt Pilots
This subtopic describes hunt pilots.
Hunt Pilots
Hunt pilots are configured with a hunt pilot number: the number that needs to be called to start a hunting
process.
- Perform digit manipulation * Point directly to a hunt list Specify the maximum hunt timer
Hunt pilots are dialable patterns, such as roule patterns and directory numbers, in the callrouting table. The hunt pilot points directly to a hunt list. Hunt lists point to line groups, which
point to endpoints.
At the hunt list, digit manipulation can be configured to transform the calling and called number before the call is passed on to line-group members. Beginning with Cisco Unified CallManager Version 4.1, calls can be redirected to a final destination when the hunting fails because of one or both of these reasons:
AH hunting options have been exhausted and the call still is not answered. A maximum hunt timer that is configured at the hunt list has expired.
"fhis call redirection is configured in the Hunt Forward Settings section ofthe Hunt Pilot Configuration page, and the destination for this redirect can be either of these options:
A specific destination that is configured globally at the hunt pilot.
A personal preference that is configured at the phone line ofthe originally called number, when hunting on behalf of that number fails. The personal preference is configured by using the Call Forward No Coverage (CFNC) settings at the phone line.
For example, you can implement the personal preferences option. To do so, configure a user
phone so that the Forward No Answer field redirects the call to a hunt pilot, which searches for someone else to answer the call. If call hunting fails because all the hunting options are
exhausted or because a timeout period expires, the call can be sent to a personalized destination for the person who was originally called. For example, if you set the Forward No Coverage field in the Directory Number Configuration page to a voice-mail number, the call will be sent to the voice mailbox of that person if hunting fails.
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These considerations applv to calls processed by hunt pilots: Call Pickup and Group Call Pickup are not supported on calls that a hunt pilot distributes. A member of die line group cannot pick up a hunt-pilot call that is offered to another member in the line group, even if both members belong to the same Call Pickup group.
The hunt pilot can distribute calls to an; of its line group members, regardless of calling-
Hunt Lists
fhis subtopic describes hunt lists.
Hunt Lists
Ahunt list is a prioritized list of linegroups. Multiple hunt pilots can point to the same hunt list.
Multiple hunt lists cancontain the same line group.
Hunt lists do not perform digit manipulation.
Ahunt list is aprioritized list ofline groups that are used for call coverage. Hunt lists have
these characteristics:
Ahunt list is aprioritized list ofline groups: line groups are hunted in the order of their
A hunt listdoes notperform digit manipulation.
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Line Groups
is subtopic describes line groups.
Line Groups
The same extension can be contained in multiple line groups. The hunt option describes how to continue hunting after trying the first member ofthe line group (stop hunting, switch immediately to
line group).
The distribution algorithm specifies theorder in which the linegroup members are hunted (circular, longest idle, broadcast or
member thatfollows the last used).
The RNAR timeout value specifies how long totry a member ofthe
line group.
Line groups control the order in which acall is distributed, and they have these characteristics: Line groups point to specific extensions, which are typically IP phone extensions or voicemail ports.
1he same extension may be present in multiple line groups. line groups are configured with aglobal distribution algorithm, which is used to select the
next line-group member for hunting,
Line groups are configured with ahunt option, which describes how hunting should be continued after trv ing the first member ofthe line group, fhe hunt option is configured per
hunt-failure event: no answer, busy, and not available.
The Ring No Answer Reversion (RNAR) timeout specifies how long the hunting algorithm rings amember ofthe line group before proceeding to hunt according to the Line Group No
Answer hunt-option setting.
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Line-Group Members
This subtopic describes line-group members.
Line-Group Members
Line-group members are the endpoints accessed by line groups; members can be any of the following
types: Any SCCP endpoints, such as Cisco Unified IP phones, Cisco VG224 Analog Phone Gateway, or Cisco ATA 188
SIP endpoints Voice-mail ports
H.323 clients
Note CTI ports and CTI route points cannol De added within a line group. Calls cannot be dstrrtjuled lo endpoints controlled through CTI appHcations (CRS, IP IVR,and so on.).
Line group members are theendpoints that linegroups access. These linegroups can be anyof
these types:
Any Skinny Client Control Protocol (SCCP) endpoints, suchas CiscoUnified IP phones,
Cisco VG224 Analog Phone Gateway, or Cisco Analog Telephone Adaptor (ATA) 188
Session Initiation Protocol (SIP) endpoints
Voice-mail ports
11.323 clients
Computer telephony integration (CTI) ports andCTIroute points cannot be added to a line group. Iherefore.calls cannotbe distributed to endpoints that are controlled through CTI applications, such as Cisco CustomerResponse Solutions (CRS) or Cisco Unified IP Interactive
Voice Response (IVR).
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Call-Hunting Flow
This topic describes the eall-hunting fiow in a Cisco Unified Communications Manager callhunting configuration.
Direct Call to Hunt Riot or Call if hunt lis! nisinniim nunt Foiwarfled from Phone
timereipires v
Try next member of this line group i1 none <e3 go lo next line group
Step 1 Step 2
A direct call is placed to the hunt-pilot number, or a call is forwarded from a phone
to the hunt-pilot number.
I he hunt pilot that is configured wilh the appropriate hunt-pilot number starts the maximum hunt timer to monitor the overall hunting time. If the timer expires,
hunting stops. The hunt pilot is associated with a hunt list. I he hunt list that is associated wilh the hunt pilot sends the call to the next line group that is configured in the hunt list (starling with the first line group).
Step 3
Step 4
fhe line group sends the call to the next line-group member, based on the
distribution algorithm thai is configured for the line group. The possible distribution
methods are as follows:
Iop down
Circular
Step 5
If the line-group member (or members, in case of broadcast) that the distribution algorithm selects do not answer the call, the hunt optionwhich is configured independently, per hunt-failure reason, for the line groupspecifies how hunting should continue. Possible hunt-failure reasons are no answer (that is. the expiration ofthe RNAR timer that is configured for the line group), busy, and not available.
[fthe hunt option that isconfigured for the appropriate hunt-failure reason is
Stop Hunting, huntingstops.
Ifthe hunt opfion that isconfigured for the appropriate hunt-failure reason is Skip Remaining Members and Go Directly to Next Group, and there are no more line groups, hunting stops. Ifthere are additional line groups, tothe
process continues with the next line group (Step 4). Ifthe hunt option that is configured for the appropriate hunt-failure reason isTry
Next Memberbut Do Not Go to Next Group, and there are no more line-group
Ifthe hunt option that is configured for the appropriate hunt-failure reason isTry
NextMember Then Try NextGroup in Hunt List, and there are additional line-
group members, the process continues with the next line-group member (Step 4). If there areno additional line-group members, the nextline group is used. If
there areadditional linegroups, the process continues with the next linegroup (Step 4). If there areno more linegroups, hunting stops.
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itl-Huntinq Fl
Hunting Slopped
Check hunt pilot
for fins'
forwarding
configuration.
Final
No Final
Forwarding
Forwarding
Use Personal
Number
Failed
Specified in
Hunt Pilot
Preferences
number
speeded at iCFNC,
Route call to
number specifies
in hunt pilot
Stop Hunting was the hunt option that needed to beapplied afler a call was notaccepted by
the last attempted line-group member.
After hunting tried the last line-group member, there were noother line-group members or
other line groups to be used. This reason is known as hunt exhaustion.
fhe ma\imum hunt timer that is configured for the hunt pilot expired.
Step 6
Rev iew the hunt pilot configuration for its final forwarding settings.
If the hunt pilot is not configured for final forwarding, the call fails and a reorder
tone is plaved.
Step 7
Review the final forwarding destination settings that are configured lor the hunt
pilot.
If a final forwarding number is specified for the hunt pilot, route the call lo the
specified number.
111 se Personal Preference is selected, route the call to the number thai is
configured for CFNC on the phone line lhat invoked the call to the pilot number.
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Call-Hunting Scenarios
Example 1: Internal and External Forwarding (No Hunting)
The figure shows an example ofinternal and external forwarding options with no hunting
enabled.
Solution:
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or
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Thefirst example is straightforward. User A atdirectory number 3000 has the configuration
that is shown in the Call Forward and Call PickupSettings windowofthe Directory Number
Configuration page:
Call Forward Busy(CFB): CFB is determined by the Forward Busy Internal and Forward Busy External settings, both setto 3001. CFB forwards incoming internal and external calls
to 3001. when 3000 is busy.
CFNA: CFNA is determined by the Forward No Answer Internal and Forward No Answer
External settings. CFNA forwards incoming internal calls to 3001 andexternal incoming
calls to 303 555-0111. when 3000 does not answer.
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Solution:
nrzzi
User A at direetorv number3000has the configuration that is shown in the Directory Number
Configuration window:
CFB: Incoming internal calls are forwarded to 3001, and external calls are forwarded lo hunt pilot 7000. when 3000 is busy.
CFNA: Incoming internal calls are forwarded to 300I. and external internal calls are forwarded to hunt pilot 7000. when 3000 does not answer.
Assume that hunt pilot 7000 is associated with hunt list abe and has four hunt parties that are distributed over I ine (iroup I and LineGroup 2. Hunt pilot 7000 has no final fonvarding fields
provisioned (default!.
Question: Which behavior results when an internal caller calls 3000 and user 3000 is busv?
Answer: The call forwards to line 300I.
Question: Which behav ior results when an external caller calls 3000 and user 3000 does not
answer?
Answer: The call forwards lo hunt pilot 7000. which causes hunting to lines 300I. 3002. 4001. and 4002, If one ofthe hunt parties answers, the caller is connected lo lhat party. If no hunt party answers, then regardless ofthe reason, the caller receives a reorder tone (or an equivalent
announcement].
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Solution:
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The Forward Ilunt No Answer field for hunt pilot 7000 is set to destination 3002, but all
Forward Hunt Busy fields are empty.
Question: Which behavior resulLs when an external caller calls 3000 and user 3000 does not
answer?
Answer: Thecall forwards to hunt pilot 7000. which causes hunting to lines 300L3002. 4001, and 4002. If one ofthe huntparties answers, the calleris connected to that party.
Ifall hunt parties are busy, the caller receives a reorder tone (oran equivalent announcement). Ifatleast one hunt party isalerted (rings), the call forwards to3002 because 3002 isthe value
that is configured forthe Forward Hunt No Answer field. Question: What if user3000is busy when an external call arrives?
Answer: In this case, the same results occur because user 3000 forwards external calls to hunt pilot 7000 for both busy and no-answer conditions.
4-261
Solution.
Configuration
number 3000
uindow
tor drectory
""""""*
D~
-*
,1
UPP Rest
Forward No Coverage
Internal is set to 3005 at line
3000
Question: Which behav ior results when an external caller calls 3000 and user 3000 does not
answer?
Answer: The call forwards to hunt pilot 7000. which causes hunting to lines 3001. 3002, 4001.
and 4002.
If oneofthe hunt parties answers, the caller is connected lo that parly. If at least one partv is
alerted, hunting exhausts because there was no answer, and the call forwards lo 3002.
[fall hunt parties are bus), the call forwards lo the Forward No Coverage Fxtenial setting of
the original called part) (user 3000). In ibis case, the call forwards to the hunt pilot 303 5550111.
hunt pilot 7000 for both busy and no-answer conditions. Note If the hunt pilot is configured to use personal preferences but the corresponding Forward No Coverage field is not set on the phone, the call fails This configuration results in the same behavior as when there is no final forwarding setting on the hunt pilot
4-262
| Hunt Li9aBc |
Assume maximum
hunttimerfor hunt
FriBW
UPP Dest
ECZJ
pilot 7000 is 25
seconds.
W'Uta.Mr.l
Question: What
happens when a
user calls that hunt
;.::z
pilot?
The RNARtimer for a line group determines how long huntingwill ring a hunt party before
moving to the next party in its list(assuming that thecustomer didnotselect the broadcast
algorithm). This timer has a default value of 10 seconds.
Question: In the examples of four hunt parties, how long will it take before hunting exhausts?
Answer: It will take 40 seconds before hunting exhausts (10 seconds RNAR * 4 hunt
members).
Assume that the maximum hunt timer for hunt pilot 7000 is set to 25 seconds. The call must be answered within this time. In this example,the hunt timer is 2.5 times the RNAR timer, which
is 10 seconds.
Question: Which behavior results when a user calls hunt pilot 7000?
Answer: The call attempts to hunt to the four parlies. If no party answers within 25 seconds, hunting terminates and the cause is treated as no answer. Hunting terminates after the third
member has been alerted for 5 seconds (10 seconds RNAR on each ofthe first two members
leaves 5 seconds before expiration ofthe 25 seconds maximum hunt time that is configured on the hunt pilot).The call then forwards to 3002 becausehunting failed with a no-answer
condition.
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Call-Hunting Configuration
This topic describes how to conligure call hunting inCisco Unified Communications Manager.
Create hunt list and add line groups. Create hunt pilot, associate hunt list, and configure huntforward settings. Configure personal preference on phones in case of no hunt coverage.
To access the Line Group Configuration. Hunt I isl Configuration, and Hunt Pilot Configuration windows in Cisco l'nified Communications Manager Administration, choose Calf Routing > Route/Hunt. When configuring hunting, follow these steps: Step 1 Step 2 Step 3 Step 4
Note
Create ihe line groups, add members, and conligure the distribution algorithm and hunt options. Create the hunt list and add the line groups. Create the hunt pilot, associate the hunt list with the hunt pilot, and conligure the
hunt forward settings.
Configure personal preferences on phone lines when hunting ends with no coverage.
Use concise and descriptive names for line groups and hunt lists. The CompanynameLocationGroup format usually provides a sufficient level of detail and is short enough to enable you to quickly and easily identify a line group. For example.
CiscoDallasAAl might identify a Cisco Access Analog line group for the Cisco office in
Dallas
[Distribution algorithm"
r Hunt Option*
Hd Answ&r*
Mat ivailaBts"
Step hurting
The director)' numbers that will become the members ofthe line group must exist in(he database before you can complete this procedure. Follow these steps to configure line groups:
Stepl
Step 2 Step 3
Enter a name in the Line Group Name field. The name can contain as many as 50
alphanumeric characters and can contain any combination ofspaces, periods (.), hyphens (-). orunderscore (Jcharacters. Ensure that each line-group name isunique
to the route plan.
Step 4
Configure the distribution algorithm, hunt options, and RNAR timeout as desired, or
leave them at their default values.
Note
Options forthe distribution algorithm are: Top Down, Circular, Longest Idle Time, and Broadcast. Huntoptions are Try NextMemberThen Try NextGroup in HuntList, Try Next Member but Do NotGo To NextGroup, Skip RemainingMembers and Go Directly to Next Group, and StopHunting. The RNAR timer specifies how long totry one member before ending ina no-answer condition The default value is 10 seconds.
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-onfiqunni
for directory
numbers lo be
added
Remove selected
members from line
group
Step5
Add members to die line group. If >ou need to locale a directory number, choose a route partition from the Partition drop-down list, enlera search string in the Director) Number Contains Held, and click Find. To find all director) numbers that
belong to a partition, leave the Directory Number Contains Held blank and click find. A list of matching director.' numbers is displayed in the Available DN/Route Partition pane.
Step 6
In the Available DN/Route Parlilion pane, selecta directory numberto add and click Add to Line Group to move ifto the Selected DN/Route Partition pane. Repeat this step for each member that vou want to add to this line group. In the Selected DN/Route Partition pane, choose the order in which the new director) numbers will be accessed in this line group. To change the order, click a directors number and use the Up and Down arrows to the right ofthe pane.
Click Save to add Ihe new director) numbers and lo updale the direetorv-number order for this line group.
Step 7
Step 8
Unified Communications
UJUvttHt
add linegroups.
Configuration page.
Step 1
Step 2
Step 3
In the Name field, enter aname. The name can contain as many as 50 alphanumeric
characters and can contain any combination ofspaces, periods (.), hyphens (-). and underscore (J Characters. Hnsure that each hunt-list name isunique to the route plan. Enter a descriptive name inthe Description field.
Step 4 Step 5
Step 6 Step 7
Choose aCisco Unified Communications Manager group from the drop-down list. The group must exist in the database; you cannot create anew group from this
window.
To add this hunt list, click Save. The Ilunt List Configuration window displays the
newly added hunt list.
Add at least one line group to the new hunt list. To add a line group, click Add Line
Group. The Hunt List Detail Configuration window isdisplayed.
From the Line Group drop-down list, choose a line group to add to the hunt list. To add the line group, click Save. The pop-up window appears, stating that, for the changes totake effect, you must reset the hunt list. Click OKtoconfirm the message. The line-group name is displayed in the Selected Group list on the right
side ofthe window.
Step 8
Step 9 Step 10
2010 Cisco Systems, Inc.
To add more line groups tothis list, click Add Line Group and repeat the previous
two steps.
When you finish adding line groups tothe hunt list, click Save, Click OK inthe pop-up window to reset the hunt list.
Single-Site Off-Net Calling 4-267
shown in the hunt list. To change the access order ofline groups, choose aline group from ihe Selected Groups pane and click the 1ip or Down arrow on the right side ofthe pane to move the
line group up or down in the list.
Cisco Unified Communications Manager accesses line groups in the order in which they are
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. or or
Step 1
Step 2
Step 3
Step 4 Step 5
Step 6
Assign the hunt pilot to a hunt list using the Ilunt List drop-down menu. Configure final forwarding settings and setthemaximum hunt timer.
When finished, click Save.
The Hunt Forward Settings area ofthe Hunt Pilot Configuration window specifies the final forwarding settings and maximum timer values, as shown in the following table.
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Description
Coverage/Destination field ofthe directory number, when a call to the directory numberfirstdiverts to coverage,
coverage either exhausts or times out, and the associated
hunt pilotfor coverage specifies Use Personal Preferences for its final forwarding.
in the Destination and CallingSearch Space fields Destination: This setting indicates the directory number to
which calls are forwarded.
When the call that is distributed through the hunt list encounters only busy lines for a specificperiod, this setting specifies the
destination to which to forward the call Choose from these
options.
Use Personal Preferences: Use this check box to enable the CFNC settings for the original called number that forwarded the call to this hunt pilot
Communications Manager ignores the settings in the Destination and CallingSearch Space fields. Destination: This setting indicates the directory number to
which calls are forwarded
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The Director. Number Configuration window provides configuration options for internal and external forwarding, based on whether a call isCFA orCFNA, as specified in the following
table.
Description
Destination: This setting indicates the directory number to whichall calls are forwarded. Use any dialable phone number, includingan
outside destination.
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Field
Description
This setting specifies the forwarding treatment forinternal or external callsto this directory number if the directory number is busy Voice Mail: Checkthis check boxto use the settings inthe Voice Mail
Profile Configuration window for internal calls
Forward No Answer
Internal
Calling Search Space: This setting applies to all devices that use this
directory number.
Forward No Answer
External
This setting specifies the forwarding treatment for internal or external calls to this directory number ifthe directory number does not answer. Voice Mail: Check this check boxto use the settings in the VoiceMail
Profile Configuration window.
Forward No Coverage
Internal
Destination: This setting indicates the directory number to which an internal call isforwarded when the call is not answered. Useany
dialable phone number, includingan outside destination. Calling Search Space: This setting applies to all devices that use this
directory number.
This setting appliesonly ifyouconfigure one ofthe other forwarding fieids CFA, CFB, or CFNAwith a hunt pilot number in the Destination directory
number field
Forward No Coverage
External
For the hunt-pilot settings, you must also configure the Forward Hunt No Answer or Forward Hunt Busy fields and check the Use Personal Preferences check box under the Hunt Forward Settings section inthe Hunt
4-272
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Cisco Unified Communications Manager offersseveral features for call coverage, including Call Forward, shared lines, Call Pickup, and call hunting. In CiscoUnified Communications Manager, IP phone lines can be configured with Call Forward All (CFA), CFB, CFNA, CFNC, and Call
Forward Unregistered.
Shared lines provide an easy way to implement call coverage byassigning one number to multiple devices.
Summary
During hunting, the hunt option, distribution algorithm, RNAR
timeout, maximum hunt timer, and final forwarding settings
are considered.
Call hunting in Cisco Unified Communications Manager usesthe following elements: hunt pilots, hunt lists, line groups, and endpoints (lines and voice-mail ports).Call-hunting implementation includes configuration ofIP phone lines, line groups, hunt lists, and
hunt pilots.
4-273
References
for additional infonnation. refer to these resources:
t isco S\ stems. Inc. ("isco Unified Communications Manager System Guide Re/ease 8.0(1)
San Jose. California. February 2010.
Cisco Sv stems. Inc. Cisco Lnified Communications System Release 8 r SRXD San lose
California. April 2010.
4-274
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Gateway integration in Cisco Unified Communications Manager can be accomplished by using gateway protocols such as MGCP,
H.323, or SIP.
CSSsand partitions are the core components to implement calling privileges in Cisco Unified Communications Manager. Digit manipulation inCisco Unified Communications Manager can be performed through several configuration elements, such as
translation patterns, route patterns, and route lists.
and other configuration elements, such as time periods and time schedules, blocked patterns,and forced authorization codes, for gateway selection and to control access to the PSTN. Cisco Unified CommunicationsManager provides various ways of providing call coverage, including Call Forward and Call Pickup features, shared lines, and the implementation of complex callhunting algorithms.
Single-Site Off-NetCalling
4-275
This module describes how to enable Cisco Unified Communications Manager for public suitched telephone nelwork (PSTN) calls and how lo implement adial plan for internal and external calls in a single-site environment. The module first describes how toimplement
gateways for PSTN access bv using Media Gateway Control Protocol (MGCP). 11.323. and Session Initiation Protocol (SIP) signaling protocols. Then, the module describes how call-
routing decisions arc made in Cisco Unified Communications Manager, based on dialed digits,
and how path selection is perlonned after anentry in the call-routing table is found. The
module then discusses implementation ofcalling privileges by using calling search spaces (CSSs) and partitions, followed b> an explanation ofthedigit-manipulation options in Cisco
Unified Communications Manager. Several examples show how lo use the available callingprivilege configuration elements toimplement classes ofservice ortoperform routing decisions that arc based onthe calling device. Finallv. the module provides an overview ofcallcoverage features and provides a detailed discussion of how to implement call hunting inCisco
UnifiedCommunications Manager.
References
For additional information, refer to these resources:
Cisco S\ stems. Inc. ('isco Unified Communications System Re/ease 8.x SRND. San .lose. California. April 201(1,
Cisco Sv stems. Inc. ('isco ( nified Communications Manager andCisco IOS Interoperability Guide. Release 15.0. Configuring MGCP Gateway Support for Cisco
L'nified Communications Manager. San .lose, California. March 2009.
Cisco Svstems. Inc. Cisco Unified Communications Manager System Guide Re/ease 8.0/1).
San Jose. California. February 2010. Imp: uuu.eiNco.com en.I S ducs.\oice_ip_eomm'ciicnvadmin/K 0 l/ccmNVs..acuii-Jwl!eni.tttml.
4-276
Module Self-Check
Use the questions here to review what you learned in this module. The correct answers and
solutions are found in the ModuleSelf-Check Answer Key.
QI)
Which two ofthe following are not core gateway requirements? (Choose two.) (Source:
Implemenling PSTN Gateways in Cisco Unified Communications Manager)
A) Supplementary Services
B)
C) D) I:)
Q2)
F) NKAS Which two features are supported for MGCP gateways inCisco Unified Communications Manager? (Choose two.) (Source: Implementing PS'fN Gateways in
Cisco Unified Communications Manager)
A) B)
C)
D)
PRI backhauling
dynamic dial peers
Q3)
Which function can be configured for H.323 gateways? (Source: Implementing PSTN
Gateways in Cisco Unified Communications Manager)
A) B) PRI backhauling H.323 call survivability
C)
D)
autoconfiguration
ASCII-based signaling
Q4)
SIP signaling isbased on which two protocols and functions? (Choose two.) (Source:
Implementing PS'fN Gateways in Cisco Unified Communications Manager)
A) B) ASN1 Q.931
C)
D) E)
TCPor UDP
ASCII EIGRP
Q5)
Which isnot considered a dial plan component? (Source: Configuring Cisco Unified
Communications Manager Call-Routing Components)
A) endpoint addressing
B)
C) D) E)
digit manipulation
cal! coverage calling privileges voice mail
4-277
Q6)
Which definition best describes off-net dialing? (Source: Configuring Cisco Unified
Communications Manager Call-Routing Components) A) calls that originate and terminate on the same telephony network B1 use of internal number to reach a PSTN phone C) calls thai originate from one telephony network and terminate on a different
telephony network
D)
Q7)
Which two ofthe following are not entries in the call-routing table ofCisco Unified Communications Manager? (Choose two.) (Source: Configuring Cisco Unified
Communications Manager Call-Routing Components)
AI B)
C)
D) E) F)
G)
route pattern
hunt pilot call park numbers
Meet-Me numbers
11)
gatew ay
Q8)
B)
C)
I))
V.)
Q9)
Which statement does not applv to urgent priority? (Source: Configuring Cisco Unified
Communications Manager Call-Routing Components)
A) B)
C)
D)
QIO)
I iigentpriority can be configured at route patterns only. Urgent priority is used to force immediate routingas soon as a match is detectedev en if other. longerroute patterns are potential matches. {Tgent priority is often used with emergency numbers. A pattern with urgent priority effectively excludes the urgent pattern from a
longer route pattern range.
Which statement describes call routing? (Source: Configuring Cisco Unified Communications Manager Call-Routing Components)
A)
B) C) D)
Call routing is the process of selecting the device where the call is sent to. Call routing is the process of finding an entry in the call routing table that
matches the called number.
Call routing is ihe process of sending VoIP RTP packets toward the destination
ofthe call.
4-278
Qll) Which ofthe following is not apath-selection configuration step? (Source: Configuring
Cisco Unified Communications Manager Call-Routing Components)
A) B) C) 0)
Add gateways and trunks. Buildroutegroups from available devices. Build route lists from available route groups. Build route patterns pointing to route groups.
Q12)
Which statement describes what calling privileges are used for? (Source: Using
Partitions andCSSs to implementing Calling Privileges for On-Net-Calls)
A)
B)
Calling privileges are used to prioritize important calls over less important
calls.
C) D)
Calling privileges give priority tovoice over data) Calling privileges give priority to on-net calls versus off-net calls.
QI3) Which two statements are true about partitions and CSSs? (Choose two.) (Source: Using Partitions and CSSs toimplementing Calling Privileges for On-Net-Calls)
A) B) When twodevices are in the samepartition, they can calleachother. When twodevices havethe same CSS,they can calleachother.
C)
D) E)
Adevice has access to only those numbers that are in partitions listed in the
CSS ofthe calling device.
If a numberis in no partition, it is accessible by all devices. If a device has no CSS, it has access to all devices.
Q14)
CSSs can beapplied to which configuration components? (Choose two) (Source: Using
Partitions and CSSs to Implement Calling Privileges for On-Net-Calls)
A)
B) C)
D)
E)
phone-line configuration
route groups
Q15)
Which function or feature cannot beimplemented using CSSs and partitions? (Source: Using Partitions and CSSs to Implement Calling Privileges for On-Net-Calls)
A) automatic alternate routing
B)
C)
D)
F.)
gateway selection
Q16)
Which ofthe following allows called and calling numbers to bemodified during call processing? (Source: Implementing Cisco Unified Communications Manager Digit
Manipulation) A) tlie use of regions digit randomization digit collection digit manipulation
B) C) D)
>2010Cisco Systems.Inc.
4-279
Q17)
Which tuo types ofdigit manipulation are commonly required on outgoing PSTN calls? (Choose two.) (Source: Implementing Cisco Unified Communications Manager
Digit Manipulation)
A) Bi C) D) F)
rernov ing the PS1 N access code from the calling-party number removing the PS'IN access code from the called-party number expanding the calling-party numberto an E.164 number adding the PSTN access code to the calling-party number adding the PS'fN access codelo the called-party number
QI8|
Which ofthe following is nota digit manipulation configuration element? (Source: Implementing Cisco Unified Communications Manager Digit Manipulation)
A) B)
C) D|
I-.) F)
translation pattern
significant digits
Q19I
Which digit manipulation feature is configured at the directory number but enabled as part ofthe calling-party transfonnation settings? (Source: Implementing Cisco Unified
Communications Manager Digit Manipulation)
A) B) external phone number mask prefix
C) D)
translation pattern
significant digits true
Q20)
again in the call routing table. (Source: Implementing Cisco Unified Communications
translation pattern route pattern
transfonnation mask
D)
transformation pattern
Q21)
Which two statements are not correct about transformation masks? (Choose two.) (Source: Implementing Cisco Unified Communications Manager Digit Manipulation) A) They can be used to modify either the calling number or called number.
B) They can contain digits 0-9. *. #. and X.
C) D)
E|
Ihey are part ofthe calling- andcalled-party Iransfonnations settings. They can be applied only to the called-party number. Ihey are configured only at translalion paltents.
022)
Which two discard digits instructions arethe only ones available for route pattenis that
do not use the a, sign? (Choose two.) (Source: Implementing Cisco Unified Communications Manager Digit Manipulation)
A) PreAt
li)
IID->10D
C) D)
I)
NoDigits IntlTollBypass
PreDot
4-280
Q23) Which statement about significant digits is correct? (Source: Implementing Cisco
Unified Communications Manager Digit Manipulation)
A)
B) C) D)
They are configurable at gateways and trunks and apply to the calling-party
Thev are configurable at route patterns and apply to both the called- and the
calling-party number,
number. number.
They are configurable at gateways and trunks and apply to the called-party They are configurable at translation patterns and apply to the called-party
number on incoming calls only.
Q24) Which statement about global transformations is not correct? (Source: Implementing
Cisco Unified Communications Manager Digit Manipulation)
A)
B) C)
D)
E)
025) Incoming calling- and called-party- settings are available at which configuration
elements? (Source: Implementing Cisco Unified Communications Manager Digit
Manipulation)
A) B) C) D)
calling party based onnumber type atSIP trunks calling party based onnumber type at H.323 gateways called party based on number type at MGCP trunks called party based onnumber type at SIP trunks
026) Which is not atypical calling privileges application? (Source: Implementing GatewaySelection and PSTN-Access Features)
A) B)
C) D) E)
Q27) Which configuration clement is not used to implement time-of-day routing? (Source:
Implementing Gateway Selection and PSTN-Access Features)
A) B) C)
[))
Q28) Which statement about the line-device approach at partition and CSS configurations is
not true? (Source: Implementing Gateway Selection and PSTN-Access Features) A) For each PSTN gateway, route patterns exist once ina different partition.
B) C) D) The device CSS is used for gateway selection. The lineCSS is used forgateway selection. The line CSS is used for class-of-scrvice implementation.
4-28.
Q29) Which two steps are not required when implementing vanity numbers'1 (Choose two.)
(Source: Implementing Gateway Selection and PSTN-Access Features)
A) R) Create a site-specific partition for each physical location. Create a service-specific partition for each different service.
C)
F.) Put the appropriate site-specific partition into the CSS ofthe phones. I) Pul the appropriate sen ice-specific partition inlo the CSS ofthe phones. Q30) Which two configuration elements are not used to implement time-of-day carrier
selection? (Choose two.) (Source: Implementing Gateway Selection and PSTN-Access
Features I
D)
For each serv ice. configure the same vanity number once per physical location.
A)
B)
(1
DI F)
0311
multiple identical route patlerns with different digit manipulation CSSs with time schedules referring totime periods
Which feature allows calls to he permitted ordenied based on end-user authorization? (Source: Implementing Gateway Selection and PSTN-Access Features)
A) CMC
B)
C)
D)
I AC ACE
CDR
032)
Which two statement;, about client matter codes and forced authorization codes are true? (Choose two.) (Source: Implemenling Gateway Selection and PSTN-Access
features)
A) B)
C)
Entering a CMC isoptional on CMC-enabled roule patterns. CMCs and FACs cannot be enabled together on a route pattern.
Valid level values for FACs are from 0 to 255.
D)
Q33)
1.) FACs can be enabled only ifCMCs are also enabled on aroute pattern. Which two ofthe following arc no-call coverage features? (Choose two.) (Source: Implementing Call Coverage in Cisco Unified Communications Manager)
A) Call Forward
B)
C)
autoregistration
shared lines
D)
E)
Call Pickup
Call Admission Control
Q34)
What happens ifsomeone calls a directory number lhat isshared by three devices? (Source: Implementing Call Coverage in Cisco Unified Communications Manager)
A) B)
C) D)
fhe phone with the lowest MAC address rings. Ihe phone with the highest MAC address rings.
All three phones ring. fhis configuration is not possible.
Q35) Which two of the following are not call hunting configuration elements? (Choose two.)
(Source: Implementing Call Coverage in Cisco Unified Communications Manager)
A) B) C) D) E) shared lines pickup groups line groups hunt lists hunt pilots
Q36) Which is not areason for hunting to stop? (Source: Implementing Call Coverage in
Cisco UnifiedCommunications Manager)
A)
B) C)
D)
The maximum number ofhunt attempts isreached. The hunt option was configured to stop hunting.
Hunt exhaustion occurs (there are no more line group members to try).
Q37) Which two statements are correct about hunt options and distribution algorithms? (Choose two.) (Source: Implementing Call Coverage in Cisco Unified Communications
Manager)
A) B)
C) D)
E)
The hunt option specifics the order in which line-group members are hunted. The distribution algorithm specifies how the maximum hunt time is calculated
The hunt option specifies how to continue hunting based on the result of the The hunt option is configured at the hunt pilot; the distribution algorithm is
configured at the hunt list.
hunted.
The distribution algorithm specifics the order in which line-group members are
4-283
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4-284
Module 51
Media Resources
Overview
This module describes the types ofmedia resources that Cisco Unified Communications
Cisco Unified Communications Manager servers provide, and how to implement Cisco
hardware-based media resources.
Manager supports, how to configure the software- and hardware-based media resources that
Module Objectives Upon completing diis module, you will be able to implement Cisco Unified Communications Manager media resources. This ability includes being able to meet this objective:
Describe Cisco Unified Communications Manager media resources, including conferences,
transcoders. and MTP. as wellas MOH and annunciator services
5-2
Lesson 1
Implementing Media
Resources in Cisco Unified Communications Manager
Overview
"fhis lesson describes available hardware- and software-based media resources and how they
are configured in Cisco Unified Communications Manager to provide features such as conferencing, transcoding, media termination, and music on hold (MOH). The lesson also explains how to perform access control to media resources by using Media Resource Groups
(MRGs) and Media Resource Group Lists (MRGLs).
Objectives Upon completing this lesson, you will be able implement Cisco Unified Communications
Manager media resources. This ability includes being able to meet these objectives:
Describe types of media resources and their functions Describe how Cisco Unified Communications Manager supports hardware- and softwarebased media resources
Describe types ofconferences (single mode and mixed mode) and explain the features of
sofivs are- versus hardware-based conference bridge media resources
Describe how to configure MRGs and MRGLs and how to assign them to devices
Cisco
Voice termination
Unified Communications
Manager Cluster
Annunciator
MOH
PSTN
A media resource is a software- or hardware-based entity thai performs media-processing functions on the data streams lo which il is connected. Media-processing functions include inking multiple streams tocreate one output stream (conferencing), passing the stream from
one connection to another (Media Tenninalion Point |M'IT|),converting ihedata stream from
one compression tvpe toanother (transcoding), echo cancellation, signaling, terminating a voice stream from a lime-division multiplexing (TDM) circuit (coding/decoding), packeli/ing a
stream, streaming audio (annunciation), and so on.
Not all media resources are needed in even deployment. Software-based features can provide the required resources, ordigital signal processors (fXSPs) can be provisioned toimplement the
resources, fhe same basic resources (DSPs and Cisco IP Voice Media Streaming Application)
can be shared to implement higher-level functions.
5-4
Media Resource
TDM legsmustbe terminated byhardware thatperforms codng/decoding and packeUzation ofthestream. Termination is performed by DSP resources inthehardware module.
Aconference bridge joinsmultpleparticipants intoa singlecall. The bridge mixes the streams and creates a unique output
stream for each connected party.
Audio conference
bridge
Transcoder
MTP
AIranscoderconverts an inputsiream lhat uses one codec into an output stream thatuses a dWerent codec. AnMTP bridgesthe mediastreamsand allows Ihem to be set up
and torn down independently.
Annunciator
MOH
The media resources in Cisco Unified Communications Manager andtheir functions are
described as follows:
Voice termination: This resource applies toa call that has two call legs: one leg on a TDM
interface andthe second on a VoIP connection. The TDM leg must be terminated by
hardware that performs coding/decoding and packetization ofthestream. This termination function isperformed by DSP resources that all reside inthe same
hardware module, blade, or platform. All DSP hardware on Cisco TDM gateways can terminate voice streams. Certain hardware can also perform other media-resource functions, such as conferencing or transcoding.
Conference bridge: This resource joins multiple participants into a single call. The resource can accept any number ofconnections for agiven conference, up to the maximum
number of streams that areallowed for a single conference on thatdevice. There is a one-
to-one correspondence between the media streams and the participants that are connected to
a conference. The conference bridge mixes the streams andcreates a unique output stream
for each connected party. The output stream for any party isthe composite ofthe streams from all connected parties, minus the input stream ofthe given party. Some conference
bridges mix only the three loudest talkers on the conference and distribute that composite stream toeach participant, minus the input stream oftheparticipant ifthe participant is one
ofthe talkers.
Transcoder: This resource takes the stream of one codec and converts it from one
compression type to another. For example, the resource could take a siream from a (1.711
codec and iranscode it in real time lo a G.729 stream. In addition, a transcoderprovides
MTP capabilities and can beused toenable supplementary services for H.323 endpoints,
when required.
Media Resources
5-5
media streams and allows them to be set up and torn down independently. "Ihe streaming data that is receiv ed from the input stream on one connection is passed to the output siream
on the other connection, and \ ice versa.
Two streams lhat use the same codec but different sampling rates can also be connected. Asingle-site deplov ment usually has no need for transcoding devices. .MTP: fhis resource accepts two full-duplex G.7! I streams. The resource bridges the
Annunciator: Ihis software function ofthe Cisco IP Voice Media Streaming Application
provides the abilitv tostream spoken messages or various call-progress tones from the sv stem to a user, fhe annunciator can send multiple one-way Real- lime Transport Protocol (RIP) streams to devices such as Cisco IP phones orgateways. The annunciator also uses
Skinnv Client Control Protocol (SCCP) messages to establish the RTP stream. The announcements can be customized bv replacing Ihe appropriate .wav file.
MOH: Ihis integral feature ofCisco Unified Communications systems provides music to
callers when a call isplaced onhold, transferred, parked, oradded loan Ad Hoc
5-6
Media Resource
Software
No
Hardware :
Yes
\foice termination
Audio conference
bridge
Transcoder
MTP Annunciator
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
No
No-
MOH
Cisco Unified Communications Manager offerssoftware-based media resources. Youcan start the Cisco IP Voice Media Streaming Application to activate the following media resources:
Hardware mediaresources can also offer audioconferencing and MTP media. MOH is a
special case: It works only in remote sites, in the Survivable Remote Site Telephony (SRST)
mode of a router.
Media Resources
5-7
All media resources register with Cisco Unified Communications Manager. Signaling between hardware media resources and Cisco
Unified Communications Manager uses Cisco SCCP. Audio streams are always terminated by media resources. No direct IP phone-to-IP phone audio streams are present
when media resources are involved.
Signaling between external (hardware) media resources and Cisco Unified Communications
Manager usualh uses Cisco SCCP.
All audio streams from any endpoint are alwavs terminated bv the media resources lhat are
involved in the call, fhere areno direct IP phone-to-IP phone audio streams if a media resource
is involved in the call How.
Voice-Termination Signaling and Audio Streams This subtopic describes voice-termination signaling and audio streams in Cisco Unified
Communications Manager.
Voice termination applies toa call with a TDM and a VoIP call leg.
PSTN;
Audio
Signaling
The voice-termination function is needed when an incoming or outgoing TDM call is created
DSPs that are installed in the gateway.
bv using agatewav. The Cisco IOS router hardware terminates the TDM leg and must perform coding/decoding and packctization functions. These functions arc performed by using hardware
There are two audio streams: one is inside the public switched telephone network (PSTN), and
the other is a VoIP audio stream that uses RTP.
Signaling messages are exchanged between agateway and Cisco Unified Communications Manager and between an IP phone and Cisco Unified Communications Manager. Ihe tigure
does not show PSTN signaling.
Media Resources
5-9
between the gateway and the conference bridge. Signaling occurs between IP phones and Cisco Unified Communications
Aconference bridge joins multiple participants in a single call. Audio streams exist between IP phones and the conference bridge and
nlegrated
Conference
Bridge
Audio
Signaling
PSTN
Aconference bridge joins multiple participants into a single call. The software conference
bridge runs on one or more Cisco Iinificd Communications Manager servers in acluster. Audio streams exist between IP phones and aconference bridge and between agalewav and a
conference bridge.
Manager, between conference bridges and Cisco Unified Communications Manager, and between agatewav and Cisco Unified Communications Manager.
use SCCP to communicate with Cisco Iinificd Communications Manager. Cisco Unified
Communications Manager does not distinguish between software- and hardware-based conference bridges, when it processes a conference-allocation request. conference, that the resource can support varies, depending on the resource.
Signaling messages are exchanged between IP phones and Cisco Unified Communicalions
All conference bridges that are under the control ofCisco Unilied Communications Manager
fhe number of individual conferences, as well as the maximum number of participants per
5-10
Audio streamsexist between IP phones and the transcoder and between the application server and the transcoder.
Audio
Signaling
Atranscoder converts aninput audio stream thatuses onecodec into anoutput stream that uses a different codec. The transcoder inthe figure is implemented by using theCisco IOS router DSP resources. "Hie example shows an application server, such asa voice-mail server that
supports only G.711 codecs. In the Cisco Unified Communications Manager network, the
G.729 codec is preferred.
Audio streams exist from the IP phones tothe transcoder and from the application server lothe
transcoder.
Signaling messages are exchanged between IP phones and Cisco Unified Communications Manager, between a transcoder and Cisco Unified Communications Manager, and between an
application server and Cisco Unified Communications Manager.
DSP resources are required toperform transcoding. Those DSP resources can beinthe voice
modules and in the hardware platforms for transcoding.
Media Resources
The MTP bridges two media streams and allows them to be set up and torn down independently
Audio streams exist between IP phones and the MTP Signaling is exchanged between IP phones and Cisco Unified Communications Manager and between the MTP and Cisco Unified Communications Manager.
Hardware MTP
Audio
Signaling
SIP
The MTP bridges two media streams and allows them lo be set up and lorn down
independently.
An MIP canbe Used as an instance of translation between incompatible audio slreams, to svnchroni/e clocking, or lo enable certain devices for supplementary services.
Audio streams exist between IP phones and an MTP.
Signaling messages are exchanged between IP phones and Cisco Unified Communications Manager and between an MTP and Cisco Unified Communications Manager.
MTPs can be used to provide ihe following general features.
MTP Types This subtopic describes the different types ofMTPs, as well as their characteristics.
MTP Types
Three MTP types exist:
Software MTP provided byCisco Unified Communications Manager
Uses same codec and packetization on both call legs. - For functions such as RSVP agents or Cisco Unified Border Element media flow-through configurations. - To Cisco Unified Communications Manager, every Cisco IOS
Software MTP is considered as a hardware MTP.
- Use of the same audio codec but different packetization on both call legs ispossible.
Software M'fP. provided by the Cisco IP Voice Media Streaming App service on Cisco
Unified Communications Manager:
This MTP type can convert G.711 mu-lawtoG.711 a-law and vice versa.
This MTP type can packetize conversion for a given codec; for example, when one call leg uses 20-ms sample size and the other call leg uses 30-ms sample size. This MTP type does notrequire any DSP resources on the Cisco router. Enable Cisco IOS Software MTPs by using the maximum session software <ri> command. As many as 500software-based sessions canbe configured.
The codec and packetization of both call legs mustbe identical.
This MTP type typically is used for Resource Reservation Protocol (RSVP) agent configurations or Cisco Unified Border Element media flow-through configurations.
Cisco Unified Communications Manager does not differentiate between softwareand hardware-based Cisco IOS MTP configurations. Every Cisco IOS Software
MTP is considered as a hardware M'I'P in Cisco Unified Communications Manager.
DSPresources are required. Configure this M'fP type by using the maximum
session hardware <n> command. The maximum number of sessions is derived
from the number of installed DSP resources on the Cisco IOS router.
Use ofthe same audio codec but differentpacketization on both call legs is possible.
Media Resources
5-13
Note
The following configuration shows a Cisco IOS Software hardware and software M'
configuration;
seep com group 1
codec pass-through
maximum sessions software 100
MTP Functions and Requirements This figure shows supported MfP functions that are based on the MTP type that is used.
Cisco IOS
Software MTP
Yes
Yes
Yes
No
No
Yes
No
No
Yes
Yes
Yes
Provide H.323v1
supplementary services
ygs
Yes
Yes
Depending on the MTP type lhat is used, different functions are provided. All three MTP types support the insertion ofdual tone multifrequency (DTMF) signaling and media termination, to provide supplementary services such as Hold and Transfer for H.323 version I (H.323vl).
Media Resources
luncii
Signaling is exchanged between IP phones and Cisco Unified Communications Manager, between the annunciator and Cisco Unified
Audio
Signaling
PSTN
An annunciator is a software function ofthe Cisco IP Voice Media Streaming Application. An annunciator prov ideslite abilitv to siream spoken messages or various call-progress tones from
the svstem to a user.
An annunciator can send multiple one-way RTP slreams to devices such as Cisco IP phones or
gateways and uses SCCP messages to establish the RTP stream. To use this feature, the device
must support SCCP. fhe system predefines tones and announcements. The announcements
support localization and can he customized by replacing the appropriate .vvav tile, fhe
annunciator can support G.71 I a-law and mu-law. (i.729. and wideband codecs, without anv transcoding resources.
Signaling messages are exchanged between IP phones and Cisco Unified Communications
Manager, between the annunciator and Cisco Unified Communicalions Manager, and between the galewav and Cisco 1inilied Communicalions Manager. The audiostream is one vvav onlv: from the annunciator to the IP phone or gatewav.
Auflio
Signaling
PSTN
MOH is an integral feature ofCisco Unified Communications systems. This feature provides
music to callers when a call is placed onhold, transferred, parked, or added to an Ad Iloc
Audio streams exist between IPphones and the MOH server and between the gateway and the
MOH server.
Signaling messages are exchanged between IP phones and Cisco Unified Communications Manager, between the MOH server and Cisco Unified Communications Manager, and between
the gateway andCisco Unified Communications Manager.
Media Resources
5-17
Resoi
Manager Server
Hardware Conference
PSTN
Manager serv ice and supports onlv single-mode conferences that use a single codec (G.711). Some hardware conference bridges can support multiple low-bil-rate (l.BR) stream Ivpes such asG.729. Global Svstem for Mobile Communications (GSM), orG.723. This capability
enables these hardware conference bridges to process mixed-mode conferences. In a mixedmode conference, the hardware conference bridge transeodes G.729, GSM. and G.723 streams
into G.71 I streams. The conference bridge then mixes the slreams and encodes the resulting
stream into the appropriate stream type for transmission back to the user. Some hardware conference bridges support onlv G.711 conferences.
Type
Ad Hoc Meet-Me
ijn.far.-h,rtfcM^-,iHMi>Kiat3
64 128
Any combination ofwideband or G.711 a-law and mu-law streams can be connected to the
same conference. Thenumber of conferences thatcan be supported on a given configuration
depends on the server on which the conference bridge software is running and on the other functionality that has been enabled for the application. The Cisco IP Voice Media Streaming Application isaresource that can also be used for several functions, and the design must
consider all functions.
Caution
If the Cisco IPVoice Media Streaming App service runson the same serveras the Cisco CallManager service, a software conference should not exceed the maximum limit of48
participants.
Media Resources
5-19
WS-X6608-T1, WS-X6608-E1
NM-HDV
WS-SVC-CMM
CiscoVideo ConferenceBridge(IP/VC-35xx)
All conference bridges lhat are under ihe control ofCisco Unilied Communications Manager
use SCCP to communicate with Cisco L'nified Communications Manager. Cisco Unified Communications Manager allocates a conference bridge from a conference bridge resource that is registered with the Cisco Unified Communications Manager cluster. Both hardware and software conference bridge resources can register wilh Cisco Unified Communications Manager at the same time, and Cisco Unilied Communications Manager can
allocate and use conference bridges from either resource. Cisco Unified Communicalions
Manager doesnot distinguish between these tvpes of conference bridges when it processes a
conference-allocation request.
fhe numberof indiv idual conferences that the resource can supportvaries, and the maximum number of participants in a singleconference varies, depending on the resource. The following Ivpes of hardware audioconference-bridge resources can be usedon a Cisco Unified Communications Manager svstem:
Cisco High-Density Voice Nelwork Module 2 (NM-l IDV2) and NM-HD-l V/2V/2V1:.
Cisco 2800 and 2900 Series Routers, and Cisco 3800 and 3900 Series Routers
Note
hardware may have become available since the writing of this course material.
5-20
The ligure describes the maximum number of conferences that different resource types provide.
Conferences Per Resource
Affect secure conferencing:
Paftidpams pMS:
- - Conference.:';
WS-X660B-T1, WS-X6608E1
32 per port
NM-HDV
The following guidelines and considerations apply to the hardware audio conference bridge
resources:
Cisco NM-HDV2 and NM-HD-1V/2V/2VE, Cisco 2800 and 2900 Series Routers, and
Cisco 3800 and 3900 Series Routers):
Based on the C5510 DSP chipset, the Cisco NM-HDV2 and the router chassis use
the packet voice DSP module, generation 2(PVDM2) modules to provide DSPs.
The Cisco NM-HDV2 has four slots that accept PVDM2 modules inany combination. The other network modules have a fixed number of DSPs. Aconference that isbased on these DSPs allows a maximum ofeight participants. When a conference begins, all eight positions are reserved. The PVDM2-8 module islisted as having one-half ofa DSP. 'fhis module has a DSP
that has half the processing capacity ofthe PVDM2-16 module. For example, ifthe
bridgesper DSP (L0.5* 8] = 4).
DSP ona PVDM2-8 module is configured for G.711, it can provide four conference
ADSP farm configuration in aCisco IOS gateway specifics which codecs the farm can accept. ADSP farm that is configured for conferencing and G.711 provides
streams.
eight conferences. When configured to accept both G.711 and G.729 calls, asingle^ DSP provides Iwo conferences because it also reserves resources for transcoding ot
Media Resources
both G.711 and G.729 codecs, then each DSP provides onlv Iwo conferences of eight participants each. In this ease, you can populate the network module completely and configure itwith 16 DSPs, which will provide 256 slreams. Conferences cannot natively accept calls that use the GSM codec. Atranscoder must
be provided separately lor these calls toparticipate in a conference.
module: (48 *8) participants =384 streams. Ifall conferencing is configured for
number ofconference bridge resources that are allocated does not cause Ihis limit to he exceeded. If G.711 conferences are configured, then allocate no more than six DSI s(for atotal ot 48 conferences with eight participants each) per network
The ,0 ot aCisco NM-IIDV2 is limited lo 400 streams, so you must ensure that the
Cisco WS-SVC-CMM-ACl":
This Cisco Catalyst hardware prov ides DSP resources that can supply conference
bridges v\ ith as many as 32 participants per bridge.
32 conference bridges.
bach module contains four individually configurable DSPs, liach DSP can support
Ihe G.711 and G.729 codecs are supported on these conference bridges, without
chipset. Conferences thai use Ihis hardware provide bridges lhal allow as many as
ihe resources are configured, on aper-DSP basis, as conference bridges.
fhe Cisco VM-1IDV can have as many as four PVDM-256K modules: the 1700
Ihis hardware uses the PVDM-256K-type modules that are based on the C549 DSP
liach DSP prov ides one conference bridge lhat can accept G.71 1orG.729 calls.
fhe Cisco I75 1Modular Access Router is limited in live conference calls per chassis: the Cisco 1760 Modular Access Router can support 20 conference calls per
chassis.
Any PVDM2-based hardware, such as the Cisco NM-HDV2. can be used simultaneously in a single chassis for voice termination hul cannot be used
simultaneously for other media-resource functionality. The DSPs that are based on PVDM-256K and PVDM2 have different DSP-farm configurations, and onlv farm
can be configured In a router at a time.
This hardware has eighl DSPs lhat are physically associated to each port, and there
are eight ports per card.
Configuration ofthe DSPs occurs at the port level, so all DSPs lhal are associated lo
a port perform the same function.
Note
Conference bridges can have as many as 32 participants, and each port supports as
many as ?>2 conference bridges.
For conferences with G.71 1orG.723. there may be 32 conferences per port. If
Some ofthelisted products are end ofsale. Refer toCisco.com as new media resource hardware may have become available since thewriting ofthis course material
Only the Barge feature invokes this built-in conference bridge, which isnot used as ageneral
conferencing resource. This type of bridge accepts only G.711 calls.
Media Resources
5-23
1 Meet-Me
Conference originator controls the conference. Originator can add and remove participants.
Advanced Ad Hoc
Any participant can add and remove other participants. Link multiple Ad Hoc conferences together.
Meet-Me conferences allow users to dial in to a conference. Ad Hoc conlcrenees allow the
ofthe conference. When a Meet-Me conference is set up.the conference controller chooses a directory number and advertises it to members ofthe group. Theusers call the directory
number tojoin the conference. Anyone who has calling privileges tocall the directory number
while the conference is active can join the conference.
'fhere are two types of Ad Hoc conferences: basic and advanced.
In basic Ad Hoe conferencing, the originator of the conference acts as the controller of the
conference and is theonly participant who can addor remove other participants. In ads anced Ad Iloc conferencing, any participant can add or remove other participants: that capability is not limited to the originator ofthe conference. Advanced Ad Hoe conferencing
also allows linking of multiple Ad Hoc conferences. Set the Advanced Ail 1loc Conference Enabled clusterwide service parameter to True, to gain access to advanced Ad Hoc
conferencing.
5-24
s ;
Implement hardware conference media resources (ifdesired). -. Configure hardware media resource in Cisco Unified
Communications Manager.
n z
Configure hardware media resource in Cisco IOS gateway. Verifythat the hardware media resource registered with Cisco
Unified Communications Manager.
Three main steps are requiredto configurethe conference bridge media resource, as shown in
the figure.
Media Resources
5-25
famint
Unified Serviceability
under Tools > Service
Activation, to enable
software media
resources on
Manager servers
.'
tl,ifi-(1j:
(1-11. "tildWia iJliHfH U3..0 Unified Mosi'E vdh:
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Activate the Cisco II' Voice Media Streaming App service in Cisco Unified Serviceability
under lools > Sen ice Activation. At the top ofthe service activation screen, choose the server
on which services should be activated or deactivated. Then, check the Cisco IP Voice Media
The next step is to configure the Cisco IP Voice Media Streaming App service parameters.
The following Cisco IP Voice Media Streaming App service parameters that relate tothe
software conference bridge:
Call Count: This parameter specifies the maximum number ofconference participants that the conference bridge will support, fhe range is0to256; the default is48. Increasing this
value above the recommended default can cause performance degradation on a Cisco
Unified Communications Manager thatis running onthesame server. If increasing this value above the default is necessary, consider installing the Cisco IP Voice Media
Streaming Application on a separate server.
Run Flag: This parameter determines whether the conference bridge functionality ofthe
Cisco IPVoice Media Streaming Application isenabled. Valid values are True (enabled) or
False. The default value is True.
Note
These settings are parameters ofthe Cisco IPVoice Media Streaming App service and can
be accessed from Cisco Unified Communications Manager Administration under System >
Service Parameters.
Media Resources
n<
Conference bridge is
automatically added with
1 IPAdd-ejs
:C.I.I 1
default configuration
parameters when the IP
Csro OWere r_p Bridge Sc.rtAare
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Apply Coifig
fhe ligure shows the default configuration of a software conference resource, fhe onlv
configurable items are Conference Bridge Name. Description. Device Pool, Common Device
Configuration, and Location.
Note
The Cisco Unified Communications Manager software conference bridge media resource
supports only the G.711 and wideband codecs. Use a transcoder to allow devices that use other codecs to participate in a conference, or use hardware conference resources that
support additional codecs
Add the Conference Bridge Name as configuredon the router. Specify the Device Pool.
Set the Device Security Mode.
hardware application, allows both Ad Hoc and Meet-Me voice conferencing. Each conference bridge canhostseveral simultaneous, multiparty conferences.
Bothhardware and software conference bridgescan be active at the same time. Software and
hardware conference devices differ in the number of streams and the types of codec that they
support.
"fhe hardware model type for Conference Bridge contains specific MAC-address anddevicepool information. Different conference bridge fields areavailable in Cisco Unified Communications Manager Administration, depending on the conference bridge typethat was
chosen.
Navigate to Media Resources >Conference Bridge and click Add New. The Conference Bridge Configuration window appears. Enter the appropriate settings, as described in the following list, and click Save.The window refreshes anddisplays the conference device that
was added. To reset the conference bridge device and apply the changes, click Reset.
Media Resources
Conference Bridge .Name: Enter a name for Ihe conference bridge. The name must match the name oftheconference bridge media resource as configured at the Cisco IOS router
(seethe next step).
Note
If the conference bridge type is Cisco IOS Conference Bridge, the nameofthe Cisco IOS
conference bridge media resource CFB followed bythe MAC address of the interface that is used for SCCP signaling Ifthe conference bridgetype is Cisco IOS Enhanced Conference
Bridge, any name for the conference bridge media resource can be configured at the Cisco IOS router The name is case-sensitive, It mustexactly match the name ofthe conference
bridge media resource in the Cisco IOS router.
contigured in the Common Dev ice Configuration window appear in the drop-down list.
Location: Choose the appropriate location for ihisconference bridge. Locations are configured in Cisco Unified Communications Manager Administration > System >
Location. 'I he selected location specifies the maximum bandwidth for calls lhal come to or
go from that location. Location is used to limit the number of calls lhal can be established
between locations (Call Admission Control [CAC|), Devices areassigned locations, which
can also be set to have unlimited bandwidth (by setting the location to Hub None).
Note For example, you can configure three locations: Lod with bandwidth of 800 kb/s, Loc2 with
bandwidth of 400 kb/s, and Loc3 with bandwidth of 400 kb/s. When one G.711 call is made
from one location to another, the call consumes 80 kb/s of location bandwidth at each
location (source and destination) In this example, 5 simultaneous calls can be established
to or from Loc2 and Loc3. 10 simultaneous calls can be established to or from Lod
Device Security Mode: This Held is available for Cisco IOS Enhanced Conference Bridge
onlv. If vou choose Non Secure Conlerence Bridge, the nonsecure conference establishes a TCP poil connection to Cisco Unified Communications Manager on port 2000. Ensure lhal this setting matches the securilv settingon the conference bridge, or the call will fail, fhe linen pted Con ference Bridge selling suppons the secure conference lealure. Refer to the Cisco I nified Communications Manager Security Guide for secure conference-bridge
configuration procedures.
dspfa dap
ices dapfarm
jl FastEthernetO/O.lOl
cp 1(
Manager
For Verification Use: show accp
Ihe following table shows the commands for configuring a Cisco IOS Knhanced Conference
Bridge.
Media Resources
5-31
Command
Command Function
dspfarm
seep ccm
This command adds a Cisco Unified Communications Manager server to the listof available servers and sets various parameters, including IP address or Domain Name System (DNS) name, port number, and
version number Use the command in global configuration mode
seep
This command enables the SCCP protocol and its related applications (transcoding and conferencing). Use the command in global
configuration mode.
This command creates a Cisco Unified Communications Manager group and enters SCCP Cisco Unified Communications Manager configuration mode. Use the command in global configuration mode This command associates a Cisco Unified Communications Manager with a Cisco Unified Communications Manager group and establishes
its priority within the group. Use the command in the SCCP Cisco
associate ccm
This command associates a DSP farm profile with a Cisco Unified Communications Manager group. Use the command in SCCP Cisco Unified Communications Manager configuration mode. This command enters DSP farm profile configuration mode and defines a profile for DSP farm services. Use the command in global
configuration mode
dspfarm profile
codec
This command specifies the call density and codec complexitythat are
based on a particular codec standard. Use the command in DSP interface DSP farm configuration mode.
associate application
seep maximum sessions
This command associates SCCP to the DSP farm profile. Use the command in DSP farm profile configuration mode
This command specifies the maximum number of sessions that the
Tip
The name that is specified in the Cisco IOS device must match the name in the Cisco Unified Communications Manager. The names are case-sensitive.
Note
When configuring a Cisco IOS Enhanced Conference Bridge, you can use the associate profile command to configure any name. When configuringa Cisco IOS conference bridge,
you cannot configure the name The name is CFB(MAC), where (MAC) is the MAC address of the interface that was specified in the seep local command
To verifv the Cisco IOS media resource configuration, use the following show commands:
show seep SCCP Admin States UP
Call Manager: 10.1.1.1, Port Number: 2000 Priority: N/A, Version: 7.0, Identifier: 1 Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.1.1.1, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1
g711alaw, Maximum Packetization Period: 30 g729ar8. Maximum Packetization Period: 60 g729abr8, Maximum Packetization Period: 60 g729r8, Maximum Packetization Period: 60 g729br8, Maximum Packetization Period: 60 rfc2833 dtmf, Maximum Packetization Period:
Registration Retries: 3, Registration Timeout: 10 sec Keepalive Retries: 3, Keepalive Timeout: 30 sec
CCM Connect Retries: 3, CCM Connect Interval: 10 sec
Codec Configuration
Media Resources
5-34
ce
You can configure the following Cisco CallManager service parameters thatrelate
conferencing:
Suppress MOH to Conference Bridge: This parameter determines whether MOI1 plays to a conference when a conference participant places the conference on hold. Valid values are True (the system does notplay MOH to theconference when a conference participant presses the Hold button) or False. Thedefault value is True.
Never (default): Theconference remains active afterthe conference controller hangs up andafterall on-net parlies hangup.If on-net parties conference in off-net parties andthen disconnect, the conference staysactive between the off-net partiesa situation that can result in toll fraud.
When Conference Controller Leaves: The conference terminates when the
Media Resources
Advanced Ad Hoc Conference Enabled: This parameter determines whether advanced Ad Hocconference features are enabled. Advanced Ad Hoc conference features include the ability for conference participants other than the conference controller lo add new participants lo an existing Ad Hoe conference, the ability for any noncontroller conference participant to drop other participants from the conference via the ConlList or RmLstC softkev. and whether Ad Hoc conferences can be linked together via features such as Conference. Join. Direct Transfer, and Transfer. Valid values are True (allowadvanced Ad
I loc conference features) or False. The default value is False.
Nonlinear Ad Hoc Conference Linking Enabled: fhis parameter determines whether more than two Ad Hoc conferences can be linked directly, in a nonlinear fashion, to an Ad
Iloc conference. Nonlinear conference linking occurs when three or more Ad 1loc
conferences are linked dircctlv to one other Ad Hoc conference. Linear conference linking
occurs when one or two Ad Hocconferences are linked directly to one other Ad Hoc conference. Ihe Adv anced Ad Iloc Conference Fnabled serv ice parameter must he set to Irue for this parameter to work proper!). Valid values are True (allow nonlinear
conference linking so that three or more Ad Hoc conferences can be linked to one other conference) or False. The default value is False. Fhe Advanced Ad Hoc Conference
Fnabled service parameter must be set to True for the Nonlinear Ad 1loc Conference Linking Fnabled service parameter to work.
participants that areallowed in a single Ad Hoc conference. The value of this field depends on the capabilities ofthe software or hardware conference bridge. Setting Ibis value above
the maximum capaeitv ofthe conference will result in failed entrance to a conference
bridge, if more ports are addedthan the specific conference bridge configuration allows.
Hie value range is 3 to 64: the default value is 4,
Mavimum Mtct-Me Conference I nkast: Fhis parameter specifies the maximum number of participants that are allowed in a single unicast Meet-Me conference. The value of this field depends on the capabilities ofthe softwareor hardware conference bridge: for example, a software conference bridge conferences as many as 128participants. When a conference is created, the sv stem automatically reserves a minimum of three slreams. so
specifv ing a value less than 3 allows a maximum of three participants. The value range is I
to 128: the default value is 4.
Note
These settings are service parameters of the Cisco CallManager service and can be
accessed from System > Service Parameters.
5-36
Meet-Me number or pattern must be configured: - Meet-Me numberrange is part ofthe dial plan and must not
overiapwith other numbers.
To configure directory numbers for Meet-Me conferences, you must first ensure that the necessary hardware and software conference bridge media resources are configured and
available.
Then, you must configure a Meet-Me number orpattern. When a pattern isconfigured, you can use the" Xwildcard to specify ranges. The Meet-Me number range is part ofthe dial plan and must not overlap with other numbers. Partitions and calling search spaces (CSSs) must be
configured ifaccess tospecific Meet-Me numbers should berestricted.
Media Resources
,.,
- H e el -H e Contij u rl tan -
C esc',;,(
lo adda number or number range to use for Meet-Me conferences, go lo Cisco Unilied Communications Manager Administration. Call Routing > Meet-Me Number/Pattern, click Add New. andconfigure the new pattern with the following data;
Directory Number or Pattern: Kntera Meel-Me number or pattern or a range of numbers. To configure a range, the dash must appear within brackets and follow a digit; lor example,
to conligure the range 10(10 to 1050. enter 10[0-5|0.
Description: fnter as manv as 30 alphanumeric characters (ora description ofthe MeetMe number or pattern.
Partition: To use a partition to restrict access to the Meet-Me number or pattern, choose
the desired partition from the drop-down list. 'lo exclude restricted access to the Meet-Me number or pallern, choose None for the
parlilion.
Note
Make sure that the combination of Meet-Me number or pattern and partition is unique within
the Cisco Unified Communications Manager cluster.
Minimum Sccurit) Level: Choose the minimum Meet-Mc conference security level for this Meel-Me numberor pattern from the drop-down list:
Choose Authenticated to block participants with nonsecure phones from joining the
conference.
Note
To use conference security, the Cisco Unified Communications Manager cluster must be enabled for secure mode. More information about security features in Cisco Unified Communications Manager is provided in the Implementing Cisco Unified Communications
Manager, Part 2 (CIPT2) course.
Media Resources
MOH Overview
This topic describes the MOH server and itscapabilities.
For special cases, external media-streaming servers can be used The Cisco Unified Communications Manager integrated MOH server supports multicast and unicast for MOH streaming
Integrated Software MOH Server in Cisco
PSTN
lor callers to hear MOI I. Cisco Unitied Communications Manager must be configured to
support the MOII feature. The MOH feature has two main requirements:
An MOH server to provide the MOH audio-stream sources
A Cisco ('nified Communications Managersystem that is configured to use ihe MOH streams lhat the MOH server prov ides when a call is placed on hold
The integrated MOH feature makes music available to any on-or off-net device that is placed on hold. On-net devices include station devices andapplications that are placed on hold,
consult hold, or park hold bv an interactive voice response (IVR) or call distributor. Off-net users include those users who are connected through Media Cialewav Control Protocol (MGCP). Session Initiation Protocol (SIP)- or H.323 gateways. The MOII feature is also available for plain old telephone service (POT'S) phones lhat connect to the Cisco IP network through foreign F.xchange Station (I'XS) ports. The integrated MOII feature includes the media server, database administration, call control. Media Resource Manager (MRM). and media
control functions. Ihe MOH server provides the music resources and streams.
In special cases, you can configure multicast MOII streaming so that external media servers can be used to prov ide the MOH stream. You can configure Cisco Unified Communications Manager Express and Cisco L'nified SRST gateways lo act as media-streamingservers for MOH by streaming audio files that are stored in the flash memory ol'Cisco IOS routers thai use multicast. Fordetailed information about this feature, consult Cisco Unified Communications
System Release 8.x SR\D.
The Cisco L'nified Communications Manager integrated MOH server supports multicast and unicast for MOH streaming. Using multicast rather than unicast for MOH streaming saves bandwidth and reduces the load on tlie MOH server. Saving bandwidth should not be a major
issue for campus I.AN environments. However, reducing load on the MOH server, by reducing
the number of media streams, isadvantageous, especially when the MOH server is coiocated on
the same server as the Cisco CallManager service.
Media Resources
5-41
MOH Sources
Cisco Unified Communications Manager automatically generates MOI I audio tiles when *,wav
audio files are uploaded to the MOII server.
10H Sources
MOH sources
One fixed source that uses a Cisco MOH USB audio sound card
50 audio file sources
MOH Audio File Management converts the audio file Codecs used for MOH are G 711, G.729. and wideband
When the administrator imports an audio source file, the Cisco Unified Communications Manager Administration interface processes the file and converts the file to Ihe proper formats
for use bv the MOH server. The recommended formal for audio source files includes the
following specifications:
If recorded or live audio is needed. MOH can be generated from a fixed source, for this type of MOII. a sound card is required. The fixed audio source is connected lo the audio input ofthe
local sound card.
"I his mechanism enables the use of radios. CI) players, or any other compatible sound source. The stream from the fixed audio source is transcoded in real time to support the codec that was configured through Cisco Unified Communications Manager Administration. The fixed audio
source can be transcoded into G.71 1 (a-law or mu-law), CS.721) Annex A, and wideband, and it is the only audio source that is transcoded in real time. The Cisco MOH I ;SB audio sound card (MOH-USB-AUDIO-) must be used lo connect a
fixed or live audio source lo the MOH server. This USB sound card is compatible with al!
Cisco Media Convergence Server (MCS) platforms that support Cisco Unified Communicalions Manager, Before using a fixed atidio source to transmit MOH, consider the legalities and ramifications of rebroadcasling copv righted audio materials. Consult the customer legal department for potential
issues.
Unicast MOH
Unicast MOH consists of slreams that are sentdirectly from the MOI 1server to theendpoint that requests an MOH audio stream.
Unicast MOH
MOH unicast characteristics:
* Separate audio stream for each connection Negative effect on netvwrk throughput and bandwidth
Useful in networks in which multicast is not enabled and
jr
; .. IP Address
Unicast MOH ^
Unicast MOH
UOHSfiSVe
A unicast MOHstream is a point-to-point, one-way audio RTP stream between the server and
the endpoint device. Unicast MOH uses a separate source stream for each user or connection. As more endpoint devices go on hold via a useror network event, the number of MOI I streams increases. Thus, if 20 endpoint devices are on hold, 20 streams of RTPtraffic are generated
over the network between the server and the devices. These additional MOH streams can have
a negative effecton network throughput and bandwidth. However, unicast MOH canbe extremely useful in networks in which multicast is notenabled or devices are incapable of
multicast. In such nelvvorLs, unicast MOH allows administrators to take advantage ofthe MOH
feature.
Media Resources
5-43
Multicast MOH Multicast MOII consists of streams that aresen! from the MOH server to a multicast group IP
address. Fndpoints that request an MOH audio siream can join multicast MOII. as needed.
i Multicast Group
Multicast MOH
A multicast MOH stream is a point-to-multipoint. one-way audio RTP stream between the MOH server and ihe multicast group IP address. Multicast MOII conserves system resources
and bandw idth because it enables multiple users to use the same audio source stream to provide MOH. Ihus. if 20 devices are on hold, as few as one stream of RIP traffic might be generated over the network. Multicast is an extremely attractive technology for Ihe deployment of a service such as MOI I because it greatlv reduces the CPU impact on the source device and the bandwidth consumption for deliverv over common paths. However, multicast MOII can be problematic in situations in which a network is not enabled for multicast or the endpoint
devices arc incapable of processing multicast.
MOH Audio-Source Selection This figure describes how the MOH audio source file and MOH audio server are selected in
Cisco Unified Communications Manager.
Make sure that configured audio fies are available on all TFTP servers.
Audio 1
PHoneB ! UserHoldAiidio2
I M0HA^jp
*srjji|j;
Lsten to
Audio 2
jhR
^^
Phone A IUserHoldAudio4
UseMRGLA.
"fhe basic operation of MOH in a Cisco Unified Communications environment consists of a holder and a holdce. The holder is the endpointuser or networkapplication that places a call on
hold, and the holdee is the endpoini user or device thai is placed on hold.
"fhe MOH stream thatan endpoint receives is determined by a combination of the User Ilold MOH Audio Sourcethat is configured for the holderand the prioritized list of MOII resources (MRGL)that is configured for the holdee. The User Hold MOH Audio Sourcedetermines
which audio file will be streamed when the holder puts a call on hold. The list of MOH
resources determines the server from which the holdee receives the MOH stream.
In the figure, if phones A and B areon a call and phone B (the holder) places phoneA (the holdee) on hold,phone A hears the MOH audio source that is configured for phone B (Audio 2). However, phone A receives this MOII audio stream from the resource or server that is
configured for phone A.
Note When more than one MOH server is active in the network, make sure that all the configured
MOH files are available for all MOH servers. You might need to copy the files manually to
the root directories of all the TFTP servers.
Media Resources
5-45
MOH Configuration
This topic describes the configuration ofthe Cisco Iinificd Communications Manager MOII
feature.
Note
When configuring multicast MOH, it is mandatory to use media resource groups and media resource group lists. At the media resource group that includes the multicast MOH server, 3
check box has to be activated which allows multicast to be used
Media resource groups and media resource group lists are discussed in more detail in later
topics of this lesson
Cisco Platform
Co-resident or standalone
G.729
G.729 Wdeband
As with all media resources, capacity planningis crucialto make certainthat the hardware, after being deployed andconfigured, cansupport the anticipated call volume ofthe network. Therefore, it is important to be awareofthe hardware capacity for MOH resources and to consider the implications of multicast and unicast MOH in relation to this capacity. Ensure that
network call volumes do not exceed these limits. When MOH sessions reach these limits,
additional load could result in poor MOH quality, erratic MOH operation, or even loss of MOH functionality. The following MOH Server Configuration parameters affect MOI I server
capacity:
Maximum Half Duplex Streams: This parameter determines the numberof devicesthat can be placedon unicast MOI 1. By default,this value is set to 250."fhe Maximum Half Duplex Streams parameter shouldbe set to the value that is derived from the following formula: (Server and deployment capacity) - ([Number of multicast MOH sources] * [Number of enabled MOH codecs]). The value of this parameter should never be set higher than the capacities lhat arc indicated in the table, according to the platform and deployment
type (coresident or standalone). Maximum Multicast Connections: This parameter determines the number of devices that can be placed on multicast MOH. By default, this value is set to 30.000. The Maximum Multicast Connections parameter should be set to a value that ensures that all devices can be placed on multicast MOH if necessary. Although the MOH server can generate only a finite number of multicast streams (a maximum of 204), many held devices can join each multicast stream. This parameter should be set to a number that is greater than or equal to the number of devices that might be placed on multicast MOH at any given time.
Media Resources
Typically. multicast traffic is accounted for according to the numberof streams that arcgenerated. However. Cisco Unified Communications Manager maintains a countofthe actual number of de\ ices that are placed on multicasl MOH or joined to each multicasl MOH stream. This method is different than the wa\ multicast traflic is normally tracked.
Note Regardingthe maximum recommended number of MOH streams (250 MOH streams on
Cisco MCS 7815 and 7625 Series and 500 MOH streams on Cisco MCS 7835 and 7845
Series) Each multicast audio source must be counted as fwo MOH streams. For example,
for a Cisco MCS 7835 and 7845 Series, if three multicast MOH audio sources and four
codecs are enabled, no more than 476 unicast MOH streams should be generated at the
same time (2 * 3 * 4 + 476 = 500).
5ar>p*&Mrf|ioSaijrTC
C^mMif. a
See lis! of files. Custom tiles
can be added or deleted.
i*-,;
Cisco Unified Communications Manager, by default, has one MOI I audio file, SampleAudioSource. To addadditional MOH audio files, go to Media Resources > MOH Audio File Management in CiscoUnified Communications Manager Administration andclick
I pload File.
The uploaded file is automatically converted intodifferent audio formats (one percodec). At
the Find and List Music On Hold Audio Files window, which is accessed via Media Resources
> MOH AudioFile Management, a file status of Translation Complete indicates that the audio file has been converted successfully. Foraudio files that have beensuccessfully converted and are alreadyconfigured as an MOH audiosource, the file status is In Use. Duringthe
conversion, the status is Open.
If anyotherstatus is displayed, or if the status remains Open for a longer period, the audio file translation fails. Depending on the size ofthe audio file and the load on the server, conversion can take as long as several minutes. The uploaded audiofile mustbe in .wav file formatand
meet the following specifications:
Delete the tiles that could not be translated from Media Resources > MOH Audio File
Management.
Media Resources
Note
Theupload of MOH files must be performed separately at each MOH server To upload the files to an MOH server, use the IP address ofthe MOH server (ratherthan the publisher IP address) in the Cisco Unified Communications Manager Administration URL (https7//P
address of MOH server/ccmadmm) before selecting Media Resources > MOH Audio File
Management
5-50
Select MOH Audio Source File for selected audio source number.
. _.
. Slfll'T ren>D
Sulti^rkljs:.
T:njl_L*Te'nlu; ula
"Ml N,s b.i
I Add Now
Upkad File
To configure MOH audio sources, in Cisco Unified Communications Manager Administration, go to Media Resources >Music On Hold Audio Source. The MOH audio sources are
identified by an MOH Audio Stream Number (1 to 51).
Inthe Music On Hold Audio Source Configuration window, first choose the MOH Audio Stream Number ofthe audio source thatyou want to configure. Then choose the MOH Audio Source File. The MOH Audio Source Name defaults to the name ofthe MOH Audio Source File and can bemodified. Finally, enable ordisable continuous playing (repeating) ofthe audio
file.
Media Resources
The fixed audio source requires the Cisco USB MOH sound adaptor, which must be ordered separately.
ftt<#MOtt*m*eS*t*atlJlBlimrilkm
iJ- X3-
-'-''--'
(;;;"-
-.-..,
|^-^
Enable the fixed MOH audio
source.
j . >-.J;:-:~.-..r,.>:rn\====-:
andcannot be modified (only one fixed MOH audio source can be configured in a Cisco
UnifiedCommunications Manager cluster). You must enler Ihe name and enable ihe fixed
MOH audio source.
5-52
Step 3: Configure
In Cisco Unified CommuricatJons Manager Administration, under
Media Resources > Music On Hold Server:
The figure shows the default configuration ofthe MOH media resource. You can modify parameters such as Name, Description, Device Pool, Location, and Maximum Half Duplex
Streams (that is. unicast MOH streams).
Ifa fixed audio source that isphysically connected tothe server is used, the name ofthe audio
source device must be specified.
Media Resources
Suppress MOH to Conference Bridge (True) Default Network Hold MOH Audio Source ID(1) Default User Hold MOHAudio Source ID (1)
Duplex Streaming Enabled (False)
"fhe default parameter values are shown in parenthesis. Theseservice parameters need to be
configured onlv if there is a need to use nondefault values.
Note These service parameters can be accessed from System > Service Parameters Note that some of the parameters are Cisco IP Voice Media Streaming App service parameters and others are Cisco CallManager service parameters
ain pieAudi=5ourJ
MOH audio sources (it used) must be enabled for multicast MOH
Click the Allow Multicasting check box for each MOI 1audio source that is allowed to be sent as a multicast stream. This setting applies to MOH audio sources and to fixed MOH audio
sources.
Media Resources
erver
The figure shows how toenable multicast MOH on an MOH server. In the Multicasl Audio
Source Information section ofthe Music On Hold (MOH) Server Configuration window,
click the Enable Multicast Audio Sources on this MOII Server check box. The Base Multicast IP Address. Base Multi-cast Port Number, and Increment Multicast On fields arc automatical!) populated after \ou enable multicast MOH onthe server. You can modifv
these values as desired.
Note
You should increment multicast on IP address instead of on port number. Doing soresults in
each multicast audio source having a unique IPaddressand helps toavoid network
saturation in firewall situations.
All MOH atidio sources that ha\e been configured to allow multicasting are listed in the Selected Multicast Audio Sources section ofthe Music On Hold (MOII) Server Configuration
window. You can set the Max Hops \aluefor each audio source (the default is2). This
parameter sets the Time to Li\ c (7 11.) value in the IP headerofthe multicast MOI I RTP
packets to the specified \aluc. Tl I. in an IP packet indicates themaximum number of routers that an audio source is allowed to cross. If Max Hops is setto 0. the multicast MOII RTP
packets remain in the subnet ofthe multicast MOI I server. IfMax Ilops is sel to 1, the audio
source can cross one router to the next subnet. The recommended setting is 2.
Note
When using multicast MOH, and when the devices that should listen to multicast MOH streams are not in the same IP network, you must enable multicast routing in the IP network. Take care when enabling multicast routing to avoid potential flooding of parts of the network
with mis-sent multicast packets (especially across WAN links). To do so, disable multosts on interfaces on which the multicast MOH packets are not required, and use the Max Hops
parameter thatwasdiscussed earlier.
Note
To use multicast MOH when MRGs and MRGLs are used to implement media-resources access control and a multicast MOH server is assigned to an MRG, you must also enable
multicastMOH for the MRG. __^_ -
.^^
r- . _
Media Resources
5-57
Movefile Multicast
Multicast MOII onl\ works ifthe Multicast enabled MOH server is assigned lo a Multicast
enabled Media Resource Group, This MRG will be configured to bea memberofa Media Resource Group I ist(MRGL). The MRCil. will then be associated with devices such as
phones.
Note
5-58
Annunciator Overview
The annunciator is partofthe Cisco IPVoice Media
Streaming App service.
An annunciator is automatically created in the system when the Cisco IP Voice Media Streaming App service is activated on a server. Ifthe Cisco IP Voice Media Streaming App
service is deactivated, the annunciator is deleted. Asingle annunciator instance can serve the
entire CiscoUnified Communicalions Manager cluster if it meets the performance
requirements: otherwise, you must configure additional annunciators for the cluster. You can
add annunciators by activating the Cisco IP Voice Media Streaming App service on other
servers within the cluster.
The annunciator registers with one Cisco Unified Communications Manager at atime, as defined by its device pool. The annunciator will automatically fail over to a secondary' Cisco
Unified Communications Manager system if one isconfigured for the device pool. Any announcement that isplaying atthe time ofan outage will not bemaintained.
An annunciator is considered a media device andcan be included in an MRG. which can
control which annunciator is selected for use by phones and gateways.
Media Resources
5-59
Tones and announcements are predefined. The announcement support localization and can be customized by replacing the appropriate .wav file. The annunciafor can support G.711, G.729. and wideband codecs, without any transcoding resources. The following features require an annunciator:
Cisco MLPP (call failure)
Integration via SIPtrunk (call progressand DTMFtones) Cisco IOS gateways and intereluster trunks (ringback)
System messages (call failure) Conferencing (Barge tone)
Cisco Multile\el Precedence and Preemption (MLPP): This feature plays streaming
messages in response to the following call-failure conditions:
MTP to generate oraccept DTMh tones when integrating with a SIP endpoint. The
following t> pes oHones are supported: Call progress tones (bus_\. alerting, and ringback)
I) IMF tones
i Cisco IOS gateways and intereluster trunks: These devices require support for the call
progress tone (ringback tone).
I System messages: During the following call-failure conditions, the system plajsa
streaming message to the end user:
A dialed numberthai the s_\slem cannot recogni/e A call that is not routed because of a servicedisruption
A number that is bus> and not configured for preemption or call wailing
Conferencing: During a conference call, the system plays a barge-in tone lo announce Lhal
a participant has joined or left the bridge.
5-60 Implementing CiscoUnified Communications Manager, Part 1 (CIPTl) v8 0
& 2010 Cisco Systems, Inc.
Annunciator Performance By default, the annunciator is configured to support 48 simultaneous streams. That number is
the maximum that is recommended for an annunciator that runs on Ihe same server (coresident) with Cisco Unified Communications Manager.
Annunciator Performance
A standalone server without the Cisco CallManager service
Multiple standalone servers can be integrated to support the required number of announcement streams.
If the server has only 10-Mb/s connectivity, lowerthe setting to 24 simultaneous streams. A standalone server without the Cisco CallManager service can support as many as 255 simultaneous announcement streams; a high-performance server with dual CPUs and a high-
performance disk system cansupport as many as 400streams. Multiple standalone servers can
be added to support the required number of streams.
Media Resources
5-61
The annunciator is automatically added with default values when the IP Voice Media Streaming App service is
activated
AflBsKHtBT CrmRgB$v6BH
SI Back To F.nd/Ust
. j Go
Pejsira'.r.r
PiJd-n-. se-ier' N^*'
Jfl_l
-.NN CjCW'.-i
o*,'-<-..'
uraton" *'"-
"i_CF
fub -Ncne
-t" o'r
Ihe figure shows the default configuration olTlie annunciator, 'fhe only configurable ileitis are
Name. Description. Device Pool, and Location.
5-62
Manager CIueIw
Solware
Conference Bridge
SW CFB 2
SW_CFB_1
Hardware
Conference Bridge
* - - - .. ^>
xv
* Conlerence Bridge
SW_CFB_2
Hardware
The figure shows a phone that needs to select a conference bridge media resource.
B; default,all existing media resources are located in a Null MRG,and use ofthe resources is
load-balanced between all existing devices. Use ofthe hardware conference resources is
preferred because of their enhanced capabilities (mixed-mode conferences) and the reduction of
load on the Cisco Unified Communicalions Manager integrated software conference bridges.
Media Resource Manager (MRM) controls and manages the media resources within a cluster, allowing all Cisco Unified Communications Manager servers within the cluster to share media
resources.
MRM enhances Cisco Unified Communications Manager features by making it easier for Cisco Unified Communications Manager to deploy transcoder, annunciator, conferencing. MTP. and MOH resources. MRM distribution throughout the Cisco Unified Communications Manager cluster uses these resources to their full potential, making the Cisco Unified Communications Manager cluster efficient and economical.
Media Resources
Sonic ofthe reasons to use media-resources access control are its follows:
lo enable hardware and software media resources to coexist within a Cisco Unilied
I o enable Cisco Unified Communications Manager to share and access the resources lhat
arc available in the cluster,
To enable Cisco Unified Communications Manager to pcrfonn load distribution within a group of similar media resources.
To allow media-resources access control that is based on type of resource: for example, to allow one user, but not another, to use a hardware conference bridge.
MAC bundles media resources in load-balanced MRGs. which are listed in prioritized MRG! s.
MRM
Resource
IL|
First
MRGL
t
Second Choice
Choice
"H
Load Sriaring
Media Resource 1 Media Resource 2
K
Load Sharing
Media Resoirce 3 Media Resource 1
MRGLs specifya list of prioritized MRGs. An application can selectthe requiredmedia resources from amongthe available resources, according to the priority order that is defined in
the MRGL. MRGLs. which are associated with devices, provide MRG redundancy.
The figure shows the hierarchical ordering of media resources and how MRGs and MRGLs are like route groups and route lists.
Note When a device needs a media resource, it searches its own MRGL first. If a media resource is not available, the device searches the default list, which includes all of the media
resources that have not been assigned to an MRG. After a resource is assigned to an MRG.
it is removed from the default list.
Media Resources
Conf 1
Conf 2
Conf. 3
Conf. 4
Conf. 5
Conf 6
vv (..I !i z
SW CFB 3
MRG HW-CFB
MRGL CFB
1. MRG_HW-CFB
2 MRG SW-CFB
(DefaultNo MRG)
SW_CFB_3(1 Conf.)
S\V_CFB_2: fhis bridge has capacity for one conference. S\Y_CFB_3: This bridge has capacity for one conference.
IIWCTB I and H\\_CFB_2 arc in MRCilIW-CFB. SW CFBJ and SW_CITS_2are in MRG_S\\'-CFB. SW_CFB_3 is not assigned lo an MRG. MRGL MRGL CI-'B has MRCi MRGJIW-CFB listed before MRG MRG_SW-CFB. Ifsk conferences are established from devices that all use the MRGL MRGL_CFB. the
conference bridges will be allocated in the following way:
The first conference uses conference bridge IIW_CFB_L The second conference uses
conference bridge H\V_CFB_2 because Ihe resources within an MRCi are load-shared and not used in the configured order. Because ofthe load-sharing algorithm, the third conference uses 1IW_CFB_I again.
Because no resource is left in the first MRCi ofthe MRCil.. the fourth conference uses a
resource ofthe second MRCi: conference bridge SW_CFB_t. lite lillh conlerence uses SW_CFB_2. The sixth conference does not find a free resource in any MRG ofthe MRCil . Rather, the conference finds a conference resource in the default list (that is, the list of resources that have
not been assigned to am MRCi). Thai resource is SW CFB 3.
Implementing Cisco Unified Communications Manager. Part 1 (CIPT1) v8 i 2010 Cisco Systems, Inc
from the configured MRGL. iftwo or more oftheoriginal conference participants are videoenabled.
conference bridge from the configured MRGLof the conference initiator. Ifa video conference bridge needs to be allocated but none is available,
an audio conference bridge for the conference is allocated, and vice versa.
MR Audio j j MR Video |
MR = Media Resource
If there are one or no video participants, Cisco Unified Communications Managerselectsan audio conference bridge from the configured MRGL. Cisco Unified Communications Manager selects an audioor a video conference bridge from the configured MRGL ofthe conference initiator. If no MRGL is configured for die conference initiator. Cisco Unified Communications Manager allocates the video or audio conference
bridge from the default MRGL,
If a video conference bridge needs to be allocated but none is available. Cisco Unified Communications Managerallocates an audioconference bridge for the conference. Similarly, if an audio conference bridge is needed but is unavailable, Cisco Unified Communications
Manager allocates a video conference bridge.
Note The IntelligentBridge Selection feature is applicable only to Ad Hoc conferences and does not affect how conference bridges are allocated for Meet-Me conferences. The conference bridge for a Meet-Me conference is allocated based on the configured MRGL for the endpoint that initiates the conference. When allocating a conference bridge for Meet-Me
conference calls, Cisco Unified Communications Manager does not take into account
whether the conference initiator is video-capable.
Media Resources
5-67
ienl
Navigate to the Cisco CallManager service paramelers and set the Intelligent Bridge Selection parameters:
' Encrypted video conference bridges are not supported. Choose between an encrypted audio CFB and an unencrypted video CFB.
Specify the numberof video-capable conference participants that must
be present in a conference to allocate a video CFB Choose a video CFB, when available, for an audio conference when the
determines whether Cisco Unified Communicalions Manager chooses an encrypted audio conference bridge or an unencrypted video conlerence bridge for an Ad Hoc conference call, when the conference controller Device Security Mode is set to either Authenticated or Lncrvpted and at least two conference participants are video-capable. Because enervpted video conference bridges arc not supported. Cisco Unified Communications Manager must choose between an encrypted audio conference bridge and an unencrypted video conference bridge. Valid values specify True (allocate an encrypted audio conference bridge), which is the default, or False (allocate an unencrypted video conference bridge). Minimum Video Capable Participants to Allocate Video Conference: This parameter specifies the number of video-capable conference parlicipanls that must be present in an Ad 1loc conference to allocate a video conference bridge. If the number of video-capable participants is less than the number that is specified in this parameter. Cisco Unitied Communications Manager allocates an audio conference bridge. If Ihe number of videocapable participants is equal to or greater than the number that is specified in Ihis parameter, a video conference bridge is allocated (when available) from the configured MRGL. Specif) ing a value of 0 means that video conference bridges will alwavs be allocated, even when none ofthe participants on the conference are video-capable. When a conference has been established by using an audio bridge and additional video-capable
participants join the conference, the conference remains on the audio bridge and does not convert to \ ideo. The default value is 2: the range is 0 to 10.
Allocate Video Conference Bridge for Audio Only Conferences when the Vdeo Conference Bridge Mas Higher Priority: This parameter determmes whether (_ sco
an Ad Hoe audio-onlv conference call when the video conference bridge has ahigher
UnTlS Communications Manager chooses avideo conference bridge (when avada e) Ior
priority in the MRGL than an audio conference bridge has. Valid values specify True (allocate avideo conference bridge) or False (allocate an audio conference bridge), which is default Ifan audio conference bridge has higher priority than any video conference bridge in the MRGL. the Cisco CallManager service ignores this parameter Ihe parameter
is useful when the local conference bridge is avideo bridge (and configured in the MRGL with the highest priority) and audio conference bridges are available only in remote locations In lhat situation, enabling this parameter means that Cisco Unified Communications Manager attempts to use the local video conference bridge first, even tor
audio-onlv conference calls.
Media Resources
5-69
Ihe figure slums the three configuration steps that are required to configure media-resources
access control.
5-70
To add an MRG. go to Media Resources >Media Resource Group in Cisco Unified Communicalions Manager Administration. In the Media Resource Group Configuration window, enter aname and description for the MRG. and add the desired media resources to the
MRCi.
Note
If theMRG includes one ormore multicast MOH servers and should allow multicast MOH,
check the Use Multicast for MOH Audio check box.
Media Resources
5-71
* In Cisco Unified Communications ManagerAdministration, under Media Resources >Media Resource Group List
Add or remove
selected media
resource to or
from MRGL.
To add an MRGL. goto Media Resources >Media Resource Group List inCisco Unified Communications Manager Administration. In the Media Resource Group List Configuration
window, enter a name for the MRGL and add ihe desired MRGs to the MRCil..
The order of MRGs within an MRCil. specifies the priorities ofthe MRGs. solisting the MRGs in the desired order is important. In the example, hardware conference bridges should be used
before sofiv\are conference bridges. Note Theorder ofMRGs is relevant only ifmultiple MRGs with thesame type ofmedia resources exist In the example, only one MRG includes annunciators and MTPs (SWANNJvlTPMRG) IfCisco Unified Communications Manager searches for an MTP, the firsttwo MRGs
are ignored because they do not include an MTP resource. If a conference resource must be allocated, the two MRGs that include conference bridges are searched in order
5-72
You can assign MRGLs to devices(such as phones, trunks, or gateways) or to device pools. In the example, the previously configured MRGL is assigned lo an IP phone.
Media Resources
5-73
Summary
This topic summarizes the kev points lhat were discussed in ihis lesson.
>til\
Media resources in Cisco Unified Communications Manager are voice termination, audio conference bridge, transcoder, MTP,
annunciator, and MOH.
Only some hardware-based conference bridges support mixed-mode conferences with participants that use different
codecs.
It is possible to configure external conference bridges to enhance the conference bridge capabilities of Cisco Unified Communications Manager.
If the IP Voice Media Streaming App service is activated, the conference bridge needs few additional configuration steps.
Summary (Com
A maximum of 51 unique audio sources counts for a cluster. For a fixed
Hold MOH Audio Source of the device that places the endpoint on hold and the configured MRGL ofthe endpoint that is placed on hold.
The annunciator streams spoken messages and vanous call-progress
To limit media resources access. MRGs and MRGLs must be configured and assigned
References
For additional infonnation. refer to these resources:
Cisco Systems. Inc. Cisco Unified Communications Manager System Guide Release8.0(1).
San Jose. California. February 2010.
http:/;'\\ww.eLsco.com/eii/US/does/'voice_ip eomm/cuem/admin/iS_0_l/ccmsvs/accm-801un.html.
Cisco Systems. Inc. Cisco UnifiedCommunications Manager Features and Services Guide, Re/ease 8.0(1). San Jose. California. April 2010. hitp:'www.cisco.coni/en/US/patlner/d<K's/voicc ip comni/cuem/adtnin/S_0_l/cctnfe;t!/fsg
d-801-cm.html.
Cisco Systems. Inc. Cisco Unified Communications Manager SecurityGuide, Release 8.0(1). San Jose. California, February 2010. littp:.''v\wvv.ci>eo.com/en/LIS/partner/docs/voiY,ejp_comm/cuem/securily/8 0 l/seeugd/se e_801_cm.html. Cisco Svstems. Inc. Cisco Unified Communications System Release 8.x SRND. San Jose. California. April 2010. http:;;vvvvvv.ciscii.com/eii/US/d()cs/voice_ip_c()nim/cuciTi/srnd/8x/uc8.\srnd.pdf.
Media Resources
5-75
5-76
Module Summary
This topic summarizes tlie key point that was discussed in this module.
Module Summary
Cisco Unified Communications Manager supports software media resources, provided by Cisco Unified Communications
This module describes Cisco Unified Communications Manager support for internal and external media resources and their implementation.
References
For additional infonnation. refer to these resources:
hup:/Avwu .cisco.com/en/US/docs/voiccip comtn/euem/drs/8 0 l/drsag801.hliiil. Cisco Systems. Inc. Cisco Unified Communications Manager System Guide Release 8.0(1).
San Jose. California. February 2010.
hitp:.'vvvvvv.cisco.coin/en/|IS/docs/vtsiee_ip_comiTi/CL!Ciii/admin/K_0 l/ccmsyv'accm-SOIem.htmL
Cisco S\ stems. Inc. Cisco Unified Communications Manager Features andServices Guide,
Release 8.0(1). San Jose. California. April 2010.
Cisco Systems. Inc. Cisco Unified Communications Manager Security Guide, Release
8.0(1). San Jose. California. February 2010.
Cisco Systems. Inc. Cisco Unified Communications System ReleaseS.x SRND. San Jose.
California. April 2010.
Imp:.vv\\vv.ciseo.e()ni/,en/US/docs/v(>iee_ip_comm/cttcin/sriKl/8\/uc8\snui.pdf.
2010 Cisco Systems. Inc Media Resources 5-77
5-78
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.
OD
Which ofthese isnot a supported media resource inCisco Unified Communications Manager? (Source: Implementing Media Resources in Cisco Unified Communications
Manager) A) audio conferencing
B) C) D) L) F)
transcoding Media Termination Point annunciator Media Encryption Point Music on Hold
Q2)
Which two media resources use one-way audio only? (Choose two.) (Source: Implementing Media Resources in Cisco Unified Communications Manager)
A) B) C) D) F) audio conferencing transcoding annunciator Media Termination Point Music on Hold
Q3)
Which two statements about conference resources are true?(Choose two.)(Source: Implementing Media Resources in Cisco Unified Communications Manager) A) A software conference bridge can be provided on a CiscoUnified
B) C)
D) Communications Manager server supporting mixed conferences. A hardware conference bridge can supportmixedconferences.
E) Q4)
Which configuration stepis notapplicable when implementing hardware conference bridges? (Source: Implementing Media Resources in Cisco Unified Communications
Manager)
A) B)
Activate the IP Voice Media Streaming Application service. Configure hardware media resources in Cisco Unified Communications
Manager.
C)
D)
05)
B) C)
D)
2010 Cisco Systems. Inc
Meet-Me conferences work only on software conference resources. Meet-Me conferences must be enabled by configuring a Meet-Me number range (pattern). Meet-Me conferences support only G.711.
Media Resources 5-79
Q6)
MOH supports which ofthe following? (Source: Implementing Media Resources in Cisco Unified Communications Manager)
A) B) C) 5 1 fixed audio sources 51 audio source files 50 audio source files and one fixed audio source 50 fixed audio sources and I audio source file
D)
07)
Whicli two extra steps are required when enablingmulticast MOH? (Choose two.) (Source: Implemenling Media Resources in Cisco Unilied Communications Manager)
A)
B)
C)
D) L)
08)
Enable multicast MOH at the device pools used by the IP phones lhat should
be able to listen to multicast MOH.
Which statement about the annunciator media resource is not true? (Source:
fhe annunciator streams spoken messages in order to inform callers about the
call progress.
The annunciator is available onlv as a soil ware media resource,
Q9)
Which ofthe following load-share within their members? (Source: Implementing Media Resources in Cisco Unified Communications Manager)
A)
B) C)
D)
Media Resource Group Lists media resource pools Media Resource Groups
media resource route lists
010)
How can Media Resource Group Lists be applied lo devices? (Source: Implementing Media Resources in Cisco Unified Communications Manager)
A) bv a dev ice pool or at the line with priority to the device pool
B) C) I))
bv a device pool oral the line with priority to the line bv a dev ice pool oral the device wilh priority to the device pool bv a device pool or at the device with priority lo the device
q:i
03)
CE
B.I)
A
04)
Q5)
C C
Q6i
0?)
A. B
13 C
D
08) 09)
QIO)
Media Resources
5-81
Module 6
Implementation
Overview
Cisco Unified Communicalions Manager provides various features and services to support the current needs and demands of both single-site and multisite IP telephony environments.
This module describes how to configure Cisco IP Phone Services and how to implement presence functionality. Cisco Unified Mobility feature is also discussed.
Module Objectives
Upon completing this module, you will be able to implement Cisco Unified Communications Manager features and applications. This ability includes being able to meet these objectives:
Describe and configure Cisco IP Phone Services Describe and configure presence-enabled speed dials and lists
5-2
Lesson 1
implement ihem in Cisco Unitied Communications Manager. The lesson also explains how
administrators andend users can subscribe CiscoIP Phone Services to CiscoUnified IP
phones.
Objectives
Upon completing this lesson, you will be able to describe and configure Cisco IP Phone
Services. This ability includes being able lo meet these objectives:
Describe Cisco IP Phone Services
Cisco IP Phone Senices are applications that use the web client orserver and XML capabilities
ofthe Cisco Unilied IP phone. The Cisco Unified IPphone firmware contains a microhrowser
that enables limited web-browsing capability, liy running directly on the desktop phone of
users, these phone-sen ice applications prov idethe potential forvalue-added services and produclivitv enhancement. (For the purposes of this lesson, the term -phone service" refers to an application that transmits and receives content to and from the Cisco Unified IPphone.)
fhese phones support Cisco IP Phone Sen ices.
Cisco f/nified Wireless IP Phone 7921G
Cisco IP Phone Sen ices can also run on the following Cisco Unified IP phones. However,
these phone models supportonlv le\l-based XML applications.
Cisco Unified IP Phone 7905G Cisco Unified IP Phone 7906G Cisco Unilied IP Phone 7911G
Cisco Unified IP Phones 79I2G and 7912G-A Cisco l'nified Wireless IP Phone 7920 Cisco l'nified IP Phone 6900 Series
All these Cisco Unified IP phones can process alimited set ofXML objects that Cisco has defined for enabling the user interface between the phone and the web server that contains the
running phone senice. Note that these phones support phone services for both Skinny Client
6-5
IP phones need to subscribe to Cisco IP Phone Services to make those services accessible at the phone:
Administrator and end users can configure subscriptions. End user cannot change subscriptions configured by
administrator.
Cannot be changed by end user - Subscribed phone services are explicitly provisioned to the phone, in the phone configuration file
Iheadministrator orend user can subscribe to Cisco IP Phone Services. After subscription, users canaccess these services by pressing the Services. Directories, or Messages buttons by
utilizing the following mechanisms:
fhe listof subscribed Cisco IP Phone Services is part ofthe IPphone configuration tile.
A senice tvpc is present to allow services to be provisioned to the Services. Directories, or
Messages button.
Foreasier access, subscribed Cisco IP Phone Services can also be bound lo phone buttons, "Ihe administrator can also provision services wilh enterprise subscriptions that applv to all
devices and lhal the user cannot override.
Additional Cisco IP Phone Sen ices parameters allow provisioning of applications, such as
Java MIDIet.v thai persist in flash on the phone.
Cisco IP Phone Sen ices can select iv el v be enabled and disabled.
6-6
This topic explains the three methods for Cisco IP Phone Services provisioning in Cisco
Cisco IP Phone Services Provisioning
. External' Phone retrieves the list of configured services by accessing the phone services URLs specified in the phone
URLenterprise parameters.
- Both- Service information received via configuration file is displayed first, followed by service names retrieved by
accessing the phone URL
Several pertinent enterprise parameter relate to Cisco IP Phone Services. In Cisco Unified
Communications Manager, the new Services Provisioning enterprise: parameter affects how sen ices are provisioned with IP phones. The following options can be configured. . Internal: The administrator provisions Cisco IP Phone Services, and the IP phone.receives ts It of configured services from ite configuration file. That file ts downtoaded through TFTP during the phone registration cycle. The Services, Messages, and Dtrec ones URLs that might be specified with the phone URL enterprise parameters are no used. Any v.ahd Java MIDlet services thai are provisioned are installed and are available to run W, hs setting. IP phones no longer need to contact the Cisco IP Phone Services hst URL firs to receive alist of configured services. Instead, the phones can d.rectly access the des.red
sen ice."fhissetting is the default.
. External URL- Cisco IP Phone Services are not provisioned in the configuration file that is obtained via TFTP. The phone uses only the phone services URLs that are specified in the phone URL enterprise parameter. Java MIDlets do not run because they must be
provisioned internallv to install and execute. This behavior is identical to release ot Cisco
Unilied Communications Manager prior to Cisco Unified Communications Manager
Version 7.0.
6-7
The Serv ices Prov isioning enterprise parameter can be set to one ofthe three values that were configuration hie,. Eternal URL fa lis, ofprovisioned phone services is specified in the phone
mentioned prcv iousIv :Internal (a lis, of prov isioned phone services is received u
URL enterprise parameters), or Both. p
On the Phone Configuration page and the Common Phone Profile Configuration page the
no S
HTrTH ' used ' ,S, on T'd ni 'hC ""fi^ **" ofnevl-lower precedence o em example, fDeau1,s the Common Phone Profile Configuration page, the setting
iiS'm,,g Par.etCT Ca" als bC set [ ;i va,ue of M^ ' 'his value Lt cts the
"
The following represent apartial lis. of configuration parameters that are in the Phone I'Rl Urometers seeiion ot the Cisco Unified Communications Manager hnlerprise Parameters
n"hones:']0n "^
authenticate,^ serv ,ce on Cisco Unified Communications Manager. 1 his service provides an authentication proxy service between Cisco Unified IP phones and Cisco Unified C ommunications Manager. The URL is used to validate push requests that the phone services make dircctlv to the phone. The service is configured automatical at installation If no value ,s specified for ihis parameter, phone services cannot push content to the phone.
I'RI. Directories: (Default value is
Directories (or Book icon) button on the phone, fhe URL is automatically configured at installation. II no value is specified for this parameter, the direetorv menu is o, available
when the user presses the Directoriesbutton.
generates and returns the direetorv ment, that is presented when the user pushes the
Imp: -CM IP_addre,s>.X()X(.eemcip/Mi1]direc1or>.isp.)This URL points to the xmldireetory .jsp serv ice on Cisco Unified Communications Manager This serv ice
I Rl- Idle: (Default value is <blank> > This URI , if specified, points to aservice lhat prov ides text or images to be displayed on the phone screen when the phone is idle This
blank (not configured) bv default at installation.
parameter ,s closely coupled with the URL Idle Time parameter. This parameter is left "
URL Idle Time: (Default vulue is 0.) This parameter indicates the lime, in seconds that a phone waits before initiating the URL Idle service. The parameter is set to 0(zero) bv
delaull at installation: this setting indicates that the phone never becomes idle
Imp: <CM IP address--:8()80/ecmcip/GelTeleeasterHelpTexl,isp.) This URL points to the GetTelecasterHelpText.jsp service on Cisco Unified Communications Manager. This service generates and returns on-screen phone help for phone keys and call statistics, when the user presses the Help (i or ?) button to the right ofthe keypad. The URL is configured
automatically at installation. Ifno value is specified for this parameter, no help information
is displaved when the userpushes the Help button.
URL Services: (Default value is
hitp: .<CM_1P address>:8080/eemcip/gelservieesmenu,jsp.) This URL points to the getserviccsmenu.jsp service on Cisco Unified Communications Manager. This service provides a list ofuser-subscribed phone services for the phone, when the user presses the
Senices (orGlobe icon) button. The service is configured automatically at installation. If no value isspecified for this parameter, a list ofsubscribed services isnot provided when
the user presses the Services button.
Cisco
The Services button or a preconfigured phone button can be used to
accessthe Services menu
Phone receives Iss; ot services v.a configuration file o-fromphc-ie service URL
Service .s selected HTT request 5 se^t to
service UR.
Application Server ^
*^^^
Unified Commun.calion
Manager
Cisco
^:<~^_
I'sers have two ways to accessa service from supported phone models. Users can press the Serv ices button or can use a preconfigured phone button. When a user presses the Services
button, the phone eitheruses the configured Cisco IP Phone Services listthat the phone received wilh its configuration lile or uses its HTTP client to load a specific URL lhat contains
a list of serv ices to which the user has subscribed. The user then chooses a service from the
listing. When the user chooses a service, the URL is requested via HTTP anda server provides
the content, which then updates the phone display.
Typical sen ices that might be supplied to a phone include weather information, slockquotes, and news quotes Cisco IP Phone Sen icesare deployed by using the HTTP protocol from
standard web servers Mich as Microsoft Internet Infonnation Services (US).
Users can subscribe onlv to sen ices that are configured through Cisco Unified
Communications Manager Administration.
After the svstem administrator configures the services, users can log in lo the Cisco Unified IP Phone User Options and subscribe lo any sen ice on their phones. Subscriptionsoccur on a perdevice basis.
Users can also subscribe to services by using Cisco L'nified Communications Manager Administrationor by using the Cisco Unified Communicalions Manager Rulk Administration
Tool (HA I).
6-10
UoWOfiJil
, Srka"_[Owl* J.pi^.SeWttS..!
From there, you can add a new Cisco IP Phone Service orreview these preconfigured Cisco IP
Phone Services:
Corporate Directory
Intercom Calls Missed Calls
Personal Directory
Placed Calls Received Calls Voicemail
6-11
The Corporate Directory serv ice configuration includes these parameters: Senice Name: Inter the name of the serv ice as it will display on the menuof available services in Cisco Citified Communications Manager User Options, [nter as many as 32 characters for the serv ice name. Lor Java MIDlet services, the service name must exactly
match the name that is defined in the Java Application Descriptor (JAD) tile.
ASCII Senice Name: Litter the nameofthe service lo display if the phone cannotdisplay
Unicode.
application i-, located. Make sure that this server remains independent ofthe servers in the Cisco Unified Communications Manager cluster.
Senice Category: Select a service application type: XML or .lava MlDiet. Service lype: Select whether tlie service will be provisioned lo the Services. Directories,
or Messages button.
Scnicc \ cndor: I or .lav a MIDlet services, enter the service vendor thai exactly matches
the vendor that is defined in the JAD lile. for XML services, this field can be blank.
Service Version: I his field can be blank for XML and Java MIDlet services. If you enter a
value for a Java MIDlet service, the value must match the version that is defined in the JAD tile. Otherwise, the MIDlet will not install or execute.
6-12
Knable: Check this check box toenable the service, or uncheck the cheek box todisable
the service without deleting it. Default services cannot be deleted. Use this field ifa default
sen ice exists but should not be available for subscription.
the enterprise, without requiring individual subscription. Ifthis option is selected, the
sen ice automatically is provisioned and is notpresented for user subscription.
2010Cisco Systems.Inc
6-13
IS CO
If high-availability of Cisco IP Phone Services is required, the following redundancy options can be
used:
The DNS server can return multiple IP addresses for a single hostname. - IP phones must use DNS.
If high availability of Cisco IP Phone Services is required, options are available to provide
redundancy:
Cisco IOS server load balancing (SI.II): HTTPrequests from IP phones are directed to a virtual IP address that is configured on a Cisco IOS Sener Load Balancer. The requests are
then forwarded to the real IP addresses ofthe web servers that host the Cisco IP Phone
Services, lo avoid making the Cisco IOS Server Load Balancer a single pointof failure. Cisco [OS redundancy options such as Hot Standby Router Protocol (IISRP) should also be
implemented.
Using Domain Name System (DNS) as a redundancy mechanism: fhe URLs for Cisco
IP Phone Serv icesthat are configured on Cisco Unified Communicalions Manager use hostnames instead of IP addresses. The DNS server that is responsible for hostname resolution is contigured to return multiple IP addresses for a given hostname, fhis redundancy method requires D\S support on the IP phones.
Note Another option to provide redundancy is an environment with a Network Address Translation
6-14
Cisco IOS SLB This figure shows a Cisco IOS SLB environment.
E3
CUCM-1
Real IP 1011 1
Virtual IP
101 5 1
Manager Systems
CUCM-2
HeaMP 10 1 1 2
When implementing SLB toprovide Cisco IP Phone Services redundancy, the Service URL
parameter ofaCisco IP Phone Service points to avirtual IP address that isconfigured on the
Cisco IOS Server Load Balancer. The Cisco IOS Server Load Balancer then forwards HTTP
requests that itreceives on these virtual IP addresses tospecific real IP addresses ofmultiple
web servers, thus providing redundancy.
6-15
CUCM-1
Manager Systems
CUCM-2
When you use DNS to implement Cisco IP Phone Services redundancy, the Service URI parameter of Cisco IP Phone Serv ices points to a hostname that one or more DNS servers will
resolve. "fhi> DNS server is configured so that a single hoslname refers lo mulfiple IP
addresses, thus prov iding redundancy.
Step 1
Step 2
Step3
6-17
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Step 1
Before adding a new Cisco IP Phone Service, verify and. if necessary, change the
relevant enterprise parameters:
Sen ices Provisioning: This new device-configuration parameter controls whether the phone uses the servicesthat are provisioned in the configuration file (inlemal). the services lhat are received from URI s (Lxternal URLs), or both. Ihis parameter is required for backward-compatibility wilh third-partv provisioning servers, primarily to disablethe new provisioning mechanism so that the phone presents only services from Ihe Service URL parameter. I RL Authentication: This parameter specifies a URL that points to a web page in one ofthe Cisco CallManager Cisco IP Phone (CCMCIP) web services in the cluster. Ihis URL provides an authentication proxy service between Cisco Unified IP phones and the l ightweight Directory Access Protocol (LDAP) director.. This URL is used to validate requests that are made directly to the phone. This URL is automatically configured at installation. If the 1 Rl. is removed, the push capabilities to the Cisco Unified IP phones are disabled.
I'RI. Directories: '['his parameter specifies the URL lhal Cisco Unified IP phones use when users press the Director;' button. This URL must return a CiseollThoneMenu object even if no Meinillems are specified in the object. Tlie Menultems that are specified and the inlemal directories are appended to the directory list on the Cisco Unitied IP phones.
I RL Idle: "Ihis parameter specifies the URL lhat a Cisco Unified IP phone uses to display information on the screen when the phone remains idle lor the time that the URL Idle Time parameter specifies.
phones will remain idle before displaying the URL that the URL Idle parameter specifies. Ifthe time is set to 0(zero), the URL that the URL Idle parameter
specifies is notdisplayed.
I'RL Idle Time: This parameter specifies the time that the Cisco Unified IP
URL, Information: This parameter specifics a URL that points to a page in the
CCMC1P web service and returns the requested help text tothe Cisco Unified IP
phone display. This information is displayed when auser presses the ior ?
button on the phone.
phones should call when users press the Messages button. When called, the URL
must return a CiscoIPPhoneMenu object. The returned Menultems are appended
lo the built-in items on Cisco Unified IP phones.
URL Messages: This parameter specifies a URL that the Cisco Unified IP
address and port; for example, proxy.cisco.com:8080. Ifa proxy server is specified, the Cisco Unified IP phones use that server torequest all URLs. Leave this setting blank toinstruct the phones toattempt toconnect directly to
all URLs. If a server name is used insteadofan IP address, configure phones with valid DNS servers,to allow name-to-IP resolution. Confirm that the proxyserver is listening at the specified destination.
URL Services: This parameter specifies the URL that a Cisco Unified IP phone calls when a user presses the Services button. The initial request by the phone
passes the device name as a parameter. The default page in the CCMCIP web
Secured Authentication URL: This parameter specifies the URL that points to
a web page inone ofthe CCMCIP web services inthe cluster. This URL provides an authentication proxy service between secured Cisco Unified IP
phones and the LDAP directory. This URL isused lovalidate requests that are made directly tothe phone. This URL isconfigured automatically at installation.
If the URL is removed, the push capabilities to the CiscoUnified IP phones are
disabled.
Secured Idle URL: This parameter specifies the URL that a secured Cisco
Unified IPphone uses todisplay information onthe screen when the phone
remains idle for the time that the URL Idle Time parameter specifies.
page inthe CCMCIP web service and returns the requested help text to the
secured Cisco Unified IP phone display. This information displays when a user pressesthe i or ? button on the phone.
Secured Messages URL: This parameter specifies a URL that thesecured Cisco IP Unified phones should call when users press the Messages button.
When called, the URL must return a CiscoIPPhoneMenu object. The returned
Secured Services URL: Thisparameter specifies the URL that a secured Cisco
Unified IPphone calls when a user presses the Services button, 'fhe initial request by the phone passes the device name as a parameter. The default page in
the CCMCIP web service returns a CiscoIPPhoneMenu object that includes a list ofthe services that are subscribed to the device. If no subscriptions exist, the return text indicates that no subscriptions exist for the device.
)2010 Cisco Systems. Inc. Feature andApplication Implementation
11*Photo SCrvtce
ft - * / 7)
Step 2
In Device "~- Device Settings^- Phone Services, click the Add New butt*
6-20
1
Check Enable to enable the service. Service Name;
\
A (meaningful) name for the service
Name for ASCI l-only phone displays
What the service does Where the service can be found UsedforsecureURLs
Check Enterprise
Service Description:
Service URL
Subscription to
auto sub scribe this
service to devices.
Secure-Service URL:
Step 3
Parameter
Define the required Cisco IP Phone Services parameters and click Save to complete
configuration.
Description
Service Name
Enter the name of the service. Ifthe service is not marked as an enterprise
subscription, the servicename will be displayed inareas inwhich youcan subscribe to a service; for example, under Cisco Unified Communications Manager User Options. Enter as many as 32 characters for the service name. For Java MIDlet services, the service name must match the name that is defined in
the JAD file.
ASCII Service
Name Service
Enter the name of the service to display if the phone cannot display Unicode.
Description
Enter a description of the content that the service provides. The descriptioncan include as many as 50 characters inany languagebutcannot include quotation
marks (") or apostrophes (').
6-21
Description
Enter the URL ofthe server on which the Cisco IPPhone Services application is
located. Make sure that this server remains independent of the servers inthe Cisco
For tne services to be available, the phones in the Cisco Unified Communications Manager cluster must have network connectivity to the server. For Java MIDIets that are signed by Cisco, enter the location where the JAD file
Fordefault servicesthat Cisco provides, the Service URL parameter is enteredas Application Cisco/name of servicebydefault; forexample, Applicatton:Cisco/CorporateDirectory Ifyou modify the Service URL parameter for these default services, verify thatyou configured Both for theServices Provisioning
setting in the Phone, Enterprise Parameter, and Common Phone Profile
Enter the secure URL of the server on which the Cisco IP Phone Services
Service URL
application is located. Make sure that this server remains independent of the
servers inthe Cisco Unified Communications Manager cluster. Do notspecify a Cisco Unified Communications Manager server or any server that is associated with Cisco Unified Communications Manager (suchas a TFTP server or publisher
database server).
For the services to be available, the phones in the Cisco Unified Communications
Manager cluster must have network connectivity to the server. Note: Ifyou do not enter a Secure-Service URL parameter, the device uses the
Service URL parameter Ifyouenter both a Secure-Service URL parameter and a Service URL parameter, the device chooses the appropriate URL, based on its
capabilities
Service
Category
Service Type
Choose whether the service is provisioned to the Services, Directories, or Messages button or option on the phone, that is, ifthe phone has these buttons o options. To determine whether a phone has these buttons or options, refer to the Cisco Unified IPPhone Administration Guide that supports the phone model
This field allows you to specify the vendor or manufacturer for the service This field is optional for XML applications but is required for Java MIDIets that are signed by Cisco. For such Java MIDIets, the value that you enter in this field must
match the vendor that is defined in the MIDlet JAD file.
Service Vendor
This field displays as blank for default services that Cisco provides.
You can enter as many as 64 characters.
6-22
Parameter
Description
Enter the version number for the application.
^^
Service
Version
For XML applications, this field isoptional and isinformational only. For Java MIDIets thataresigned by Cisco, consider thefollowing information: If you enter a version, the service version must match the version that is defined inthe JAD file. If you entera version that is different from the version thatis installed on the phone, the phone attempts lo upgrade or downgrade
the MIDlet if the version.
If the field is blank, the version is retrieved from the ServiceURL. Leaving the
field blankensures that the phone attempts to downloadthe JADfile every
the phone always runs themost recent version oftheJava MIDlet, without the
Service Version field being updated manually.
time that the phone reregistersto CiscoUnified Communications Manager, as well as every time thatthe Java MIDlet is launched. This action ensures that
Thisfield displays as blank fordefault servicesthat Ciscoprovides. You can enter numbers and periods inthisfield (as many as 16 ASCII characters).
Enable
This check box allows you toenableor disable the service, without removing the
Unchecking thecheck box removes theservice from the phone configuration file
Enterprise Subscription
This check boxallows you to automatically provision the serviceto alldevices in the cluster thatcan support theservice. If you checkthischeckbox, you (or an end
user) cannot subscribe to the service. Ifthis check box is unchecked, you must manually subscribe to the service for it to
Tip: This setting displays only when you configure a service for thefirst time. After
yousave the service, the check boxis not displayed inthe window. Toidentify whether the service is provisioned to all devices inthe cluster thatcan support the service, goto the Find and List IP Phone Services window and display
the services. IfTrue is displayed in the Enterprise Subscription column,you cannot
This pane lists the service parameters thatapply tothis Cisco IP Phone Service.
Use the following buttons to configure service parameters for this pane: New Parameter: Clickthis button to display the Configure Cisco Unified IP Phone Service Parameter window, in which you can configure a new service parameter for this Cisco IP Phone Service. Edit Parameter: Choose a service parameter that is displayed in the
Parameters pane. Then,click this button to display the Configure Cisco Unified
IP Phone Service Parameter window, in which you can edit the selected service parameter for this Cisco IP Phone Service. Delete Parameter: Choose a service parameter that is displayed in the
ManagerAdministration web page Bythe end user, via the user web page
lo use Cisco IP Phone Services, vou need to subscribe tlie configured services to Cisco Unified IP phones. You can configure a Cisco IP Phone Services subscription via the Cisco Unified Communicalions ManagerAdministration web page, or the end usercandirectly configure the subscription on the Cisco Unified Communications Manager User web page.
5-24
To subscribe to a phone service, open the phone configuration web page for the phone that should have a service subscription. Choose Subscribe/Unsubscribe Services from the Related
Links drop-down list.
Step 1
Step 2
6-25
Administrator (Cont)
- Subicrrbed Servires -
Substr.be
Baik
Step 3
Click Subscribe to add the selected serviceto the service list for this phone.
6-26
to a phone service.
SL.-.
y> ^-1
** 5*-fll--
La-j ~
End users can configure phone service subscriptions by logging into the Cisco Unified Communications Manager User Options web page. End users should then follow this
procedure:
Step 1 Step 2
Open the Cisco Unified Communications Manager User Options web page at
https:/AS'm'er //Vccmuser.
From the User Options menu choose Device, and then click the Phone Services
button.
6-27
to a phone service.
PfciM5lWW*5**a1p8MElflqwto*i
1,.....
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Step 3
Step 4
6-28
subscription.
^^
Step 5
After the Cisco IP Phone Services subscription has been completed, the new Cisco IP Phone Service will show up on the Cisco Unified IP phonewhen the user presses the Services button.
6-29
Summary
This topic summari/es the kev points lhat were discussed in this lesson.
Cisco Unified IP phones can use Cisco IP Phone Services for a variety of functions. Cisco IP Phone Services redundancy can be provided via
DNS or via Cisco IOS SLB.
Cisco IP Phone Services are added and updated via the Cisco Unified Communications ManagerAdministration web
page.
References
For additional information, refer to these resources:
Cisco Svstems. Inc. Cisco I nifiedCommunications System Release S.x SRXD. San Jose.
California. April 2010.
Imp: www.cKco.com en t S doc- voice_ip_comm.vuc,m,srnd.',8\/iteS\srrid.pdf. Cisco Svstems. Inc. Cisco Unified Communications Manager Administration Guide.
Release 8.0(2) San Jose. California. March 2010.
hup: www.ci-eo.cotu eiv'l. S Joes voice ip eomnv'cucni/admin/K 0 "/eenief;:/becni.pdf. Cisco Svstems. Inc. Cisco Unified ('ommunications Manager Features and Services Guide.
Release 8.0/2/. San Jose. California. March 2010.
6-30
Lesson 2
Cisco Unified Communications Manager presence, an integrated part of Cisco Unified Communications Manager, allows IPphone users to monitor thestatus ofdirectory numbers.
This lesson describes how Cisco Unified Communications Manager native presence works and how it is configured.
Objectives
Upon completing this lesson, you will be able todescribe and configure presence-enabled
speed dials and lists. This ability includes being able to meet these objectives:
Presence policy
* Cisco Unified Presence: User status information
Cisco Unified Communications includes multiple options to integrate presence information. Cisco Unified Communications Manager presence, a nativepresence feature, includes the
following capabilities:
Presence-enabled speed dials: Speed-dial buttons that indicate the status ofthe targel of
the speed dial
Presence-enabled call and directory lists: Call lists and directory entries that indicate the
status of each list entry
When vou use Cisco f inilied Presence, many features arc added to those thai the Cisco Unilied Communications Manager nativepresence feature provides, including these:
Standards-based Session Initiation Protocol (SIP) and SIP for instant Messaging and
Presence Leveraging Extensions (S1MPLF) network interface
Cisco (Unified Personal Communicator, a client tool lhat integrates voice, video, and 1M
communications
Note
This lesson discusses Cisco Unified Communications Manager native presence only
Allows an interested party (a watcher) to monitor the real time status of a directory number (a presence entity)
Watcher subscribes to status information ofthe presence entity
- Entity is registeredoff-hook
Cisco Unified Communications ManagernativelysupportsCisco Unified Communications Manager presence, so no additional products or servers arerequired. Tliefeature allows an interested partythe watcher (subscriber)to monitor the real-time statusof a directory number, a presence entity, or a subscribee.
A watcher subscribes to the status information of one or more presence entities. The watcher can view the status infonnation of a presence entity by using presence-enabled speed dials or
presence-enabled lists: public directory lists and call lists such as placed, received, or missed
calls.
6-33
Messenger
Manua.ly ovemdes presence stalus (Available. Busy Do Not Disturb) Manages 'he contact list from phone and user pages
A*-, i SOA^i
Ihe Cisco iP Phone Messenger application serves as a protocol translator between 11TTP and SIP messaging. Cisco IP Phone Messenger uses XMI over HTTP to communicate with the Cisco Unified IP phones: it uses SIP to communicate with the Cisco Session Initiation Protocol (SIP) Pro\v Server (which also functions as a registrar server). Cisco IP Phone Messenger can
distinguish between two devices that have the same directory number but that are in different partitions. The application can also function when the user is logged in via Cisco Intension Mobility. However, the application does rely on the availability ofthe Cisco Unilied Presence
publisher for new user logins.
The Cisco IP Phone Messenger application prov ides the following presence functionality:
Shows aggregated presence status of other users
Supports manual override of ihe presence status (Available. Busy, or Do Not Disturb)
Updates user presence status in Cisco IP Phone Messenger
Manages the contact list from both the phone and the web user interface
&/
'
In the example. John's phone subscribes tothe status of Bryan's phone. (More precisely, the phone subscribes tothe status ofthe directory number that belongs to Bryan.) The subscription
occurs either becausethe Cisco Unified Communications Manager administrator configured a
presence-enabled speed dial for Bryan's extension, orbecause John is browsing through acall
listthat includes Bryan's directory number. When thecall listis viewed, John'sphone automatically subscribes to thestatus ofthe other entries in the list. Cisco Unified Communications Manager presence now keeps John'sphone updated about the status ofthe subscribed presence entity. If Bryan goes off-hook while John is browsing thecall list that includes Bryan's directory number, the status information isdisplayed. IfJohn has a presence-enabled speed dial for Bryan's directory number, the speed dial permanently displays the statusof Bryan's directory number.
6-35
Cisco L'nified Communications Manager presence allows CiscoUnified IP phones and hv SIP devices to watch director} numbers, through a SIP trunk. Endpoints lhatcan be reached
through a SIP trunk can be watched bv Cisco Unified IPphones and by SIP devices, through
other trunks. Cisco L'nified IP phones that run Skinny Client Control Protocol (SCCP)and Cisco Unilied IP phones that run SIP can watchpresence entities and can be watched. If presence subscriptions are sent over a SIP trunk. Cisco LJnified Communications Manager takes care of protocol conversion between SCCP and SIP. If only IP phones lhalare registered within
the Cisco Unitied Communications Manager cluster arc involved, there isno need for endpoinlto-endpoint communication; Cisco L'nified Communications Manager is aware ofthe state of
all registered IP phones.
Note
When watching the presence status of an entity through a SIPtrunk, some digit
manipulation features that apply to call routing do not apply to presence subscriptions. For example, significant digits configured at the SIP trunk do not apply to presence
subscriptions
Cisco Unified IP phones can display the status information (unknown, on-hook, or off-hook) of
presence entities by using presence-enabled speed dials orcall and directory list entries. Presence-enabled speed dials show a symbol inthescreen ofthe IPphone, at theappropriate speed-dial button. Some phone models (Type-B Cisco Unified IPphones) have an LED inside the speed-dial button and indicate the status by using red(off-hook) or green (on-hook) lights.
When users browse through a directory or calllist, eachentry displays a symbol that indicates
its status.
Cisco
Unified IP Phone
Models All modem Cisco Unified
IP phone models
Yes
Yes
79l4Exparision Module,
Cisco Unified IP Phones 7940Gand7960G Yes No
(SCCP)
Cisco Unified IP Phone
(SIP)
As shown in the tabic in the ligure. Cisco Unified IP Phone 7914 Expansion Module and Cisco Unilied IP Phones 7940G and 7960G do not supportpresence at all when running SIP. When running SCCP. thev support onlv presence-enabled speed dials but no presence-enabled call and direetorv lists. Tvpc-B Cisco Unified IP Phones 7941. 7942. 7945. 7961. 7962. 7965. 7970. 7971. and 7975. 8900. and 9900 Scries support both presence-enabled call and directory li^ts and presence-enabled speed dials, regardless ofthe protocol (SIP or SCCP).
Note Cisco IP Communicator also supports both presence-enabled speed dials and presenceenabled call and directory lists
Presence-enabled speed dials are configured statically by the Cisco Unified Communications Manager administrator andcannot be configured or modified by a user. In this way. the administrator has control overthemonitored presence entities for each watcher. However, partitions andsubscribe catling search spaces (CSSs) alsoapply to
presence-enabled speed dials.
Access control forpresence-enabled callanddirectory listscan be provided by partitions andsubscribe CSSs. and by presence groups. Each ofthe two methods can be used independently of each other. If both are used, bothmust permita subscription for
successful watching ofthe presence-entity status.
Like with traditional CSSs. a presence entity can be watched only ifthe watcher has the presence entity partition in its
subscribe CSS.
The (standard) partition that is applied to a line or a route pattern that refers to a trunk is used for both calling privileges
and presence,
Calling priv ileges are implemented bv using partitions and CSSs. Presence policies are implemented bv tiding the same partitions (applied lo directory numbers and roule patterns) thai are used for calling-priv ilege configuration. The CSSs. however, are separated. Rather than the
(standard) CSS configured on IP phones. lines, and trunks, dedicated subscribe CSSs are used.
A subscribe CSS is applied to a watcher, 'fhis watcher can be a SIP trunk (assuming that subscriptions have been enabled, in general, on the trunk), a phone, or an end user. Subscribe CSSs do not use the concept of a device CSS and a line CSS. Watching a presence entity is always a global function ofthe IP phone, not of a certain line. Therefore, subscribe CSSs are applied to IP phones, not to lines. When a subscribe CSS is applied to an end user, this subscribe CSS is used in case of Cisco Intension Mobility or if the end user is associated with
a dev ice.
Like standard CSSs. the subscribe CSS determines which presence entities a watcher is allowed to monitor. A subscription is permitted only if the watcher has the partition of the desired presence entity in its subscribe CSS.
The (standard) partition that is applied lo a line or a route pattern lhal refers to a SIP trunk is used for both calling privileges and presence policies. If no partition is applied to the desired presence entity, the presence entity is available to all watchers.
6-40
- Route patterns
- Anychanges to partition configuration affect calling privileges (standard CSSs) and presence policies
(subscribe CSSs).
Design and implementation of calling privileges and presence policies must be performed together.
Presence policies andcalling privileges sharea configuration element. The partitions that arc applied to lines or route patterns apply to thepolicies and privileges. Therefore, implemenling
presence policiesaffectsexistingcallingprivileges, and vice versa.
Whenever partition configuration is changed because of the implementation ofoneofthe two features (calling privileges or presence policies), the other feature is affected. Therefore, calling
priv ileges and presence policiesmustbe designed and implemented together.
6-41
Baseline configuration does not include any partitions (no calling privileges and no presence policies are in place). If partitions and (standard) CSSs are implemented for calling privileges, subscriptions will fail. Lines and route patterns now have partitions.
Devices (phones and trunks) do not have subscribe
CSSs.
Lines and route patterns now have partitions. Devices (phones, lines, and trunks) do not have CSSs.
In the example scenario, the baseline configuration does not include any partitions or CSSs. Neither are any callingprivileges or presence policiesin place. All directory numbers and route patterns are in the null partition and can be accessed by all devices. All devices can place calls to all destinations. Presence subscriptions are also possible to all supported targets, such as directory numbers and devices that arc reached through SIP trunks. If callingprivileges (partitions and CSSs)are implemented withoutconsidering presence (adding subscribe CSSs). presence subscriplions will stop working for all presence entities lhat were put into partitions when calling privileges were implemented.
Note The recommendation when implementing partitions and CSSs is not to leave any targets in the null partition, but lo assign a partition to all call deslinations Therefore, in the sample scenario, usually no targets are left where subscriptions still work.
Or. if the baseline configuration is modified so that presence policies (partitions and subscribe CSSs) are implemented without considering calling privileges (adding standard CSSs). all calls will fail. Lines and route patlerns now have partitions, but calling devices do not have CSSs that allow access to some partitions, 'fhe devices have only subscribe CSSs. so only presence information can be obtained: no calls can be placed.
6-42
C-1: P-1.P-2
(CSS)
1003
-StP
-**:-#
Phone3
Effective Permissions:
Phonel to 1002: Permitted Phonel to 1003: Denied Phone2 to 1001: Permitted Phone2 to 1003: Permitted Phone3 to 1001; Permitted Phone3to1002: Denied
The configuration consists ofthree CSSs: C-1, C-2, and C-3. C-1 contains partitions P-l and P2. C-2 contains partitions P-l. P-2, and P-3. C-3 contains partition P-l only. Phonel has partition P-l applied toits line, which is configured with directory number 1001.
CSS C-1 is assigned to Phonel.
Phone2 has partition P-2 applied toits line, which isconfigured with directory number 1002.
CSS C-2 is assigned to Phone2.
Phone3 isa SIP phone with directory number 1003 and can bereached through a SIP trunk. The corresponding route pattern 8.1003 isin partition P-3. CSS C-3 isassigned tothe SIP
trunk.
The effective permissions for presence subscriptions are asfollows: Phone 1is allowed towatch
the status of 1002 butnot of 1003. Phonc2 is allowed to watch both 1001 and 1003. Phone3 is allowed to subscribe to presenceinfonnation of 1001 but not of 1002.
Note
In the figure, "(CSS)" refers tothe standard CSSsthatare used for the implementation of calling privileges. These CSSsare notrelevant to the discussion ofpresence subscription permissions. However, because they alsodepend ontheconfigured partitions, theCSSs are added to the figure to illustrate that theymust be considered inthe overall configuration.
Note
6-43
Presence Groups
Presence policies can be implemented by partitions and subscribe CSSs or by presence groups. This subtopic describes how presence policies are implemented by using presence groups.
Presence Groups
Presence groups can be used to implement presence
policies:
Watchersand presence entitiesare put into presence groups. Subscnptions are permitted within presence groups. Subscrptionscan be allowed or denied between presence
groups
IP phones have separate presence groups: Line presence group (presence entity) Phone preserce group (watcher) SIP trunks have only one presence group: Used for both watcher and presence entity Presence groups apply only to presence-en a bled call lists, not to
presence-enabled speed dials
When implementing presence policies, watchers and presence entities areputinto presence groups. Subscriptions can beallowed or denied at an intergroup level; within a presence group, subscriptions are always permitted (unless Ihcy aredenied by partitions andsubscribe CSSs). IP phones arecontigured with two or more presence groups. One presence group is applied to the device (in the role ofa watcher), and each line can be configured with a presence group in
its role as a presence entity.
On SIPtrunks, only one presence group is configured. This group is used in both roles: walcher and presence entity . You cannotassigna presencegroup lo a route pattern. Like subscribe CSSs. presence groups can alsobe assigned lo end users, fhe groups are used when the end users log into the phone by using Cisco Extension Mobility or when the users are
associated with a dev ice.
Note
The configuration uses three presence groups: G-l. G-2. and G-3. Interpresence group subscriptions arc permitted from G-2 to G-3 and from G-3 to G-l. All other interpresence
group subscriptions aredenied.
Phonel has presence group G-l applied to its line, which is configured with directory number
1001. Presence group G-2 is assigned to Phonel.
Phone2 has presence group G-2 applied to its line, which is configured with directory' number
1002. Presence group G-2 is alsoassigned to Phone2.
Phone3. a SIP phone with number 1003, can be reached through a SIP trunk. Presence group G3 is assigned to the SIP trunk.
fhe effective pennissions for presence subscriptions are as follows: Phone 1is allowed to watch
the status of 1002 and 1003. Phone2 isallowed towatch 1003 but not 1001. Phone3 isallowed
to subscribe to the presence information of 1001 but not of 1002. Note Presence groups apply only to presence-enabled call lists. Presence groups donot affect
presence -enabledspeed dials.
6-45
Ihis subtopic describes the interaction ofpresence groups, subscribe CSSs. and partitions.
ion of Pres
* Both must permit subscription, for successful watching Provides two levelsof hierarchyuseful in larger deployments.
* Example
Requirements
Use one presence group per department - Deny interpresence group subscriptions.
Include manager partition in the subscribe CSS ofthe assistant only
F.ach feature can be used standalone, or the features can be combined. If both uses are
implemented, then both mechanisms must permit the subscription to allow successful watching.
Combining both presence-policy mechanisms provides two hierarchy levels, which are useful
in larger deployments or complex scenarios.
The following example illustrates how subscribe CSSs and partitions and presence groups can
be effectively combined to fulfill the given requirements,
Solution: One presence group perdepartment is configured. Interpresence group subscriptions are denied by setting the default interpresence group policy accordingly. One
partition per manager is configured, hach partition is listed only in the subscribe CSS ofthe
respective manager assistant.
In the example, presence groups arc used for the tirsl level ofpresence policies (at department level). Subscribe CSSs and partitions are used for additional access control within a department
(or presence group).
Note
2. Apply phone button templates to phones. 3 Configure presence-enabled speed-dial buttons. 4 Apply subscribe CSSs to phones.
'lite Cisco Unified Communications Manager presence configuration procedure includes these
three tasks:
Step 1
Step 2 Step3
Step 4
Caution
Enable presence-enabled call lists: Enable the BLF for Call Lists enterprise parameter.
Note In CiscoUnified Communications Manager configuration, presence-enabled call lists are referred to as busy lamp field (BLF) call lists.
6-47
Ihis subtopic shows how to implement presence-enabled speed dials. The first step is the
Configure presence-enabled
speed-dial buttons in phone button
template.
B.H.O If..
The first step in implementing presence-enabled speed dials is to configure aphone button template that includes presence-enabled speed dials. To configure a phone button lemplate. go to Device >Device Settings >Phone Button lemplate and either add anew template orcopy
a default phone button template and save it with a new name. Configure the phone button
template with the desired number of presence-enabled speed dials.
Note
6-48
ilSTLjJtfllll
Assign the previously configured phone button template to the IP phone that should be configured for presence-enabled speed dials. Go to the Phone Configuration page and select the appropriate template from the Phone Button Template drop-down list.
6-49
displayed on phone.
W1W fiMI StSTi Dbl (MOpirau SPW)4t55lE6
U- &<- . ?~
-..u.
./ /
/i
&*
-.,1 ,...
wd m.i *,,
,.*.--
'
""""""""'~.:
",
""
After applv ing the new phone button template, the presence-enabled speed dials are displayed in the Association Infonnation area ofthe Phone Configuration window, 'fhephone can now use buttons for presence-enabled speed dials, but the buttons must be configured appropriatelv.
To configure presence-enabled speed dials, click the Add a New BIJ-'-SD link. The Busv
I amp Field Speed Dial Configuration window appears. In Ihis window, configure the larget (the presence entity to be watched) ofthe presence-enabled speed-dial button, as well as a label lhatwill be displaced on the phone screen next to the corresponding button.
Note In Cisco Unified Communications Manager configuration, presence-enabled speed dials are
referred to as BLF speed dials
6-50
paranwUrVvlDt
CS3(M*dencB 3! DSC
SCCP
.-Kc-T C-^^/Jii'ar or *
If call lists should also provide presenceinformation, the appropriate enterprise parameter must be enabled, as shown in tlie figure. After changing the BLF for Call Lists enterprise parameter to Enabled, you must reset all phonesthat supportpresence-enabled call lists, for the changeto
become effective.
Note
6-51
nq Pn
System > Security Profile > SIP Trunk Security Profile
SIP tun* S*w*y pioMb CorJWHMitw
k Snunn Proftle JBfofaum
E^L
if presence subscriplions arc possible over a SIP trunk, presence needs to be enabled on the SIP trunk. Presence is not enabled direct]; at the SIP trunk but via a SIP trunk sccurit)' profile. To configure a SIP trunk securitv profile, go to System > Security Profile > SIP Trunk Security Profile and verifv that the Accept Presence Subscriplions and the Accept Unsolicited Notification check boxes are checked. Then, apply the SIP Trunk Security Profile to the SIP
trunk, as shown in the figure.
6-52
Tins topic describes how to implement presence policies in Cisco Unified Communications
Manager.
1 Configure partitions and CSSs. 2 Assign partitions to lines and route patterns.
Step 3 Assign subscribe CSSs to phones and trunks. Implement presence policies that arebased onpresence groups:
Step 1 Configure presence groups.
Step2
Step 3
Note
presence-enabled call lists and subscribe CSSs. Partitions apply to presence-enabled call
lists and presence-enabled speed dials.
6-53
The first two steps of implementing presence policies that are based on partitions and subscribe
The figure shows how CSSs are assigned to IP phones and SIP trunks as SUIiSCRIBl: Calling
Search Spaces.
6-54
Presence groups apply only to presence-enabled call lists and are ignored by presence-enabled speed dials. The first step when implementing presence groups is to add and configure them, as
shown in the figure.
Individually configure
Permission to unlisted
presence groups is
determined by service
parameter.
Presence groups can beadded and configured under System > Presence Group. One presence group, the standard presence group, exists by default and cannot be deleted. All phones, lines, and SIP trunks are. by defaultmembers ofthe standard presence group. The standard presence group can be modified to setpermissions to other groups butcannot be deleted. When adding a new presence group, enter a name and description and configure the permission for subscriptions toward other presence groups. The permission toward all other (unconfigured) groups does not need tobe entered. The permission for subscriptions towards unconfigured presence groups is determined by system default, which is configurable as a Cisco CallManager
sen ice parameter. Note Subscription permissions are configured in a unidirectional manner, between pairs of presence groups. Youcan permitsubscriptions fromone group to another but to deny
subscriptions in the opposite direction.
6-55
Dirtllrj* 5yli5Ci"i^llcn
The Default Inter-Presenee Group Subscription service parameter specifies the svslem default for presence subscriptions, fhis default is applied forsubscriplions toward presence groups for which no explicitpermission is set in the configuration ofthe presence group from which ihe
subscription request is sourced.
fhe Default Inter-Presenee Ciroup Subscription parameter is a service parameter ofthe Cisco
CallManager serv ice and so is configured under System > Service Parameter.
6-56
groupto adirectory
number (in
presence-entity role).
Presence groups allow the implementation ofpresence policies by checking the permission for subscriptions that go from one presence group to another presence group. This means that each
subscriber and each presence entity must be in a presence group.
IPphones (and their lines) act asboth: The IP phone generates subscriptions (when using presence-enabled speed dials orpresence-enabled call lists) and their directory numbers can be watched by other subscribers. Therefore, presence groups are applied to both the phone (in the
role of subscriber) and all phone lines (in the role of presence entity).
Note
Bydefault, all phones and all lines are in the Standard Presence group.
Note
Presence groups apply to presence-enabled call lists only. Therefore, subscriptions that presence-enabled speed dials cause ignore all presence group-based policies.
6-57
Fru!
The presence group configured on a SIP trunk applies to both subscriptions being sent out and subscriptions being received on the trunk.
Device > Trunk
Tnesame presence
group is used in the
subscriber and
presence-entity rotes
Cisco Unified Communications Manager can send out subscribe messages on a SIP trunk (when watching a presence entitv on the other side ofthe trunk) and can receive subscriptions on a SIP tntnk (when a local directory number is walched over the SIP trunk by a subscriber on ihe other side of tlie trunk). The taink. iherefore. can act in both the subscriber and presence entitv roles. However, on a SIP trunk, only one presence group can be configured. Iherefore, this single presence group applies to both sent and received subscriplions.
Summary
This topic summarizes the key points lhat were discussed in this lesson.
Summary
Native Cisco Unified Communications Manager presence
allows lines or endpoints that are reachable through SIP trunks to be monitored for their status (on-hook versus offhook).
Most IP phones support presence-enabled speed dials; TypeB Cisco Unified IP phones using SIP also support presenceenabled call and directory lists.
Cisco Unified Presence policies can be applied to control presence subscriptions. Cisco Unified Communications Manager native presence policy configuration includes implementing partitions and subscribe CSSs and presence
g roups.
configuration includes implementing presence-enabled speed dials and enabling presence-enabled calland directory lists.
References
For additional information, refer lo these resources:
Cisco Systems. Inc. Cisco Unified Communications Manager Features and Services Guide,
Release 8.0(2). San Jose. California. March 2010.
littp:,',vvwvv.cise{>xx)m/en.TUS/docs/v()iccjp_comm/cucm/adniin/X_0_2/ecmfeat/lsgd.pdf.
Cisco Systems. Inc. Cisco Unified Communications Manager Administration Guide,
Release 8.0(2). San Jose. California. March 2010.
hitp:;/www.cisco.coni/ct^'US/docs/voice ip_conim/cucin/admin/8_0_2/ecmefg.;bccni.pd!'.
Cisco Sy stems. Inc. Cisco Unified Communications System Release 8.xSRND. SanJose.
California. April 2010.
htlp:';vvvvw.cisco.com 'en/lJS/does/voice_ip_comm/cucm/srnd/8x/uc8\smd.pdf.
6-60
Lesson 3
CiscoUnified Mobility allows usersto be reachable at a single number, regardless ofthe device they use. This lesson describes the features of Cisco Unified Mobility, as well as how these
features work and how to configure them.
Objectives
Uponcompleting this lesson,you will be able lo describe and configure Cisco Unified Mobility. This abilityincludes being able to meet theseobjectives:
Describe thepurpose of Cisco Unified Mobility, howit works, and when to use it
Analyze call flows that involve Cisco Unified Mobility List the requirements for implementing and installing Cisco Unified Mobility
Describe considerations when using Cisco Unified Mobility MVA Describe how lo configure Cisco Unified Mobility
:ssco Umfie'
With Mobile Connect, calls placed to office phones ring the office
phones anc associated remote phone.
MVA allows users to call into the enterprise from any phone and
place outgoing calls that appear to come from the office phone.
Cisco unified Commumcalioris
MVA establishes a
system to create
Manage'
Cisco Unified Mobilitv consists of two main components: Cisco Mobile Connect and Cisco
Unified Mobile Voice Access (MVA):
Mobile Connect allows an incoming call to the enterprise phone number of a user lo be offered to the office phone ofthe user. The call can also be offered to as many as 10 configurable remote destinations. Such remote destinations hpically are mobile or cellular telephones and home office phones. MVA provides similar features for outgoing calls. With MVA enabled, users who are outside the enterprise can make calls as if they were directly connected to Cisco Unified Communications Manager. This functionality is commonly referred to as Direct Inward Svstem Access (D1SA) in traditional telephony environments.
Both features allow active calls to be switched between the IP phone and the remote phone, for example, if users can initiate calls from a mobile phone while on the vvav to the office, then switch the calls to an office phone once they arrives at their desks.
Answer incoming calls on office or remote phone Switch active calls between officeand remote phone
MVA characteristics:
Mobile Connect enables users toreceive business calls at a single phone number, regardless of the device that isused toreceive the call. Mobile Connect allows users toanswer incoming calls on the office phone orataremote destination and pick up in-progress calls on the office
phone or remote destination, without losing the connection. When the call is olTcred to the desktop and remote-destination phone or phones, the user can answer at any ofthose phones. After answering the call on aremote-destination phone, the user can hand offthe call to the
office phone. Active calls on the office phone can be handed offto aremote phone.
Kor example, when auser receives acall that is placed to the business number ofthe user, the office phone and the cell phone ofthe user ring. Ifthe user is traveling to the office, the user can accept the call on the cell phone. After arriving at work, the user can pick up the inprogress call at the office IP phone, by pressing asingle key at the office IP phone. The call
continues without interruption on the office IP phone; the other party ofthe call does not notice
the handover from the cell phone to the IP phone. When MVA is used, after the call is connected, users can invoke midcall features. Users can
also pick up the call on their desk phones, just like they can with received Mobile Connect
calls, fhese actions arcpossible because thecall is anchored at theenterprise gateway.
Cisco Unified Mo I
6 Single (office) numberformultiple devices:
Enterprise caller ID preservation
Single enterprise voice mailbox
Single enterprise number: Regardless ofthe device that is used (enterprise phone, cell
phone, home phone, orother), calls can bereceived ona single number: the number ofthe
enterprise phone, fhe caller ID ofthe enterprise phone isalso preserved on outgoing calls, regardless ofthephone from which the call is initiated. Ilaving a single enterprise number for incoming calls and alvvav s using the same enterprise number for outgoing calls also
allows the use of a single voice mailbox. Theenterprise voice mailbox can serve as a single, consolidated voice mailbox for all business calls. Incoming callers have a predictable mean- ofcontacting employees, and employees do not need to check multiple
voice-mail svstems.
Access lists: Cisco Unified Mobility users can configure access lists to permit or den;
callingnumbers to ring remotedestinations. If a pennit access list is used, unlisted callers arenotallowed to ring remote destinations. If a deny access list is used, only unlisted callers are allowed to ring remote destinations.
User interfaces for enablingand disabling Cisco Inificd Mobility: Users canturn Cisco Unified Mobility on and off b; using a telephone user interface (TUl) lhat MVA provides.
A GUI for Cisco Unified Mobility userconfiguration is available on the Cisco Unified Communications Manager user web pages.
Access to enterprise features: Cisco Unified Communications Manager features can be accessed b; using dual lone multiirequency (DTMI;) feature access codes. The supported
features include hold (default *81). exclusive hold (default*82).resume (default*83).
transfer (default *84). and conference (default *85).The feature codes can be configured as
Cisco Unified Communications Managerservice parameters.
6-64
Smart client support: On phones on which smart clients are installed, softkeys can be
used to access features such as hold, resume, transfer, andconference. Users can also enableor disable Cisco Unified Mobility from a smart client.
Call logging: Enterprise calls are logged regardless ofwhich device (enterprise phone or
remote phone) is used.
Manage;
Gateway
79 565-1555
MoEile Conned
Outside caller calls office phone 2001 (dials 1 511 555-2001). Mobie Connect rings office phone and remole phone.
The figure illustrates the call fiow when Mobile Connect is used. The figure shows an IP phone with extension 2001 and a mobile phone that belongs to the user ofthe IP phone. In this example, a public switched telephone network (PSTN) user calls the office number of
the user. Because Mobile Connect is enabled, both the desktop phone 2001 and the configured
remote destination (mobile phone 408 555-1001) ring simultaneously, 'fhe call is presented to the remote phone, with the original caller ID (479 555-1555). As soon as the call is accepted on one ofthe phones, the other phone stops ringing, fhe user can switch the call between the office phone and the mobile phone (and vice versa) during tlie call, without losing the
connection.
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Manager
Gateway
Mobile Connect
40B555-1001
^Mr
LI
Kerrwtf>r*uieol
Mobile Connect influences the calling-number presentation. If a call is received from a recognized remote destination, the corresponding internal directory number, not the E.164 number ofthe remote device, is used as the calling number. In the example, extension 2001 has a Mobile Connect remote destination 408 555-1001 (cell
phone ofthe user of 2001). The user places a call from the mobile phone to an enterprise PSTN number of a colleague (by dialing 1511 555-2002). The called colleague sees the call as
coming from the internal directory number 2001 instead ofthe external mobile-phone number. The same applies to calls that are placed lo other internal destinations, such as voice mail. If the user of extension 2001 places a call from the cell phone to Cisco Unity, Cisco Unity sees directory number 2001. not the PSTN numberof the cell phone (408 555-1001), as the source
ofthe call. Cisco Unity can identify the user by lhat directory number and can provide access lo the appropriate mailbox instead of playing a generic welcome greeting. To recognize Mobile Connect remote destinations, the Mobile Connect remote destination
number must match the Automatic Number Identification (ANI) ofthe incoming call. Mobile
Connect remote destinations typically include an access code; for example, 9 in the number 9 I 408 555-1001. The access code 9 and the long distance I must be prefixed to the incoming ANI
408 555-1001 to recognize the source as a Mobile Connect remote destination. Alternatively, the Cisco CallManager service Matching Caller ID with Remote Destination parameter can be set to Partial Match, and the Number of Digits for Caller ID Partial Match value can be set. This value specifies how many digits ofthe incoming AN! (starting with the least significant digit) must match a configured remote-destination number.
6-67
If the source ofthe call is not recognized as a Mobile Connect remote desfination. the PSTN number of the remote destination is used for the calling number and is not changed to the
internal direetorv number.
Cisco Unified
Communications
Outside
Manager
^^
V|I&^
:",<: i>r y >: IjC!
When MVA is used, users can place calls from a remote destination to the outside, as if they were dialing from the desktop phone. In the example, the user ofthe IP phone with directory number 2001 uses a cell phone (408 555-1001) to dial the PSTN number ofthe headquarters, extension 2999. The gateway is configured to start an interactive voice response (IVR) call application for calls that are placed to lhal number. The call application, which is based on Voice Extensible Markup Language (VoiceXML, also known as VXML), offers a prompt and asks for the remote destination number and the PIN ofthe user. After login, the user can activ ate and deactivate MVA and can initiate a call from the enterprise network. The call is set up with the H.164 PSTN number of directory number 2001, instead of with 408 555-1001. This action allows the called party to identify the caller by the (single) office number ofthe user. That the call is actually placed from a mobile phone instead ofthe office IP phone does not matter: the call appears to come from the office phone.
After the user has used MVA to initiate a call from a remote destination, the user can switch the
call to the office phone and back again as needed, without losing the connection.
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To implement (isco l'nified Mobility features, you must start the Mobile Voice Access service, which interacts with the call application that runs on a Cisco IOS gateway, on at least
one Cisco l'nified Communications Manager svstem.
MVA requires an 11.323 or Session Initiation Protocol (SIP) gateway to provide a VXML. call application to remote callers who dial a certain number. Media Gateway Control Protocol (MGCP) is not supported because it does not support call applications.
DTMF must be sent out-of-band (OOB). for MVA to work.
The remote destination cannot he an IP phone within the enterprise, fhe remote destination
must be an external device. tvpicallv a PSTN number. As many as 10 remote destinations can be configured. Class of service (CoS) can he configured, lo limit access to the PS'IN.
6-70
Configuration Element
Function
The end user Is referenced by the officephona and remote destination profile.
Mobie Connect or MVAmustbe enanled.
The office phone needs to be configuredwithan owner (that is, the end user)
prone
End user, (device)CSSs, end MOH audio soureea are specified.One or more
remote destinations are adfied,
Remote destnation
Associaed with shared line(s) of remote destination profile. Configured wilh destination number. Optionally, access lists can be applied. Mobile phone and
Mobile Conned functions are selectively enabled
Fitters used to permit or deny sicoming calls placet)to the officephone to ringa remote destination. PennKted or denied callerIDs are specif sd.
Media resource used to interact witi the VXML call application running on a Cisco IOS router. Required for MVAonly.
MVAmediaresource
The following are configuration elements: End user: Each end user must have a configured PIN. which is used for authentication when MVA is used. Three important Cisco Unified Mobilily-rclated settingsthat can be
configured for the end user:
Enable Mobility: Thischeck box mustbe checked to allow the userto use the
Mobile Connect feature (that is. to receive enterprise calls at one or more remote
destinations and to place calls from a remote phone into the enterprise).
Enable Mobile Voice Access: This check box must be checked to allow the user to
place MVA calls, "fhese calls arcoutgoing enterprise calls from a remote phone that
should be placed on behalf of the office phone. Remote Destination Limit: This setting is used to limit the number of remote destinations that can be configured. The maximum is 10.
1P phone: "fhe office phone of a Cisco Unified Mobility usermustreferlo theend-user name, fhis task is done by setting the owner in the PhoneConfiguration windowto the
user ID ofthe end user.
Note
In the End User Configuration window, the end user can be associated with one or more
devices, such as IP phones. Such an association allowsthe end user to configure the device
from the Cisco Unified Communications Manager user web pages, but it is not relevant for Cisco Unified Mobility. The mapping of the IP phone to the end user must be done by setting the owner in the Phone Configuration window.
6-71
Remote destination profile: This setting creates a virtual phone that is linked to the end
user andthat represents all remote destinations that areassociated with the user, fhe profile
includes phone device-level configuration settings, such as user and network Music on
Hold (MOID audio sources andcalling search spaces (CSSs). Kor each office phone thatan
end user should he able to use for Cisco Unified Mobility, a shared line with the line or
lines ofthe office phone or phones must be added to the remote destination profile. In
addition, the remote destination profile is configured with remote destinations.
Remote destination: A remote destination is associated with one or more shared lines of a
remote destination profile, for each remote destination, the remote destination number, as
dialed from within the enterprise, must be specified. The rerouting CSS ofthe specified
remote destination profile is used to look up the configured remote destination number.
Note The remote destination profile has two CSSs that are used for call routing One standard CSS is used for outgoing calls that are initiated by using MVA and the rerouting CSS. The rerouting CSS is used to place a call to the remote destination (eitherwhen receiving a call to the number of the line that the office phone and the remote destination profile share, or when a call is handed over from the office phone to the remote destination). Therefore, the remote destination number must be reachable by the rerouting CSS. For MVAcalls, the rerouting CSS is composed of the CSS that is configured at the shared line and the CSS of the remote destination profile (with priority to the CSS ofthe shared line)
Access list: Access lists can be configured to permit or deny calls that are to be placed to a remote destination when the shared line is called. The filter is based on the calling number. An access list is configured with one or more numbers that specify the calling number that should be permitted or denied. Access lists are also configured with an owner (end-user ID)
and are applied to remote destinations. An allowed, a blocked, or no access list can be applied. Ifan allowed access list is applied, all calling numbers that are not listed in the access list are blocked. If a blocked access list is applied, all unlisted numbers arc allowed. If no access list is applied, all calling numbers are allowed to ring the remote destination.
MVA media resource: This media resource interacts with the VXMI, call application that runs on the Cisco IOS galewav. The resource is required for MVA only. The number at which the Cisco IOS router can reach the media resource must be specified, a partition can be applied, and one or more locales must be chosen.
The CSS of the gateway that runs the VXML call application must include the partition that is
applied to :he number of the MVA media resource
Note
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Office Phone 1
MAC Address
Linel: 2002
Partition
CSS
CSS
Owner
Etc.
CSS
Etc. Shared Line
Etc,
Shaied t.iiie
Remote
Remote
Destination
Profile User ID
destinatbnfs)
associated with
CSS Etc,
Line2: 2002 Partition
CSS
corresponding
line of remote
DestJnation2:
9 1 479 5551555
Rerouting CSS
Etc.
CSS
Etc.
destination profile.
A remotedestination profile is associated with one or more II' phones, liach phone line that an end user should be able to use with Cisco Unified Mobility must be added to Ihe remote
destination profile that is associated withthe end user. Thedirectory number for the useris thus
associated with two devices: the IP phone and the remote destination profile. Such a directory number is also called a shared line. The IP phone or phones that share a line with the remote
destination profile must be owned by the end userwho is associated withthe remote destination
profile.
Remote destinations are associated with one or more shared lines that are configured at remote
destinations.
As described earlier, the settings ofthe shared directory number (including the partition and CSS) apply to al! associated devices. The remotedestination profile is configured with a (standard) CSS. which is used for calls that a remote phone places when it uses MVA, and a rerouting CSS. which is applicable to calls that are placed to a remote destination.
Forexample, if a call is placed to directory number2002. Unci at Office Phone2 and all
remote destinations that are associated with Line2 ofthe remote destination ring. For the call to the remote destination number, the rerouting CSS is used.
If the remote phone with number 9 I 479 555-1555 calls in to the mobile voice application and requests an outgoing call to be placed, the CSS of Linc2 and the CSS ofthe remote destination profile are used for the outgoing enterprise call thai Remote Destination2 initiates.
lonship of Cm
Service Activation
Service Parameter
nst4e MUA.ffdtrptt*: realum Act.-
Media Resources
IVR Application
VoiceXML
t- 323 Galenas
To use Cisco l'nified Mobility, the Cisco Unilied Mobile Voice Access service must be activated if MVA is desired in addition to Mobile Connect functionality.
When the Cisco Unified Mobile Voice Access service is activated, a corresponding media resource is automatical!) added. The media resource must be configured with the MVA number, a partition, and locales.
The configured number must be reachable from the Cisco IOS router that provides remote
phones access to a VXMI IVR call application. Incoming MVA callers are authenticated by remote destination number. Callers are also
authenticated b> the PIN that is configured for the user who is associated with die remote destination profile that the corresponding remote destination number references. When Mobile Connect is used and incoming calls are sent to a line that is shared by an IP phone and a remote destination profile (both referring to the same end-user 11)), access lists lhat are applied to remote destinations can be used to control which callers are allowed to ring the
remote destination, 'fhe access list must refer to the end user who is configured in the remote
phone must have the Mobility soltkey configured for the Connected call stale. If the Mobility
softke\ is also added to the On Hook call state, the softkey can be used to check the stalus of
Cisco Unified Mobility (Mobile Connect on or off). In summary, the end user is the central element that is associated with IP phones (at which the user is configured as the owner), access lists, and remote destination profiles. Remote destinations arc associated with shared lines of remote destination profiles and access lists. For MVA. the appropriate service must be activated, and the automatically generated media resource is made available to a router that runs the VXML call application.
6-74
- Dial peer configurations are ignored. - MVA call application cannot be triggered for incoming
PSTN call.
Onthe voice gateway, the MVA application is configured and triggered as part of a voice dial peer application. Dial peer matching takes place only if the gateway provides call control functionality. When MGCP or Skinny Client Control Protocol (SCCP) is used to control voice
interfaces that receive incoming PSTN calls, the gateway no longerhas complete call control.
Call control is passed overto the Cisco Unified Communications Manager. Inthis case, the MVA application cannot be started because no dial peer matching process takes place.
To use MVA in such an environment. Cisco Unified Communications Manager must forward calls that were received from an MGCP- or SCCP-controlled interface to an H.323 gateway, to
startthe MVA application. From then on, the call treatment is like an H.323-only environment, except that the outbound PSTN call is establishedvia the MGCP gateway.
6-75
Manager
Communications Manager forwards the call to an 11.323 gateway. On the 11.323 gateway, the
MVA application is started and the caller can be authenticated and can define the final destination for the call. The caller is then connected to the MVA media resource, from which
the outgoing call is placed on behalf of the caller office phone (2001). Cisco Unified Communications Manager establishes the outgoing call via the MGCP gateway.
Note The H 323 gateway functionality can be combined on the gateway that receives the PSTN calt on the MGCP-controlled interface. In this case, only one gateway that provides MGCP and H 323 signaling is required
Trunk
- Gateway
Depending on the origin of a call that uses the Mobile Connect feature, different CSSs are used:
For an incoming PSTN call to an office phone that is associated with a remote destination, the rerouting CSS at the remote destination needs access to the mapped remote destination
number.
For an incoming call from a remote phone (remote destination) to an internal destination, the CSS ofthe receiving device (trunk, gateway) needs access to the called internal
number.
6-77
CSS Handling In
CSS used for incoming MVA calls (gateway call application to MVA media resource) needs access to partition that includes MVA media resource number:
Two options-depending on Cisco CallManager "erv^o
(default value}
CSS used for outgoi ig MVA calls (MVA media resource to PSTN)
(priority to line CSS)
"fhe incoming and outgoing call legs ofan MVA call are treated independent!}, 'fhe incoming call leg is the call leg from the gateway where the MVA call application is running to the MVA media resource in Cisco L'nified Communications Manager. The CSS that is used for this call leg depends on a Cisco CallManager serv ice parameter. This service parameter is called Inbound CSS for Remote Destination. The parameter can be set lo one these values:
Trunk or Gateway Inbound CSS: This value is the default value in Cisco Unified Communications Manager. If this option is chosen. Cisco Unified Communications Manager uses the CSS ofthe trunk or gateway from which the MVA call arrived, 'fhe CSS
ofthe shared line and the CSS that is configured at the remote destination profile are not considered for the incoming call leg ofan MVA call.
Remote Destination Profile + Line CSS: If this option is selected, the CSS ofthe shared line and the CSS that is configured at the remote destination profile are combined (with
priority given to the partitions ofthe shared-line CSS). The outgoing call leg ofan MVA call is the call leg from the MVA media resource to the PSTN
destination that is called from the MVA call application. The CSS that is used for this call leg is alwavs the combination ofthe CSS ofthe shared line and tlie CSS that is configured at the remote destination profile (with priori!} given to Ihe partitions ofthe shared-line CSS).
6-78
In Cisco Unified Communications Manager, the end user and the administrator can control
To support time ofday-based access to remote destinations, the remote destination configuration page allows the configuration ofaring schedule. This schedule applies to the
remote-destination configuration page on both the administrator and user web pages. The remote destination can be generally enabled {enabled all the time), or explicit time ranges can be configured. The default is to enable Ihe remote destination all the time.
When an explicit time range is configured, each day orthe week can be disabled, enabled for
the whole day (24 hours), orconfigured with a From/To time range.
Access lists can limit caller IDs. These lists are applied at the remote-destination configuration
page:
The Allowed Access List Access List setting iscalled Ring This Destination Only ifCaller
Is in <Access List>.
The Blocked Access List Access List setting iscalled Do Not Ring fhis Destination if
Caller Is in <Access List>.
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PerformIhe following
checks per configured
remote destination.
Rfifl.feriiote
destination.
Basicallv. two things must be considered when using time-of-dav access control toremote
destinations:
The remote destination rings only when the call is received during the specified ring
schedule. Ihis first decision is independent ofthe access-list configuration.
If no access list fs configured, all callers are permitted. Ilowever. this permission applies
permitted according to anaccess-list configuration but the call is received outside ihe configured ring schedule, the call is not extended to the remote destination.
destination profilearc processed.
onlv after the first check (the call received during ihe specified ring schedule). Ifacaller is
fhe figure shows how calls lhat are received at a shared line that is configured at a remote
For each remote destination that is associated with the called line, the ring schedule thai is configured at the remote destination is checked in the following way: Ifthe call isreceived outside the configured ring schedule, the remote destination does not
ring.
Ifthe call is received within the configured ring schedule, the access-list configuralion of the remote destination is checked. Ifihe caller ID is permitted, the remote destination rings. If the caller ID is lot permitted, the remote destination docs not ring.
"fhe caller ID is permitted inthe following scenarios:
The Alwavs Ring the Destination parameter is selected.
An access list is applied bv using the Ring This Destination Only ifCaller Is in<Aecess I.ist> parameter, and the caller II) is found in the specified access lis!.
An access list is applied bv using the Do Not Ring'fhis Destination if Caller Is in <Access
List> parameter, and the caller ID is not found in the specified access list.
6-80 Implementing Cisco Unified Communications Manager, Part1 (CIPT1) v8.0
w CtmiplctB Matcli
liSS-B.jaJslJttiSi.!
For Mobile Connect and for MVA, the calling line ID ofan incoming call is compared against configured remote destinations, to identify the end user and the associated office phone. This matching process can easily fail because incoming PSTN calls typically do not contain prefixes such as access or long-distance codes. To allow successful number matching, even if not all digits ofan incoming caller ID and configured remote destinations match, the following two Cisco CallManager service parameters exist:
Matching Caller ID with Remote Destination (Partial Match or Complete Match [Default|)
Number of Digits for Caller ID Partial Match
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Configuration Procedure
Configure Mobile Connect:
/ :< -' Add mobility softkey to IP phone softkey templates. Configure end user. Configure IP phone. Configure remote destination profile with shared line. Configure remote destination(s) to remote destination profile. Configure service parameters Optional Implement access lists to specify which caller ID is allowed to ring a remote destination when a call to the office phone is received. Configure access lists Apply access lists to remote destination.
>'
"fhis list summarizes the steps for configuring Mobile Connect and MVA:
Step 1
Step 2 Step 3
Step 4
Step 5
Step 6
Step 7
Optional: Implement access lists to specify which caller ID is allowed to ring a remote destination when a call to the office phone is received.
Configure access lists.
Note
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3
Selected Softkeys Ordered by post*
Forward AllICfwdAII)
MofiilUVlMpMrMI
Settings > Softkey Template and configure a softkeytemplatethat includes the Mobility
softkey for the following call states:
On Hook Connected
6-83
b JH |rrcden|i*ll
Hr.hHr| Infn
""""
j : r r -V!" ",..
|<C ,ML*K,
:>- ?e-.;re I>er.^ ' c - :ic>ks
lo configure end users, choose I scr .Management > Lnd I'ser. Configure Cisco Unified Mobility parameters in the Mobility Infonnation section ofthe find User Configuration
window.
Enable Mobility: Check this cheek bo\ to enable Mobile Connect, which allows the user to receive calls on multiple dev ices lhat are placed to a single enterprise phone number and to hand over in-progress calls between the desktop phone and a remote phone. Mobile Connect also allows users to place calls from remote phones into the enterprise: for
example, to voice mail or internal directory numbers that are signaled with the internal
direetorv number ofthe user.
Knable Mobile \ oice Access: Check this check box to allow the user to use the MVA
incoming calIs that are placed to the enterprise phone number of the user can be sent, fhe range is from I to 10: the default value is A.
Remole Destination Profiles: fhis read-only field lisls the remole destination protilcs that
have been created for this user.
Access Lists: Thi* read-onlv field displays the access lisls lhat have been created for this
user.
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destination profle.
Assign softkey
template.
Set Owner User ID
value.
As shown in the figure, two parameters must beconfigured inthe Phone Configuration window
ofthe office IP phone ofthe user:
Softkey Template: Apply the softkey template (which you created inStep 1) tothe IP phone sothat Ihe user can access the Mobility softkey inthe On Hook and Connected
states.
Owner t'scr ID: Choosethe end-username that you configured in Step 2. This action enablesCisco Unified Communications Manager to locaterelated configuration elements,
such as the remote destination profile ofthe end user.
Note
As the line is shared with the line of the office phone, the same partition that is applied to the
line ofthe office phone has to be set here. The screenshot does notshowa partition, so in
this case, the office line would also have no partition assigned. This is not a common
configuration.
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hate Line is snare J with office phone Configured line CSS and partition apply to both
devices office phone and remole destination profile The line CSS and device CSS are combined lor MVA.Ire partitions of Ihe line CSS are considered first) The rerouling CSS is
not combined with 'He line CSS
To configure remote destination profiles, go lo Device > Device Settings > Remote Destination Profile, fhe remote destination profilecontains the parameters lhal apply to all of the remote destinations ofthe user, [intera name,description, device pool. CSS. rerouting CSS. and network and user MOH audiosources for the remote destination profile. Also enter these
mobility-specific parameters:
User ID: Choose the user to whom this profile is assigned, 'fhe choice must match die ID ofan end user for which the Enable Mobility cheek box is checked.
Privao: Choose a privacy option for this profile. Possible values are On. Off. or Default. Ignore Presentation Indicators: Cheek this cheek box to ignore the connected line ID
presentation. This setting is recommended for internal calls.
Calling Search Space: This CSS (combined with the line CSS) is used for outgoing enterprise calls that arc placed front a remote destination by using MVA. The setting has no
relevance to Mobile Conned.
Rerouting Calling Search Space: Set the CSS that should be used when sending calls that
are placed to the enterprise phone number ofthe user to the specified remote destinations.
Ihis CSS i>. also used when an active call is handed over from lite office phone to a remote
phone.
After a remote destination profile is created, one shared line must be configured lor each direetorv number lhal is used at the office phone or phones ofthe user. To add a shared line, click Add a New DN at the appropriate phone link.
6-86
iSOQ
19000
This setting allows calls placed to office phones to also ring the remole
destination.
desfination profile.
To configure remote destinations, choose Device > Remote Destination. Alternatively, you
can click the Add a New Remote Destination link in a remote destination profile. Enter a name for the remote destination and configure the following parameters:
Destination Number: Enter the telephone number for the remote destination. Include the area code and any additional digits that are required to dial the remote phone from within the enterprise. The maximum field length is 20 characters.
The destination number must not be an internal directory number; the destination number must be an external number. The number must be entered as it would be if it was being dialed from an IP phone: Use a complete E.164 number that includes the access code so that the number matches a route pattern that points to the PSTN. The rerouting CSS that is configured in the remote destination profile will be used to look up the specified number in
the call-routing table.
Note
Remote Destination Profile: The remote destination profile must be chosen, if you created a new remote destination after choosing Device > Remote Destination. If you open the Remote Destination Configuration window by clicking the Add a New Remote Destination link in the Remote Destination Profile window, or if you are editing an existing remote destination, the remote destination profile is already set up and cannot be changed.
If you want to associate a remote destination that is already associated with one remote destination profile with another remote destination profile, you must delete and recreate the
remote destination.
Note
6-87
Mobile Phone: Check this check box to allow active calls to be handed over from ihe
office phone lo this remoledestination when the user presses the Mobility softkey at the
office phone.
Knable Mobile Connect: Check this cheek box to allow calls to be placed to ihis remote destination when there is an incoming call to a shared-line directory number ofan office
phone.
Note
End users can create their own remote destinations on the Cisco Unified Communications
Finally. the remole destination must be associated with one or more shared lines ofthe specified remote destination profile. From now on. the remote destination rings if a call is placed lo the appropriate shared line ofan office phone. When a call is placed from a recognized remote destination to an internal
destination, the calling number is modified from the remole phone number to the office-phone direetorv' number. However, in most cases, the caller ID of that incoming call is a 10-digit number: the remote destination number usually has a PS'fN access code (for example, 9) and then an 11-digit number (trunk prefix I lollowed by the 10-digit number). If the incoming calling number is not prefixed with 91. inlemal phones see the call coming from the F.164 number ofthe remote phone instead of from the associated internal directory number. The next step shows how to resolve such issues.
6-88
To set partial matches so that a calling numbercan be recognized as a remotedestination, you can configure Cisco CallManager service parameters. To access Cisco Unified CallManager service parameters, choose System > Service Parameters and choose Cisco CallManager. Configure the following parameters to allow incoming caller IDs that do not includethe 91 prefix that is used in the remote destination to be recognized:
Matching Caller ID with Remote Destination: Set this parameter to Partial Match
(default is Complete Match).
Number of Digits for Caller ID Partial Match: Set this parameter to the numberof digits that must match (beginning wilh the least significant digit) when comparing the incoming calling number against the configured remote destination number.
Alternatively, choose Call Routing > Transformation Pattern to configure caller ID transformations. Each pattern can be assigned a partition. The Calling Party Transformation CSS, which is configured in the remote destination profile, is used to control access to the configured transformation patterns
Note
configuration of an access-list
member.
existing member
"to configure access lists, choose Device > Device Settings > Access Lists. Hnter a name and a description for the aetess list. In the Owner drop-down list, choose the user to whom the access list applies. 1hen check the Allowed check box lo create a list of phone numbers that should be allowed to ring a certain remole destination when a call is placed to tite office phone number of the user. Io block ihe numbers that are listed in the access list from ringing the remote destinations to which the access lisl will be applied, leave this cheek box unchecked. After saving the access list, the window reopens lo display the Access Lisl Member Infonnation area. Click Add Member to add a member, and then choose an option from the
Filler Mask drop-down list in the Access List Member Detail window. Choose lo enter a
directory number or to filter out calls that do not have caller ID (the Not Available option) or do not display their caller ID (the Private option). You can also change existing members by
clicking the appropriate link.
In the Access I i^t Member Detail window, if filter Mask is sel to Directory Number, enter a phone number or lilter in the DN Mask field. You can use the following wildcards: \: Matches a single digit
The X wildcard must be entered in uppercase, Cisco Unified Communications Manager
displays a syntax error message otherwise
Note
Note
6-90
_,,r.
Either allowed or blocked access lists car be set. If blocked access list is set. all numbers not listed in access list are allowed
"Er :EESE
:::EE
\
^
.rEErr:.
:rr:
To applv anaccess listto a remote destination, open the Remote Destination Configuration
window. Choose Device > Remole Destination or click the appropriate link in the Remote Destination Profile window. Then choose the access list from the drop-down list under Ring
'fhis Destination Only if Caller Is In (allowed access list)or DoNot Ring This Destination if
Caller Is In (blocked access list).
Note Only an allowed (Ring this destination only if caller is in}, blocked (Do not ring this destination ifcaller is in),or no access list(Always ringthis destination) can be applied to a remote destination; calling numbers that are not listed in an allowed access list are denied, and calling numbers that are not listed in a blocked access listare allowed.
:,
Step.
Step 2
hnable MVA per end user. Configure the MVA media resource.
6-92
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Sertpr
10.1.1.1
- |_to_
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i unrted Motto voir? J
Open the Cisco Unified Serviceability window. Choose Tools>Senice Activation and check
the Cisco Unified Mobile Voice Access Service check box. When the service has been
activated, verify that it is started by following ihe Control CenterFeature Services link.
6-93
'si | enterpnsefeature
codes and change if required
Enable access to
enterprise features
Enable MVAgbbally.
lo configure Cisco I 'nified Mobilit; service paramelers, choose System > Service Parameters and choose a server. Then choose the Cisco CallManager service, fhe parameters that are shown in the figure are clusterwide parameters, which apply to all servers. You can enable access lo enterprise features h; setting the Enable Enterprise Feature Access
parameter to True. In this ease, the following features can be used from a remote destination phone, and the corresponding feature access codes can be modified from their default values:
These parameters must be unique and two or three digits or letters long. Allowed values are 0
through 9. A through 1). and *. To enable MVA. set the Enable Mobile Voice Access parameter lo True.
Note By setting the Enable Mobile Voice Access parameter to True, you enabled MVA in general only To allow MVA to be used, you must enable it individually for each end user, in the End User Configuration window
fPM'J
^ Edit Crwletrttaf j
In the End User Configuration window, check the Enable Mobile Voice Access check box to
allow the end user to use MVA.
Note
All other Cisco Unified Mobility parameters were discussed earlier in this topic.
6-95
The H.323 gateway running the MVA call application needs lo have access to the partition
of the MVA number (if set)
"fhe VIVA media resource is automatically added when the Cisco Unified Mobile Voice Access Serv ice is activ ated. I he resource can be configured from Media Resources > Mobile Voice Access. The following configuration options e\isl:
Mobile Voice Access Director; ISumber: Remole users who want to use the MVA feature must dial a certain PS'fN number at an H.323 gateway that provides access by a call application to the MVA feature. The call application will route the incoming calls to ihe MVA media resource, fhe number that is used for this call leg (gatewav to media resource) is the Mobile Voiee Access Directory Number that is configured at the MVA media resource, fhe VXML call application resides on Cisco Unified Communications Manager and is accessed from the gateway bv HTML. Therefore, the local VXML application code can refer to this configuration parameter, which is stored in the Cisco Unified Communications Manager configuration database. Ilowever. the galewav must have a dial peer for this number, and that dial peer must poinl to the Cisco Unified Communications Managersvstem or svstems on which the Cisco Unified Mobile Voice Access service has
been activated.
Mobile Voiee Access Partition: Assign a partition to the Mobile Voice Access Director; Number. Make sure that the CSS ofthe gateway has access to this partition.
Selected Locales: Choose at least one locale from lite lisl of available locales.
By default, only U S. English is available.
Note
6-96
voice-port 0/0/0:23
voice
translation-rule
port 0/0/0:23
1
dtmf-relay h245-alphanumar ic
codec g711ulaw
no vad
codec g771ulaw
The figure shows a sample configuration ofan H.323 gateway. Inthe example, an incoming translation profile, which strips the called number down to four digits, is applied to the voice port. Therefore, all other dial peers thatareapplicable to calls from the PSTN refer to fourdigit, called numbers only.
"fhus. the following happens when a remote user dials the MVA number 1511 555-2999. The call is routed to the voice portof Ihe router, and the PSTN delivers a 10-digit national number thatthe translation profile then strips down to 4 digits. The called number 2999matches the incoming plain old telephone system (POTS) dial peer 29991, which is configured by using the call application mva service command. The Mobile Voice Access service isconfigured with
the URL ofthe MVA VoiceXML call application. This application is on the Cisco Unified Communications Manager server on which the Cisco Unified Mobile Voice Access service has
been activated.
Note
The MVA application URL can be found inthe Cisco Unified Communications Manager Help
pages.
6-97
Tip
When a Cisco IOS release earlier than 12.3(12) is used, the following syntax changesmust
be considered
incoming called-number 2 99 9
direct-inward-dial
This syntax applies to versions earlier than Cisco IOS Software Release 12.3(12).
call application voice MVA http:// 10.1.1.1:8080/ccmivr/
pages.'IVRMainpage.vxml
When the call i passed on to the MVA media resource, the number lhal was configured at the MVA media resource during the previous slep (in this case, also 2999) is u.sed.
Note The number that is used to start the call application on incoming PSTN calls (1 511 5552999) does not need to match (or partiallymatch) the number that is used for the call leg from the H 323 gateway to the Cisco Unified Communications Manager MVA media
resource. However, you should use the same number to avoid confusion.
The outgoing VoIP dial peer lhat is used for this call leg (dial peer 29992) must be configured for DTMF relav. and voice aelivitv detection (VAD) must be disabled.
All oilier dial peers that are shown in the example apply to incoming PS'fN calls to direelor> numbers other than 2999 (dial-peer voiee 1 pots and dial-peer voice 2 voip command sections, which are ouilined in the boltom right section ofthe figure) and ouigoing PSTN calls (all received VoIP calls use incoming dial-peer 2 and outgoing dial-peer 1). fhese outgoing PS'fN calls include normal calls that are placed from internal devices as wel! as calls that arc initialed from remote phones thai use MVA lo place enterprise calls lo the PSTN.
Note More information about incoming and outgoing dial peer matching is provided in the Implementing Cisco Voice Communications and QoS (CVOICE) course
6-98
Summary
This topic summarizes the key points that were discussed in this lesson.
Summary
Mobile Connect enables users to receive calls that are placed to their enterprise number at the enterprise phone and
enterprise number ofthe user as the calling number. MVA requires an H.323 gateway that provides an IVR
application to MVA users.
The Cisco Unified Mobile Voice Access service must be
Summary (Cont.)
Ifan MGCP gateway is used for PSTN access, an additional H.323gateway is required for the MVA feature. Proper CSS and access-list configuration is required for MVA and Mobile
Connect.
Implementation of Cisco Unified Mobility includes the configuration of access lists, remote destination profile, and
remote destinations.
6-99
References
I or additional infonnation. refer to these resources:
Cisco Svstems. Inc. Cisco Unified Communicalions System Release S.x SRND. San Jose. California. April 2010. Imp: umv.ui-.eo corri'en I Sdoc>\oiee ip eomm/cucm4nid/8 vue8\M-nd.pdf. Cisco Svstems. Inc. Cisco Unified Communications ManagerAdministration Guide
Release 8.0(2). San Jose. California. March 2010.
http:1 wvvu.cKco com'en I S'docvAoiuejp comm/ciicm/adinin/K 0 2'ccmcf!.,/bcc!n.pd!'. Cisco Svstems. Inc. Cisco l'nified Communications Manager features and Services Guide
Release 8.0(2). San Jose. California. March 2010.
6-100
Module Summary
This topic summarizes the key points that were discussed in this module.
Module Summary
Cisco IP Phone Services can be configured to allow access
any phone (PSTN phone, cell phone, or officephone) to place and receive calls from a single (office) number.
This module describes howto configure CiscoIP Phone Services as well as howto implement presence functionality. Cisco Unified Mobility feature is alsocovered. References
For additional information, refer to these resources:
Cisco Systems. Inc. Cisco Unified Communications System Release S.xSRND. San Jose.
California. April 2010.
http:'/\v\v\v.eiseo.com/en/US/docsA;oiee_ip_eomm/eucm/srnd/'8\Ate8\srnd.pdf.
Cisco Svstems. Inc. Cisco Unified Communications Manager Administration Guide
Release 8.0(2). San Jose. California, March 2010.
http:.v\wvv.eiseo.com/en/l ;$/docs.'VoiceJp_eomm/cuem/admin/8J)_2/ccmieat/fsgd.pdf.
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6-102
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.
Ql)
A)
B) C) D)
Q2)
Ilow can redundancy beprovided to Cisco IP Phone Services? (Choose two.) (Source:
Configuring Cisco IP Phone Services)
A) B)
C)
Configure backupserviceson a Cisco IOS gateway. Configure Cisco IOS server load balancing.
Configure SRST.
D)
Q3)
Which isnot a valid service provisioning mode? (Source: Configuring Cisco IP Phone
Services)
A) B)
internal external
C) D)
Q4)
How can configured Cisco IP Phone Services be subscribed? (Choose two.) (Source:
Configuring Cisco IP Phone Services) A)
B)
C) D) hi)
By the end user, via a phone soflkey By the administrator, via the administration web page Cisco IP Phone Services are subscribed automatically.
Q5)
Which two presence features arenatively supported by Cisco Unified Communications Manager? (Choose two.) (Source: Cisco Unified Communications Manager Native
Presence)
A) B)
C) D) E)
presence-enabled speed dial third-party presence server integration presence-enabled directories and call lists
6-103
Q6)
Which twoendpoints arc supported by the Cisco Unified Communications Manager presence feature? (Choose two.) (Source: Cisco Unified Communications Manager
Native Presence)
A)
B) C) D) 1:|
Cisco IP phones devices that are reached through an SIP trunk MGC P gatewav endpoints 11.32.* gateways voice-mail ports
Q7|
Which two statements are true about presence policies? (Choose Iwo.) (Source: Cisco
Unified Communications Manager Native Presence)
A) li)
C)
I))
k) Q8)
Which ihnot a configuration step when enabling presence? (Source: Cisco Unified
Communications Manager Native Presence)
A|
B) C)
f.nable the BLF for Call Lists enterprise parameter. l.nahlc Cisco Unilied Communications Manager presence on SIP trunks.
D) Q9-)
Cisco I nitied Mobility consists of which two features? (Choose two.) (Source: Configuring Cisco I Inificd Mobility)
A)
B) C) I)) f)
QIO)
Which number is indicated as the calling number for a call that is placed from a remote destination to an internal directory number? (Source: Configuring Cisco Unified Mobility)
A) B) C) the Mobile Voice Access number the number ofthe remote destination
the dircclon numberof the office phone lhal the remote destination is
associated with
D)
the directory numberof the called office phone, if associated wilh ihe calling
remote destination
Q11)
Which is not a requirement for Cisco Unified Mobility? (Source: Configuring Cisco
L'nified Mobility)
A) remote destinations that have to be external numbers
B)
C) D)
H.323 or SIP gateway that provides the Mobile Voice Access IVR application
out-of-band DTMF transcoder that runs at the gateway providing Mobile Voice Access IVR
application
Q12) What must be considered when implementing Cisco Unified Mobility in an environment with MGCP-controlled PSTN gateways? (Source: Configuring Cisco Unified Mobility)
A)
B)
C)
D)
PSTN calls that arrive at the MGCP gateway must be sent to an H.323 gateway by Cisco Unified Communications Manager. MGCP gateways cannot receive Cisco Unified Mobility calls.
QI3)
D)
E) F)
QI4)
Dial via Office exists in which two variants? (Choose two.) (Source: Configuring Cisco
Unified Mobility)
A) remote
B)
C) D)
forward
backward reverse
E)
transparent
6-105
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i)
A. I) c.v
A. U A.l
QS)
Qf) 1 OKI) (.111) g i: i
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014)
! B. 0
6-106
Table of Contents
Lab Guide
Overview Outline
2
2 2 2
3
Task 4- Configure Cisco Unified Communications Manager Enterprise and Service Parameters 7 Lab 2-2: Managing User Accounts in Cisco Unified Communications Manager 9
Activity Objective Visual Objective Required Resources g g 10
,,
Task 2: Manage Network and Feature Services Task 3' Configure Cisco Unified Communications Manager to Provide DHCP Services
J b
Job Aids
Task 2: Manage User Accounts by Using Cisco Unified Communications Manager BAT
Activity Objective Visual Objective Required Resources
Job Aids
Task 1: Configure System Parameters
AT
15 15 ^
16
Task 2:Add IP Phones by Using Autoregistration Task 3: Manually Add and Configure an IP Phone
20
Task 4(Optional): Prepare Cisco Unified Communications Manager BAT for Adding Cisco IP
by Using Cisco Unified Communications Manager BAT Task 6: Configure Cisco Unified Communications Manager to Support a Third-Party SIP
Task^Optional): Insert IP Phones into the Cisco Unified Communications Manager Database
Lab 4-1: Implementing PSTN Gateways
Activity Objective Visual Objective
Required Resources
Command List
25
l'
27 27
._
__
Task 1: Configure an MGCP Gateway by Using T1/E1 PRI to the PSTN in Cisco Unified
Communications Manager
Jj
33
34
Job Aids k_ M Task 1: Configure a Route Plan That Uses Both Gateways and Includes Two Route Groups, a Task 2(Optional): Enable Cisco Unified Communications Manager Dialed Number Analyzer ana
Use It for Dial Plan Verification
Jt, ._ ,
3?
37
^ ^ f?
-Q
Used for Called Numbers on Outgoing Calls 40 Task 2: Configure Cisco Unified Communications Manager to Extend Callinq Numbers in Outgoing Calls to Fully Qualified PSTN Numbers 41 Task 3: Configure Cisco Unified Communications Manager to Reduce the Called PSTN Number
of Incoming Calls to Directory Numbers luinuar
y
Task 1: Configure Cisco Unified Communications Manager to Strip Access Codes That Are
Task 4: Configure Cisco Unified Communications Manager to Prefix Access Codes to Callina
Numbers on Incoming PSTN Calls
Task 5(Oplional). Implement E.164 Pius Dialing and Phone Number Localization
44
4fi
Required Resources
Job Aids
Task 1: Configure Partitions and CSSs Task 2: Implement CoS for Internal Calls Task 3: Implement CoS for Incoming PSTN Calls Task 4: Implement CoS for Outgoing PSTN Calls Task 5(Optional): Implement Time-of-Day-Based CoS Task 6(Optional): Implement FACs Lab 4-5. Implementing Call Coverage in Cisco Unified Communications Manaqer Activity Objective y
48 49 50 51
54
c2
Visual Objective
jS
Required Resources
Task 1: Create a Line Group and Add Members Task 2: Create a Hunt List and Add Line Groups
57 58
Task 4: Test Call Distribution Task 5(Optional): Configure Final Forwarding for Busy and No-Answer Conditions Lab 5-1: Implementing Media Resources
Required Resources Command List
Task 2: Add a Hardware Conference Media Resource Task 3: Configure Meet-Me Conferences in Cisco Unified Communications Manaqer Task 4: Manage MOH Audio Files Task 5(Optional): Configure Multicast MOH Task 6 (Optional): Implement Media Resources Access Control Lab 6-1: Configuring Cisco Unified Communications Manager Native Presence
Activity Objective Visual Objective Required Resources Task 1: Configure Presence-Enabled Speed Dials Task 2: Implement Presence-Enabled Call Lists Task 3 (Optional): Configure Presence Policies
Task 3: Createa Hunt Pilot Number and Associate It with a Hunt List
Visual Objective
Activity Objective
59 60 62
62 63
g4 66 67 69 71 75
fi?
Task 1: Add aSoftware Conference Media Resource in Cisco Unified Communications Manager
7c 75 75 76 77 77
Lab 6-2: Configuring Cisco Unified Mobility Activity Objective Visual Objective Required Resources Task 1:Add the Mobility Softkey to IP Phones
Task 2: Associate an End User Account with the IP Phone and Enable the Use of Mobility
79 79 79 79 80
80
Task 3: Configure Remote Destination Profiles and Remote Destinations Task 4: Configure Ring Schedulesand Access Lists for Remote Destinations
Task 5: Enable MVA
81 84
85
Task6: Configure Cisco Unified Mobility Media Resources Task 7: Configure the CiscoIOS Gateway for CiscoUnified Mobility Answer Key Lab 2-1: Configuring Cisco Unified Communications Manager Initial Settings Lab 2-2: Managing UserAccounts in Cisco Unified Communications Manager Lab3-1: Implementing IP Phones Lab 4-1: Implementing PSTN Gateways Lab 4-2: Configuring Cisco Unified Communications Manager Call-Routing Components Lab4-3: Implementing Digit Manipulation Lab4-4: Implementing Calling Privileges inCisco Unified Communications Manager Lab4-5: Implementing Call Coverage in Cisco Unified Communications Manager Lab 5-1: Implementing Media Resources Lab 6-1: Configuring Cisco Unified Communications Manager Native Presence
Lab 6-2: Configuring Cisco Unified Mobility
86 87 92 92 92 92 92 93 93 93 93 93 95
95
CIPT1
Lab Guide
Overview
This guide presents the instructions and other information concerning the lab activities for this
course. You can find the solutions in the lab activity Answer Key.
Outline
This guide includes these activities:
Lab 2-1: Configuring Cisco Unified Communications Manager Initial Settings Lab 2-2: Managing User Accounts in CiscoUnified Communications Manager
Lab 3-1: Implementing IP Phones
Activity Objective
In Ihis aeti\it\. >on will configure Cisco L'nified Communications Manager initial settings to aclhate sen ices. You will use the Cisco Unified Communications Manager IP address rather than its hoslname. After completing this acli\ in. you will be able to meetthese objectives:
Change the hostname of Cisco Unified Comnitinicatioiis Manager into its IP address
Acthate and manage required sen ices
Visual Objective
The figure illustrates what >ou will accomplish in this activity.
Required Resources
These resources and equipment are required lo complete this activity:
Cisco IP phones
Device Role in the Activity Cisco Unified Communications Manager publisher Cisco Unified Communications Manager subscriber Student computer with web browser
CUCM1-X CUCM2-X
PC-x
Access
https://10.x. 1.1/ccmadmin
ManagerAdministration GUI
Cisco Unified Communications
cucmpassi
cucmadmin
https://10.x. 1.1/ccmservice
cucmpassi
Job Aids
Thisjob aid is available to helpyou complete the labactivity. Theaddressing of labdevices usesthe IP allocation scheme that is shown in the table.
IP Allocation Scheme
Parameter Voice server network Value 10.X.1.0/24 10.X.3.0/24
Data network
Default gateways
CUCMl-x
CUCM2-X PC-x
Activity Procedure
Complete these steps:
Step 1
from PC-.v. access Cisco Unified Communications Manager Administration using the infonnation provided in the "Credentials for CUCMI -x Application Access"
table.
Step 2
Step 3
Go to System > Server and click Find to list all the servers in your cluster.
Click CUCM 1-.V and change the Host Name/IP Address field from CUCMKv to
KU.1.1.
Step 4
Step 5
>2010 Cisco Systems, Inc.
Add a description for your server. Enter Publisher in the Description field.
Click Save and then click OK.
Lab Guide
Note
The subscriber has already been configured with its IP address during the installation
procedure because installation requires the subscriber to be added as a server before
installing it
StepG
Step 7
Change ihe hostname part of all phone URLs to the appropriate IP address and click Sa\e (for example changehttp://CrC'M1-A-:8080/ccmcip/authcnticate.jsp to
Imp://t0a-.l.I:8080/ccmcip/authentieate,jsp).
Activity Verification
You have completed this task when \ou attain these results:
IP addresses are used for the server names.
Go to System > Server and click Kind to list all the servers in your cluster.
Both the publisher and subscriber servers should be listed, with an IP address in the Host Name'IP Address column and with a Description.
In the Phone URL Parameters pane, all URLs should contain (he IP address ofthe publisher server {10.v. 1.1. where a is vour pod number).
Activity Procedure
Complete these steps:
Configure Cisco Unified Communications Manager Servers Step 1 from PC-.v. access Cisco Unified Communicalions Manager Administration.
Step 2 Go to System > Cisco I nified CM and click Kind.
Step 3
Step4
Step 7
Go back through the Related Links menu to Back To Find/List and repeal the last tliree steps for the subscriber CM_10..v.1.2. Rename the subscriber to CUCM2-A and enter Subscriber for the description.
Step 10
4
At the Select Server page, choose KLv.1.1 and then click do.
2010 Cisco Systems, Inc.
Note
The DHCP service can be activated on multiple servers. You will enable the DHCP service
at the subscriber in a later task.
Step 11
Note
Step 12
From the list ofservices, check Cisco CallManager, Cisco TFTP. and Cisco
DHCP Monitor Service.
You will need the Cisco DHCP Monitor Service toperform the next task in this lab.
Click Saveto activate these services. A pop-up window appears and informs you that service activation will take a while. Confirm by clicking OK,and then wait
until die Status (on top-lea comer ofthe page) changes from Ready toUpdate
Operation Successful.
Step 13 Step 14
Step 15
Using the related links or Tools > Control CenterFeature Services, go to the
Control Center for feature services.
Verify that the Cisco CallManager. Cisco TFfP. and Cisco DHCP Monitor Service
serv ices are started and activated.
Repeat the previous steps toactivate the Cisco CallManager service on the
subscriber. Thisservice is theonly one thatyou will activate on the subscriber.
Step 16
Activity Verification You have completed this task when you attain these results:
Verifv that the new names are configured.
Verify that all required services are started. In the Cisco Unified Serviceability, go to Tools > Control CenterFeature
Services.
Make sure thatthe Cisco CallManager, Cisco TFTP. and Cisco DHCP Monitor
Service services are shown as started services on the publisher.
Make sure that the Cisco CallManager serviceis shown as a startedservice on the
subscriber.
Lab Guide
Activity Procedure
Complete these steps:
Step 1
Step 2
Step 3
Step 4
Primary I"FTP Server IP Address (Option 150): KLr.Ll Keep tne default values for all other parameters.
Step5
In the l)\K'V Subnet Configuration window, choosethe newly created DHCP server and enter the following parameters:
Subnet IPv4 Address: HLv.2.0
Primary Start IPv4 Address: HLv.2.11 Primary End IPv4 Address: HLv.2.19
Note
Byspecifying option 150 at the DHCP server, you made the configured value the default value for all DHCP subnets Therefore,this parameter does not need to be set again during
the DHCP subnet configuration
Step 6
Step7
Open a lelnet session to reachyour HQ-a router(IP address I0a.250.I01). Log in and change to enable mode. The login password and enable secret password are
cisco.
Step 8
ip helper 10.x.1.1
Note
Use the Ethernet interface that connects to the HQ-x Phones network. This is the interface
with IP address 10x2.101
Step 9
Activity Verification
You have completed this task when you attain these results:
6
The HQ-a router is configured to act as a DHCP relay at the Phones network. Further verification, to verify that IP phones obtain an IP address from the Cisco Unified
Communications Manager publisher, is performed in Lab 3-1.
Activity Procedure
Complete these steps:
Step 2
Step 3
Note
phones) usea particular CSS. If you need todelete a record from Cisco Unified
Communications Manager, you can use dependency records to show which records are associated with the record that you wantto delete. You then can reconfigure thoserecords so that theyare associated with a different record. Configure Service Parameters
Step 4
Step 5
Note
Navigate to System > Service Parameters and choose the publisher server. Wail for
the windowto refreshand then choosethe Cisco CallManager servicefrom the
drop-down list.
Set the CDR Enabled Flag value to True in the System parameters.
The CDR Enabled Flag parameter determines whether CDRs are generated. Valid values
specify True (CDRs are generated) or False (CDRs are not generated).
Step 6 Step 7 Set the CDR LogCalls with Zero Duration Flag to True intheSystem
parameters,
SettheCMR parameter Call Diagnostics Enabled value to Enabled Only When CDR Enabled Flag is True in the Clusterwide Parameters (DeviceGeneral).
TheCall Diagnostics Enabled parameter determines whether CMRsalso called diagnostic
recordsare generated. Generating CMRswithout corresponding CDRs can cause
Note
uncontrolled disk-space consumption. Therefore, you should always enableCDRs when CMRs are enabled. If youchoose Enabled Only WhenCDR Enabled Flagis Trueand the CDR Enabled Flagserviceparameteris set to False, no CMRs will be generated.
Step 8 Save the changes.
Lab Guide
Step 9
Note
Repeat Steps 5 and 6 for the subscriber server and save the changes
There is no need to repeat Step 7 for thesubscriber server because the parameter thatis
changed n Step 7is a clusterwide parameter and is automatically applied to all servers.
Activity Verification You have completed diis task when you attain these results:
fhe Dependencv Records parameter isenabled. In System > Lnterprise Parameters, verifv
that the Unable Dependency Records parameter is set to True.
fhe CDR and CMR parameters arc enabled.
do to Svstem > Service Parameters and choose the publisher server. Wait for the
window to refresh and then choose ihe Cisco CallManager service from the drop
down list.
The CDR Lnabled Flag and the CDR LogCalls with Zero Duration Flag arcset to
I rue. and the CMR parameter Call Diagnostics Lnabled is set to Fnabled Only When CDR Fnabled Flag Is True.
Repeal ihe previous steps at the subscriber.
Activity Objective
Inthis activity, you will practice how tomanage user accounts. You will add administrators with different administrative privileges. After completing thisactivity, youwill be able to meet
these objectives:
Manage useraccounts by using the administration GUI Manage useraccounts by using CiscoUnified Communications Manager BAT
Visual Objective
fhe figure illustrates what you will accomplish inthisactivity.
DHCP I 10,30^
ioi i
Required Resources
These resources and equipment are required to complete this activity:
Lab Guide
Use the credentials in thetableto log in lo the labdev ices that require authentication.
Credentials for CUCMI-x Application Access
Device
Access
https://10.x. 1.1/ccmadmin
cucmpassi
Job Aids
Ihisjob aid is available to help vou complete the lab activity.
'fhe addressing of lah dev ices uses the IP allocation scheme that is shown in the table.
IP Allocation Scheme
Parameter
Voice Data server network
network
Value
10x1.0/24 10.x.3.0/24
should have full access: the other should have read-only access.
Activity Procedure
Complete these steps:
Add New User Through Cisco Unified Communications ManagerAdministration In Cisco Unified Communications Manager, conligure a user:
Step 1 Step 2 from PC-.v access Cisco Unified Communications Manager Administration. Go to I'&er Management > End I'sur and click Add New.
Step 3
Configure a user with the attributes thai follow, and save the newly created account by clicking Save at the bottom ofthe page or the Save symbol at the top ofthe hnd
I'scr Confiauration window.
User ID: fserl
Password: password
PIN:54321
Step 4
Click Add New again and add a second user with these attributes:
UserID:User2
Password: password
PIN:12345
Step 5
In User Management > End User, verify that the two end users, Userl and User2. areconfigured in Cisco Unified Communications Manager Administration.
Step 6
The first user. Userl, isassigned the Standard CCM Super Users access rights. Click
the User ID in the Find and List Users list to open the End User Configuration
window.
Step 7
Step 8
Click Add to l]ser Group inthe Permissions Information box on the bottom ofthe
page.
Step9 Step10
Step 11
Choose the Standard CCM Super Usersparameter from the resulting list, and click Add Selected. The selectedparameter is visiblein the Groupsbox. Inthe End User Configuration window, clickSave.Thestatuschanges to Update
Successful. Go back to the Find and List Users list.
The second user, User2. is assigned Standard CCM Read Only access rights. Go to the Permissions Information box in the End User Configuration window and click
Add to User Group.
Step 12
Step 13
Step 14
Choosethe Standard CCM Read Only parameter from the resultinglist, and click Add Selected. The selected parameter is visible in Ihe Groups box.
Click Save.
Step 15
Step 16
On the top of the page, in the right corner, click the Logout link to logout the
cucmadmin user.
User and verify that the Add and Delete buttons are shown.
Step 17 Click the Logout link to log out Userl.
Step 18
On the Cisco Unified Communications Manager Administration login page, log in as User2 and access some administrator menus. Go to User Management > End User and verify that User2 has only read access rights (no buttons to add or delete
users are shown).
Lab Guide
11
Step 19
Log in as Userl again and navigate to the End User menus. Click Find and verifv lhat two usersare configured and that you can see the infonnation that is offered on
the Find page. Click User I and change the password to cisco and the PIN to 12345.
Step 20
Activity Verification
You have completed .his task when you attain these results:
The first end user is assigned to the standard groupStandard CCM Super Users.
Navigate to User Management > Lnd User. Select the user, enter the Fnd User
'fhe second end useris assigned to the standard group Standard CCM Read Only.
In the User Management > End User window, enler the End UserConfiguration window and go lo the Permissions Information box. Verify lhal Standard CCM Read
Onlv is assigned to the user.
When logging in with the first username. vou have full access to Cisco Unilied Communications Manager Administration.
When logging in wilh the second useniame. you have read-only access lo Cisco Unified Communications Manager Administration.
You wereable to change the password ofthe new ly createdusers when vou were logged in
as the first end-user (Userl).
--
Navigate to User Management > End User and choose Userl. In the User Configuration window, change the password and PIN.
Task 2: Manage User Accounts by Using Cisco Unified Communications Manager BAT
In this task, vou will use Cisco Unified Communications Manager BAT to add users and you
will configure user templates to assign rights to users.
Activity Procedure
Complete these steps: Activate the Bulk Provisioning Service The Bulk Prov isioning Service must be activated to work with Cisco Unified Communications Manager BAT.
Step 1
Step 2 Step 3
In the Cisco Unified Serviceabililv window, navigate to Tools > Service Activation.
In the Serv ce drop-down box. choose the publisher server HLv.I.I. In the Database and Admin Serv ices area, activate the Cisco Bulk Provisioning
Sen ice and click Save. The w indow refreshes, and the Activation Status
After you download the bat.xlt file, you can enter user infonnation in the tile. The data will be exported to a.csv file and imported in Cisco Unified Communications Manager through Cisco
Unified Communications Manager BAT.
Step 4
Step 5 Step 6
Click the check box for the bat.xlt file and click Download Selected. Download the Cisco Unified Communications Manager BAT spreadsheet tothe C directory and open the bat.xlt file. Ifprompted, click Enable Macros to use the
spreadsheet capabilities.
Note
The bat.xlttilewill be automatically renamed at download. The new filename includes the Cisco Unified Communications Manager Version; for example, bat_8.0.1.10000-40.xlt.
Step7
Step 8
Complete all mandatory fields ineach row, providing the following information:
First Name: User3
Step 9
Step 10
Step 11
Scroll to the right andclick Export to BAT Format lo transfer the data from the
bat.xlt spreadsheet into a CSV-formatted data file.
Note
Ifyou enter a comma in one of the fields, batxlt encloses that field entry in double quotes when you export to CSVformat. Ifyou enter a blank row in the spreadsheet, the system
treats the empty row as the end of the file. Data that is entered after a blank line is not
converted to CSV format.
Add a New User Template in Cisco Unified Communications Manager User templates are used lo define common features for end users. Step 12 Step 13 Navigate to Bulk Administration > Users > User Template and click Add New. In the User Template Configuration window, enter the following parameters, and
then click Save:
Lab Guide
Lser Locale: Fnglish, United States User Ciroup: Standard CCM \:n<_] Users
fhe newly created ,l>y file will be imported inCisco Unified Communications Manager
through the Bulk Administration menu.
Step 14
Step 15
Step 16
In the File text box. click Browse and locate the file C:\XLSDalafiles\Users'timestam/?>.l\t.
Step 17
Step 18
Step 19
Step 20
Step 21
In the File Name field, choose the CSV data file lhal you created for this bulk transaction (Users-<//wt\s/</w/j>.txl).
Step 22
Step 23 Step 24
Step 25
Choose the user template (CIPTl sers) lhat vou created, from the User Template
Name drop-down box.
In the Job Infonnation area, enter as Job Description CTP'Tl Insert User and click
ihe Run Immediately radio button.
Click Submit to create the job for inserting the user records. The Status should displav tha'. adding the job was successful and thai the job request is submitted.
Use the Job Scheduler, in Bulk Administration > Job Scheduler, lo see the status of \ our job.
Step 27
Step 28
fhe two newlv created users. Useo and User4. should appear in the user lisl.
Browse to https://10_i.l.l/ci:musei- and log in with users User3 and User4, using the defined password cisco.
You will be prompted to change the password You can change the password to the same password that was initially configured: cisco.
Note
Activity Verification
You have completed this task when vou attain ihis result: The two new users. User.l and User4. appear in the user list.
\a\ iaate to User > End User and click the Find button.
Activity Objective
In this activ ity. you will manually add IP phones toCisco Unified Communications Manager,
by using Cisco Unified Communications Manager BAT and autoregistration. Further, you will harden IP phones by disabling web access and GARP, and by blocking access to the voice
VLAN. After completing this activity, you will be able to meet these objectives:
Configure system parameters, including the Cisco Unified Communications Manager group, and device pools, to prepare the Cisco Unified Communications Manager cluster to
autoregister IP phones
Use Cisco Unified Communications Manager BAT to add IP phones to Cisco Unified
Communications Manager
Visual Objective
The figure illustrates what youwill accomplish inthisactivity.
Required Resources
These resources and equipment are required to complete this activity:
Cisco Unified Communications Manager cluster
Lab Guide
Job Aids
Ihese job aids are a\ .lilable to help vou compleie the lab activity.
DHCP Addressing and Subnet Information for Cisco Unified Communications
Manager Server
DHCP Server
Subnet
IP Address
Primary
Start IP Address
End IP Address
Primary
Router IP
Subnet
Mask
Primary
TFTP
Server
Address
(Option
150)
CUCMI-x
10 x 2 0
10x2 11
10x2.20
10x2 101
(10x.1 1}
255.255 25 5.0
10.x. 1 1
Manager Name
CUCM1-X 10x 1.1
Publisher
2001-2002
Directory Numbers
Device Phonel-x Phone2-x
Phone3-x
Directory Number
2001
2002 2003
IP Allocation Scheme
Parameter Voice
Data
Value
10.x. 1.0/24
server network
network
10.x.3.0/24
Default, gateways
CUCMl-x
PC-x
Activity Procedure
Complete these steps:
Inthis section, vou will update the default Cisco Unified Communications Manager group to
use the publisher as first choice and the subscriber as second choice. You will add another
Cisco Unified Communications Manager group, with the subscriber as first choice and the
publisher as second choice.
Step 1
Step 2
Step 3
In the Cisco Communications Manager Group Members area, the publisher (CUCM l-.v) should already be intheSelected Cisco Communications Managers
pane.
Highlight the subscriber (CCCM2-X) inthe Available Cisco Communications Manager pane. Use thearrow between Ihe twoboxes to move the subscriber server
to the Selected Cisco Communications Manager pane.
Step 4
Use the Up arrow and Down arrows to place Ihe publisher Cisco Unified Communications Manager at the top ofthe list,making it the primary Cisco Unified
Communications Manager for the group.
Step 5 Step 6
Click Save. You might benotified about the reset ofdevices. Click OK ifa pop-up
window appears.
Click Add New or Copy and repeat theprevious procedure to create a second group
that is called SubPub, which has the two serversthat are listed in reverseorder
(CUCM2-.V before CUCM \-x).
Step 7
Verify thatboth groups areshown in the Find and List Cisco Unified CM Groups
page.
In this section, you will create device pools that arenamed to reflect their Cisco Unified Communications Manager group. Onedevice pool is named Default because it uses thedefault Cisco Unified Communications Managergroup.The other devicepool is called SubPub
because it uses the SubPub Cisco Unified Communications Manager group.
Step 8
Step 9
Step 10
Date/Time Group: CMLocal (use the date, time, and time zone ofthe Cisco
Unified Communications Manager server) Region: Default
SRST Reference: Disable
Step 11
Click Copy.
Step 12
Change the device pool name lo SubPub and the Cisco Unified Communications Manager Groupto SubPub. Leave all other parameters at ihe displayed value and
click Save.
Lab Guide
Activity Verification
You have completed this task when vou attain these results:
There are two Ci^co Unified Communications Manager groups. Uoth include both servers
(in different order).
fhere are two device pools. hach one uses a different Cisco I Inilied Communications Manager group.
Activity Procedure
Complete these sleps:
Cisco Unified Communications Manager Configuration Step 1 fo watchthe registration process per phone, unplug the Hthemet cable from Phone2.v and Phore3-.v. Keep Phone I-.v (the phone to which PC-.t is connected) plugged in.
Step 2 Go to System > Cisco Unified CM and click Find.
Step 3
Step4
first enter 2002 for the ending directory number and then 2001 for thestarting directory numberin the Auto-Registration Information area. Afteryou leavethe endingdirectory number field (for example, by usingthe Tab key) the system will
automatically uncheck the Auto-Registration Disabled on This Cisco Unified
Communications Manager check box.
Step 5
Step7
Step 8
Observe as the IP phone evclesthrough the registration process. Whenthe IP phone has successfully registered, il will display the date, time, and directory number.
On the registered IP phone, press the Sellings button, and then enter 3 on your kevpad lo view the Device Configuration ofthe IP phone. Press I to view the Cisco Unitied Communications Manager settings. Ihe II) ofthe Cisco Unified Communicalions Manager that is designated as the primary Cisco Unified Communications Manager is displayed first, with the word "Active" to the right. I he 11) of the backup (secondary) Cisco Unified Communications Manager is displayed next, with the word"Standbv"to the right.
Plug the Pthernet cable of Phone2-.Y back into the phone.
Step 9
Step 10
Repeat Step 8 for Phone2-v. Press the Settings key and then enter 3 on your keypad to view the Dev ice Configuration, hnlcr I on your keypad or press the Select softkey. Again. \ou will see the primary and secondary Cisco Unified Communic.itions Manager svstems and their status.
Plug the IEthernet cable of Phone3-.Y back into the phone. Observe as the IP phone cycles through the registration process. This IP phone should fail .o obtain the directory number and should be unable to autoregister
Step 11 Step 12
because the autoregistration directory number range was limited totwo directorynumbers.
Step 13
The device pool and IP address ofthe Cisco Unified Communications Manager to which the IP phone iscurrently registered isdisplayed. The IP address ofthe Cisco Unified Communications Manager is listed under the Status column. Note that the
Description field shows "Auto" with the directory number, to indicate that a phone
is autoregistered.
Step 14
Change the descriptions ofthe two registered phones to Phonel-* and Phone2-x. It
Phone I-.t (Cisco Unified IPPhone 7965) does not have directory number 2001
assigned, change the directory number to 200I. IfPhone2-.x (Cisco Unified IP Phone
7965) does not have directory number 2002 assigned, change the directory' number
lo 2002.
Activity Verification
You have completed this task when you attain these results: The first two Cisco IPphones have directory numbers, and you are able to call from one IP
phone to the other.
Place a call between Phonel-* and Phone2-*.
In the Device > Phone menu, afteryou click Find, the twonewly registered IP
phones appear inthe list, asdescribed inthe activity procedure. The third IP phone attempts toregister, but fails with a message that registration is rejected. This rejection happens because autoregistration does not have any directory
numbers left to assign to new phones.
Onthenew page thatis shown, from the Phone Type drop-down menu, choose the actual type of Phonc3-^ (forexample, Cisco 7965). ClickNext. Onthe new page, confirm thatthe SCCP device protocol is selected and click Next.
Step 4
Obtain the MAC address from Phone3-;c. On Phone3-.y, click Settings, then choose
the Network Configuration, and find the IP phone MAC address at entry 3. Another way to obtain the MAC address is to look on the barcode sticker that is
located on the bottom back ofthe Cisco IP phone. Record the MAC address here:
Step 5
Step 6
Click Save. (You will configure additional parameters laterin the course.) Inthe
pop-up window that appears, notify ing you about the reset, click OK.
Step 7
Steps
On the new page, click the Tine 11 JAdd a New DN link tocontinue to configure
the line 1extension. Fnter 2003 in the Directory Number field.
Click Save.
Step 9
Use the Related Links to get back lo ihe device-configuration level. Then reset the
IP phone.
Step 10
Step 11
Observe as the IP phone cycles through the registration process. When the IP phone
hassuccessfully registered, it will display the date, time, anddirectory number. Verifv thai Phonc3-.Y hasthe subscriber (CUCM2-x) as its primary Cisco Unified Communications Manager (in Activestale) and the publisher (CUCM I-.v) as its secondarv Cisco Unified Communications Manager (in Standby state).
Use Cisco Unified Serviceability to stop the Cisco CallManager serviceat the subscriber. Phone3-.v should now show the secondary Cisco Unified Communications Manager (CUCM I-.y) as Active because the primary Cisco Unified Communitations Manager (CI!CM2-.v) is not running the Cisco CallManager
sen ice.
Step 12
Step 13
Activity Verification
You have completed this activity when vou attain these results: Phone3-_Y registers with directory number 3001.
Phone.V.v has the subscriber as its primary Cisco Unified Communications Manager and the publisher as its secondary Cisco Unified Communications Manager.
Task 4 (Optional): Prepare Cisco Unified Communications Manager BAT for Adding Cisco IP Phones
In this task, you will prepare Cisco l'nified Communicafions Manager to use Cisco Unified Communications Manager BAT to add IP phones. Activity Procedure
Complete these steps:
Activate the Cisco Bulk Provisioning Service on the Publisher Step 1 In Cisco I inilied Serviceability, make sure thai the Cisco Bulk Provisioning serv ice is activ ated on the publisher.
Create an IP Phone Template for Use with Cisco Unified Communications Manager BAT
Step 2
Step 3 Step 4
20
Step 5
I.eave all other settings at their default values and click Save.
Step 6
Step 7 Step 8
Click the Line |1] link in the left column, enter linel for Line Template Name, and
click Save.
Highlight BAT-7965 in the Associated Devices pane, and click Edit Device oruse
the Related Links to getback to thedevice-configuration level.
Click the Line|2| link in the left column, enter Iine2 for Line Template Name, and
click Save.
Createthe CSV File Using the Cisco Unified Communications Manager BAT Spreadsheet Step 9 In Cisco Unified Communications Manager Administration, choose Bulk
Administration > Upload/Download Files and click Find.
Step 10 Step 11
Check the check box next tothe bat.xlt file and click Download Selected. In the new: dialog window, click Save, and then navigate tothe C directory onyour
local PC.
Step 12
Step 13 Step 14
Change the filename in the Kile Name field tobat7965.xlt and click Save. Wait until
the file is downloaded to your local PC.
Open the file onyour local PC. using Microsoft F.xcel. On the Phones spreadsheet tab. click tlie Phones radio button, then click Create File
Format.
Step 15
In the new dialog window, highlight the Directory Number inthe Line Fields pane. and then click the Right arrow button to move the entry to the Selected Line Fields
pane.
Step 16
Step 17
Click Create, and then click Yes onthe pop-up window, tooverwrite the existing
file.
In the Maximum Number ofPhone Lines field, enter 2 to create two lines for new IP phones. Then click into any other portion ofthe spreadsheet to leave the field sothat
the sheet is updated with the selected number of lines.
Step 18 Step 19
Check the check box nextlo the Dummy MAC Address field. On the Phones spreadsheet, enter five newIP phones withthe following parameters,
and leave the MAC Address column empty.
Lab Guide
Description
IPPhonee IPPhone" IPPhcneS IPPhoneS IPPhonelO
Directory Number 1
2006 2007 2008 2009
Directory Number 2
3006 3007
3008
3009 3010
2010
Step 20
Step 21
Step 22
Save the new file with the suggested filename lo ihe suggested folder. You should be notified that the file was successfully exported. Click OK.
IXit Microsoft Fxcel without saving the changes to the .xll file.
Step 23 Step 24
Step 25
Step 26
Step 27 Step 28
Choose Insert PhonesSpecific Details from the Select Transaction Type drop
down menu and click Save.
Choose Bulk Administration > Phones > Validate phones. from the file Name drop-down menu, choose the previously uploaded CSV file.
Step 29
Step 30
from the Phone Template Name drop-down menu, choose the BAI-7965 template
that y ou created earlier.
Click Submit.
Step 31
Step 32
Step 33
In thejob list that is shown, click the Job Id link thai has "Validate Specific
Phones'" in the Description column. fhe job results should display the validation status without errors.
Activity Verification
You have completed this task when you attain this result:
The validation status is Validate Completed, and when you click (he Log FileName link, the Result Summary message reads "Validatefor 5 Phones passed. Validate for 0 Phones
failed.""
Note
22
Activity Procedure
Complete these steps:
Step 1
Step 2
Step 3 Step 4
Step 5
Step 6
Choose the previously uploaded CSV file from the File Name drop-down menu.
Choose BAT-7965 from the Phone Template Name drop-down menu. Check the check box next to Create Dummy MAC Address.
Click the Run Immediately radio button.
Click Submit.
Step7
Step 8 Step 9
Ifthe Status column ofthe phone insertion jobdocs not show Completed, click Find
again. Repeat this step until youseethestatus Completed.
Click the phone insertion Job Id link and make sure that the jobhas completed
successfully. You can also reviewthe log file.
Activity Verification
You have completed this task whenyou attain these results: The phones insertion job has completed with success.
This verification step was part of the lab activity.
Note
When you go lo Device > Phone and click Find, you see inthe list the five new IPphones that you bulk-added. Notice that the telephones that you added with Cisco Unified Communications Manager BAT all start with BAT. followed by a dummy MAC address in
the Device Name field. Click one ofthe device names and view the results. You can
To prepare for future labs, follow these steps to delete the Cisco Unified Communications Manager telephones that you added by using Cisco Unified Communicalions Manager BAT.
Step 1 Cio to Device > Phone and click Find.
Step 2
Step3
Check the check boxnext to each bulk-added IPphone (their names start with BAT)
and click Delete Selected. Click OK in the pop-up window.
Make sure that the only IP phones that remain are Ihe three active IP phones (their
names start with SEP).
Lab Guide
Activity Procedure
Complete these steps:
Step 3
Step 4
Step5
For the Phone Iype. choose Third-party SIP Device (Basic) from the drop-down
menu. Click Next.
On the Phone Configuration page for the third-party SIPdevice, add a phone with
these parameters, and click Save:
MAC Address: Fnter any dummy MAC address; third-parly SIP phones do not
register by MAC address, so the value is irrelevant. Description: Phnnc4-.v
Device Pool: Default
Step 6
On the new page, click the Fine [1|Add a new DN link to continue to configure
the line ! extension. Fnter 2004 in the Directory Number field. Click Save.
Configure the Third-Party Phone to Register with Cisco Unified Communications Manager
Step 7 At PC-.v. start the X-f ite SIP softphone. If the SIP Accounts window is not automatically shown, right-click in the phone display, choose SIP Account
Settings, and click Add.
Step 8
Step 9
Step 10
Communications Manager publisher, and then place test calls to other IP phones in
your group.
Wait until the third-party SIP phone registers with the Cisco Unified
Step 11
Step 12
only Cisco IP phones can be reset remotely from Cisco Unified Communications
Manager Administration.
fry to reset the phone from Cisco Unified Communications Manager Administration. The SIP softphone application at the PC is not restarted because
Activity Verification
You have completed this task when you attain these results: On PC-*, the third-party SIP softphone isregistered wilh the Cisco Unified
Communications Manager publisher.
You can place calls to other phones in your pod from the third-party SIP phone and receive
calls from IP phones in your pod.
The third-party SIP phone is listed with its IP address as aregistered phone in the Find and List Phones page. However, the phone cannot be reset from Cisco Unified Communications
Manager Administration. You must manually reboot the phone from the phone.
Activity Procedure
Complete these steps:
Set a Password at the SIP Phone
Step 1
Step 2
Close and reopen the X-Lite application to reset the SIP softphone.
Step 3
Note
Place acall from the SIP softphone. The call should succeed, although you did not
seta digest password at Cisco Unified Communications Manager.
This step shows that a digest password that the phone provides isignored as long as Cisco
Unified Communications Manager is not configured to use digestauthentication forthe thirdparty SIP phone.
Enable Digest Authentication for the Third-Party SIP Phone Step 4 InCisco Unified Communications Manager Administration, go to System > Security Profile> Phone Security Profile andclick Find.
Step 5
Step 6
Click the Copy button tothe right of Third-Party SIP Device BasicStandard
SIP Non-Secure Profile.
Lab Guide
25
Step 7
Activate the Enable Digest Authentication check box. Then click Save.
Step 8 Step 9
Step10
Go to the phone configuration page ofthe SIP phone (go lo Device >Phone). Change the Dev ice Security Profile to Third-Party SIP (Basic) wilh Digest
Authentication, click Save, and reset the phone.
Place a call from the SIP softphone. The call should fail because Cisco Unified
Step 11 Step 12
Step 13 Note
Go to User Management >End User and set Ihe digest authentication password at User4 lo digestpass. In Cisco Unified Communications Manager, the term for digest
password is Digest Credentials.
At PC-.v. close the third-party SIP phone and reopen itto enable the phone to register
again. The phone should now be able to register.
Place a call from ihe SIP softphone. Ihe call shouldsucceed. Cisco Unifed Communications Manager isconfigured to usedigest authentication for the third-party SIP phone, and now thedigest authentication password thatisconfigured at the phone and at thecorresponding end user in Cisco Unified Communications Manager are
identical
Activity Verification
You have completed this task when vou attain these results:
fhe SIP softphone worked with different passwords configured on both ends when Cisco
Unified Communications Manager was not configured touse digest authentication. The SIP softphone did not work with different passwords configured on both ends when Cisco Unified Communications Manager was configured lo use digesl authentication.
TheSIPsoftphone worked with the same passwords configured on both endswhen Cisco Unified Communicalions Manager was configured to use digest authentication.
26
Activity Objective
In this activity, vou will configure Cisco Unified Communications Manager to use aCisco IOS
MGCP gatewav as well as an H.323 gateway to connect to the PS'fN, by using adirect Tl/E I
PRI connection tothe PSTN. After completing this activity, you will be able tomeet these
objectives:
Configure an MGCP gateway using aTl/El PRI to connect to the PSTN in Cisco Unitied
Communications Manager
Configure aCisco IOS gateway for MGCP and obtain MGCP configuration from Cisco
Unified Communications Manager
Visual Objective
The figure illustrates what you will accomplish in this activity.
Configure HQ-x as
MGCP gateway
Required Resources
"fhese resources andequipment arerequired to complete this activity:
Cisco IP phones
Lab Guide
27
Command List
The table describes the commands that are used in this aclivilv.
Description
Displays information about call setup and teardown of ISDN network connections (Layer 3} between the local
router (user side) and the network
Specifies the TFTP serverfrom which the MGCP gateway downloads Cisco Unified Communications Manager XML
configuration files, and enables download of the configuration
Configures the IP address or logical name of the TFTP server from which the XML configuration files are
downloaded
mgcp
show mgcp
Job Aids
Thesejob aids are available to help you complete the labactivity.
MGCP Gateway Information
Call Agent
10x.1 1
Redundant Host
10.x 1 2
Description
Cisco Unified Communications Manager Group
Module
Subunit
Port
in Slot
1
Channel Selection
Order
Data
HQ2-x IP address
Add an MGCP Gateway in Cisco Unified Communications Manager Step 1 InCisco Unified Communications Manager Administration, go to Device > Gateway. On the new page that is shown,click Add New.
Step 2
Choose the gateway platform (for example, Cisco 2811) thai isused for Cisco IOS MGCP gateway HQ-* (where x isyourcluster number) from theGateway Type
drop-down list and click Next.
Step 3
From the Protocol drop-down list, choose theprotocol type MGCP and click Next.
Configure an MGCP Gateway in Cisco Unified Communications Manager Step 4 Hnter the following parameters in theGateway Configuration window, then click
Save:
The name has to match the hostname of the router and is case-sensitive.
Description: UQ-x MGCP gateway Cisco Unified Communications Manager Group: Default
>2010 Cisco Systems. Inc.
Lab Guide 29
Global ISDN Switchtype: NI2 Caution Thesesteps are platform-dependent. Ask your instructor todetermine the actual hardware
that is used for the MGCP gateway This lab guide is based on the Cisco 2811 router
platform with T1/E1 interfaces You can usetheshow version, showdiag, and show inventory commands tosee details about thehardware that isactually used in your lab
environment
Add MGCP Endpoints by Selecting Modules and Voice Interface Cards Step 5 In the Configured Slots. VICs and F.ndpoints pane, from the Subunit 0 in Slot 0,
choose module VWIC2-1MFT-T1E1-TI. Click Save.
Step 6
Click the port icon 0/0/0 (ihe far-left endpoint with the question mark).
Step 7
Step8
In the ne.\t window, from the Device Proloeol drop-down list, choose Digital Access
PRI. and click Next.
Step 9
Activity Verification
Vou have completed Ihis task when vou attain litis result:
fhe MGCP gateway appears in the listwhen you choose Device > Gateway and click find.
Access the Cisco IOS MGCP Gateway Step 1 Open a Telnet session lo reach your MGCP gateway I IQ-v (II* address 10..T.250.101). Log in and changeto enable mode. The login password and enable
secret password are cisco.
Step 2
In the enable mode, enter the show running-con fig command. No MGCP commands are eurreotlv configured on the gateway.
The Cisco :OS commands shown in this lab exercise depend on the hardware that is used in your class Ask your instructor for any changes based on the actually used hardware
Note
Configure the Cisco IOS MGCP Gateway to Use the Configuration Server Method Step 3 In global configuration mode, enter ihe following commands:
card type cl 0 0
net work-;: lock-participate wic 0 30 Implementing Cisco Unifed Communications Manager, Part 1 (CIPT1) vB0 2010 Cisco Systems, Inc.
The gateway will now pull its MGCP configuration from the Cisco Unified Communications
Manager TFTP server.
Step 4 Enter the show running-conf.g command. You should see more than the two
configured MGCP commands.
configuration mode,
Step 6 In global configuration mode, enler the following commands to shut down the voiceport associated with the Tl PRI.
voice-port 0/0/0 :23
shutdown
Step 7 Step 8
no pri-group timeslots 1-24 service mgcp pri-group timeslots 1-8 service mgcp
no shutdown
Step 9
Note
As you deactivated the configuration server feature, the MGCP process at the Cisco IOS gateway is not automatically reset anymore when you reset the gateway in Cisco Unified
Communications Manager. You have to manually reset the MGCP process atthe Cisco IOS
gateway every time after you reset the gateway in Cisco Unified Communications Manager.
Enter the command no mgcp command followed by mgcp command in order to reset the
MGCP process at theCisco IOS router.
Activity Verification
You have completed this task when you attain these results:
You can verifv that your MGCP gateway has successfully registered lo the Cisco Unilied
controlled Tl/El PRIendpoint channels are up.
Lab Guide
Communications Manager by using the show ccm-manager hosts command (Status shows
as Registered) and by using the show mgcp endpoint command to check that the MGCP-
Verity Hie MGCP status by using the show mgcp command. The Admin Stale and the
Operational State are Active.
Verifv that the MGCP gateway and the MGCP endpoints are registered at Cisco Unified
Communications Manager:
Step 1
Step 2
Step 3
In the Find Gatewav ssection, choose Show Kndpointe and click Kind.
Step 4
The status ofthe MGCP galewav endpoint should be "Registered with 10 vI I"- the
galewav IP address should be IO.t.I.IOL
In this task, vou will configure Cisco Unified Communications Manager lo route calls that start
Step 2
Step 3 Step 4
Click Save. You are notified bv a pop-up window thai ihe authorization code will
not he activated. Click OK,
Click OK.
You arc notified by another pop-up window that changes will reset the galewav
b
.-
Activity Verification
You have completed Uiis task when vou attain these results:
from aCisco IP phone in your pod. you can successfully reach (he PSTN emergency
number (pan ofthe simulated PST N). Specifically, perform these steps:
At PC-.v. si art Cisco IP Communicator. Step 1
Step 2
Cisco IP Communicator should display PSTN-Phone,v in the top-right corner ofits display. On top ofthe softkeys. Cisco IP Communicator should display PS 1N Phone. IfCisco IP Communicator does not display this information, ask your
instructor lor help.
Step 3
Step 4
At the PSTN phone (Cisco IP Communicator running on PC-.v), accept the call. This
call was sent through your IIQ-.v PSTN gateway.
32
In this task, you will configure Cisco Unified Communications Manager to use the H.323
proloeol toward the H.323 galewav HQ2-* in your cluster.
Note The H323 gateway will be added to Cisco Unified Communications Manager only to show the necessary configuration steps. The gateway is not physically present. It isonly added in
Cisco Unified Communications Manager andwill be used in thenext task todemonstrate
that if the preferred gateway (HQ2-x) isnot running, a backup gateway (HQ-x) can be used.
Activity Procedure
Complete these steps:
Step 1
Step 2
Choose H.323 Gateway from the Gateway Type drop-down list, and click Next.
Configure an H.323 Gateway in Cisco Unified Communications Manager Step 3 Enter the following parameters in the Gateway Configuration window, then click
Save:
Device Name: lO-v.1.102 (where x is your pod number) Description: UQl-x H.323 gateway
Device Pool: Default
Activity Verification You have completed this task when you attain this result:
"fhe H.323 gateway appears in the list when you choose Device >Gateway and click Find.
Cleanup
Delete the route pattern that you created inthis lab exercise. In the next lab, you will create route patterns that refer toa route list with redundant gateways. To delete the route pattern,
follow these steps:
Step 1 Step 2
Goto Call Routing> Route/Hunt >Route Pattern, and click Find. Select thenewly created route pattern 911 and click Delete Selected.
Lab Guide
33
Activity Objective
In this activ ity. you will configure Cisco Unilied Communicalions Manager for PSTN calls that use multiple gateways. Alter completing this activity, you will be able to meet these objectives: Configure a route plan that includes a route group, roule list, and route pattern that enables
calls to Ihe PSTN
Hnable Cisco Unified Communicalions Manager Dialed Number Analyzer and use it for
dial plan verifkaiion
Visual Objective
1he figure illustrates what you will accomplish inthis activity.
Lab 4-2: Configuring Cisco Unified (
Configure a route
Manager Dialed
Required Resources
"fhese resources and equipment are required to complcle this activity:
Cisco IP Phones
Job Aids
This job aid is available tohelp you complete the lab activity.
PSTN Numbering Plan
Pattern
911
Destination
Description
PSTN phone PSTN phone PSTN phone PSTN phone PSTN phone
PSTN phone
Emergency number
XXX xxxx
011<as many as 14
digits>
1 800 XXX XXXX
Note
When dialing from the Cisco Unified Communications Manager cluster tothe PSTN, a PSTN
access code 9 must be prefixed to all PSTN numbers. This PSTN access code should not
be sent to the PSTN.
Step 1 Step 2
Step 3
Step 4
Navigate to Call Routing >Route/Hunt >Route Group and click Add New. Enter the following parameter in the Route Group Information window:
Route Group Name: RG_MGCP_GW
hi the Available Devices field, choose the UQ-x MGCP gateway identifier
(SO/SCO/DSl-0>HQ-a) and click Add to Route Group to add the HQ-.t gatewav
Click Save.
Step 5
Step6 Step7
StepS
In the Available Devices field, choose the HQ2-.* gateway identifier andclick Add
to Route Group to add the IIQ2-* (10x1.102) gateway.
Click Save.
Lab Guide
35
Step 10
Step 11
Click Save.
Step 12
Step 13
In the updated Route I ist Configuration page, click Add Route Group inthe Route I isl Member Infonnation pane and choose the RG_MGCP_GVV route group.
Choose Discard Digits Instruction NA.NP:PreDot as Called Partv Transformation
foriheRG MGCPGW.
Step 14
Step 15
Repeat Steps 12 and 14 for the RG_1I323__GW route group, hut do notchoose anv entrv in the Discard Digit Instructions for the route group.
Typically the PSTN accesscode isstripped at theH.323 gateway using Cisco IOS digit
manipulation features. Therefore, you have lo send PSTN numbers with PSTN access code 9 to the Cisco IOS gateway.
Note
Step 16
Step 17
Change the orderofthe route groups in the route listso lhatroute group RG_H?23. (iW is listed before routegroup RG MGCP_GW.
Click Save. Click OK in the pop-up window.
Step 19
Note
Step 20 Step 21
Step 22
Click Save You arc notified by a pop-up window thai the authorization code will
not be activated. Click OK.
You are notified bv another pop-up window that changes will reset the galewav.
Click OK,
Repeat the Steps 16 through 19 to create another route pattern, but enler 9.! as the Route Pattern parameter and enter PSTN Access with Interdigit Timeout as the
Description.
Note
You can use the Copy icon to create a copy of the existing route pattern, which you can
modify as required
Activity Verification Youhave completed thistask when you attain these results:
From a Cisco IPphone in your pod, you can successfully reach a PS'fN number using the prefix code 9 (for example. 9-555-7890). When dialing the number, you can either wait for the interdigit timeout toexpire orpress the ft key to instruct Cisco Unified Communications Manager tostop digit collection. Verify that the call isreceived at PSTN-Phone-.r and
accept the call at the PSTN phone.
Task 2 (Optional): Enable Cisco Unified Communications Manager Dialed Number Analyzer and Use It for Dial Plan
Verification
Inthis task, you will install Cisco Unified Communications Manager Dialed Number Analyzer and use it to verify that gateway HQ2-x is preferred overgateway HQ-x for callssentto the
PSTN.
Activity Procedure
Complete these steps:
Activate Cisco Unified Communications Manager Dialed Number Analyzer Service Step 1 Access Cisco Unified Serviceability, and go to Tools > Service Activation. Choose ihepublisher server IOjc.1.1 from the Server drop-down list. ClickGo.
Note The Cisco Unified Communications Manager Dialed Number Analyzer service must be activated on the publisher server. The service can also be activated on the subscriber server, in case the Cisco Unified Communications Manager Dialed Number Analyzer service on the publisher is unavailable. Activatingthe service on the subscriber only does not enable the Cisco Unified Communications Manager Dialed Number Analyzer web page.
Step 2
Note
ChooseCisco Dialed Number Analyzer from the CM Services list, and click Save.
Ifthe service is already activated, the ActivationStatus displays as Activated.
Step 3
fhe service is activated and the Activation Status column displays the status as
Activated.
Access Cisco Unified Communications Manager Dialed Number Analyzer Step 4 To access Cisco Unified Communications Manager Dialed Number Analyzer, go lo Tools > Dial Number Analyzer in Cisco Unified Serviceability.
Step 5 A pop-up window appears. If you are prompted to enter the username. enter cucmadmin as the user ID and cucmpassi for password. Click OK. You are now logged in to Cisco Unified Communications Manager Dialed Number Analyzer.
You can also use the URL https://10.x.1.1/dna to access Cisco Unified Communications
Note
Manager Dialed Number Analyzer. You do not need to access it from Cisco Unified
Serviceability.
Perform a Simple Analysis by Using the Analyzer Window Step 6 Navigate to Analysis > Analyzer.
Lab Guide
Step 7
Step 8
fnter ihe calling-party digits 2002 in the Calling Party field. By default, 1000
displays in this field.
filter ihe digits 9911 (or am other valid PSTN number starting with 9). in the
Dialed Digits field.
Step 9
Click Do Analysis to start the analysis. Cisco Unitied Communications Manager Dialed Number Analyser analyzes the dialed digits and displays the results in a new
window, the DNA Analysis Output window. In the DNA Analysis Output window, click Kxpand All. Under Call Flow, at the end of the Route List output section, the gateways are listed in the order in which
thev are used to route the call.
Step 10
Activity Verification
You have completed ihis task when you attain these results:
In Tools > Control CenterFeature Services, the Cisco Unified Communications Manager
Dialed Number Analyzer sen ice is activated and started.
Front a Cisco IP phone in vour pod. vou can successfullv place a call to the PS'fN phone by dialing a valid PS IN number (starling with prefix digit 9). You performed call analysis by using Cisco Unified Communicalions Manager Dialed Number Analyzer, fhe output shows that gateway 11Q2-.Y is preferred for calls to the
PSTN.
Note
Cisco Unifed Communications Manager Dialed Number Analyzer is not aware that the H 323 gateway (HQ2-x) is not running and therefore considers the primary path only. However, because you were able to place the call to the PSTN (using the backup gateway HQ-x). yoL successfully verified your route list and route group configuration
Cleanup
Remove the PreDot d:git stripping instruction at the Route List for the RG_M(iCP_GW Roule Ciroup. Step 1
Step 2 Step 3
Navigate to (all Routing > Route/Hunt > Route L.ist and click Find.
Choose the RL HQ C\\ s route list. Click the RC-_M(;CP_(;\V link in the Route List Details Held.
Step 4
Step 5
Verify Cisco Unified Communications Manager configuration for stripping access codes
lhat are used for called numberson outgoingcalls
Configure Cisco Unified Communications Manager to extend calling numbers for outgoing
callsto fully qualified PSTN numbers
Visual Objective
The figure illustrates what you will accomplish in this activity.
to be placed to numbers in
+ format
Translate called
number of incoming
PSTN calls tram 52x555-3XXX to 2XXX
Change callingnumber of
outbound PSTN calls from
Required Resources
Theseresources andequipment are required to complete this activity:
Cisco IP Phones
Lab Guide 39
Job Aids
Ihis job aid is av aikble to help you complete the lab activ ity.
Transformation Masks (Where x Is Your Pod Number)
Cluster Name
Pod*
IP Phones
Note
Be aware that the DID number range is different to the internally assigned directory
numbers For example, to reach Phone1-1 (2001) from thePSTN, the DID number would be
5215553C01
Step 2
Step 3 Step 4
Steps
Go to Call Routing >Class of Control >Calling Search Space, and click Add
New.
Liner MGCP calledjrans as the CSS name, and move the M(iCP_called trans
partition to the Selected Partitions field.
Click Save.
Step 6
Step 7
Step 8
Define the pattern 9.! and choose MGCP called trans from ihe Partition drop
down li^t.
In the Discard Digit Instructionsdrop-down list, choose PreDot and click Save.
Step9 Step 10
Click Add New and then repeal Step7 and Step 8 fordefining Transfonnation
Pattern 9.!?.
Click Add New and then repeat Step 7 and Step8 defining Transformation Pattern
911 without any digit manipulation.
Note
Tne 911 Transformation Pattern is required to avoid matching the 9.!Transformation Pattern
where the leading 9 is stripped off.
Step 11
Step 12 At the listed MGCP Gateway (HQ-*). open the See Endpoints link and open the
displayed endpoint.
Step 13 Scroll down to the Call Routing InformationOutbound Calls field and uncheck
I se Device Pool Called Party Transformation CSS.
Step 14 Step 15
Access the HQ-.v router via telnet and reset the MGCP process using the no mgcp
Step 16
Note
At gateway HQ-.v. enter the debug isdn q931 command in enable mode.
As you access HQ-x via Telnet, do not forget to use the terminal monitor command to see
the debug output.
Step 17
From one of your IP phones, dial aPSTN number and verify that the dialed PSTN
access code 9 is not sent to the PSTN.
Activity Verification You have completed this task when you attain this result:
At gateway HQ-.v. you see that the access code was removed from the called number.
Task 2: Configure Cisco Unified Communications Manager to Extend Calling Numbers in Outgoing Calls to Fully Qualified
PSTN Numbers
In this task, you will configure external phone number masks or transformation masks to extend
directory numbers to fully qualified PSTN numbers.
Activity Procedure
Complete these steps:
Step 1
Step 2
Step 3
Step 4
Step 5
At the Phone Configuration page, click Line Il| to get to the Line Configuration
page ofthe phone.
Step 6
Step 7
Note
Go toCall Routing > Route/Hunt> Route Pattern and click Find. Enter the Route Pattern Configuration window by clicking the route pattern 9.!. In the Calling Party Transformations pane, check the Use Calling Party's External
Phone Number Mask check box.
Lab Guide
Step 11
Click Save.
Step 12
Step 13
Place acall to the PSTN phone from one ofyour IP phones. ISefore accepting the call at the PSTN phone, verifv thai the calling number is shown as a 10-digit PSTN
numberrather than a 4-digitextension.
Activity Verification
You have completed this task when vou attain ihis result:
On outgoing PSTN calls, the calling number is shown as ]0-digil PS'fN number (52r5553XXX). This number can be verified at the PS fN phone that receives the call orbv examining the debug isdn q93I command output at the gateway (IIQ-.v).
Step 1
Try placing a call from the PSTN phone to Phone I-x (2001) by dialing I-52.T-5553001. The call will fail. When looking at debug isdn q931 output at HQ-x you will realize thai the PSTN tries lo set up the call to52*5553001. Cisco Unified ' Communicalions Manager rejects the call because no phone is configured with such a long number: internally, four-digit directory numbers are used. Also, the internally
assigned direetorv number range2XXXdoes not match the extensions used bv DID
calls (3XXX).
Step 2
Toallow incoming calls from the PS'fN. goto Call Routing >Translation Pattern
and click Add New. This translation pattern should translate called numbers for calls
coming from the PSTN from 10-digil PSTN numbers tothe 4-digit directory
number.
Step 3 Step 4
Step 5
In the Translation Pattern field, enter 51v55S3\XX (where x is your pod number). In the Description field, enter Translation of incoming 10-digit PSTN calls.
Make sure that the Provide Outside Dial Tone box is unchecked.
Step 6
Step 7
Step 8
Click Copv to create a copy ofthe translalion pattern. This translation pattern should translate called numbers for calls coming from the PSTN from seven-digit PS'fN
numbers lo the four-digit directory number.
Change the I ranslation Pattern field lo 5553XXX.
Step 9
Step 10
Step 11
42
Note
When internal directory numbers match the DID extensions, setting the significant digits at
the HQ-x gateway to 4would be easier. This setting will trim the called number on incoming
calls to the last four digits. ___
Step 12 At the Translation Pattern Configuration page, click Add New to create another
Step 13 In the Translation Pattern field, enter 2XXX.
Step 14 Step 15
Step 16
Step 17
In the Description field, enter Translation for Unassigned Directory Numbers. Make sure lhat the Provide Outside Dial Tone box is unchecked.
In the Called Party Transform Mask field, enter 2002.
Click Save.
Activity Verification
You have completed this task when you attain this result:
You can place calls from the PSTN phone to the IP phones in your pod.
Dial 555-3009. Phone2-* rings because 2009 isan unassigned directory number and
therefore, after perfonning the first translation from 5553009 to 2009. the called
number matches translation pattern 2XXX, which translates the called number to
2002.
Note
Depending on the line that you use at the PSTN phone, calls are received from different calling numbers. However, all calling numbers use the PSTN format, so they do not include
access code 9 (or 1for long-distance numbers), which are required for callback from call lists. The displayed number in call lists must be edited to be able to call back a number.
Task 4: Configure Cisco Unified Communications Manager to Prefix Access Codes to Calling Numbers on Incoming PSTN
Calls
In this task, vou will configure Incoming Calling Party Settings on the MGCP gateway, so that received ormissed PSTN calls can be called back from call lists without the need toedit the
number.
Activity Procedure
Complete these steps:
Step 1 Step 2
Step 3
Co to Device > Gateway and click See Endpoints atthe IIQ-.* MGCP gateway. Select the shown device name and scroll down to the Incoming Calling Party
Settings pane.
For the National Number Type enter 91 atthe Prefix Held, for Ihe Subscriber Number Type enter 9 atthe Prefix field, and for the International Number Type
enter 9011 at the Prefix field.
Lab Guide 43
Step 4
Step 5
Also reset the MGCP process at the IIQ-v router by issuing the no mgcp followed
bv the mgcp command in global eon figuration mode.
Activity Verification You have completed this task when vou attain ihis result:
You can place acall from the PSTN phone lo one ofthe IP phones in jour pod. 1romjine 2(national) ofthe PSTN phone, dial the long-distance numberof Phone 1
r (1-52.V-555-3001). You see thai Ihe call isreceived from 916065554444. From line 1(local) ofthe PSTN phone, dial Ihe local number of Phone2-.T (5553002). You see that the call is received from 95554444.
In Ihis task. >on will add an F. I64 phone number uiih a+prefix lo your personal direetorv and
then locah/e the number when sending the call to the PS'fN.
Activity Procedure
Complete these steps: Stepl
Step 2
Step3
At the Dev ice Information pane click the Device Association, button, then check
Step 4
Steps
Step 6
Step 7
Step 8
Step 9
Step 10
Step 11
Step 12
Step 13
Activity Verification
You have completed this task when you attain these results: The call rings on the international line on the PSTN phone.
Step 1
Step 2
Step 3
On Phonc3-.t. press the Directories button and open the personal directory.
Login I ser3 and select Personal Address Book.
Press the Submit softkey to display the previously created contact entry and dial the
number that is associated with it.
Step 4
Observe the localization ofthe called E.164 number while the call rings atthe
nternational PSTN line.
Lab Guide
45
Activity Objective
In this activity. \ou will implement calling privileges in Cisco Unitied Communications
Manager. Afler completing this activity, vou will be able to meet these objectives:
Configure partitions and CSSs
Implement CoS i'or internal calls
Implement CoS ,br outgoing PS'fN calls Implement CoS lor incoming PSIN calls
Implement time-of-dav-based CoS Implement FAC
Visual Objective
'fhe ligure illustrates what vou will accomplish inthis aclivilv.
Required Resources
Ihcsc resources and equipment arerequired to complete this activity:
Cisco L'nified Communications Manager cluster
Job Aids
These job aids are available to help you complete the lab activity.
Partitions
Partition Name
Description
Lobby phones
Assigned to Devices
2001 2002 2004
Lobby-Phones
Phones
Translation patterns:
Manager phones Description PSTN: Local and long distance PSTN: International, office hours only
PSTN: International, FAC
2003
9.1I2-9]XX[2-9]XXXXXX
9.011!
PSTN-lntl ToD
9.011!#
PSTN-lntl FAC
9.011! 9.011!#
911
PSTN-Emergency
PSTN: Emergency
9.911
PSTN-Free
9.1[800]XXXXXXX
CSS Name
Contains Partitions
Phones
Lobby IP phone
(2001)
Lobby_css
PSTN-Emergency
Phones ess
Employee IP phone
(2002, 2004)
Lobby-Phones
Phones
Manager-Phones
PSTN-Emergency
PSTN-Free
PSTN-Local_LD
PSTN-lntl ToD
Manager IP phone
(2003)
Manageress
Lobby-Phones
Phones
Manager-Phones
PSTN-Emergency
PSTN-Free
PSTN-Local_LD PSTN-lntl_ToD
PSTN-lntl FAC
Lab Guide
47
Devices
CSS Name
Description
Contains Partitions
Phones
To-Phones_css
patterns
Route Patterns
Route Pattern
911 9911
Route Partition
Description
Gateway/Route List
RL HQ GWs
PSTN-Emergency PSTN-Emergency
PSTN-Free
PSTN Emergency
PSTN: Emergency
PSTN Toil free
PSTN Local
RL_HQ_GWs
RL_HQ_GWs RL_HQ_GWs RL_HQ__GWs RL_HQ_GWs RL_HQ_GWs RL_HQ_GWs RL_HQ_GWs
9 1800XXXXXXX
9[2-9]XXXXXX 9 1[2-9]XX(2-9jXXXXXX
90111
9 011 !#
901V
9 011!#
PSTN-lntl_FAC
Activity Procedure
Complete these steps: Configure Partitions
Step 1
(io to Call Routing > Class of Control > Partition, and click Add New.
Step 2
Using the partition conllguration data from ihe "Partitions" table (in the Job Aids
section ofthis lab exercise), enter all partition names and their descriptions, using
this format:
<partit.LonName> , <description>
<parr_it-.onName> , <description>
Step 3
Click Save
Configure CSSs
Step 4
(io to Call Routing >Classof Control >Calling Search Space, and click Add New to open the CallingSearch Space Conllguration window. Using the-Calling Search Spaces" table(in theJob Aids section of this labactiv in ).
enter the first CSS name and description.
Step 5
Step 6
Using the information in the "Calling Search Spaces" table column Contains Partitions "highlight the appropriate partitions in the Available Partitions pane and
add them to the Selected Partitions by using the Down arrow. To remove a partition from the list ofSelected Partitions, highlight the partition and click the Up arrow.
Note
Step 7
Use the Shift key to highlight multiple contiguous entries and the Ctrl key to choose multiple
noncontiguous entries.
Click Save.
Step 8
Note
Repeat the previous steps lo create the remaining CSSs that are listed in the "Calling
Search Spaces" tabic.
When you configure the Manageress CSS, make sure that the PSTN-lntl_ToD partition is
setbefore the PSTN-lntl_FAC partition. This order isimportant when time-of-day routing is
implemented and aFACmust be entered if international calls are placed outside business
hours.
Activity Verification
You have completed this task when you attain these results:
When you navigate to Call Routing >Class of Control >Partition and click Find, you see
all tlie partitions thatyou added.
When you navigate to Call Rouling >Class of Control >Calling Search Space and click
Find, vou see all the CSSs that you added.
Note
Activity Procedure
Complete these steps:
Assign Partitions and CSSs to IP Phones Step 1 Goto Device > Phone andclick Find.
Step 2 Step 3
Step 4
Step 5
Click the IP phone wilh the directory number 2001, to open the Phone Configuration
page.
Click Line |1|2001 from the left page column, to access the Directory Number
Configuration page.
Choose Lobby-Phones for the Roule Partition, and in tlie Directory Number
Settings, choose Lobby_css for the Calling Search Space parameter.
Click Save.
Step 6
Repeat Steps I through 5for the other three IP phones in your cluster. Refer to the
tables "Partitions" and "Calling Search Spaces" inthe Job Aids section to locate Ihe
partition and the CSS configuralion parameters that are applicable lo these
remaining IP phones.
Lab Guide
Step 7
(io to Device >Phone and click Find, then choose all four IP phones and click Reset Selected. Click Reset again, and click Close in the pop-up window.
Activity Verification
You have completed this task when you attain these results:
fhe lobby IP phone (2001) is able to call only employee IP phones (2002 and 2004) and
cannot dial the manager IP phone (2003).
Note
At ,hls lime- the translation pattern 2XXX will match for a call from 2001 to2003.
The employee IP phones (2002 and 2004) are able to call all other IP phones.
The manager IP phone (2003) is able tocall all other IP phones.
Step 2
Step 3
Click the translation pattern 2XXX to open ihe franslation Pattern Configuration
window.
Step 4
Step 5
Step 6
Repeat the pre\ ious steps for the 52y5552XXXand the 5553XXX translation
patterns.
Assign Partitions and CSSs to Gateways Step 7 Choose Device >Gateway, and click Find to lisl all the gateways. Step8 Note At the HQ-x MGCP gateway, click the Sec Kndpoints link in the Slatus column. Each endpoint ofan MGCP gateway iscontrolled separately, therefore CSSs areapplied
individually per endpoint and not at the gateway level
Step 9 Step 10
section, inwhich the CSS can beconfigured. Set the appropriate CSS according, to
the job aid tabic.
Step 11
Note
You also have to reset the mgcp process directly atthe HQ-x gateway by issuing no mgcp command followed by mgcp command in global configuration mode. __
Activity Verification
You have completed this task when you attain these results:
The PSTN phone is able to call employee phones (555-3002 or 555-3004). Ifthe PSTN dials the manager or lobby IP phone (555-3001 or 555-3003), the call is sent to
the attendant (2002).
Ifan unassigned directory number is dialed from the PSTN phone (for example, 555-3010),
the call is sent to the attendant (2002).
In this task, vou will configure Cisco Unified Communications Manager to provide CoS for
Activity Procedure
Complete these steps:
Create and Update Route Patterns for CoS inCisco Unified Communications Manager In this step, you will create different route patterns for different call types (emergency, toll-free, local, long-distance, and international). The existing route patterns (.!# and 9.!) will be changed to emergency route patterns; all other route patterns will be added.
Step 1
Step 2
Step 3
Step 4
Change the route pattern to911 and the description to PSTN Emergency.
Set the partitionto PSTN-Emergency.
Step 5
Note
Step 6
Step 7
Step 8
Repeat the previous steps but change the 9.!# route pattern to9.911. Set the partition to PSTN-Emergency, change the description to PSTN Emergency, and check the
l/rgcnt Priority check box. Donot forget to save yourchanges.
Click the Copy button in the Route Pattern Configuration window and update the following parameters with the values of the next route pattern that isshown in the
"Route Patterns" table ofthe Job Aids section.
Route Pattern: 9.1800XXXXXXX Route Partition: PSTN-Free
Lab Gulde
Note
Step 9
You do not need to configure PreDot digit stripping at the route pattern because Ihis is
performed directly at the gateway, using the called-party transformation CSS
Step 10 Note
Repeat the last two steps tocreate ihe remaining route patterns as described in the
"Route Patterns" table in the Job Aids section.
If you decide to add the new route patterns from scratch instead of copying them from the
existing route pattern, make sure that you click the Use Calling Party's External Phone
Number Mask check box at all route patterns thatyou add
Activity Verification
You have completed this task when vou attain these results:
When \ on go loCall Routing > Route/I lunl > Roule Pattern and click Kind, vou see the newly creaied route patterns in the appropriate partilions.
(9-9II or 9 N).
The lobby IP phone (Phoncl-.v) can dial onlv one PSTN destination: emergenev numbers
fhe employee IP phones (Phone2-.v and Phone4-.r) and the manager IP phone (Phonc3-.v) are able to call all supported PS'fN destinations (for example, local: 9-555-4444. long distance: 9-1-666-555-4444. international: 9-011-43-555 4444. emergency : 9-911 or91!
and toll free: 9-1-800-555-4444).
Note
At this stage, ignore the route pattern in partition PSTN-lntl_FAC. That pattern is irrelevant here because it isidentical tothe route pattern PSTN-lntl_JoD Also, note that partition PSTN4ntl_ToD does not have a time schedule that isapplied at this point and therefore is
always active inthe CSSs that includethis partition.
Step 1
Go to Call Routing > Class of Control > Time Period and click the Add New
button.
Step2
Step 3
Click Save.
Note
The requirement in this task is to permit calls during business hours, such as 9:00 a m. (0900) to 500 p.m. (1700), as indicated in the name ofthe time period. However, to easily
simulate a call that is placed within business hours and a call that is placed outside business hours, the period is first set to 24 hours, repeated from Monday to Sunday. This will ensure that the call will match the configured time range and allows the simulation of calls that are
placed within business hours. Then, the time period will be changed to avery small time
slot, which will not match the current day and time. This allows the simulation of calls that are placed outside business hours. In reality, you would configure something like 9:00 a.m (0900) to 5pm. (1700) hours, Monday to Friday, to achieve the requirement of limiting
international calls to business hours.
Create a Time Schedule
Step 4 Step 5
Go to Call Routing >Class ofControl >Time Schedule and click Add New. hi the Time Schedule Configuration window, enter BusinessHours in the Name
field, and click Save.
Step 6
Step 7
Move the 9a-5p_Mo-Fr time period from the Available Time Periods pane in ihe
Selected Time Periods.
Click Save.
Assign the Time Scheduleto the ToD Partition Step 8 Navigate to Call Routing >Class ofControl >Partition and click Find.
Step 9 Click the partition with the name PSTN-Intl_ToD.
Step 10
Step 11
In the Partition Configuration window, choose the newly created time schedule
BusinessHours in the Time Schedule field.
Click Save.
Activity Verification
Employee phones can dial international destinations during business hours only. To verify
yourconfiguration, follow thesesteps:
Stepl
From Phone2-.x orPhone4-.x, dial any international number (for example. 9-011-43555 4444). Al the moment, the time period isconfigured tocover ihe whole week (0000-2400 hours, Monday toFriday). Because the call isplaced within the specified time, itisconsidered to be within business hours and therefore is
permitted.
Step 2
To simulate acall outside business hours, go to Call Routing >Class ofControl >
Time Period and click Find. ChooseIhe 9a-5p_Mo-Fr time period and changethe
Step 3
time range tovalues that do not include the current time as displayed at your IP phones (for example, 00:00-00:15, Monday to Monday), and click Save. Try placing an international call again from Phone2-x orPhone4-jc. Because the call isnot placed within the specified time, it isconsidered to be outside business hours
and is denied.
Lab Gulde
Note
At this stage, manager phones have access to international route patterns mpartition PSTNlntl_ToD, which is now limited to business hours and in partition PSTN-lntl_FAC, which is not limited atall atthis point If partition PSTN-lntLJoD is removed from the manager phone CSS the route pattern is still visible from the PSTN-lntl_ToD partition Therefore, manager
phones can still call international destinations at anytime.
Step 1
Go to Call Routing > Forced Authorization Codes and click Add (New.
Step 2
Step 3
Click Save.
Step 4
Repeat the previous steps to add another FAC with name Too I.ow I evel code
9998. and level 4.
Step 5
Go to Call Routing > Route/Hunt > Route Pattern and click Find. From the
displayed list, click the 9.011! route pattern, which is in the PSTN-lntl_FAC
partition.
Step6
In the Route Pattern Configuration window, check the Require Forced Authorization Code check box. Set theminimum required Authorization I.evel to
5.
Step 7
Click Save.
Steps
Repeat the prev ioussteps for the 9.011 lit route pattern in the PSTN-lntl FAC
partition.
Activity Verification
You have completed this task when you attain these results:
Outside business hours, manager phones should be able to dial international destinations
only afterentering a valid FAC with a high-enough authorization level. To verify your
configuration, follow these steps:
Step 1
Dial an international destination (for example. 9-011-43-555 4444) from the manager IP phone. Assuming thai vour lime period configuration is still set lo a rangethat does not includeIhe currentday and time, a beepshould be heardand "Inter .Authorization Code" is displayed on the IP phone, indicating thai an FAC
needs to be entered.
Step 2
Step 3
Dial the same number again, but this time enter FAC 9998 followed by #. The call
should be denied because the authorization level configured for FAC 9998 is not
high enough for the matched route pattern.
Step 4
Dial the same number again, but this time enter an invalid FAC (for example, 9997)
followed by U, The call should bedenied.
Step 5
Change the time period back to 00:00 to 24:00. Monday to Friday (as described in
the previous task).
Step 6
Dial the same number again. This time there will be no prompt for an FAC. The CSS of the manager phone has partition PSTN4ntl_ToD listed before partition PSTNlntl_FAC. If the ToD partition is active {that is, during business hours) aroute pattern that does not require an FAC is matched. If the call is placed outside business hours, the partition PSTN-lntl_ToD is not active in the phone CSS and therefore the route pattern that
is in the PSTN-lntl_FAC partition is matched. This route pattern requires an FAC to be entered. If the partitions were configured in the wrong order <PSTN-lntl_FAC before PSTNlntl_ToD) in the CSS, the manager phone would always require an FAC to be entered, even
during business hours. _^
Note
Lab Guide
55
Activity Objective
In this task, you will configure hunt groups that consist ofline groups, hunt lisls. and a hunt pilot number with internal and external forwarding sellings for busy, no-answer, and no-callcoverage conditions. After completing ihis activity, you will be able to meet these objectives: Configure call hunting, including line groups, hunt lists, and hunt pilots
Configure final forwarding on hunt exhaustion
Visual Objective
fhe figure illustrates what you will accomplish in this activity.
Configure line groups, hunt list, hunt pilot, and final forwarding
Required Resources
Ihese resources and equipment arerequired to complete this activity:
Cisco Inificd Communications Manager cluster
Step 1
Step 2
Choose Call Routing >Route/Hunt >Line Group and click Add New.
Fnter IstLG in the Line Group Name field.
Step 3
For now, leave the distribution algorithm at the default (Longest Idle Time). Leave the hunt options for Busy. No Answer, and Not Available at their default values
(Try Next Member; Then, Try Next Group in Hunt List).
Step 4
Note
Change the Ring NoAnswer Reversion (RNAR) Timeout from its default value of
10 seconds to 5 seconds.
This value might not beappropriate in a call center environment, however, in a classroom or
test environment, a shorter timeout enablesyou to validate call-distribution behavior more
quickly.
You will add the 200I and 2003directory' numbers to the linegroup.
Step 5
Step 6
The order oftlie directory numbers in the Selected DN/Route Partition pane
determines the order in which the directory numbers are accessed in this linegroup.
Change the order ofthe line group members so that 2003 isthe first member and
2001 is the second member. To change the order, click a directory number and use the Up and Down arrows orchoose Reverse Order of Selected DNs. Step 7 Click Save to add the new directory numbers to the line group.
Step 8
Repeat the previous steps to create asecond line group named 2ndLG with member
2002/Phones.
Activity Verification
Iwo line groups with their respective members have been added to Cisco Unified Communications Manager. You can seethese twoline groups if youchoose Call Routing >
Route/Hunt > Line Group and click Find.
Lab Guide
57
Activity Procedure
Complete Ihese steps: Step 1
Step 2
Step 3 Step 4
Step 5
In the Hunt List Name field, enter the name IstHL. In the Description field, enter
First Hunt List.
Check the Lnable this Hunt List check box and click Save.
Step 6
Step 7
Add the prev iously created line groups to Ihe new hunt list. Click Add Line Group.
ihe Hum List Detail Configuration window appears.
From the Line Group drop-down list, choose the IstLG line group, and ihcn click Save. Click OK on the pop-up window. The line group name will appear in the Hunt
List Configuration window in the Selected Groups pane.
StepS
Click the Add LineGroup button again, and repeat the previous step to add the
remaining line group. 2ndL(i. to the hunt list.
Note
Cisco Unified Communications Manager accesses line groups in the order in which they
appearin the hunt list You can changethe access order ofline groups, ifnecessary, by choosing a line group from the Selected Groups listand clicking the Upor Down arrow on
trie right side of the pane to move the line group up or down in the list.
Step 9
Step 10
Click Save inthe Hunt List Configuration window. Then click OK on ihe pop-up
that reminds you lo reset the hunt list.
Click Reset to reset the hunt lisl. When the dialog window appears, click Reset, and
then click Close.
Activity Verification
You have completed this task when you attain this result: One hunt list thai contains two line groups has been added to Cisco Unified Communications Manager, You can verify lhat by choosing Call Routing > Route/Hunt ~>
Hunt Lisl and then clicking Find.
Step 3
Step 4 Step 5
Enter 2111 in the Hunt Pilot Number field and Hunt Pilot 1 in the Description field. Choose partition Phones from the Route Partition drop-down list.
Assign the hunt pilot to the IstHL hunt list using the Hunt List field drop-down
menu.
Step 6
Click Save.
Activity Verification
You have completed this task when you attain this result: You created a hunt pilot number 2111 and assigned itto the hunt list: Go to Call Routing >Route/Hunt >Hunt Pilot and click Find toverify that your
hunt pilot has been createdcorrectly.
Task 4: Test Call Distribution In this task, you will test and validate the call-distribution behavior to ensure that itoperates as
desired.
Activity Procedure
Complete these steps:
Step 1
Prom the IP phone with the directory number 2004 (Phonc4-.x), call the hunt pilot
number 2111 andobserve the calldistribution behavior. Answer the call when it
rings on 2003.
Note
Directory number 2001 should now bethe member with the longest idle time in the first line group (1 stLG) and therefore should ring first when the next call isplaced tothe hunt pilot
number.
Step 2
Call 2111 again and verify that the call isfirst sent to 2001. Do not answer the call.
Afier 5 seconds (RNAR timeout), the call should besent to 2003. Again, do not
answerthe call. After another5 secondsthe call is sent to 2002. a memberofthe
Step 3
second line group (2ndLG). The call ispassed on toa member ofthe second line group because "Try Next Member, Then, Try Next Group in Hunt List" is specified as the hunt option. The number 2002 will ring for 5 seconds, then hunting fails. To beable torun into a busy condition ona phone line, you need tosetthe busy trigger to I (at the Line Configuration page). This configuration disables call waiting, allowing a directory' number lo receive only one call ata time and generate
a busy signal for additional callers. Perform the following steps to enable a busy
condition on line 2001:
At the Line Configuration page,scroll down to the Multiple Call/Call Waiting settings. Change the value ofthe BusyTriggerparameterto I.
Click Save and then reset the line.
Step 4
Lab Guide
59
Step 5
Place acall from 2003 to 2001 and keep the call open to gcncrale a busy condition.
What do you expect will happen when you call the hunt pilot from 2004? Write
dow n your assumption and then test yourhypothesis.
Step 6
Hang up the call between 2001 and 2003. Spend a few moments experimenting with
other line group distribution algorithms (Circular. Broadcast, orTop-Down) and other hunt options {Stop Hunting: Skip Remaining Members, and Cio Directly to Ne\t Group: and Try Next Member, but Do Not Goto the Next Group).
Activity Verification You have completed this task whenyou attain this result:
Calls to the hunt pilot will hunt and achieve call distribution according to the configured
huntoption and distribution algorithm.
Morespecificverification was part ofthe activity procedure.
Note
Task 5 (Optional): Configure Final Forwarding for Busy and NoAnswer Conditions
In thistask, you will configure final forwarding on the hunt pilotnumber so lhal a call that is forwarded to the hunt pilotand that is notanswered (because of no-answer or busv conditions)
is forwarded to a local PSTN number. In a production nelwork. the final forwarding destination is usually selto the number ofan auto-attendant or voice-mail system.
Activity Procedure
Complete these steps:
Stepl Step 2
Choose (all Routing > Route/Hunt > Hunt Pilot, and then click Find. Choose 2111 logo to the Hunt Pilot Configuration window.
Step3
Inthe 1lunt forward Settings section, configure these final forwarding settings:
Forward Hunt No Answer Destination: 2001
Step 4
Click Save.
Choose Call Routing > Route/Hunt > Line Croup, and click Find. Choose IstLG to go to the Line Group Configuration window. Set the call-distribution algorithm to Top Down, the hunt option for No Answer and Not Available to Try Next Member, Then, Try Next Group in Hunt Lisl. and the hunt option lor Busy to Try Next Member, Rut Do Not Go to Next Group.
Click Save.
Step 8
Step 9
From Phone4-.v (2004). call the hunt pilot number 2111 and do not answer the call. What do you expect will be the call-distribution and final forwarding behavior?
Step 10
Place acall from 2003 to 2001 and keep the call open, to generate abusy condition.
From Phone4-,\. call the hunt pilot number 2111. What doyou expect will bethe call-distribution and final forwarding behavior? Write down your assumption and
then test your hypothesis.
Step 11
Activity Verification
Cleanup
When not answering the call, final forwarding todirectory number 2001 isperformed after 12 seconds of hunting. At Phonel-.r, "Forwarded for 2111" appears onthe display.
When Phone 1and Phone3 arebusy, the call will notbe sent to thenexl group (2002) but
will be forwarded to the local line ofthe PSTN phone.
To prepare for future labs, follow the procedure tliat was described in the previous task to change the Busy Trigger parameter atline 2001. This time set the parameter back to2.
Lab Guide
Activity Objective
In this activity, you will configure media resources in Cisco Unified Communications Manager nd control access lothese media resources. After completing this activity, you will be able to
meet these objectives:
Visual Objective
fhe figure illustrates whatyou will accomplish in this activity.
c_c.v->
I control
Required Resources
These resources and equipment are required to complete this activity:
Command List
The table describes the commands that are used in this activity.
DSP Farm Cisco IOS Commands
Command
Description
voice-card cardnuwber
dspfarm
Activity Procedure
Complete these steps:
Step1
Step 2 Step 3
Note
Configure the Software Conferencing Media Resource Step 4 In the Cisco Unified Communications ManagerAdministration, go to Media
Resources > Conference Bridge and click Find.
Step 5
You should see one software conference bridge per server. These servers are generated automatically during installation, with a description of CMJservername as entered during installation). The name is usually CFB_2 for the first installed
Lab Guide 63
server. CFR_3 for the next installed server, and so on. Fhe conference bridges are
running only when the Cisco IPVoice Media Streaming App service isactivated on
the appropriate server.
Step 6
Step 7
Click the conference bridge name CFB_2 (the bridge ofthe publisher) to enter the
Conference Bridge Configuration window.
Change the name lo SW-CFB CI CM1-a\
Step 8
Step 9
Activity Verification You havesuccessfully completed this task when you attain these results:
In Tools > Service Activation, iheCisco IP Voiee Media Streaming Application service is
activated.
In Media Resources ^ Conference Bridge, you can see the automatically generated
conference bridge.
When you go lo Media Resources >Conference Bridge and click Find, vou seethat SWCFBJXCMI-.r is registered.
Iry io establish an ad-hoc conference with three participants. Specifically, perform these
steps:
Step 1
Step2
Step 3 Step 4
Step 5
At Phone2-.r. press ihe More softkey lo browse through theavailable softkeys. Press
the Confrn softkey and dial 2003.
members. Use ihe ConfLisl softkey at Phone 1-v and Phone2-.vto show the
Conference I ist. Notethat the creatorofthe conference can also remove participants
from the conlerence.
Activity Procedure
Complete these steps:
Configure a Cisco IOS Router as a Hardware Conference Media Resource Step 1 Connect io your HQ-.r router and enter enable mode. Step 2 Discover the current MAC address on the Fasll;.lhenielO/0 interface of your transcoder router and record il on the iine that follows the example. F.ntcrthe show interface fastethernetO/0 command to see the output, which will he similar to this example. Make note of ihe MAC address that is shown in your output.
(output truncated)
Note
You must use the MAC address of the interface on which SCCP will be enabled inthe name
oftheconference bridge, when using a Cisco IOS conference bridge. In this lab, theMAC addressis notrequired becausea Cisco IOS enhanced conference bridge is configured.
Step 3
Make router DSP resources available as a hardware conference bridge and have them registered at your Cisco Unified Communications Manager system. In global
configuration mode, enterthissequence of commands:
voice-card 0
dspfarm
seep local FastEthernetO/O.lxl seep ccm 10.x.1.1 identifier 1 version 7.0+ seep ccm 10.x.1.2 identifier 2 version 7.0+
Note
When entering the seep ccm command, use ? afterthe keyword version to find outwhich
Cisco Unified Communications Manager versions are supported by this Cisco IOS version Choose the one that is closest to your Cisco Unified Communications Manager version.
SCCp
Note
Inthis lab, a Cisco IOSenhanced conference bridge is configured. Therefore, the name of the conference bridge can be freely chosen. When configuring a Cisco IOS conference
bridge, the name mustbe CFB<MAC >,where <MAC> is the MAC address of yourSCCP
interface (as determined in Step 2), without the dots. The name that is based on the output in Step 2 for example, would be CFB000F34D90D0O.
dspfarm profile 1 conference
codec g711ulaw codec g7llalaw
maximum sessions 2
exit
Lab Guide
65
Add the Cisco IOS Hardware Conference Media Resource to Cisco Unified Communications
Manager
Step 4
Connect to the publisher and. inCisco Unilied Communicalions Manager Administration, choose .Media Resources >Conference Bridge to open the Find
and List Conference Bridges page.
Click Add New.
Steps
Step 6
From Ihe Conference Bridge Iype drop-down menu, choose Cisco IOS Enhanced
Conference Bridge.
Step 7
specified at the Cisco IOS router in the associate profile command (II\V-CrBJ1Qx).
Tip
Step 8
Step 9
Step 10
Step 11
Choose Non Secure Conference Bridge for the Device Security Mode.
Click Sa\e.
Activity Verification You ha\e successfully completed this task when you attain these results:
Connect to \our conference bridge router HQ-.r and enter ihe show seep command. Verifv lhat the Conferencing Oper State is Active and that the TCP Fink Status is Connected. Fnter the show seep ccm group 1 command. Verify tliat the group is associated with both Cisco Unified Communications Managers and the dspfarm profile. Fnter the show dspfarm profile 1 command. Verily the status, number of supported sessions, and the lisl of supported codecs.
When you go to Media Resources > Conference Bridge and click Find, vou see HWCFB_HQ-.r registered.
Communications Manager
In this task. \ou will configure Cisco Unified Communicalions Manager to support Meel-Me
conferences.
Activity Procedure
Complete these steps; Step 1
Step 2
Step 3
In the Meet-Me Number Configuration window, enter the following parameters: Meet-Me Number Configuration: 45XX Description: Meet-Me Range
66
Partition: Phones
Step 4
Click Save.
Activity Verification
You have successfully completed this task when you attain these results: You configured a Meet-Me number range in Call Routing > Meet-Me Number/Pattern.
At Phonel-*. go off-hook and press the MeetMe softkey (use the More softkey to see all
available softkeys). Dial a number from the MeetMe number range (for example, 4511). A
Meet-Me conference will be opened.
At all other phones (Phonc2-.r to Phone4-jc), join the Meet-Me conference by dialing the
Meet-Me numberofthe previously opened conference (4511).
Step 1
Step 2
Step 3
Click Browse and choose a .wav file thatis stored onyourcomputer (forexample. Windows XP Logon Sound.wav in C:\WINDOWS\Media\). Alternatively, ask your
instructor for a MOH .vvav file.
Note
The filename must not contain spaces. Therefore you will have to copy the file first, rename
it, and then refer to the renamed file.
Step 4
Step 5
Step 6
Update the browser page by navigating again to Media Resources > MOII Audio File Management. The uploaded fileshould nowbe shown in the file list andits
status should be Translation Complete.
Note
The upload of MOH files has to be performedper server that provides MOH services. Inthis
lab, the IP Voice Media Streaming App service is only activated at the Publisher. However, it is recommended that you always upload all MOH files to all servers. This ensures that the MOH files are in sync on all servers of the clusters when the IP Voice Media Streaming App
service is activated at another server at a later time,
Step 7
Administration web page ofthe Publisher using the URL https://iOr. l.2/cemadmin.
)2010 Cisco Systems, Inc. Lab Guide 67
Note
Uploading MOH files is the only task performed from the Cisco Unified Communications Manager Administration web page, which only applies tothe server that is specified in the
URL Usually per-server configuration tasks areperformed from the Cisco Unified Operating
System Administration web page
In general, all configurations performed from the Cisco Unified Operating System
Aammistration web page apply onlyto the server that is specified in the URL. All
Managing MOH files is the onlyexception. It has to be performedfrom the Cisco Unified Communications Manager Administration web page but onlyaffects the server that is
specified in the URL
Add a New MOH Audio Source Step 8 (io to Media Resources > Music On Hold Audio Source and click Find.
Step 9
Step 10
You will see oneentry (MOII Audio Stream Number 1the SainpleAudioSource).
Click Add New,
Step 11
Click Save.
Fnter the configuration for Phone I-.v. Change the L;ser Hold MOH Audio Source from <None> lo 2-Cuslom MOH. Sa\e \our changes and reset the phone.
The default User Hold MOH Audio Source and Network Hold MOH Audio Source value is 1
By modifying the User Hold MOH Audio Source value at Phone1-x only, the default MOH file
is played, unless the user at Phone1-x puts a call on hold.
Activity Verification
You have successfullx completed this task when you attain these results:
You have uploaded a new MOH file in Media Resources > MOH Audio File Management.
You have added a new MOH audio source for the uploaded MOII file. All phones use MOH audio source I for Network Ilold events. MOI 1audio source 2 is used for User Hold at Phonel-v. All other phones use MOH audio source 1 for User Hold. Perform the following steps to \erif\ your configuration:
Stepl
Establish acall between Phonel-* and Phone2-*. Press the Hold softkey at Phonclx. Tlie call to Phone l-jc is put on user hold and you should hear the uploaded MOH
file atPhone2-*. Press the Resume softkey atPhonel-* and keep the call open.
Step 2
Press the Hold softkey at Phone2-jr. The call to Phonel-* is put on user hold and you
should hear the default MOH file atPhonel-*. Press the Resume softkey atPhone2* and keep the call open.
Step 3
Press the Transfer softkey at Phonel-* and dial 2004. The call to Phone2-* is put on
network hold and you should hear the default MOH file at Phone2-*. Accept the incoming call at Phone4-*. Press the Transfer softkey at Phonel-* and keep the
transferred call (between Phone2-* and Phone4-*) open.
Step 4
Click the Hold button atthe X-Lite application onPC-* (Phonc4-*). No MOH is
played at Phone2-*. This is because MOH is not supported for the third-party SIP
phone. Click the Hold button again al Phone4-* to resume the call. Keep the call
should hear the default MOH file at Phone4-*.
open.
Steps
Press the Hold softkey atPhone2-*. The call to Phone4-* isput on user hold and you
As you have seen during the activity verification, the MOH Audio Source is selected based
on the configuration of the party that puts the other party on hold. The third-party SIP phone (Phone4-x) can listen to MOH when being held, but no MOH is played ata phone held by
the third-party SIP phone.
Note
Step 1
Step 2
Step 3
Fnter the configuration ofMOH Audio Stream Number I and check the Allow Multicasting check box. Click Save.
Repeat theprevious step for MOH Audio Stream Number 2.
Step 4 Step 5
Go to Media Resources > Music on Hold Server and click Find. You will see one MOH server perCisco Unified Communications Manager server.
The\ havebeen automatically configured during installation ofthe server. Their
Step 6
Step 7
In die Multicast Audio Source Information pane, check the Enable Multicast Audio
Sources on this MOH Server check box.
Step 8
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Step 9
Step 10 Step 11
Step 12
Verity that the Max Hops parameter for Multicast Audio Source I and 2 is sel to 2
Click Save and. in the pop-up window, click OK. Click Reset and. in the pop-up window, click Reset and then Close.
Connect to \ our I \Q-.x router.
Step 13
Note
You must enable multicast routing at both theincoming interface (FaO/0 x01) and the
outgoing interface (FaO/0 1x2)
Activity Verification You ha\e successful^ completed this task when you attain this result:
When MOH is played, multicasl MOH. rather than unicast MOII, isused. To verify \our
configuration. >ou can watch the packets that are sent out on interface FastFthernetO/0.1*2. Perform the following steps for verification:
Step 1
At the HQ-* router, enter the following commands in global configuration mode:
access-list 101 permit udp host lO.x.l.l 239.1.1.0 0.0.0.255
Note
This access listeffectively permits all IP traffic. You can use it to see the match counts per
access list line In this lab, multicast MOH traffic will match the first access list line, and unicast MOH traffic will match the second access list line. All other IP traffic will match the
third access list line.
Step 2
Cse the show access-lists 101 command several limes and look al the numberof
matches for each access list line:
Extended IP access list 101
(8 matches)
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Note
At this stage, you should not see a significant increase in any access list line. Occasionally,
the first access list line is incremented byabout 50 matches per second, but notforlonger than 10 to 15 seconds. Thiseffectcan be ignored forthe verification of multicast MOH.
Step 3
Step 4
Establish a call between Phonel-* and Phone2-.v. Put the call on hold from any side.
In your Telnet session to HQ-*, enter the show access-lists 101 command several
times. Although you expect multicast MOH tobe played, only the second access list line isincremented (by about 50matches persecond), while MOH isplayed at the
held phone. Obviously thephones still use unicast MOH.
Note Occasionally, thefirst accesslist line isalso incremented by about 50matches per second, but not longer than 10to 15seconds. This effect can be ignored for theverification of
multicast MOH.
Note
Multicast MOH is not used because for multicastMOH, MRGsand MRGLs are required,and the Use Multicastfor MOH Audio check box at the MRG must be checked. MRGs and
MRGLs are implemented inthe nexttask, so you will be ableto verify your multicast MOH configuration only inthe activity verification ofthe nexttask.
Step 2
Name: HVV-CFB_mrg
Step 3
Step 4
From the Available Media Resources, choose the Cisco IOS hardware conference
bridge HWCFBJIQ-a.
Click Save.
Step 5
Step 6
Step 7
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From the Available Media Resources, choose the software conference bridge SWCFB_CUCM1-jc.
Click Save.
Lab Guide 71
Step 8
Step 9
From the Available Media Resources, choose ANN_2, MOII 2{Multicast| and
MIP 2.
Note
The software media resources that end with _3 are not active because the Cisco IP Voice
Media Streaming App service has not been activated on the appropriate server (CUCM2-x;
Therefore, there is no need to add them into any MRG.
Step 10
Check the I se Multicast fur MOII Audio (If at Least One Multicast MOH
Resource Is Available) check box.
Note
This setting is required to allow MOH serversthatare partofthe MRG to use multicasting (if
enabled at the MOH servers).
Step 11
Click Save.
Configure MRGLs
Step 12
Go to Media Resources > Media Resource Group Fists and click Add New.
Step 13
Step 14
From the Available Media Resource Groups, choose all groups and make sure to
order them as follows:
HVV-CFB mrg
SU-CFIimrg
Note
Other-SW-MR_mrg The hardware conference bridge should be used before the software conference bridge,so
make sure that the HW-CFB_mrg is listed before the SW-CFB_mrg Other media resource
types (MOH server, annunciator, and MTP) are made available by adding the Other-SWMR_mrgl to the MRGL. The position of this MRG within the MRGL is not relevant because
this is the only MRG that contains such types of media resources.
Step 15
Click Save.
Assign the MRGLs to Phones Step 16 Go to System > Device Pool and click Find and choose Default.
Step 17
Step 18
Step 19 Step 20
Click Reset and reset all devices using this device pool. From the Related Links, choose Dependency Records and click Go. You will see a record summan lhat indicates which device types (and their number) arc using this dev ice pool.
Click Record Tvpe Phone to see the list of phones that use this device pool. Note lhat Phonej-.i uses a different device pool (SubPub).
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Step 21
72
Activity Verification
You have successfullv completed this task when you attain these results:
From Phone l-.v and from Phone2-.r. you can initiate ad hoc and Meet-Me conferences. You
cannot initiate ad hoc or Meet-Me conferences anymore from Phone3-.r.
Verify that the hardware conference mediaresource is used before the software conference
media resource, by performing the following steps:
End all active calls.
Step 1
Step 2
From Phonel-*. set up a Meet-Me conference at number 450I. From Phonc4-*. join
the Meet-Me conference by dialing 4501. At Phonel-.*, leave the Meel-Me conference by ending the call.
Note
put a Meet-Me conference in place, with Phone4-x as the only member {after Phone1-x
drops out)
Step 3
In a Telnet session to HQ-x, use the show dspfarm dsp active command to verify that the hardware conference bridge is used for the conference. Keep the call active
at Phone4-.r.
Step 4
Create another Meet-Mc conference from Phonel-j:, but this time use 4502 for the number. From Phone3-*. join the Meet-Me conference. At Phonel-*, leave the
Meet-Me conference.
Note
Phone3-x cannot create conferences because it does not have an MRGL assigned, and all
media resources are put into MRGs. However, the Meet-Me conference was created by another phone, so a second conference is in place with Phone3-x as the only member (after
Phone1-x drops out).
Step 5
Again, use the show dspfarm dsp active command at HQ-* to verify that the hardware conference bridge is also used for the second conference, Keep the call
active at Phone3-*.
Step 6
StepS
When dialing an invalid number (for example, 4444) from Phonel-* or I'hone2-.v.you will hear an annunciator message. At Phone3-*. you only hear the appropriate call progress
tone.
Lab Guide
Note
Phorte3-x cannot listento MOH because it does not have an MRGL assigned and all media
resources (including the MOH server) are put into MRGs. Therefore, Phone3-x does not
have access to any MOH server and TOH is played instead of MOH.
When Phone2-* or Phone3-.v is put on hold, it hears MOII. Use the show access-lists 101 command at HQ-x to verifv that multicast MOH is used this time.
74
phones.
Required Resources
These resources and equipment are required to complete this activity:
Lab Guide
75
Activity Procedure
Complete these steps:
Nav igate to Device >Device Settings >Phone Button lemplate. Choose the Standard 7%5 SCCP Template and click Copy. fnter Standard 7965 SCCP Presence as the name ofthe new template and. for
phone button 3. choose Speed Dial BLF. Click Save.
Open the phone configuration for Phone2-.v.
Step 5
Step 6
Step 7
Choose the Standard 7%5 SCCP Presence Phone Button Template and click Save.
Click the Add a new RLF SI) link next to phone line 3.
hnter 2003 in the Destination Held and click Save.
Step 8
Repeat Steps 4 through 7 for Phone3-* but enter 2002 at Ihe Destination field.
Step 9
\\ rite down the partitions lhat arc assigned lo the first directory numbers ofthe
following phones:
Phone2-.v:
Phone3-.v:
Step 10
Note
Place calK between phones and verify that you do not see presence information on
ihe corresponding presence-enabled speed dials.
You will not see presence information because the watched directory numbers are in
partitions but no subscribe CSS isapplied tothe IP phones
Step 11
Step 12
Create a CSS called Presence_css. which includes all partitions that were discovered
in Step 9.
Open the phone configuration for phones Phone2-\ and Phone3-\, scroll down to the
Protocol Specific Infonnation iield and assign the previouslv created CSS as a
SI BSCRIRF. Calling Search Space.
Step 13 Reset both phones.
Step 14 Note
Place calls between phones and verify lhal you now see presence information on the
corresponding presence-enabled speed dials.
When implementing presence in anenvironment that already has partitions in use (because
ofcalling-privilege implementation), presence isalsoaffected bythese partitions Therefore. subscribe CSS must beassigned even if norestrictions should beapplied to presence.
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Activity Verification
You have completed this task when you attain these results: Phone2-.x and Phone3-.t see accurate presence information atthe presence-enabled speed
dials.
Step 1 Step 2
Step 3
Place acall from each phone to the other two phones and do not answer the call. Go to System >Enterprise Parameters and enable the BLF-" for Call Lists enterprise parameter in Cisco Unified Communications Manager Administration.
Click Save and Reset.
Step 4
Step 5
On the Phone2-* and Phone3-*, press the Directory button and choose Missed
Calls.
You should see the presence status also onthe call lists.
Step 6
Step 7
Press the Directory button at the Phonel-x and choose Missed Calls. Observe that presence infonnation is not available to this phone. This is because you have not
configured a Subscribe CSS for thisphone.
Step 8
Note
Repeat Step 6 for Phonel-* and verify that presence-enabled call lists are now
visible for Phonel-*.
numbers are in partitions. Asubscribe CSS wasalready assigned toall phones, in the
previous task
Activity Verification
Youhave completed thistaskwhen you attain this result: At Phonel-*. Phone2-.v and Phone3-*. you can viewpresenceinformation in call lists.
Lab Guide
77
In this section, you will apply the following presence policies: The manager phone (Phonc3-.v) should be able to see presence-enabled call list infonnation
for all other phones.
All other phones will have access to presence-enabled call-list information between each
other but not for the manager phone.
The employ ee phone (Phone2-*) will still be able io use the presence-enabled speed dial.
follow these steps to implement presence policies for speed dials:
Step 1
Step 2
Step 3
Step 4
Select the Standard Presence group, choose Allow Subscriptions from the
Subscription Permission drop-down menu, and click Save.
In the Related Links menu, leave Back to Find/List selected and click Go.
Step 5 Step 6
Step 7
Click Find and open the Standard Presence (iroup configuration. Select Manager ps and choose Disallow Subscriptions from the Subscription
Pennission drop-down menu, and click Save.
Steps
Step 9
Step 10
At the Line Configuration window for line 2003. choose Manager pg from the
Presence Group drop-down menu.
Click Save and Reset.
Activity Verification
You have completed this task when vou attain this result:
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Activity Objective
In Ihisactivity, you will implement Mobile Connect (Tasks I to 4) and MVA(Tasks5 to 7). Afler completing this activity, you will be able to meet these objectives:
Add the Mobility softkey to IP phones Associate an end-user account with the IP phone and enable the use of mobility
Configure Cisco Unified Mobility media resources Configure the Cisco IOS gateway for Cisco Unified Mobility
Visual Objective
The figure illustrates what you will accomplish in this activity.
Allow remote
destinations (mobile
Required Resources
These resources and equipment are required to complete this activity:
Cisco IOS gateway (MGCP and H.323) PSTN with PSTN phone
Lab Guide
79
Activity Procedure
Complete these steps:
Locate and click the Copy icon to the right ofthe Standard User.
l.nter the name Standard User Mobility.
Hnter Mobility Softkey Template for the Description and click Save. from the Related Links, choose Configure Softkey Layout and click Go. Verifv that On Hook is chosen in the Select a Call State to Configure drop-down menu. Click the Mobility entrv in tlie Unsclcctcd Soflkcys pane and move il to the Selected Softkeys pane by using the arrow link.
Save tlie conllguration.
Step 7
Step 8
Repeat the previous steps to add the Mobility softkey lo tlie Connected call slate.
Assign the Softkey Template to the IP Phone Step 9 Navigale lo Device > Phone and click Find.
Step 10 Choose Phone3-.v.
Step 11
Step 12 Step 13
Choose Standard User Mobility from the Softkey lemplate drop-down menu.
Click Save and click OK in the pop-up window. Reset the phone.
Activity Verification
You have completed this task when um attain these results:
Press the Mobility Softkcv on Phone3-r. 'fhe "You are not a valid Mobile User" error message should be displayed.
Activity Procedure
Complete these steps:
Configure an End User for Device Mobility Step 1 Navigate to the User Management > Lnd User and click Kind.
Step 2 Choose user I seri.
Step 3
In the Mobility Infonnation area, cheek the hnable Mobility check box.
2010Cisco Systems, Inc
Step4
Step 5
Step 8
Step 9
Step 10
In the Device Information pane choose User3 from the Owner User ID drop-down
menu.
Activity Verification
Press the Mobility Softkey on Phone3-x The"NoMobile Remote Destination found" error
message should be displayed.
Activity Procedure
Complete these steps:
Configure a Remote Destination Profile Step 1 Navigate to Device> DeviceSettings > Remote Destination Profile and click Add
New.
Step 2
Calling Search Space: Phones_css Privacy: On Rerouting Calling Search Space: Phones_css
Lab Guide
Note
TheRerouting Calling SearchSpace parameter is the CSS thatis used for ringing the
remote destination when a call is received at the office phone. The parameter is also used for handing calls that are active at the office phone to a remote destination
The Calling Search Space parameter is the device CSS of the virtual phone that represents the remote destinations In other words, this CSS is used when placing outgoing enterprise
calls from a remote destination. The MVA feature is used only later in this lab exercise
(Tasks 5 to 7).
Step 3
Step 4
Check the Ignore Presentation Indicators (internal calls only) check box.
Click Save.
Step 5 Step 6
Click Line 11]Add a New DN. At the Direetorv Number Configuration window, enler 2003 for the Direetorv'
Number. Then click in any other input field so lhal the conllguration of this
direetorv number is loaded.
Step 7
Step 8
Step 9
Click Save.
Step 10 Step 11
Verify that User3-rdp shows up at the Associated Devices section. From the Related Links, choose Configure Device (User3_rdp) and click Go. to
return to the Remote Destination Profile Configuration window.
Step 13
Step 14
Step 15
Note
Step 16
Note
Step 17
Step 18
Step 19
In the .Association Information pane check the check box at the right of Line |1|
2003 in Manager-Phones. Click Sa>e and click OK in the pop-up window.
Step 20
Navigate to Call Routing > (lass of Control > Calling Search Space and select
the To-Phones_css.
Step 21
Add the Manager Phones Partition to this Calling Search Space and click Save.
Note
For the verification of this lab, you need to allow calls from the PSTN to the Manager phone
(Phone3-x).
Activity Verification
You have completed this task when you attain these results:
Verify that both the office phone and the PSTN phone ring when internal calls are made to
Phone3-.t:
Step1
Step 2
Step 3
The call should be presented to Phone3-x line 1and lo the PSTN phone (at line 2).
Answer thecallon the PSTN phone.
Step 4
Step 5
Step 6
Look at the line 1button on Phone3-x. Note the color is red, indicating that acall is
active at a remote destination on the shared line 2003.
Hand the call over to the office phone by ending the call atthe PSTN phone and then
pressing the Resume softkey on Phone3-x.
find the call.
Verify that calls from the PSTN phone line 2(remote destination) are presented as calls
from the office phone when calling internal directory numbers:
Step 1 Step 2
Step 3
At the PSTN phone, press the National button toplace a call with a long distance
calling number, and dial 152x5553002.
Verify that the call ispresented with the internal number of Phone3 (2003) atthe
receiving phone (Phone2-x).
While Phone2-x isringing, look al the line 1button on Phone3-x. Note that the color is red. indicating thatthe remote destination hasa call. End the call onthe PSTN
phone.
Verity that cails that are made from PSTN phones to Phone3-* will ring at 2003 and at the remole destination (PSTN phone line 2). Verify thattheremote phone (PSTN phone line 2)
is showing the caller ID ofthe PSTN phone that called Phone3-x.
Step 10 Step 11
Step 12
Step 13 Note
From the PSTN phone line 1.make a call to Phone3-x (151x5553003). fhe call should be presented to Phone3-* line l and to the PSTN phone (atline 2).
Check the callerIDofthe call ringing at the PSTN phone line2. ThecallerID
should be the number ofthe PSTN phone fine 1. End the call without answering. In many countries, youare not allowed to set the calling number ofoutgoing PSTN callsto a
number that is different from your actual PSTN number. Therefore, the preservation of the
Verifv that the office phone can hand an answered call over to the PSTN phone:
From Phone2-.t.make a call to 2003. The call should be presented to Phone3-.r line I
and PS'fN phone line 2.
Answer the call at Phonc3-.x.
LabGuide 83
Step 1
Step 2
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Step 3
Step 4
Step 5
Step 6
Step 7
Step 8
Keep the call active between Phone2-.v and the PSTN phone, and make a call from
Phone3-.i to 2001. answer the call on Phone I-v.
In this task, vou will configure a ring schedule to ring the remole destination only during a specified time range. Further calls from PSTN phone line 9 (emergency) will beblocked even
during the allowed time schedule.
Activity Procedure
Complete these steps: Configure an Access List
Step 1
Step 2
Step 3
Open the I'ser Web Page (htlps://l(U.I. l/ccmuser) and log on User3
Nav igate to I ser Options > Mobility Setting > Access Lisl and click Add New.
Enter the Access List Name ACL911.
Make sure the Blocked radio button is selected and click Save. C lick Add Member and enter 911. Click Save.
Navigate to I'ser Options > Mobility Setting > Remote Destinations and click
Find.
Step 9
At the Ring Schedule Pane, choose the As Specified Below radio button.
Step 10
Checkall check boxes from Monday to Friday and specify 09:00 to 18:00 as begin
and end times.
Step 11
At the \\ hen receiving a call during the above ring schedule pane, choose the Do
not ring this destination if caller is in radio button and choose the ACL911 access list from the drop-down menu.
Step 12
Click Save.
Activity Verification
You have completed this task when vou attain these results:
Step 1
Verifv that the current dale and lime lhat is displayed on Phone3-.v is within Ihe specified schedule (Monday through Friday. 9:00 to 18:00).
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84
From the PSTN phone line 1. call Phone3-JC (152x5553003). The call should be presented to Phonc3-x line 1and to the PSTN phone (at line 2). From the PSTN phone line 6(emergency), call Phonc3-:v (151v5553003). The call should be presented to Phone3-;t line 1but not to the PSTN phone line 2.
Navigate to User Options >Mobility Setting >Remote Destinations and click
Find.
Choose the configured remote destination and set the Ring Schedule ofthe current
weekday toa value outside the current time (for example, 23:00 to23:05).
Click Save.
Repeat Steps 2 and 4. The remote destination (PSTN Phone line 2) should never
rim
Cleanup
To prepare for future labs, follow these steps to re-enable the remote destination at all times.
Step 1 At the Remote Destination Configuration select All the time atthe Ring Schedule pane and Ahvay ring this destination atthe When receiving a call
during the above ring schedule pane.
Step 2 Click Save.
Task 5: Enable MVA In this task, you will activate the Cisco Unified Mobile Voice Access Service feature service. You will configure Cisco Unified Mobility service parameters toenable the MVA feature globally, and then you will allow individual end users touse MVA.
Activity Procedure
Complete these steps:
Step 1 Step 2
Step 3
Note
In Cisco Unified Serviceability, navigate to Tools>Service Activation. Select the 10.x. 1.1 server and check Ihe Cisco Unilied Mobile Voice Access
Service check box.
Click Save.
The Cisco Unified MobileVoice Access Service can be activated only on the publisher
server.
Configure Cisco Unified Mobility Service Parameters In the following steps, you wilt enable Enterprise Feature Access and writedown the corresponding feature access codes. Then you will enable MVA globally.
Step 4 Navigate to System > Service Parameters.
Step 5
Step 6
Step 7
Lab Gu'de
Note
Enterprise Feature Access allows Cisco Unified Communications Manager features such as
hold, resume, transfer, and conference to be controlled from a remote phone by using DTMF
tones
Step 8
Feature
Hold
Exclusive Hold
Resume
Transfer
Conference
Step 9
Step 10
Selthe Enable Mobile Voice Access and the Fnablc Enterprise Feature Access
parameter to True.
Navigate to the I'ser Management > End Lser and click Find.
Choose user I ser3.
in the Mobility Infonnation area, cheek the Enable Mobile Voice Access check
box.
Step 14
Click Save.
Activity Verification
You have completed this task when vou attain these results:
In Cisco I nified Serviceabilitv under Tools > Control CenterFeature Services, the Cisco
Activity Procedure
Complete these steps:
Step 1 Navigate to Media Resources > Mobile Voice Access.
Step 2
Step 3
Step 4 Choose the English United States locale in the list ofAvailable Locales and move
it to the Selected Locales by using ihearrow link.
Step5 Save the configuration.
Activity Verification
You have completed this task when you attain this result: The resource is configured under Media Resources >Mobile Voice Access.
In this task, vou will configure the Cisco IOS gateway with acall application that allows PSTN calls to be placed from the remote phone as ifthey originated from the office phone. Remember that vour Tl/El PSTN connection is MGCP-controlled. To direct calls to the IVR
application, vou'will need to send calls that are received on Cisco Unified Communications Manager via the MGCP-controlled interface back to the gateway using H.323 signaling
(hairpinning).
Activity Procedure
Complete these steps:
Configure H.323 Gateway Functionality for the IVR Application and Hairpinning Step 1 Log in to the HQ-x gateway and enter the following commands in conllguration
mode.
application
Step 2
Configure an incoming VoIP dial peer for the MVA number (2998) and associate
the IVR call application with it.
dial-peer voice 2998 voip
service MVA
dtmf-relay h245-alphanumeric
no vad
exit
Step 3
Configure a VoIP dial peer to enable the call application that is running in Cisco IOS
Software to contactthe MVA media resource in Cisco UnifiedCommunications
Manager.
Lab Guide
87
Note
The destination pattern must match the MVA directory number that is configured atthe MVA
media resource The pattern does not need to match the last digits ofihe PSTN number that
is used forMVA (52x5553998 inthis case).
session target ipv4:10.x.1.1
codec g7llulaw
Step 4
Add the HQ-x Gateway in Cisco Unified Communications Manager as H.323 Gateway
Step 5 Navigate to Device> Gateway and click Add New.
Step 6
Step 7
Choose 11.323 Gatewav from the Gateway Type drop-down menu and click Next.
Enter the following parameters:
Device Name: ICv.LlOl
Step 8
Step 9
Step 10
Click Save.
CheckWhether the Remote Destination Is Recognized at the H.323 Gateway It isimportant to prov ide the correct calling number lothe IVR application thai isconfigured
on the H.323 gatewav. Similar to Mobile Connect. MVA needs to detect thai the call is coming
from a configured remote destination. If a remote destination cannot be detecled. the MVA
callerhasio authenticate to MVA by enteringa valid remotedestination numberand the PIN of the associaied user. If a remote destination is detected, only the PIN has lo be entered. Check if the remote destination is recogni/.ed bv performing the following steps:
Step 11 From PS IN phone line 2. dial 15Z\S553998.
Step 12
Step 13
I isten to tlie IVR script. You will be prompted to enler your remotedesfination
number (916065554444) and your PIN (12345).
End the call.
Note
In this case, the remote destination was not recognized and therefore the remote destination number had to beentered before being prompted for the PIN. This happens because all calls from the PSTN arrive atCisco Unified Communications Manager through the MGCPcontrolled interface.
If a call arrives at Cisco Unified Communications Manager and originates from a configured remote destination number, thecalling number istransformed to theoffice extension (2003)
by Mobile Connect. When sending the MVA call received from the remote destination to the IVR application, the application isno longer able to recognize the received call asa call from
a remote destination becausethe calling number is 2003 instead ofthe actual remote destination number. Asa consequence, the MVA user is prompted to entera valid remote
___
Configure Digit Manipulation to Provide Correct Calling Number Information tothe IVR
Application
In order to send the original calling number ofthe remote destination to the IVR call
application, you will configure voice translation rules at the H.323 gateway. The voice
translation rule will match onthe four-digit directory number ofthe office phone and translate this internally used office number tothe associated remote destination number. Step 14 Log in tothe HQ-x gateway and enter the following commands in global
configuration mode
i
voice translation-rule 1
Activity Verification You have completed this task when you attain these results:
An outgoing PSTN call can be placed from the remote phone butappears to be initiated
from the office phone.
Step 1
Listen to the IVR script prompt. The remote destination number (PS'fN 916065554444) is recognized, and only the PIN is requested from the IVR script. Enter the PIN 12345 followed by #, when prompted by the IVRscript. Listen to the
IVR script prompt.
Choose option 1 to initiate a call, from the remote phone to a PSTN destination, that
looks like a call from the office phone.
LabGuide
Step 5 Step 6
Step 7
Enter a PSTN directory number as ilwould be entered from Phoncl-\ (for example
9-911). followed bv #.
Verifv that the incoming call that is received atthe PS'fN phone presents the full direetorv number ofPhone3-x. The call should be received al the PSTN phone at
line 6. the emergenev line.
Fnd the call.
Make a cal! from a nonremole destination and verify that the remote desfination is
unknown.
Step 1
Step 2
Step 3
Step 4
The IVR script should now prompt for the remote destination. Enter your remote destination 916065554444 and then your PIN. as prompted by Ihe script.
Place a call to a PSl N destination (lorexample. 9-911). 'fhe call should be received at the PSfN phone line 5 (emergenev): the calling number should be 51x5553003.
End the call.
Step 5 Step 6
Step 7
Step 8
Step9
Verifv thai the call is received at Phone I-x. with the internal director) number of
Phone.Vx (2003) as the calling number.
End the call.
Dial the MVA number again, and activate (option 2) and deactivate (option 3) Mobile
Connect capabilities via the IT'I. From PSTN phone line 2. dial 152x5552998.
Step 1
Step2
Step 3
Step 4
Step5
From Phone2-.v. call 2003. Note lhal the call is not sent lo the remote destination.
End the call.
Step 6
Step 7
Step 8
Step 9
Step 10
From Phone2-.\. call 2003. Note that the call is sent again to the remole destination.
End the call.
Dial the MVA number and use Enterprise Feature Access to place a call on hold and to
resume the call.
Step 1
Step 2
Step 3 Step 4
Step 5
Dial Feature Access Code *81 to place the call on hold. Phone2-.v should play MOH. At Phone3-.v. the display indicates that the call at the remote phone has been put on
hold.
Step 6
Lab Guide
Answer Key
Tlie correct answers and expected solutions for the activities lhat are described in this guide
appear here.
network-clock-participate wic 0
0/0/0
source
line
Note
Some of these commands are default commands and therefore may not be shown
interface SerialO/0/0:23
no ip address
no cdp enable
voice-port 0/0/0:23
cciTi-mar.ager redundant-host 10.2.1.2
92
ccm-manager mgcp
Note
mgcp
mgcp call-agent 10.2.1.1 2427 service-type mgcp version 0.1 mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
Lab Guide
ip multicast-routing
voice-card
dspfarm
interface FastEthernetO/0.lxl
ip pin sparse-dense-mode
interface FastEthernet0/0.1x2
seep local FastEthernetO/0.lxl seep ccm 10.x.1.2 identif ier 2 version 7.0 seep ccm lO.x.l.l identifier 1 version 7.0
seep
access-list 101 permit udp host lO.x.l.l 239.1.1.0 0.0.0.255 access-list 101 permit udp host lO.x.l.l any access-list 101 permit ip any any
Note
94
voice translation-rule 1
application
service MVA
http://10.x.l.l:8080/ccmivr/pages/IVRMainpage.vxml
i
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
Note
Lab Guide
95
96