Cisco CallManager Express 3.

3
System Administrator Guide

May 2005

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Cisco CallManager Express 3.3 System Administrator Guide

Copyright © 2005 Cisco Systems, Inc. All rights reserved.
 CONTENTS

Feature Map xiii

Preface xvii

Documentation Objectives xvii

Audience xvii

Documentation Organization xviii

Document Conventions xix

Obtaining Documentation xx
Cisco.com xx
Documentation CD-ROM xxi
Ordering Documentation xxi
Documentation Feedback xxi
Obtaining Technical Assistance xxi
Cisco TAC Website xxii
Opening a TAC Case xxii
TAC Case Priority Definitions xxii
Obtaining Additional Publications and Information xxiii

Cisco CallManager Express Overview 1

Contents 8
Cisco CallManager Express Description 9
Cisco CallManager Express Network Scenarios 9
Additional Features 10
Provisioning 10
Connecting Cisco IP Phones 10
Prerequisites 11
License Prerequisites 11
Memory Prerequisites 11
Network Prerequisites 11
Software Prerequisites 12
Cisco IOS Software 12
Cisco CallManager Express Files 12

Restrictions 17

Information About Setting Up a Cisco CME System 19

Before You Start: Basic Cisco CME Concepts 22

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Contents

Ephones 24
Ephone-dns 24
Single-Line Ephone-dn 25
Dual-Line Ephone-dn 25
Two Ephone-dns with One Number 26
Dual-Number Ephone-dn 27
Shared Ephone-dn 28
Overlay Ephone-dn 28
Phone Number Plan 30
Direct Inward Dialing 31
PBX or Keyswitch Model 32
What to Do Next 32
Additional References 32
Related Documents 33
Related Websites 34
Standards 34
MIBs 35
RFCs 35

Setting Up a Cisco CME System 37

Contents 37

Information About Setting Up a Cisco CME System 37

Prerequisites 38

Setting Network Parameters for Cisco CME 38

Setting Up DHCP Service for Cisco CME 38
Defining a Single DHCP IP Address Pool 39
Defining a Separate DHCP IP Address Pool for Each Cisco IP Phone 40
Defining a DHCP Relay Server 41
Configuring Network Time Protocol 42
Downgrading the Cisco IOS Software Image to an Earlier Version 43

Verifying the Configuration 44

What to Do Next 44

Setting Up Phones in a Cisco CME System 45

Contents 45

Information About Setting Up Phones in a Cisco CME System 45

Specifying Cisco CME Parameters 46

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Setting Up Initial Extensions and Phones 50

Automated Phone Setup Using the Cisco CME Setup Tool 50
Prerequisites 51
Restrictions 51
Cisco CME Setup Tool Dialog 52
Configurations Resulting from the Cisco CME Setup Tool Dialog 55
Partially Automated Setup Using the Router CLI 56
Prerequisites 57
Restrictions 57
Manual Setup Using the Router CLI 59
Prerequisites 60
Specifying Phone-Related Parameters 64
DTMF Relay for H.323 Networks 64
Dial-Plan Pattern 65
Translation Rules 68
Verifying Translation Rules 69
Translation Profiles 69
Verifying Translation Profiles 71
International Language, Date Formats, and Tone Support 72
Restrictions 72
Changing the TFTP Server Address 74
Changing the TFTP Server Address When the Cisco CME Router is the DHCP Server 74
Changing the TFTP Server Address When the Cisco CME Router is Not the DHCP Server 74
Resetting and Restarting Cisco CME Phones 75
Using the reset Command in Ephone Configuration Mode 76
Using the reset Command in Telephony-Service Configuration Mode 76
Using the restart Command in Ephone Configuration Mode 78
Using the restart Command in Telephony-Service Configuration Mode 79
Verifying Cisco CME Phone Configuration 80

Configuration Examples for Setting Up Phones in a Cisco CME System 83

What to Do Next 86

Configuring Call Transfer and Call Forwarding 87

Contents 87

Information About Call Transfer and Call Forwarding 87

Background 88
Strategies for Call Transfer and Call Forwarding 89
H.450.2 and H.450.3 Support 89
H.450.12 Support 92

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Hairpin Call Routing 93

H.450 Tandem Gateways 95
Typical Network Scenarios for Call Transfer and Call Forwarding 98
Cisco CME 3.1 or Later and Cisco IOS Gateways 98
Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, and Cisco IOS Gateways 99
Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways 99
Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, Non-H.450 Gateways, and Cisco IOS
Gateways 100
Cisco CME 3.1 or Later, Cisco CallManager, and Cisco IOS Gateways 101
Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, Cisco CallManager, and Cisco IOS
Gateways 102
Configuring Call Transfer and Call Forwarding 103
Enabling or Disabling H.450.2 and H.450.3 Capabilities 103
Enabling or Disabling H.450.12 Capabilities 108
Enabling H.323-to-H.323 Connection Capabilities 111
Restrictions 111
Enabling Interworking with Cisco CallManager 112
Configuring Cisco CME 3.1 or Later to Interwork with Cisco CallManager 113
Configuring Cisco CallManager to Interwork with Cisco CME 3.1 or Later 116
Setting Up Dial Peers 118
Verifying and Troubleshooting 118
Configuration Examples for Call Transfer and Call Forwarding 119
Cisco CME 3.1 or Later and Cisco CallManager in the Same Network: Example 120
H.450 Tandem Gateway Working with Cisco CME 3.1 or Later and Cisco CallManager:
Example 122

Transcoding Between G.729 and G.711 125

Contents 125
Prerequisites 125

Restrictions 125

Information About Transcoding Between G.729 and G.711 126

Configuring Transcoding Between G.711 and G.729 127

Determining Digital Signal Processor Resources 127
Determining the Correct DSP Allocation for Transcoding 130
Provisioning NMs or NM Farms for Transcoding 130
Setting Up DSP Farms for NMs 130
Setting Up DSP Farms for NM-HDVs 131
Setting Up DSP Farms for NM-HDs and NM-HDV2s 132
Changing the Number of Transcoding Sessions 136
Configuring Cisco CME to Act as the DSP Farm Host 136

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Verifying That the DSP Farm Is Registered and Running 139

Tuning Performance 142
Configuration Examples for Transcoding Between G.711 and G.729 143
NM-HDV: Example 143
NM-HD and NM-HDV2: Example 144

Setting Up the Cisco CME GUI 145

Contents 145

Information About Setting Up the Cisco CME GUI 145

Prerequisites 146
Restrictions 146
Setting Up GUI Access for the System Administrator 147
Setting Up the HTTP Server 147
Setting Up GUI Access for the System Administrator 148
Troubleshooting Tips 150
Accessing the Cisco CME GUI 150

Setting Up GUI Access for Customer Administrators and

Phone Users 152
Setting Up GUI Access for Customer Administrators 152
Creating and Loading an XML Configuration File 153
Defining a Customer Administrator 156
Method 1: Using the Cisco CME GUI to Define a Customer Administrator 156
Method 2: Using the Cisco IOS CLI to Define a Customer Administrator 157
Setting Up GUI Access for Phone Users 158
Method 1: Using the Cisco CME GUI to Define a GUI Account for a Phone User 158
Method 2: Using the Cisco IOS CLI to Define a GUI Account for a Phone User 160
What to Do Next 160

Setting Up Optional Cisco CME System Features 161

Contents 161

Call Park 161

Secondary Dial Tone 165

Busy Timeout 166

Interdigit Timeout 167

Ringing Timeout 168

Music on Hold 169

Configuring Music on Hold from an Audio File 169
Prerequisites 170
Restrictions 170

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 Contents

Configuring Music on Hold from a Live Feed 172

Restrictions 172

Configuring Cisco CME Phone Features 177

Contents 177

Related Features 177

Dial Features 178

Callback Busy Subscriber 178
On-Hook Dialing 178
Speed-Dial Features 179
Monitor-Line Button Speed Dial 179
Local Speed Dial Menu 180
Prerequisites 180
Personal Speed Dial Menu 182
Speed-Dial Buttons and Abbreviated Dialing 183

Automatic Line Selection 187

Call-Transfer and Call-Forward Features 189

Consult Transfer Support for Direct Station Select 189
CFwdAll Soft-Key Restriction Control 190
Do Not Disturb Features 191
DND Call Forward 192
DND Disable for Feature Ring 193

Conference Call Features 194

Three-Party G.711 Conference Calls 194
Examples 195
Conference Initiator Drop-Off Control 196
Examples 197
Configurable Phone Displays and Sounds 197
Call-Waiting Beep 198
Restrictions 199
Examples 199
Call-Waiting Ring 200
Restrictions 200
Called Name and Ephone-dn Name Display 201
Configuring for the Display of Called Names for Overlaid Ephone-dns 201
Restrictions 202
Configuring for the Display of Called Names for Nonoverlaid Ephone-dns 206
Configuring for the Display of Ephone-dn Names for Overlaid Ephone-dns 210
Restrictions 210

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 Contents

Caller ID Blocking on Outbound Calls 213

Restrictions 214
Caller ID Blocking per Call 215
IP Phone Header Bar 216
Soft-Key Selection and Order 218
System Text Message for the Cisco IP Phone 7905G, Cisco IP Phone 7912G, Cisco IP Phones 7940 and
7940G, and Cisco IP Phones 7960 and 7960G 222
System Display Message (Idle URL) for the Cisco IP Phones 7940 and 7940G and Cisco IP Phones 7960
and 7960G 223
URL Provisioning for Customized Function Buttons 225
Direct FXO Trunk Lines 227
Restrictions 228
Examples 231
Cisco IP Phone 7970G and 7971G-GE Settings 232
Restrictions 233

Integrating Voice Mail with Cisco CME 235

Contents 235

Cisco CME Integration with Cisco Unity 235

Cisco CME Integration with Cisco Unity Express 236

DTMF Integration for Legacy Voice-Mail Devices 236

Configuring DTMF Patterns on the Router 236
Configuring Integration Files on Legacy Voice-Mail Systems 238

Configuring an Attendant for Primary Call Coverage 239

Contents 239
Related Features 239

Label Support 240

Monitor Lamp and Direct Station Select 241

Silent Ring 243

On-Hold Call Notification 244

Night Service 246

Configuring Call Blocking 251

Contents 251

Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date 251

Call-Blocking (Toll Bar) Override 254

Restrictions 254
Do-Not-Disturb Service 256

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Contents

Class of Restriction 257

Configuring Secondary Call Coverage 261

Contents 261

Related Features 261

Information About Configuring Call Coverage 262

Overlaid Ephone-dns 262

Restrictions 263
Call Waiting for Overlaid Ephone-dns 267
Ephone-dn Dial-Peer Preference 270

Huntstop 271

Ephone Hunt Groups 273

Configuring Sequential Ephone Hunt Groups 277
Configuring Peer Ephone Hunt Groups 280
Configuring Longest-Idle Ephone Hunt Groups 282
Automatic Hunt Group Logout 284
Restrictions 285
Call-Pickup Groups 286

Configuring Directories 289

Contents 289

Local Directory Order 289

Additional Directory Entries 290

Local Directory Disable 291

Configuring Productivity Tools 293

Contents 293
Flash Soft Key for Hookflash Functionality 293

Intercom 294

Paging 298
Restrictions 299
Configuring Paging for a Single Group 300
Configuring Paging for a Combined Group 303
Prerequisites 303
Account Code Entry by User 306

Applications Integration with Cisco CME 306

Installing Cisco IOS TSP 307
Prerequisites 307
Modifying a TSP Configuration 309

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 Contents

Removing Cisco IOS TSP 311

Prerequisites 311
Verifying Basic TAPI Operation 311
Troubleshooting Cisco IOS TSP 312
Troubleshooting Cisco IOS TSP on the Cisco CME Router 312
Troubleshooting Cisco IOS TSP on a PC 312
XML API 313

Related Features 315

Monitoring and Managing a Cisco CME System 317

Troubleshooting a Cisco CME System 321

Appendix A: Configuring Loopback Call Routing 323

Contents 323

Information About Loopback Call Routing 323

Configuring Loopback Call Routing 324

Appendix B: Providing Cisco CME Support for SIP 329

Contents 329

DTMF Relay for SIP Applications and Voice Mail 329

DTMF Relay Using SIP RFC 2833 329
DTMF Relay Using SIP Notify (Nonstandard) 331

SIP Register Support 333

Call Transfer over SIP Networks 335

Call Forwarding over SIP Networks 337

INDEX

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Cisco CallManager Express 3.3 System Administrator Guide

xii Cisco CME Tcl IVR scripts version 2.1 and later versions
 Feature Map

Callback Busy Subscriber (Configuring Cisco CME Phone

A Features) 178
Called Name and Ephone-dn Name Display (Configuring
Abbreviated Dialing (Configuring Cisco CME Phone
Cisco CME Phone Features) 201
Features) 183
Caller ID Blocking on Outbound Calls (Configuring
Account Code Entry (Configuring Productivity Tools) 306
Cisco CME Phone Features) 213
Additional Directory Entries (Configuring Directories) 290
Caller ID Blocking Per Call (Configuring Cisco CME Phone
After Hours Call Blocking (Configuring Call Blocking) 251 Features) 215
API, TAPI (Configuring Productivity Tools) 306 Call-Waiting Beep (Configuring Cisco CME Phone Features)
API, XML (Configuring Productivity Tools) 313 198
Applications Integration (Configuring Productivity Tools) 306 Call-Waiting Ring (Configuring Cisco CME Phone Features)
Automatic Hunt Group Logout (Configuring Secondary Call 200
Coverage) 284 Cisco IOS TSP (Configuring Productivity Tools) 306
Automatic Line Selection (Configuring Cisco CME Phone Cisco IP Phone 7970G and 7971G-GE Settings (Configuring
Features) 187 Cisco CME Phone Features) 232
Cisco CME System Parameters (Setting Up Phones in a
Cisco CME System) 46
B Conference Gain Control (Configuring Cisco CME Phone
Features) 194
Busy Timeout (Setting Up Optional Cisco CME System Conference Initiator Drop-Off Control (Configuring
Features) 166 Cisco CME Phone Features) 196
Conference, Three Party G.711 (Configuring Cisco CME
Phone Features) 194
C

Call Blocking by Time and Date (Configuring Call Blocking) D

251
Call Blocking Override (Configuring Call Blocking) 254 Dedicated FXO Trunk Lines (Configuring Cisco CME Phone
Call Forwarding (Configuring Call Transfer and Call Features) 227
Forwarding) 87 DHCP Setup (Setting Up a Cisco CME System) 38
Call Park (Setting Up Optional Cisco CME System Features) Dial Plan Pattern (Setting Up Phones in a Cisco CME System)
161 65
Call Pickup Groups (Configuring Secondary Call Coverage) Direct Station Select (Configuring an Attendant for Primary
286 Call Coverage) 241
Call Transfer (Configuring Call Transfer and Call Direct Station Select Transfer (Configuring Cisco CME
Forwarding) 87 Phone Features) 189
Call Waiting for Overlaid Ephone-dns (Configuring Directory Disable, Local (Configuring Directories) 291
Secondary Call Coverage) 267 Directory Entries, Additional (Configuring Directories) 290

Cisco CallManager Express 3.3 System Administrator Guide

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 Feature Map

Directory Order, Local (Configuring Directories) 289

H
Do Not Disturb Call Forward (Configuring Cisco CME
Phone Features) 192
Hairpin Call Routing (Configuring Call Transfer and Call
Do Not Disturb Service (Configuring Call Blocking) 256 Forwarding) 93
Do Not Disturb, Allow Feature Ring (Configuring Hookflash for IP Phones (Configuring Productivity Tools) 293
Cisco CME Phone Features) 193
Hunt Group Member Logout (Configuring Secondary Call
DSP Farms (Transcoding Between G.729 and G.711) 130 Coverage) 284
DTMF Integration for Legacy Voice-Mail Devices Hunt Groups, Ephone (Configuring Secondary Call
(Integrating Voice Mail with Cisco CME) 236 Coverage) 273
DTMF Relay for H.323 Networks (Setting Up Phones in a Huntstop (Configuring Secondary Call Coverage) 271
Cisco CME System) 64
DTMF Relay for SIP and Voice Mail (Providing Cisco CME
Support for SIP) 329
I

Intercom (Configuring Productivity Tools) 294

E Intercom No-Mute (Configuring Productivity Tools) 294
Interdigit Timeout (Setting Up Optional Cisco CME System
Ephone Hunt Groups (Configuring Secondary Call Coverage)
Features) 167
273
International Support (Setting Up Phones in a Cisco CME
Ephone-dn Dial-Peer Preference (Configuring Secondary
System) 72
Call Coverage) 270
IP Phone Header Bar Display (Configuring Cisco CME
Phone Features) 216

Feature Ring (Setting Up Phones in a Cisco CME System) 63 L

Flash Soft Key for Hookflash (Configuring Productivity
Label Support (Configuring an Attendant for Primary Call
Tools) 293
Coverage) 240
Forward, Do Not Disturb (Configuring Cisco CME Phone
Line Selection, Automatic (Configuring Cisco CME Phone
Features) 192
Features) 187
Forward, Restrict Number of Digits (Configuring Cisco CME
Local Directory Disable (Configuring Directories) 291
Phone Features) 190
Local Directory Order (Configuring Directories) 289
Forwarding (Configuring Call Transfer and Call Forwarding)
87 Local Speed Dial (Configuring Cisco CME Phone Features)
180
Forwarding over SIP Networks (Providing Cisco CME
Support for SIP) 337 Loopback Call Routing (Configuring Loopback Call Routing)
323
FXO Lines, Dedicated (Configuring Cisco CME Phone
Features) 227

M
G
Monitor Lamp (Configuring an Attendant for Primary Call
Coverage) 241
Graphical User Interface (GUI) (Setting Up the Cisco CME
GUI) 145 Monitor Mode (Setting Up Phones in a Cisco CME System)
63
Monitor-Line Speed Dial (Configuring Cisco CME Phone
Features) 179
Music on Hold (Setting Up Optional Cisco CME System
Features) 169

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 Feature Map

Phone, Monitor Lamp (Configuring an Attendant for Primary

N Call Coverage) 241
Phone, Silent Ring (Configuring an Attendant for Primary
Network Time Protocol Setup (Setting Up a Cisco CME
Call Coverage) 243
System) 42
Pickup Groups (Configuring Secondary Call Coverage) 286
Night Service (Configuring an Attendant for Primary Call
Coverage) 246 Preference, Ephone-dn Dial Peer (Configuring Secondary
Call Coverage) 270

O
R
On-Hold Notification (Configuring an Attendant for Primary
Call Coverage) 244 Resetting Phones (Setting Up Phones in a Cisco CME
System) 75
On-Hook Dialing (Configuring Cisco CME Phone Features)
178 Restarting Phones (Setting Up Phones in a Cisco CME
System) 75
Overlaid Ephone-dns (Configuring Secondary Call
Coverage) 262 Ringing Timeout (Setting Up Optional Cisco CME System
Features) 168
Overlaid Ephone-dns, Call Waiting (Configuring Secondary
Call Coverage) 267
Overlaid Ephone-dns, Called Name and Number Display
(Configuring Cisco CME Phone Features) 201 S

Secondary Dial Tone (Setting Up Optional Cisco CME

System Features) 165
P
Setup Tool (Setting Up Phones in a Cisco CME System) 50
Paging (Configuring Productivity Tools) 298 Silent Ring (Configuring an Attendant for Primary Call
Coverage) 243
Personal Speed Dial (Configuring Cisco CME Phone
Features) 182 SIP Network Call Transfers (Providing Cisco CME Support
for SIP) 335
Phone Button Labels (Configuring an Attendant for Primary
Call Coverage) 240 SIP Networks, Call Forwarding (Providing Cisco CME
Support for SIP) 337
Phone Buttons, Overlaid Ephone-dns (Configuring
Secondary Call Coverage) 262 SIP Register Support (Providing Cisco CME Support for SIP)
333
Phone Display, Header Bar (Configuring Cisco CME Phone
Features) 216 Soft Key Selection and Order (Configuring Cisco CME
Phone Features) 218
Phone Display, System Display Message (Configuring
Cisco CME Phone Features) 223 Speed Dial Buttons (Configuring Cisco CME Phone
Features) 183
Phone Display, System Text Message (Configuring
Cisco CME Phone Features) 222 Speed Dial, Local (Configuring Cisco CME Phone Features)
180
Phone Function Buttons, Customizing (Configuring
Cisco CME Phone Features) 225 Speed Dial, Monitor Line (Configuring Cisco CME Phone
Features) 179
Phone Setup, Basic (Setting Up Phones in a Cisco CME
System) 50 Speed Dial, Personal (Configuring Cisco CME Phone
Features) 182
Phone Setup, Using Setup Tool (Setting Up Phones in a
Cisco CME System) 50 System Display Message for IP Phones (Configuring
Cisco CME Phone Features) 223
Phone Soft Key, Account Code (Configuring Productivity
Tools) 306 System Text Message Display on IP Phones (Configuring
Cisco CME Phone Features) 222
Phone Soft Key, Selection and Order (Configuring
Cisco CME Phone Features) 218

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 Feature Map

Tandem Gateway (Configuring Call Transfer and Call

Forwarding) 95
TAPI Applications (Configuring Productivity Tools) 306
Timeout, Busy (Setting Up Optional Cisco CME System
Features) 166
Timeout, Interdigit (Setting Up Optional Cisco CME System
Features) 167
Timeout, Ringing (Setting Up Optional Cisco CME System
Features) 168
Toll Bar Based on Time and Date (Configuring Call
Blocking) 251
Toll Bar Override (Configuring Call Blocking) 254
Transcoding Between G.729 and G.711 (Transcoding
Between G.729 and G.711) 125
Transfer (Configuring Call Transfer and Call Forwarding) 87
Transfer, Direct-Station Select (Configuring Cisco CME
Phone Features) 189
Transfer, SIP Networks (Providing Cisco CME Support for
SIP) 335
Translation Profiles (Setting Up Phones in a Cisco CME
System) 69
Translation Rules (Setting Up Phones in a Cisco CME
System) 68

URL Provisioning for Customized Function Buttons

(Configuring Cisco CME Phone Features) 225

Voice-Mail Integration (Integrating Voice Mail with

Cisco CME) 235

XML API (Configuring Productivity Tools) 313

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xvi
 Preface

This preface discusses the objectives, audience, organization, and conventions of this document. It also
provides sources for obtaining documentation and technical assistance from Cisco Systems.

Documentation Objectives
This document describes the tasks and commands necessary to configure and maintain
Cisco CallManager Express (Cisco CME).

Audience
This document is intended primarily for system administrators who configure and maintain Cisco CME
but who may not be familiar with the tasks, the relationship between tasks, or the Cisco IOS software
commands necessary to perform particular tasks. This configuration guide is also intended for those
users experienced with Cisco CME who need to know about new features, new configuration options,
and new software characteristics in the current Cisco IOS software release.
System administrators who are setting up a Cisco CME system should be familiar with the following:
• TCP/IP fundamentals: IP addressing, routing, DHCP, HTTP, NTP, TFTP.
• Cisco IOS fundamentals: CLI operation, VLAN configuration, Flash memory and TFTP file
management.
• VoIP fundamentals: Configuring and verifying dial peers and voice ports.

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 Preface
Documentation Organization

Documentation Organization
This document includes the following sections:

Table 1 Document Organization

Title Description
Feature Map Alphabetically organized links to all the features
in the System Administrator Guide.
Cisco CallManager Express Overview High-level description of Cisco CME procedures
and concepts. Includes software prerequisites and
download instructions.
Setting Up a Cisco CME System Basic steps to prepare a router for Cisco CME.
Setting Up Phones in a Cisco CME System Steps to set up initial Cisco CME phones.
Configuring Call Transfer and Call Forwarding Description of call transfer and forwarding
options and configuration.
Transcoding Between G.729 and G.711 How to configure Cisco CME to transcode G.729
voice signals to G.711, and vice versa, for various
Cisco CME features
Setting Up the Cisco CME GUI How to set up the Cisco CME GUI for a
web-browser-based interface for administrators
and phone users.
Setting Up Optional Cisco CME System Features Features that affect all Cisco CME users on a
systemwide basis.
Configuring Cisco CME Phone Features Features that are set up on individual phones.
Integrating Voice Mail with Cisco CME Features that allow integration with voice-mail
systems.
Configuring an Attendant for Primary Call Features that assist in setting up an attendant to be
Coverage the single initial source of incoming call
coverage.
Configuring Call Blocking Selective restrictions on outgoing calls.
Configuring Secondary Call Coverage Features that provide flexibility to manage
coverage of incoming calls.
Configuring Directories Directories that are maintained in Cisco CME.
Configuring Productivity Tools Features that improve employee efficiency.
Monitoring and Managing a Cisco CME System Commands to help observe system functioning.
Troubleshooting a Cisco CME System Commands to help diagnose problems.
Appendix A: Configuring Loopback Call Routing Software-based, limited emulation of connected,
back-to-back physical voice ports to provide a
loopback call-routing path for voice calls.
Appendix B: Providing Cisco CME Support for Special configurations for Cisco CME systems on
SIP SIP networks.
Index Links to key terms and commands in the text.

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 Preface
Document Conventions

Note For a list of related documents, see the “Additional References” section in the “Cisco CallManager
Express Overview” chapter.

Document Conventions
Within Cisco IOS software documentation, the term router is generally used to refer to a variety of Cisco
products (for example, routers, access servers, and switches). Routers, access servers, and other
networking devices that support Cisco IOS software are shown interchangeably within examples. These
products are used only for illustrative purposes; that is, an example that shows one product does not
necessarily indicate that other products are not supported.
The Cisco IOS documentation set uses the following conventions:

Convention Description
^ or Ctrl The ^ and Ctrl symbols represent the Control key. For example, the key combination ^D or Ctrl-D
means hold down the Control key while you press the D key. Keys are indicated in capital letters but
are not case sensitive.
string A string is a nonquoted set of characters shown in italics. For example, when setting an SNMP
community string to public, do not use quotation marks around the string or the string will include the
quotation marks.

Command syntax descriptions use the following conventions:

Convention Description
boldface Boldface text indicates commands and keywords that you enter literally as shown.
italics Italic text indicates arguments for which you supply values.
[x] Square brackets enclose an optional element (keyword or argument).
| A vertical line indicates a choice within an optional or required set of keywords or arguments.
[x | y] Square brackets enclosing keywords or arguments separated by a vertical line indicate an optional
choice.
{x | y} Braces enclosing keywords or arguments separated by a vertical line indicate a required choice.

Nested sets of square brackets or braces indicate optional or required choices within optional or required
elements. For example:

Convention Description
[x {y | z}] Braces and a vertical line within square brackets indicate a required choice within an optional element.

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Obtaining Documentation

Examples use the following conventions:

Convention Description
screen Examples of information displayed on the screen are set in Courier font.
boldface screen Examples of text that you must enter are set in Courier bold font.
< > Angle brackets enclose text that is not printed to the screen, such as passwords.
! An exclamation point at the beginning of a line indicates a comment line. (Exclamation points are also
displayed by the Cisco IOS software for certain processes.)
[ ] Square brackets enclose default responses to system prompts.

The following conventions are used to attract the attention of the reader:

Caution Means reader be careful. In this situation, you might do something that could result in equipment
damage or loss of data.

Note Means reader take note. Notes contain helpful suggestions or references to materials not contained in
this manual.

Timesaver Means the described action saves time. You can save time by performing the action described in the
paragraph.

Obtaining Documentation
Cisco provides several ways to obtain documentation, technical assistance, and other technical
resources. These sections explain how to obtain technical information from Cisco Systems.

Cisco.com
You can access the most current Cisco documentation on the World Wide Web at this URL:
http://www.cisco.com/univercd/home/home.htm
You can access the Cisco website at this URL:
http://www.cisco.com
International Cisco websites can be accessed from this URL:
http://www.cisco.com/public/countries_languages.shtml

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Documentation CD-ROM
Cisco documentation and additional literature are available in a Cisco Documentation CD-ROM
package, which may have shipped with your product. The Documentation CD-ROM is updated regularly
and may be more current than printed documentation. The CD-ROM package is available as a single unit
or through an annual or quarterly subscription.
Registered Cisco.com users can order a single Documentation CD-ROM (product number
DOC-CONDOCCD=) through the Cisco Ordering tool:
http://www.cisco.com/en/US/partner/ordering/ordering_place_order_ordering_tool_launch.html
All users can order annual or quarterly subscriptions through the online Subscription Store:
http://www.cisco.com/go/subscription

Ordering Documentation
You can find instructions for ordering documentation at this URL:
http://www.cisco.com/univercd/cc/td/doc/es_inpck/pdi.htm
You can order Cisco documentation in these ways:
• Registered Cisco.com users (Cisco direct customers) can order Cisco product documentation from
the Networking Products MarketPlace:
http://www.cisco.com/en/US/partner/ordering/index.shtml
• Nonregistered Cisco.com users can order documentation through a local account representative by
calling Cisco Systems Corporate Headquarters (California, USA.) at 408 526-7208 or, elsewhere in
North America, by calling 800 553-NETS (6387).

Documentation Feedback
You can submit comments electronically on Cisco.com. On the Cisco Documentation home page, click
Feedback at the top of the page.
You can send your comments in e-mail to [email protected].
You can submit comments by using the response card (if present) behind the front cover of your
document or by writing to the following address:
Cisco Systems
Attn: Customer Document Ordering
170 West Tasman Drive
San Jose, CA 95134-9883
We appreciate your comments.

Obtaining Technical Assistance

For all customers, partners, resellers, and distributors who hold valid Cisco service contracts, the Cisco
Technical Assistance Center (TAC) provides 24-hour, award-winning technical support services, online
and over the phone. Cisco.com features the Cisco TAC website as an online starting point for technical
assistance.

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Cisco TAC Website

The Cisco TAC website (http://www.cisco.com/tac) provides online documents and tools for
troubleshooting and resolving technical issues with Cisco products and technologies. The Cisco TAC
website is available 24 hours a day, 365 days a year.
Accessing all the tools on the Cisco TAC website requires a Cisco.com user ID and password. If you
have a valid service contract but do not have a login ID or password, register at this URL:
http://tools.cisco.com/RPF/register/register.do

Opening a TAC Case

The online TAC Case Open Tool (http://www.cisco.com/tac/caseopen) is the fastest way to open P3 and
P4 cases. (Your network is minimally impaired or you require product information). After you describe
your situation, the TAC Case Open Tool automatically recommends resources for an immediate solution.
If your issue is not resolved using these recommendations, your case will be assigned to a Cisco TAC
engineer.
For P1 or P2 cases (your production network is down or severely degraded) or if you do not have Internet
access, contact Cisco TAC by telephone. Cisco TAC engineers are assigned immediately to P1 and P2
cases to help keep your business operations running smoothly.
To open a case by telephone, use one of the following numbers:
Asia-Pacific: +61 2 8446 7411 (Australia: 1 800 805 227)
EMEA: +32 2 704 55 55
USA: 1 800 553-2447
For a complete listing of Cisco TAC contacts, go to this URL:
http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml

TAC Case Priority Definitions

To ensure that all cases are reported in a standard format, Cisco has established case priority definitions.
Priority 1 (P1)—Your network is “down” or there is a critical impact to your business operations. You
and Cisco will commit all necessary resources around the clock to resolve the situation.
Priority 2 (P2)—Operation of an existing network is severely degraded, or significant aspects of your
business operation are negatively affected by inadequate performance of Cisco products. You and Cisco
will commit full-time resources during normal business hours to resolve the situation.
Priority 3 (P3)—Operational performance of your network is impaired, but most business operations
remain functional. You and Cisco will commit resources during normal business hours to restore service
to satisfactory levels.
Priority 4 (P4)—You require information or assistance with Cisco product capabilities, installation, or
configuration. There is little or no effect on your business operations.

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Obtaining Additional Publications and Information

Obtaining Additional Publications and Information

Information about Cisco products, technologies, and network solutions is available from various online
and printed sources.
• The Cisco Product Catalog describes the networking products offered by Cisco Systems, as well as
ordering and customer support services. Access the Cisco Product Catalog at this URL:
http://www.cisco.com/en/US/products/products_catalog_links_launch.html
• Cisco Press publishes a wide range of networking publications. Cisco suggests these titles for new
and experienced users: Internetworking Terms and Acronyms Dictionary, Internetworking
Technology Handbook, Internetworking Troubleshooting Guide, and the Internetworking Design
Guide. For current Cisco Press titles and other information, go to Cisco Press online at this URL:
http://www.ciscopress.com
• Packet magazine is the Cisco quarterly publication that provides the latest networking trends,
technology breakthroughs, and Cisco products and solutions to help industry professionals get the
most from their networking investment. Included are networking deployment and troubleshooting
tips, configuration examples, customer case studies, tutorials and training, certification information,
and links to numerous in-depth online resources. You can access Packet magazine at this URL:
http://www.cisco.com/go/packet
• iQ Magazine is the Cisco bimonthly publication that delivers the latest information about Internet
business strategies for executives. You can access iQ Magazine at this URL:
http://www.cisco.com/go/iqmagazine
• Internet Protocol Journal is a quarterly journal published by Cisco Systems for engineering
professionals involved in designing, developing, and operating public and private internets and
intranets. You can access the Internet Protocol Journal at this URL:
http://www.cisco.com/en/US/about/ac123/ac147/about_cisco_the_internet_protocol_journal.html
• Training—Cisco offers world-class networking training. Current offerings in network training are
listed at this URL:
http://www.cisco.com/en/US/learning/index.html

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 Cisco CallManager Express Overview

Prior to Version 3.0, Cisco CallManager Express was known as Cisco IOS Telephony Services
(Cisco ITS). Cisco CallManager Express (Cisco CME) is a call-processing application in Cisco IOS
software that enables Cisco routers to deliver key system or hybrid PBX functionality for enterprise
branch offices or small businesses. Cisco CME is ideal for customers who have data connectivity
requirements and also have a need for a telephony solution in the same office. Whether offered through
a service provider’s managed services offering or purchased directly by a corporation, Cisco CME offers
most of the core telephony features required in the small office, as well as many advanced features not
available with traditional telephony solutions. Being able to deliver IP telephony and data routing using
a single converged solution allows customers to optimize their operations and maintenance costs,
resulting in a very cost-effective solution that meets office needs.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

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Feature Specifications

Feature History
Release Version Modification
12.1(5)YD Version 1.0 Cisco IOS Telephony Services was introduced on the Cisco 2600 series,
Cisco 3600 series, and Cisco IAD2420 series.
• Support for Cisco IP Phone 7910, Cisco IP Phone 7940, and
Cisco IP Phone 7960
• Multiple lines per Cisco IP phone
• Multiple shared-line appearances across phones
• Call forwarding for all calls or for busy and no-answer conditions
• Call transfer
• Distinctive ringing for external and internal calls
• Dial-plan class of restriction (COR)
• Call hold and retrieve
• Call pickup of on-hold calls
• Caller identification display and blocking
• Function keys
• Speed dialing
• Cisco IP phones derive the date and time from the router through
Network Time Protocol (NTP)
• Interworking with Cisco gatekeeper
• Analog foreign exchange station (FXS) and foreign exchange office
(FXO) ports
• On-net calls using Voice over IP (VoIP) H.323, Voice over Frame
Relay (VoFR), and Voice over ATM (VoATM)

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12.2(2)XT Version 2.0 This service was implemented on the Cisco 1750 and Cisco 1751.
• Cisco IP Conference Station 7935 support
• Two-call support for Cisco IP Phone 7910
• Three-party conference (G.711 calls)
• Intercom for Cisco IP phones
• Paging for Cisco IP phones and for external system
• Call transfers across an H.323 network
• Music on hold (MOH)
• Graphical user interface (GUI) using a standard web browser
• Recent call history and activity display
• Call forwarding enhancements (huntstop)
• Digit manipulation using translation rules
• Enhancements to distinctive ringing for internal and external calls
• Cisco Unity voice-mail integration including message-waiting
indication
• On-hold call timeout alert
• Session Initiation Protocol (SIP) unsolicited message-waiting
notification support
• Local phone directory display and search on Cisco IP phone
• XML services support on Cisco IP phones
• Basic Telephony Application Programming Interface (TAPI)-aware
PC application support
• Interactive voice response (IVR) and auto-attendant support using
Tool Command Language (Tcl)
12.2(8)T Version 2.0 This service was integrated into Cisco IOS Release 12.2(8)T and
implemented on the Cisco 3725 and Cisco 3745.
12.2(8)T1 Version 2.0 This service was implemented on the Cisco 2600XM and Cisco 2691.
12.2(11)T Version 2.01 • This service was implemented on the Cisco 1760, and support for
Cisco 1750 was removed.
• Support was added for an increased number of directory numbers or
virtual voice ports on Cisco IP phones.
• Support was added for ATA-186.
• Support was added for top-line display description on the
Cisco IP Phones 7940 and 7940G and the Cisco IP Phones 7960
and 7960G.
12.2(13)T Version 2.02 • This service was implemented on the Cisco Catalyst 4000 family,
Cisco Catalyst 4224, and Cisco 3640A. Support was removed for
the Cisco 2610, Cisco 2611, Cisco 2620, Cisco 2621, and
Cisco 3620.
• Support was added for an increased number of directory numbers or
virtual voice ports on Cisco IP phones.

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12.2(11)YT Version 2.1 • Consultative transfer using the ITU-T H.450.2 standard
• Call forwarding using the ITU-T H.450.3 standard
• Support for French, German, Italian, and Spanish languages for
phone displays and call progress tones
• eXtensible Markup Language (XML) scripting for administrative
customization
• Cisco IP Phone Expansion Module 7914
12.2(11)YT1 Version 2.1 The reset command was modified and the restart command was
introduced to provide more options when IP phones are rebooted after
configuration updates.
12.2(15)T Version 2.1 ITS Version 2.1 was integrated into Cisco IOS Release 12.2(15)T.
12.2(15)ZJ Version 3.0 This service was implemented on the Cisco 3640A and the
Cisco IAD2430 series.
Support was added for the following features:
• ITS setup tool for quick installation
• Automatic assignment of free extension numbers to new IP phones
• Night service
• Call blocking (toll bar) based on time of day, day of week, or date
• Call blocking (toll bar) override
• Call pickup and call-pickup groups
• Hunt groups
• Secondary dial tone
• Three types of speed dial: speed-dial buttons, local speed-dial
numbers common to all users, and personal speed-dial numbers that
can be updated by an administrator or from the phone.
• Cisco IP Phone 7902G, Cisco IP Phone 7905G, Cisco IP
Phone 7912G
• Account code entry
• Callback busy subscriber
• Do not disturb
• International date format, language, and call-progress tone support
• Call-forward-all soft key on Cisco IP phones
• Flash soft key for hookflash functionality for the public switched
telephone network (PSTN)
• Dual-line mode to support call waiting and other features
• Extension overlays for better call handling and distribution
• Fast-dial support
• GUI enhancements
• Label support
• Busy lamp monitor and direct station select

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12.2(15)ZJ Version 3.0 • Phone directory entry

continued continued
• Silent and feature ring options
12.2(15)ZJ1 Version 3.0 The name keyword was added to the clid strip command.
12.2(15)ZJ3 3.0 • The product name was changed from Cisco IOS Telephony Services
(Cisco ITS) to Cisco CallManager Express (Cisco CME).
• Support was added for an increased number of directory numbers or
virtual voice ports on Cisco IP phones.
• Limited support was provided for the Cisco Wireless IP Phone 7920
in 7960-emulation mode.
• Local hairpin call routing was supported as an option for networks
that cannot support H.450 call transfer and forwarding. This feature
requires installation of the Tcl script app_h450_transfer.2.0.0.8.tcl
or a later version.
12.3(4)T 3.0 Cisco CME 3.0 was integrated into Cisco IOS Release 12.3(4)T.
12.2(15)ZJ4 3.0 The secondary keyword was added to the calling-number local
command.
12.3(7)T 3.1 • Basic call handling, call transfer, and call
forwarding—Enhancements for VoIP networks containing a mix of
platforms that support H.450.2 and H.450.3 standards, such as
Cisco CME 3.1, Cisco CME 3.0, and Cisco ITS V2.1, and
platforms that do not support H.450.2 and H.450.3 standards, such
as Cisco CallManager, Cisco BTS Softswitch (BTS), and
Cisco PSTN Gateway (PGW).
– Support for H.450.12 standards.
– Automatic detection of Cisco CallManager endpoints.
– Hairpin VoIP-to-VoIP call routing and routing to an H.450
tandem gateway.
• Call park allows calls to be commonly held and retrieved by anyone.
• Automatic line selection enhancement specifies a particular button
as the line automatically used for outgoing calls.
• CFwdAll soft key restriction control restricts the number of digits
that can be entered using the CFwdAll soft key.
• Ephone-hunt group enhancements allows secondary numbers for
pilot numbers.
• Language display localization and directory search are supported
on Cisco IP Phone 7905G and Cisco IP Phone 7912G. Call progress
tone localization is supported on Cisco IP Phone 7902G, Cisco IP
Phone 7905G, and Cisco IP Phone 7912G.
• The Cisco Wireless IP Phone 7920 and Cisco IP Conference
Station 7936 are fully supported in the Cisco IOS command-line
interface (CLI) and Cisco CME GUI. The Cisco Wireless IP
Phone 7920 internally supports display translation to French and
German from English.

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12.3(11)T 3.2 Network features

• Transcoding between G.711 and G.729; see the “Transcoding
Between G.729 and G.711” chapter.
• H.323-to-SIP Call routing to Cisco Unity Express; see Integrating
Cisco CallManager Express with Cisco Unity Express.
• DTMF relay enhancement for SIP calls; see “DTMF Relay Using
SIP Notify (Nonstandard)” section on page 331.
Phone features
• Customization of soft-key display; see the “Soft-Key Selection and
Order” section on page 218.
Call center
• Dynamic hunt group login and logout; see the “Do-Not-Disturb
Service” section on page 256.
• Addition of the statistics and statistics last keywords to the show
ephone-hunt command; see show ephone hunt in the
Cisco CallManager Express 3.3 Command Reference.
• Addition of the longest-idle keywork to the ephone-hunt
command; see the “Configuring Longest-Idle Ephone Hunt
Groups” section on page 282.
System features
• Call-waiting beep customization; see the “Call-Waiting Beep”
section on page 198.
• Called name display for overlay ephone-dns and dialed number
identification service (DNIS) calls; see the “Called Name and
Ephone-dn Name Display” section on page 201.
• IP phones display the original calling parties’ IDs automatically.
• Conference initiator drop-off control; see the “Conference Initiator
Drop-Off Control” section on page 196.
• Consult transfer support for direct station; see the “Consult Transfer
Support for Direct Station Select” section on page 189.
• Direct FXO trunk line support; see the “Direct FXO Trunk Lines”
section on page 227.
• Immediate call forward to voice mail or other call-forward no
answer (CFNA) target numbers; see the “Do Not Disturb Features”
section on page 191.
• When a local IP phone calls another local IP phone that is in the do
not disturb (DND) state, the message “Ring out DND” is displayed
on the calling phone, indicating that the target phone is in the DND
state.

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12.3(11)T 3.2 continued • Monitor-line button speed dial; see the “Monitor-Line Button Speed
continued Dial” section on page 179
• Night service call notification is sent automatically every 12
seconds until the call is either answered or aborted.
• Translation profile support for ephone-dn; see the “Translation
Profiles” section on page 69.
12.3(11)XL 3.2.1 • A basic automatic call distribution (B-ACD) and auto attendant
(AA) service is available to provide the following:
– A menu for outside callers with options that allow one-key
dialing and extension-number access
– Call queuing
– Tools for obtaining call statistics
See the “Cisco CME Basic Automatic Call Distribution and
Auto-Attendant Service” chapter in Cisco CME B-ACD and Tcl
Call-Handling Applications.
• The Cisco IP Phone 7970G is supported. See the “Phone Firmware
Files” section on page 14 and “Cisco IP Phone 7970G and
7971G-GE Settings” section on page 232.
• Call Waiting for overlaid ephone-dns. See the “Call Waiting for
Overlaid Ephone-dns” section on page 267.
• Ringing for call-waiting notification per ephone-dn. See the
“Call-Waiting Ring” section on page 200.
• Do not disturb (DND) can be blocked from phones. See the “DND
Disable for Feature Ring” section on page 193.
• An ephone-dn of an ephone hunt group (see the “Ephone Hunt
Groups” section on page 273) can be configured to log out
automatically after a call to the ephone-dn is unanswered. See the
“Automatic Hunt Group Logout” section on page 284.
12.3(11)XL1 3.2.2 • The Cisco IP Phone 7971G-GE is supported. See the “Phone
Firmware Files” section on page 14 and “Cisco IP Phone 7970G
and 7971G-GE Settings” section on page 232.
• A conference gain control for external calls has been added. See the
“Three-Party G.711 Conference Calls” section on page 194.
• An intercom no-mute function has been added. See the “Intercom”
section on page 294.
• Call-park slot status can be observed using monitor mode. See the
“Call Park” section on page 161

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Contents

12.3(14)T 3.3 • Cisco CME B-ACD has a new mode called drop-through mode, in
which incoming calls to the B-ACD AA are put directly through to
an agent without encountering an interactive menu.
• Cisco CME B-ACD now supports multiple AA applications per
Cisco CME system.
Note For more information, see the “Cisco CME Basic Automatic
Call Distribution and Auto-Attendant Service” chapter in
Cisco CME B-ACD and Tcl Call-Handling Applications.

• The maximum number of members in an ephone-hunt group has

increased to 20.
• The maximum number of ephone-dns that can be overlaid on a
single phone button has increased to 25.
Supported Platforms
Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image
support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on
Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel
at the login dialog box and follow the instructions that appear.
Supported Cisco IP Phones and Phone Devices
See the Cisco CallManager Express 3.3 Specifications.

Contents
• Cisco CallManager Express Description, page 9
• Prerequisites, page 11
• Restrictions, page 17
• Information About Setting Up a Cisco CME System, page 19
• Before You Start: Basic Cisco CME Concepts, page 22
• What to Do Next, page 32
• Additional References, page 32

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Cisco CallManager Express Description

Cisco CallManager Express Description

Cisco CallManager Express is a feature-rich entry-level IP telephony solution that is integrated directly
into Cisco IOS software. Cisco CallManager Express allows small business customers and autonomous
small enterprise branch offices to deploy voice, data, and IP telephony on a single platform for small
offices, thereby streamlining operations and lowering network costs.

Cisco CallManager Express Network Scenarios

Figure 1 shows a typical deployment of a Cisco CallManager Express (Cisco CME) router with several
Cisco IP phones connected to it. The Cisco CME router is connected to the PSTN. The router can also
connect to a gatekeeper and a RADIUS billing server in the same network.

Figure 1 Cisco CallManager Express for the Small- and Medium-Size Office

Telephone Telephone

Fax

Cisco CME router

PSTN

RADIUS
billing
IP IP IP Cisco IP phones
server

PCs
62142

Gatekeeper

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Cisco CallManager Express Description

Figure 2 shows a branch office with several Cisco IP phones connected to a Cisco IAD2430 series using
Cisco CME. The Cisco IAD2430 series is connected to a multiservice router at a service provider office.
The multiservice router at the service provider office provides connection to the WAN and PSTN.

Figure 2 Cisco CallManager Express for Service Providers

Telephone Telephone
IP
PSTN network
Fax

Voice
switch
Cisco IAD2430
Service
T1/DSL/Cable
provider
IAD V office

IP IP IP Cisco IP phones
Voice-mail
Gatekeeper server
PCs

62145
Additional Features

Provisioning
The router provides a mechanism to provision Cisco CallManager Express, which allows you to perform
the following functions:
• Assign extension numbers to the line appearances on each Cisco IP phone.
• Assign numbers to the speed-dial buttons on each Cisco IP phone.
• Assign caller identification information to each extension number.
• Assign extension numbers to phones other than Cisco IP phones attached to the system by using the
standard voice-port and dial-peer configuration command-line interface (CLI).
• Provide dial-plan information to route calls to either PSTN lines or voice network connections.
For more information, see the “Setting Up Phones in a Cisco CME System” chapter.

Connecting Cisco IP Phones

Cisco IP phones can be connected to and disconnected from the Cisco CallManager Express router
without requiring a router reboot or manual status reset, a process sometimes called “hot-plugging.”

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Prerequisites

Prerequisites
Prerequisites for installing Cisco CallManager Express are grouped into the following categories:
• License Prerequisites, page 11
• Memory Prerequisites, page 11
• Network Prerequisites, page 11
• Software Prerequisites, page 12

License Prerequisites
You must purchase a base Cisco CallManager Express feature license and phone user licenses that entitle
you to use Cisco CallManager Express.

Note To support H.323 call transfers and forwards to network devices, such as Cisco CallManager, that do not
support the H.450 standard, a tandem gateway is required in the network. The tandem gateway must be
running Cisco IOS software 12.3(7)T or higher and requires the Integrated Voice and Video Services
feature license (FL-GK-NEW-xxx), which includes H.323 gatekeeper, IP-to-IP gateway, and H.450
tandem functionality.

Memory Prerequisites
• For information about the maximum number of Cisco IP phones, maximum number of directory
numbers (ephone-dns) or virtual voice ports, and memory requirements for Cisco CME, refer to
Cisco CallManager Express 3.3 Specifications at the following URL:
http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cme33/cme33spc.htm
• Disable Smartinit and allocate ten percent of the total DRAM to support Cisco CME with the
following command:
Router(config)# memory-size iomem 10

Network Prerequisites
• IP routing must be enabled.
• VoIP networking must be operational. For quality and security purposes, it is recommended to have
separate virtual LANs (VLANs) for data and voice. The IP network assigned to each VLAN should
be large enough to support addresses for all nodes on that VLAN. Cisco CME phones receive their
IP addresses from the voice network, whereas all other nodes such as PCs, servers, and printers
receive their IP addresses from the data network.
• The voice VLAN should be configured to receive IP addresses from a Dynamic Host Configuration
Protocol (DHCP) server. A DHCP server for Cisco CME phones is designated during Cisco CME
setup. For more information, see the “Setting Up DHCP Service for Cisco CME” section in the
“Setting Up a Cisco CME System” chapter.

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Prerequisites

• The clock on the router must be configured to the proper date and time. All IP phones connected to
the router will receive their time and date settings from the router clock. To keep the router clock
accurate, configure the router for Network Time Protocol (NTP).
• Trivial File Transfer Protocol (TFTP) must be enabled on the router to allow IP phones to download
phone firmware files.

Software Prerequisites
To use Cisco CME, you need to download Cisco IOS software and Cisco CME files, which are available
through the Cisco Software Center. The files that you need and the URLs where they can be obtained are
described in the following sections:
• Cisco IOS Software
• Cisco CallManager Express Files

Cisco IOS Software

• Download and install the Cisco IOS Release 12.3(14)T, 12.4, or later version IP VOICE image from
the Cisco Software Center. Please consult Feature Navigator to find the appropriate Cisco IOS software
feature set if you wish to enable other Cisco IOS features with Cisco CME.
• Cisco CME 3.3 requires a minimum of an 12.3(14)T IP VOICE image on the Cisco 2600XM, 2800,
3600, 3700, and 3800 series, and an IP/ADSL VOX PLUS image on the Cisco 1751-V and Cisco
1760 series. If you are using AIM-ATM-VOICE-30 on Cisco 2600XM, 3600, and 3700 series, a
minimum of SP SERVICES image is required.

Cisco CallManager Express Files

For Cisco CME, certain files must be downloaded and installed in the Cisco CME router flash memory.
The process and files are discussed in the following sections:
• How to Download and Install Cisco CME Files, page 12
• List of Individual Cisco CallManager Express Files, page 13

Note If you are downgrading or upgrading Cisco CME and use the Cisco CME GUI, you must downgrade or
upgrade your GUI files. For more information, see the “GUI Files” section on page 15.

Note Customers who purchase a Cisco CME-enabled router bundle will have the necessary Cisco CME files
installed at time of manufacture.

How to Download and Install Cisco CME Files

Install Cisco CME files from the Cisco CME Software Download website using one of the following
methods:
• Zipped, Compressed Archive, page 13
• Tar Archive, page 13

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Prerequisites

Note The software version numbers that are listed in this section were current when this document was
written. Later versions may be available when you prepare to download files. You should use the latest
software version that is available on the software download site.

Zipped, Compressed Archive

The zipped, compressed archive contains all the Cisco CME files in a single archive file. To install files
from this archive, follow these steps:
1. Download the cme-xxx.zip file that is supported for the Cisco IOS release that is running on your
Cisco CME router from http://www.cisco.com/cgi-bin/tablebuild.pl/ip-key. The site lists the .zip file
that corresponds to the Cisco IOS release that is running on your router. Using a .zip file that does not
match the Cisco IOS release on your router can cause features to malfunction, such as the Cisco CME
GUI.
2. Uncompress cme-xxx.zip using the WinZip program.
3. Copy each of the resulting files except phone firmware files and CiscoIOSTSP.zip to router flash
memory. For phone firmware files, copy only the firmware files for phone types that you have at
your site. For CiscoIOSTSP.zip, refer to the installation instructions in the “Cisco IOS TSP
Download and Setup” section in the “Configuring Productivity Tools” chapter.

Tar Archive
At the Cisco CME Software Download website, you can find a single tar file (cme-basic-xxx.tar) that
contains both the basic Cisco CME system files and Cisco CME GUI files. A Cisco IOS command
allows you to uncompress the tar files and copy them to router flash at the same time. The
cme-basic-xxx.tar file does not contain the bundled TSP file (CiscoIOSTSP.zip), but you can download
that separately. To install files from the Cisco CME tar archive, use the following steps:
1. Download the desired version of the Cisco CME and the Cisco CME GUI tar archive from
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp to a TFTP server that is accessible to the
Cisco CME router.
2. Uncompress and copy each archive to router flash memory using the following command:
• archive tar /xtract source-url flash:/file-url
For example, to extract contents of cme-basic-123-11T.tar from TFTP server 192.168.1.1 to router
flash memory, use this command:
archive tar /xtract tftp://192.168.1.1/cme-basic-123_11Ttar flash:

Note The file called CiscoIOSTSP.zip is not included in the tar archive. To install TSP files, download
the CiscoIOSTSP.zip file separately and follow the installation instructions in the “Cisco IOS
TSP Download and Setup” section in the “Configuring Productivity Tools” chapter.

List of Individual Cisco CallManager Express Files

This section contains a comprehensive list of the files that are used with Cisco CME. All the files listed
in this section are included in the zipped, compressed archive called cme-xxx.zip that can be downloaded
from the Cisco CME Software Download website. Descriptions of the files are grouped into the
following categories:
• Phone Firmware Files, page 14
• GUI Files, page 15

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• XML Template, page 16

• Music-on-Hold (MOH) File, page 16
• Script Files, page 16
• Bundled TSP Archive, page 16
• Cisco CME Basic Automatic Call Distribution and Auto Attendant Service Scripts, page 16

Phone Firmware Files

For a list of the firmware files for each phone type, refer to the Cisco CallManager Express 3.3
Specifications. Install only the firmware files for the types of phones that you have at your site.
Installing firmware files involves the following steps:

Step 1 Find out the names of the firmware files for your Cisco IP phones from the Cisco CallManager Express
3.3 Specifications. Note that the Cisco IP Phones 7970G and 7971G-GE use more than one firmware file
and that the firmware version for the Cisco IP Phones 7940 and 7940G and the add-on Cisco IP Phone
7914 Expansion Modules must match.
Step 2 Download the firmware files.
Firmware files can be downloaded individually from
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp or from a compressed archive, such as cme-xxx.zip
or cme-basic-xxx.tar. For more information about obtaining firmware files from archives, refer to the
“How to Download and Install Cisco CME Files” section on page 12.
Step 3 Copy each of the firmware files to the router flash. For example, for the Cisco IP Phone 7960G you
would configure the following:
Router # copy tftp://x.x.x.x/P00305000301.sbn flash:
Router (config)# tftp-server flash:P00305000301.sbn

Step 4 Repeat Step 3 for the remaining phone firmware files. Note that the Cisco IP Phones 7970G and
7971G-GE have five firmware files that you must copy to flash. These five files are listed in the
Cisco CallManager Express 3.3 Specifications.
Step 5 Load the files that you have copied to flash. Note that for the Cisco IP Phone 7970G and 7971G-GE, you
must load only the TERM70.6-0-3SR1S file. The following examples show Cisco IP Phone 7970G,
7971G-GE, and 7960G load configurations. Note that the filename extensions are not included in the
command.
Router (config)# telephony-service
Router (config-telephony)# load 7970 TERM70.6-0-3SR1S

Router (config)# telephony-service

Router (config-telephony)# load 7971 TERM70.6-0-3SR1S

Router (config)# telephony-service

Router (config-telephony)# load 7960 P00305000301

Step 6 (Required for Cisco IP Phone 7970G and 7971G-GE) Configure the type 7970 or type 7971 command
in ephone configuration mode.
Router (config)# ephone 1
Router (config-ephone)# type 7970

This command enables Cisco CME to create a Sep*.conf.xml file, which is read by the phone firmware
when a Cisco IP Phone 7970G or 7971G-GE is booted. For additional information about Cisco IP Phone
7970G or 7971G-GE configuration, see the “Cisco IP Phone 7970G and 7971G-GE Settings” section on
page 232.

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Prerequisites

Step 7 Reset the phones so they can download the specified firmware.
Router (config)# telephony-service
Router (config-telephony)# reset all

Step 8 Check the firmware versions running on your phones to ensure that the phones picked up their loads.
Router (config)# show ephone phone-load

For Cisco IP Phones 7960 and 7960G or Cisco IP Phones 7940 and 7940G, it is recommended that
unsigned loads should be upgraded to signed loads using the following steps:

Step 1 Download P00305000301.zip http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp to obtain the files

required to covert unsigned loads to signed loads.
Step 2 Copy a 5.0 phone load that has a .bin extension on to the router flash.
Router # copy tftp://x.x.x.x/P00305000300.bin flash:
Router (config)# tftp-server flash:P00305000300.bin
Router (config)# telephony-service
Router (config-telephony)# load 7960-7940 P00305000300

Step 3 Reset the phone so that it picks up the 5.0 bin file.
Router (config)# telephony-service
Router (config-telephony)# reset all

Step 4 Check the firmware version running on the phone and ensure that the phone did pick up the 5.0 load.
Router (config)# show ephone phone-load

Step 5 Disable the TFTP sharing of P00305000300.bin after all phones have upgraded to the interim image.
Router (config)# no tftp-server flash:P00305000300.bin

Note You may delete this file from flash as it is not required any more.

Step 6 Copy a 5.0 secure phone load that has a .sbn extension onto the router flash.
Router # copy tftp://x.x.x.x/P00305000301.sbn flash:
Router (config)# telephony-service
Router (config-telephony)# load 7960-7940 P00305000301

Step 7 Reset the phone so that it picks up the 5.0 sbn file.
Router (config)# telephony-service
Router (config-telephony)# reset all

Step 8 Check the firmware version running on the phone and ensure that the phone did pick up the 5.0 .sbn load.
Router (config)# show ephone phone-load

GUI Files
Table 2 lists the files that support web-browser GUIs for users and administrators. These files are
included in the tar archive called cme-xxx-gui.tar or later versions. To install the files, see the “How to
Download and Install Cisco CME Files” section on page 12.

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Prerequisites

Note Cisco CME GUI files are version-specific; GUI files for one version of Cisco CME are not compatible
with any other version of Cisco CME. When downgrading or upgrading Cisco CME, the GUI files for
the old version must be overwritten with GUI files that match the Cisco CME version that is being
installed.

Table 2 GUI Files

admin_user.html dom.js normal_user.html Tab.gif

admin_user.js downarrow.gif normal_user.js telephony_service.html
CiscoLogo.gif ephone_admin.html Plus.gif uparrow.gif
Delete.gif logohome.gif sxiconad.gif xml-test.html

XML Template
The file called xml.template can be copied and modified to allow or restrict specific GUI functions to
customer administrators, who are a new class of administrative users with limited capabilities in an
Cisco CME system. This file is included in the zipped, compressed archive called cme-xxx.zip and also
in both tar archives (cme-basic-xxx.tar and cme-xxx-gui.tar and later versions). To install the file, see
the “How to Download and Install Cisco CME Files” section on page 12.

Music-on-Hold (MOH) File

An audio file named music-on-hold.au provides music for external callers on hold when a live feed is
not used.This file is included in the zipped, compressed archive called cme-xxx.zip and also in the tar
archive called cme-basic-3.2.0.tar and later versions. To install the file, see the “How to Download and
Install Cisco CME Files” section on page 12.

Script Files
The following Tcl script files provide optional functionality for Cisco CME systems:
• cme-b-acd-2.1.0.0.tar—Cisco CME B-ACD AA and queueing script.
• app-h450-transfer.2.0.0.9.zip.tar—Script that adds H.450 transfer and forwarding support for
analog FXS ports.

Bundled TSP Archive

An archive called CiscoIOSTSP.zip contains several Telephony Application Programming Interface
(TAPI) Telephony Service Provider (TSP) files that are needed to set up individual PCs for Cisco IP
phone users who wish to make use of Cisco CME-TAPI integration with TAPI-capable PC software. This
archive is included in the zipped, compressed archive called cme-xxx.zip and is also available as an
individual archive at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp. To install the files, see the
“Cisco IOS TSP Download and Setup” section in the “Configuring Productivity Tools” chapter.

Cisco CME Basic Automatic Call Distribution and Auto Attendant Service Scripts
An archive called cme-b-acd-x.x.x.tar contains the following files required for the Cisco CME basic
automatic call distribution (B-ACD) and auto attendant (AA) service. For more information, see the
“Cisco CME Basic Automatic Call Distribution and Auto-Attendant Service” chapter in Cisco CME
B-ACD and Tcl Call-Handling Applications.
• app-b-acd-x.x.x.x.tcl (AA script)
• app-b-acd-aa-x.x.x.x.tcl (call queuing script)

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Restrictions

• en_bacd_allagentsbusy.au (audio file)

• en_bacd_options_menu.au (audio file)
• en_bacd_disconnect.au (audio file)
• en_bacd_music_on_hold.au (audio file)
• en_bacd_invalidoption.au (audio file)
• en_bacd_welcome.au (audio file)
• en_bacd_enter_dest.au (audio file)
The archive is located at http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp, and its files are
approximately 1.5 MB. To extract these files and copy them to flash, use the archive tar /extract
command in global configuration mode.

Note Because Cisco CME B-ACD's script uses time stamps, the clock on the router must be configured to the
proper date and time. To keep the router clock accurate, configure the router for NTP.

Restrictions
Note Restrictions for call transfer are listed in the “Configuring Call Transfer and Call Forwarding” chapter.

• General phone support is subject to the following restrictions:

– No more Cisco IP phones or extension numbers than the maximum specified in the
Cisco CallManager Express 3.3 Specifications are supported.
– Only the platforms and phone devices listed in the Cisco CallManager Express 3.3
Specifications are supported.
– First-generation Cisco IP phones, such as the Cisco IP Phone 30 VIP and the Cisco IP Phone 12
SP+, are not supported.
– The Cisco IP Phone 7970G and 7971G-GE do not support user and network localization. The
user-locale and network-locale command configurations must be set to their default, United
States (US).
– IP phones connected to Cisco CME systems require the use of out-of-band dual-tone
multifrequency (DTMF) relay to transport DTMF (keypad) digits across VoIP connections. For
more information, see “DTMF Relay for H.323 Networks” section on page 64.
• Analog phone support is subject to the following restrictions:
Cisco CME features such as call forward and call park are not available for analog phones connected
to FXS ports in H.323 mode. In order to support these features, SCCP supplementary features must
be enabled on the FXS ports.
Consult the Cisco CME data sheet for a list of platforms and hardware that can be configured for
SCCP supplementary features.
FXS ports on platforms that cannot enable SCCP supplementary features can use H.323 mode to
support call waiting, caller ID, hookflash transfer, modem pass-through, fax (T.38, Cisco fax relay,
and pass-through), and PLAR. These features are provisioned as Cisco IOS voice features and not
as Cisco CME features. Note that when using Cisco CME, you can configure FXS ports in H.323
mode for call waiting or hookflash transfer, but not both at the same time.

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Restrictions

The following links provide details on configuring analog phone features for FXS ports in H.323
mode:
– “Configuring Analog Voice Ports” section in Voice Ports Configuration
– “Caller ID” section of the Cisco IOS Voice Configuration Library
– “Modem Support for VoIP” section of the Cisco IOS Voice Configuration Library
– Cisco Fax Services over IP Application Guide
• Remote skinny client control protocol (SCCP) phones connected across WAN links are subject to the
following restrictions:
– Cisco TAC will not handle any voice or signaling issues for remote IP phones, unless the same
issue can be replicated for LAN phones.
– E911 or emergency calls are not supported from remote IP phones.
– All calls made to and from remote IP phones must use G.711. Cisco CME does not support the
ability to specify G.729 codec for remote IP phones.
– For inbound or outbound calls, remote IP phones cannot fail and go over to a PSTN connection.
Remote phones must use the WAN for all calls, even if available bandwidth is not sufficient to
guarantee voice quality.
– Remote IP phones do not support Network Address Translation (NAT). All Cisco CME phones
must use IP addresses that are routeable to and from Cisco CME. Remote IP phones must be
able to access the IP addresses that are used for all other local and remote phones.
– All PSTN access is through the central site only. PSTN termination at the remote site is not
supported.
– Cisco CME does not support Call Admission Control (CAC) for remote SCCP phones, so voice
quality can degrade if a WAN link is oversubscribed. High-bandwidth data applications used
over a WAN can cause degradation of voice quality for remote IP phones.
• Cisco CME cannot register as a member of a Cisco CallManager cluster.
• The only codecs supported are G.711 and G.729. For conferencing and music on hold (MOH)
support with G.729, hardware digital signal processors (DSPs) are required for transcoding G.729
between G.711.
• Cisco CME does not support the following:
– Cisco IP Communicator
– CiscoWorks IP Telephony Environment Monitor (ITEM)
– Element Management System (EMS) integration
– Media Gateway Control Protocol (MGCP) on-net calls
– Java Telephony Application Programming Interface (JTAPI) applications, such as the Cisco IP
Softphone, IPCC, or IPCC Express, Cisco CallManager Auto Attendant or Cisco Personal
Assistant
– Telephony Application Programming Interface (TAPI) Version 2.1. Cisco CME implements
only a small subset of TAPI functionality. It does support operation of multiple independent
clients (for example, one client per phone line), but not full support for multiple-user or
multiple-call handling, which is required for complex features such as automatic call
distribution (ACD) and Cisco IP Contact Center (IPCC). Also, this TAPI version does not have
direct media- and voice-handling capabilities.
• Cisco Voice Manager (CVM) does not support IP phone configurations.

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Information About Setting Up a Cisco CME System

Information About Setting Up a Cisco CME System

The complete set of tasks necessary to install and configure a Cisco CME system varies according to the
design of a particular system, but the tasks fall into the major categories listed in Table 3 on page 20.
You can choose to perform the basic initial setup of the system and phones (Tasks 2 and 3 in Table 3) in
any of the following ways:
• Automatically, using the Cisco CME setup tool. With this method, you engage an interactive script
from the router CLI. The script queries you for the system parameters that you want and then builds
the required configuration files. This method automatically identifies the DHCP server and the IP
address of the Cisco CME router to the system. It also automatically assigns one extension number
to each newly registering IP phone. With this method, you do not need to manually perform most of
the steps in the “Setting Up a Cisco CME System” chapter and the “Setting Up Phones in a
Cisco CME System” chapter of this guide. For more information, see Table 3 on page 20 and the
“Automated Phone Setup Using the Cisco CME Setup Tool” section in the “Setting Up Phones in a
Cisco CME System” chapter.
• Semiautomatically, using the auto assign command. With this method, you perform the steps in
Task 2 using Cisco IOS software CLI and then use the auto assign command to automatically assign
one extension number to each newly registering IP phone. For more information, see the
“Partially Automated Phone Setup Using Router CLI” section in the “Setting Up Phones in a Cisco
CME System” chapter.
• Manually. With this method, you perform Tasks 2 and 3 using Cisco IOS software CLI on the
Cisco CME router. This method allows you to make nondefault settings and to assign multiple
numbers and lines to phones. For more information on using router CLI for basic Cisco CME setup,
see both the “Setting Up a Cisco CME System” chapter and the “Setting Up Phones in a Cisco CME
System” chapter.

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Information About Setting Up a Cisco CME System

Table 3 Cisco CME Installation Sequence

Chapter Where the Task Is

Task Purpose How the Task Is Performed Described
1. Planning the system and • Understand design Manually, by reviewing your “Cisco CallManager Express
assembling prerequisites criteria for your requirements and by using Overview” (this chapter)
Cisco CME system Cisco IOS software CLI to
download and install the
• Download required
appropriate files
software to router flash
memory
2. Setting up a Cisco CME • Set up a DHCP server for Automatically with “Setting Up a Cisco CME
system to communicate Cisco CME Cisco CME setup tool or System”
with your network manually with Cisco IOS
• Set up Network Time
software CLI:
Protocol (NTP) on the
Cisco CME router • DHCP server
• Specify the IP source • IP source address
address for the
• Phone firmware
Cisco CME router and
create a configuration file Manually with Cisco IOS
software CLI:
• (Optional) Specify a
nondefault keepalive • NTP
interval • (Optional) Keepalive
• Install phone firmware on
the Cisco CME router

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Table 3 Cisco CME Installation Sequence (continued)

Chapter Where the Task Is

Task Purpose How the Task Is Performed Described
3. Setting up the basic • Set maximum number of Automatically with “Setting Up Phones in a
required Cisco CME ephone-dns and ephones Cisco CME setup tool or Cisco CME System”
phone configurations manually with Cisco IOS
• Set range to
software CLI:
automatically assign
numbers to newly • Maximum number of
registering phones ephone-dns and ephones
• Define ephone-dns • (Optional) Range of
numbers to automatically
• Define ephones and
assign to phones
assign ephone-dns to
ephone buttons • Ephones and ephone-dns
with default settings
• Set voice-mail number
and no-answer timeout • (Optional) Date format,
language for display, and
• Set Direct Inward
locale for tones
Dialing (DID) capability
and dial-plan pattern • (Optional) DID and
dial-plan pattern
• Specify type of DTMF
relay • (Optional) Voice-mail
number and no-answer
• Set the date format and
timeout
language for displays and
set the locale for tones • Phone restart or reset
• Define translation rules Manually with Cisco IOS
software CLI:
• Restart or reset phones
• Ephones and ephone-dns
with nondefault settings
• DTMF relay
• (Optional) Translation
rules
4. Configuring call transfer • Set up call transfer Manually with Cisco IOS “Configuring Call Transfer
and forwarding parameters software CLI and Call Forwarding”
• Define call forwarding
conditions and numbers

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Before You Start: Basic Cisco CME Concepts

Table 3 Cisco CME Installation Sequence (continued)

Chapter Where the Task Is

Task Purpose How the Task Is Performed Described
5. Setting up the • Enable an HTTP server Manually with Cisco IOS “Setting Up the Cisco CME
Cisco CME GUI and set the HTTP path software CLI GUI”
• Create a GUI account for • HTTP server and path
the system administrator
• System administrator
• (Optional) Create a GUI GUI account
account for the customer Graphically with Cisco CME
administrator GUI or manually with CLI:
• (Optional) Create GUI
• (Optional) Customer
accounts for phone users
administrator GUI
account
• (Optional) Phone user
GUI accounts
6. Configuring optional Set up and enable such Graphically with Cisco CME • “Setting Up Optional
system and phone optional features as GUI or Cisco CME System
features secondary dial tone, music on Manually with Cisco IOS Features”
hold, call blocking, caller-ID software CLI • “Configuring Cisco CME
blocking, speed dial, or silent
Phone Features”
ringing, night service, paging,
intercom, call pickup, and • “Integrating Voice Mail
more. with Cisco CME”
• “Configuring an
Attendant for Primary
Call Coverage”
• “Configuring Call
Blocking”
• “Configuring Secondary
Call Coverage”
• “Configuring
Directories”
• “Configuring
Productivity Tools”

Before You Start: Basic Cisco CME Concepts

Cisco CME systems are infinitely flexible because they are modular. A Cisco CME system consists of a
router that serves as a gateway and one or more VLANs that connect IP phones and phone devices to the
router. In addition, a Cisco CME system uses the following basic building blocks:
• Ephone—A software construct that usually represents a physical telephone instrument, although it
is also used to represent a port that connects to a voice-mail system. The ephone construct provides
the ability to configure the physical instrument using Cisco IOS software. Each ephone can have
multiple extensions associated with it in a many-to-many relationship, and a single extension can be

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Before You Start: Basic Cisco CME Concepts

associated with multiple ephones so that it appears as a shared extension. The maximum number of
ephones in a Cisco CME system is the maximum number of physical instruments that can be
connected to the system.
• Ephone-dn—A software construct that represents the line that connects a voice channel to a phone
instrument on which a user can receive and make calls. An ephone-dn represents a virtual voice port
in the Cisco CME system, so the maximum number of ephone-dns in a Cisco CME system is the
maximum number of simultaneous call connections that can occur. Note that this concept is different
from the maximum number of physical lines in a traditional telephony system and also is different
from the maximum number of telephone or extension numbers that can be assigned.
Traditional telephony systems are based on physical connections and are therefore limited in the types
of phone service that they can offer. Because the ephone and ephone-dn are software constructs and
because the audio stream is packet-based, an almost limitless number of combinations of phone
numbers, lines, and phones can be planned and implemented.
Cisco CME systems can be designed in many ways. The key is to determine how many simultaneous
calls you want to handle at your site and at each phone at your site, how many different numbers you
want to have, and how many phones you want to have. Even a Cisco CME system has its limits, however.
The following factors should be considered in your system design:
• Maximum number of ephone-dns—As noted in the Cisco CallManager Express 3.3 Specifications
(http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cme33/cme33spc.htm), there is a
maximum number of ephone-dns per system. This number corresponds to the maximum number of
simultaneous call connections that can occur.
• Maximum number of telephone numbers—Your numbering plan may restrict the range of telephone
numbers or extension numbers that you can use. For example, if you have DID, the PSTN may assign
you a certain series of numbers.
• Maximum number of buttons per phone—You may be limited by the number of buttons and phones
that your site can use. For example, you may have two people with six-button phones to answer
twenty different telephone numbers.
The flexibility of a Cisco CME system is due largely to the different types of ephone-dns that you can
assign to phones in your system. By understanding the types of ephone-dns and considering how they
can be combined, you can create the complete call coverage situation that your business requires. For
more information about types of ephone-dns, see the “Ephone-dns” section on page 24.
After setting up the ephone-dns and ephones that you need, you add optional Cisco CME features to
create a telephony environment that enhances your business objectives. Cisco CME systems are able to
integrate with the PSTN and with your business requirements to allow you to continue using your
existing number plans, dialing schemes, and call coverage patterns. The following sections explain
concepts that will help you to design and configure Cisco CME systems.
• Ephones, page 24
• Ephone-dns, page 24
• Phone Number Plan, page 30
• Direct Inward Dialing, page 31
• PBX or Keyswitch Model, page 32

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Before You Start: Basic Cisco CME Concepts

Ephones
An ephone, or “Ethernet phone,” is a single instance of the software configuration of the physical
instrument with which a phone user makes and receives calls in a Cisco CME system. The physical
ephone is either a Cisco IP phone or an analog phone equipped with an analog telephone adaptor (ATA)
device.
Each ephone has a unique phone-tag, or sequence number, to identify it during configuration.
An ephone is populated with ephone-dns and features by the ephone command, which associates the
MAC address of a physical phone with the telephone numbers associated with ephone-dns and with other
Cisco CME features.

Ephone-dns
An ephone-dn, or “Ethernet phone directory number,” is a software construct that represents the line that
connects a voice channel to a phone instrument on which a user can receive and make calls. An
ephone-dn has one or more extension or telephone numbers associated with it to allow call connections
to be made. An ephone-dn is equivalent to a phone line in most cases, but not always. There are several
types of ephone-dns, which have different characteristics. The maximum number of ephones per
Cisco CME system is indicated in the Cisco CallManager Express 3.3 Specifications
(http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cme33/cme33spc.htm).
Each ephone-dn has a unique dn-tag, or sequence number, to identify it during configuration.
Ephone-dns are assigned to line buttons on ephones during configuration.
An ephone-dn is created by the ephone-dn command, which builds one virtual voice port and one or
more dial peers for the ephone-dn, depending on your dial-plan pattern and the ephone-dn secondary
number field. The ephone-dn command automates the process of associating dial peers to an
ephone-dn’s virtual voice port and manages the numbering and configuring of virtual voice ports. Dial
peers that are created by the ephone-dn command can be reviewed using the show telephony-service
dial-peer command.
The number of ephone-dns that you create corresponds to the number of simultaneous calls that you can
have, because each ephone-dn represents a virtual voice port in the router. This means that if you want
more than one call to the same number to be answered simultaneously, you need multiple virtual voice
ports (ephone-dns) with the same destination pattern (extension or telephone number).
The ephone-dn is the basic building block of a Cisco CME system. Six different types of ephone-dn can
be combined in different ways for different call coverage situations. Each type will help with a particular
type of limitation or call coverage need. For example, if you want to keep the number of ephone-dns low
and provide service to a large number of people, you might use shared ephone-dns. Or if you have a
limited number of extension numbers that you can use but need to have a large number of simultaneous
calls, you might create two or more ephone-dns with the same number. The key is knowing how each
type of ephone-dn works and what its advantages are.
The following sections will help you understand the types of ephone-dn in a Cisco CME system:
• Single-Line Ephone-dn
• Dual-Line Ephone-dn
• Two Ephone-dns with One Number
• Dual-Number Ephone-dn
• Shared Ephone-dn
• Overlay Ephone-dn

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Single-Line Ephone-dn
A single-line ephone-dn has the following characteristics:
• Makes one call connection at a time using one phone line button. A single-line ephone-dn has one
telephone number associated with it.
• Should be used when phone buttons have a one-to-one correspondence to the PSTN lines that come
into a Cisco CME system.
• Should be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI),
loopback, and music-on-hold (MOH) feed sources.
• When used with multiple-line features like call waiting, call transfer, and conferencing, there must
be more than one single-line ephone-dn on a phone.
• Can be combined with dual-line ephone-dns on the same phone.
Note that you must make the choice to configure each ephone-dn in your system as either dual-line or
single-line when you initially create ephone-dn configuration entries. If you need to change from
single-line to dual-line later, you must delete the ephone-dn and then recreate it.
Figure 3 shows a single-line ephone-dn.

Figure 3 Single-Line Ephone-dn

ephone-dn 11
number 1001
IP V ephone 1

88888
Phone 1 button 1:11
Button 1 is extension 1001

Dual-Line Ephone-dn
A dual-line ephone-dn has the following characteristics:
• Can make two call connections at the same time using one phone line button. A dual-line ephone-dn
has two channels for separate call connections.
• Can have one number or two numbers (primary and secondary) associated with it.
• Should be used for an ephone-dn that needs to use just a single button for features like call waiting,
call transfer, or conferencing.
• Cannot be used for lines that are dedicated to intercom, paging, message-waiting indicator (MWI),
loopback, and music-on-hold (MOH) feed sources.
• Can be combined with single-line ephone-dns on the same phone.
Note that you must make the choice to configure each ephone-dn in your system as either dual-line or
single-line when you initially create ephone-dn configuration entries. If you need to change from
single-line to dual-line later, you must delete the ephone-dn and then recreate it.
Figure 4 shows a dual-line ephone-dn.

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Figure 4 Dual-Line Ephone-dn

ephone-dn 12 dual-line
number 1002
IP V ephone 2

88889
Phone 2 button 1:12
Button 1 is extension 1002

Two Ephone-dns with One Number

Two ephone-dns with one number have the following characteristics:
• Have the same telephone number but two separate virtual voice ports, and therefore can have two
separate call connections.
• Can be single-line or dual-line ephone-dns.
• Can appear on the same phone on different buttons or on different phones.
• Should be used when you want the ability to make more call connections while using fewer numbers.
Figure 5 on page 27 shows a phone with two buttons that have the same number, extension 1003. Each
button has a different ephone-dn (button 1 is ephone-dn 13 and button 2 is ephone-dn 14), so each button
can make one independent call connection if the ephone-dns are single-line and two call connections (for
a total of four) if the ephone-dns are dual-line.
Figure 6 on page 27 shows two phones that each have a button with the same number. Because the
buttons have different ephone-dns, the calls that are connected on these buttons are independent of one
another. The phone user at phone 4 can make a call on extension 1004, and the phone user on phone 5
can receive a different call on extension 1004 at the same time.
The two ephone-dns-with-one-number situation is different than a shared line, which also has two
buttons with one number but has only one ephone-dn for both of them. A shared ephone-dn will have the
same call connection at all the buttons on which the shared ephone-dn appears. If a call on a shared
ephone-dn is answered on one phone and then placed on hold, the call can be retrieved from the second
phone on which the shared ephone-dn appears. But when there are two ephone-dns with one number, a
call connection appears only on the phone and button at which the call is made or received. In the
example in Figure 6, if the user at phone 4 makes a call on button 1 and puts it on hold, the call can be
retrieved only from phone 4. For more information about shared lines, see the “Shared Ephone-dn”
section on page 28.
The examples in Figure 5 and Figure 6 show how two ephone-dns with one number are used to provide
a small hunt group capability. If the ephone-dn on button 1 is busy or does not answer, an incoming call
to extension 1003 rolls over to the ephone-dn associated with button 2 because the preference and no
huntstop commands have been used. Values assigned in the preference command are passed to the dial
peers created by the two ephone-dns. Both dial peers for the ephone-dns are matched when this extension
number is dialed. The call is connected to the ephone-dn with the highest preference. The default
preference value is 0 (the highest value), so ephone-dn 13 on button 1 gets the first call. The no huntstop
command tells the dial peers not to stop hunting for another match, so the second call to extension 1003
is sent to ephone-dn 14.

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Figure 5 Two Ephone-dns with One Number on One Phone

ephone-dn 13
number 1003
no huntstop

ephone-dn 14
IP V number 1003
Phone 3 preference 1
Button 1 is extension 1003

88891
Button 2 is also extension 1003 ephone 3
button 1:13 2:14

Figure 6 Two Ephone-dns with One Number on Two Phones

Phone 4 ephone-dn 13
Button 1 is extension 1003 number 1003
no huntstop
IP
ephone-dn 14
number 1003
IP V preference 1
Phone 5 ephone 4
Button 1 is extension 1003 button 1:13

88892
ephone 5
button 1:14

Dual-Number Ephone-dn
A dual-number ephone-dn has the following characteristics:
• Has two telephone numbers, a primary number and a secondary number.
• Can make one call connection if it is a single-line ephone-dn.
• Can make two call connections at a time if it is a dual-line ephone-dn.
• Should be used when you want to have two different numbers for the same button without using
more than one ephone-dn.
Figure 7 shows an ephone-dn that has two numbers, extension 1006 and extension 1007.

Figure 7 Dual-Number Ephone-dn

ephone-dn 15
number 1006 secondary 1007
IP V
ephone 6
88890

Phone 6 button 1:15

Button 1 is extension 1006
Button 1 is also extension 1007

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Shared Ephone-dn
A shared ephone-dn has the following characteristics:
• Appears on two different phones but uses the same ephone-dn and number.
• Can make one call at a time and that call appears on both phones.
• Should be used when you want the capability to answer or pick up a call at more than one phone.
Because these phones share the same ephone-dn, if the ephone-dn is connected to a call on one phone,
that ephone-dn is unavailable for other calls on the second phone. If a call is placed on hold on one
phone, it can be retrieved on the second phone. This is like having a single-line phone in your house with
multiple extensions. You can answer the call from any phone on which the number appears, and you can
pick it up from hold on any phone on which the number appears.
Figure 8 shows a shared ephone-dn. Extension 1008 appears on both phone 7 and phone 8.

Figure 8 Shared Ephone-dn

Phone 7
Button 1 is extension 1008
ephone-dn 16
number 1008
IP
ephone 7
button 1:16
IP V
Phone 8 ephone 8

88893
Button 1 is extension 1008 button 1:16

Overlay Ephone-dn
An overlay ephone-dn has the following characteristics:
• Is a member of an overlay set, which includes all the ephone-dns that have been assigned together
to a particular phone button.
• Can have the same telephone or extension number as other members of the overlay set or different
numbers.
• Can be single-line or dual-line, but cannot be mixed single-line and dual-line in the same overlay set.
• Can be shared on more than one phone.
Overlay ephone-dns provide call coverage similar to shared ephone-dns because the same number can
appear on more than one phone. The advantage of using two ephone-dns in an overlay arrangement
rather than as simple shared ephone-dns is that a call to the number on one phone does not block the use
of the same number on the other phone, as would happen if this were a shared ephone-dn.
You can overlay up to ten lines on a single button and create a “10x10” shared line with ten lines in an
overlay set shared by ten phones, resulting in the possibility of ten simultaneous calls to the same
number.
Figure 9 on page 29 shows an overlay set with two ephone-dns and one number that is shared on two
phones. Ephone-dn 17 has the default preference value of 0, so it will receive the first call to extension
1001. The phone user at phone 9 answers the call, and a second incoming call to extension 1001 can be
answered on phone 10 using ephone-dn 18.

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Figure 9 Overlay Ephone-dn (Simple Case)

Phone 9 ephone-dn 17
Button 1 is two appearances number 1001
of extension 1001
ephone-dn 18
IP number 1001
preference 1
IP V ephone 9
Phone 10 button 1o17,18

88894
Button 1 is two appearances
of extension 1001 ephone 10
button 1o17,18

A more complex ephone-dn configuration mixes overlay ephone-dns with shared ephone-dns and plain
dual-line ephone-dns on the same phones. Figure 10 on page 30 illustrates the following example of a
manager with two assistants. On the manager’s phone the same number, 2001, appears on button 1 and
button 2. The two line appearances of extension 2001 use two single-line ephone-dns, so the manager
can have two active calls on this number simultaneously, one on each button. The ephone-dns are set up
so that button 1 will ring first, and if a second call comes in, button 2 will ring. Each assistant has a
personal ephone-dn and also shares the manager’s ephone-dns. Assistant 1 has all three ephone-dns in
an overlay set on one button, whereas assistant 2 has one button for the private line and a second button
with both of the manager’s lines in an overlay set. A sequence of calls might be as follows.
1. An incoming call is answered by the manager on extension 2001 on button 1 (ephone-dn 20).
2. A second call rings on 2001 and rolls over to the second button on the manager’s phone
(ephone-dn 21). It also rings on both assistants’ phones (also ephone-dn 21, a shared ephone-dn).
3. Assistant 2 answers the call. This is a shared overlay line (one ephone-dn, 21, is shared among three
phones, and on two of them this ephone-dn is part of an overlay set). Because it is shared with
button 2 on the manager’s phone, the manager can see when assistant 2 answers the call.
4. Assistant 1 makes an outgoing call on ephone-dn 22. The button is available because of the
additional ephone-dns in the overlay set on the assistant 1 phone.
At this point, the manager is in conversation on ephone-dn 20, assistant 1 is in conversation on ephone-dn
22, and assistant 2 is in conversation on ephone-dn 21.

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Figure 10 Overlay Ephone-dn (Complex Case)

Manager phone
Button 1 is extension 2001 ephone-dn 20
Button 2 is extension 2001 number 2001
no huntstop
IP ! Manager number

ephone-dn 21
number 2001
IP V preference 1
Assistant 1 phone ! Manager number
Button 1 is extension 2001
and extension 2002 ephone-dn 22
number 2002
! Assistant 1 personal number
IP
Assistant 2 phone ephone-dn 23
Button 1 is extension 2003 number 2003
Button 2 is extension 2001 ! Assistant 2 personal number

ephone 8
button 1:20 2:21
! Manager phone

ephone 9
button 1o22,20,21
! Assistant 1 phone

ephone 10

88895
button 1:23 2o20,21
! Assistant 2 phone

Phone Number Plan

If you are installing a Cisco CME system to replace an older telephony system that had an established
telephone number plan, you can retain the old number plan. Cisco CME supports flexible extension
number lengths and can provide automatic conversion between extension dialing and E.164 public
telephone number dialing.
A successful Cisco CME system requires a telephone numbering plan that supports future expansion.
The numbering plan also must not overlap or conflict with other numbers that are on the same VoIP
network or are part of a centralized voice mail system.
Cisco CME supports shared lines as well as multiple lines configured with the same extension number.
This means that you can set up several phones to share an extension number to provide coverage for that
number. You can also assign several line buttons on a single phone to the same extension number to
create a small hunt group. For more information about types of line configurations, see the “Ephone-dns”
section on page 24.
If you are configuring more than one Cisco CME site, you need to decide how calls between the sites
will be handled. Calls between Cisco CME phones can be routed either through the PSTN or over VoIP.
If you are routing calls over VoIP, you must decide among the following three choices:
• You can route calls using a global pool of fixed-length extension numbers. For example, all sites
have unique extension numbers in the range 5000 to 5999, and routing is managed by a gatekeeper.
If you select this method, you should assign a subrange of extension numbers to each site so that
duplicate number assignment does not result. You will have to keep careful records of which
Cisco CME system is assigned which number range.

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• You can route calls using a local extension number plus a special prefix for each Cisco CME site.
This choice allows you to use the same extension numbers at more than one site.
• You can use an E.164 PSTN phone number to route calls over VoIP between Cisco CME sites. In
this case, intersite callers use the PSTN area code and local prefix to route calls between Cisco CME
systems.
If you choose to have a gatekeeper route calls among multiple Cisco CME systems, you may face
additional restrictions on the extension number formats that you use. For example, you might be able to
register only PSTN-formatted numbers with the gatekeeper. The gatekeeper might not allow the
registration of duplicate telephone numbers in different Cisco CME systems, but you might be able to
overcome this limitation. Cisco CME allows the selective registration of either 2- to 5-digit extension
numbers or 7- to 10-digit PSTN numbers, so registering only PSTN numbers might prevent the
gatekeeper from sensing duplicate extensions.
To properly configure your Cisco CME system to handle direct calls, call forwarding, and call transfers
between Cisco CME sites, make sure that you understand and configure the dialplan-pattern command,
which is described in the “Setting Up Phones in a Cisco CME System” chapter.
In addition, your selection of a numbering scheme for phones that can be directly dialed from the PSTN
is limited by your need to use the range of extensions that are assigned to you by the telephone company
that provides your connection to the PSTN. For example, if your telephone company assigns you a range
from 408-555-0100 to 408-555-0199, you may assign extension numbers only in the range 100 to 199 if
those extensions are going to have Direct Inward Dialing (DID) access. For more information about DID,
see the “Direct Inward Dialing” section on page 31.
Also note that the mapping of public telephone numbers to internal extension numbers is not restricted
to simple truncation of the digit string. Digit substitutions can be made using the extension-pattern
keyword in the dialplan-pattern command. For more information, see the “Dial-Plan Pattern” section
in the “Setting Up Phones in a Cisco CME System” chapter.

Direct Inward Dialing

When you define an ephone-dn (extension number instance) using the ephone-dn command, the
Cisco CME system automatically creates a POTS dial peer with the ephone-dn endpoint as a destination.
The default behavior is for the Cisco CME system to create a single POTS dial peer for each ephone-dn.
If the dialplan-pattern command is set and matches against the ephone-dn number, two POTS dial peers
are created, one for the local extension and one for the complete E.164 direct-dial telephone number. For
example, an ephone-dn extension number is 1234, and the dial-plan pattern is “dialplan-pattern 1
4085551... extension-length 4.” One POTS dial peer is created for “1234” and a second POTS dial peer
is created for “4085551234.” A third POTS dial peer is created if an ephone-dn secondary number is
defined, and a fourth dial peer is created for this number if the secondary number also matches a
dial-plan pattern. When the PSTN connects a DID call for “4085551234” to the Cisco CME system, it
also forwards the extension digits “1234” to allow the system to route the call. Dial peers that are created
by the ephone-dn command can be reviewed using the show telephony-service dial-peer command.
For more information, refer to Dial Peer Configuration on Voice Gateway Routers, Cisco IOS
Release 12.3.

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PBX or Keyswitch Model

When setting up a Cisco CME system, you need to decide if call handling should be similar to that of a
PBX, similar to that of a keyswitch, or a hybrid of both. Cisco CME provides a significant amount of
flexibility in this area, but requires that you have a clear understanding of the model that you choose.
The simplest case is the PBX model, in which most of the IP phones in your system have a single unique
extension number. For this model, it is recommended that you configure ephone-dn entries to use the
dual-line configuration option. With this setting, each button that appears on an IP phone can handle two
concurrent calls. The phone user toggles between calls using the blue navigation button on the phone.
Dual-line ephone-dn entries enable your configuration to support call waiting, call transfer with
consultation, and three-party conferencing (G.711 only). If you assign strictly sequential extension
numbers to newly registering IP phones, the Cisco CME setup tool can assist you in creating a PBX-style
configuration automatically.
In a keyswitch type of system, you can set up most of your phones to have a nearly identical
configuration, in which each phone is able to answer any incoming PSTN call on any line. For example,
you have four incoming PSTN lines that each appear as shared lines on four different phones. Each
phone has the same shared lines. You configure this model by creating a set of ephone-dn entries that
correspond one-to-one with your PSTN lines. Then you configure your PSTN ports to route incoming
calls to the ephone-dns. The maximum number of PSTN lines that you can assign in this model might
be limited by the number of available buttons on your IP phones. If so, the ephone-dn overlay option may
be useful for extending the number of lines that can be accessed by a phone.
In the keyswitch model, the same ephone-dn entries are assigned to all IP phones. This means that when
an incoming call arrives, it rings all available IP phones. When multiple calls are present within the
system at the same time, each individual call (ringing or waiting on hold) is visible and can be directly
selected by pressing the corresponding line button on an IP phone. In this model, calls can be moved
between phones simply by putting the call on hold at one phone and selecting the call using the line
button on another phone. In a keyswitch usage model, it is often not appropriate to use the ephone-dn
dual-line option because the PSTN lines with which the ephone-dn entries correspond do not themselves
support dual-line configuration. Use of the dual-line option also makes configuration of call-coverage
(hunting) behaviors more complex.
You can mix PBX and keyswitch configurations on the same IP phone and include both unique per-phone
extensions for PBX-style calling and shared lines for keyswitch-style call operations. You can also
combine single-line and dual-line ephone-dns on the same phone.

What to Do Next
After downloading Cisco IOS software and Cisco CME software, you are ready to set up an Cisco CME
system. For instructions, see the “Setting Up a Cisco CME System” chapter.

Additional References
• Related Documents, page 33
• Standards, page 34
• MIBs, page 35
• RFCs, page 35

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Additional References

Related Documents

Related Topic Document Title

Cisco CallManager Express 3.3 Specifications • Cisco CallManager Express 3.3 Specifications
Cisco CME Basic Automatic Call Distribution • Cisco CME B-ACD and Tcl Call-Handling Applications
(B-ACD) and Auto-Attendant (AA) service
Cisco IP phones end-user instructions • Cisco CME 3.0 Quick Reference Cards
Cisco IP phones administration, installation, and • Cisco IP Phones and Services
regulatory information
Cisco Unity integration • Cisco CallManager Express 3.0 Integration Guide for Cisco
Unity 4.0
XML guidelines for Cisco IP phone services • Cisco IP Phone Services Application Development Notes
XML application programming interface (API) • XML Provisioning Guide for Cisco CME/SRST
TAPI development • TAPI Developer Guide for Cisco CME/SRST
Default session application • Default Session Application Enhancements, Cisco IOS
Release 12.2(15)ZJ
Domain management with Cisco Packet Telephony • Provisioning Manager - Managed Cisco CallManager Express
Center - Virtual Switch (Cisco PTC - VS) Router
Dynamic Host Configuration Protocol (DHCP) • "DHCP" part of the Cisco IOS IP Addressing Services
Configuration Guide
Dial peers, DID, and other dialing issues • Dial Peer Configuration on Voice Gateway Routers
• Understanding One Stage and Two Stage Dialing (technical
note)
• Understanding How Inbound and Outbound Dial Peers Are
Matched on Cisco IOS Platforms (technical note)
• Using IOS Translation Rules - Creating Scalable Dial Plans for
VoIP Networks (sample configuration)
Network Time Protocol (NTP) • “Performing Basic System Management” chapter of Part 3:
System and Network Management in the Cisco IOS
Configuration Fundamentals and Network Management
Configuration Guide
H.323 • Cisco IOS H.323 Configuration Guide
SIP • Cisco IOS SIP Configuration Guide
• SIP Gateway Enhancements, Cisco IOS Release 12.2(15)ZJ
ATAs • Cisco ATA Release Notes
Analog phone features for FXS ports in H.323 mode • “Configuring Analog Voice Ports” section in Voice Ports
Configuration
• “Caller ID” section of the Cisco IOS Voice Configuration
Library
• “Modem Support for VoIP” section of the Cisco IOS Voice
Configuration Library
• Cisco Fax Services over IP Application Guide

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Additional References

Related Topic Document Title

Additional Cisco IOS Voice Configuration Library Cisco IOS Voice Configuration Library at
documents, including library preface and glossary http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/
123cgcr/voice_c/vcl.htm
Cisco IOS command references • Cisco IOS Debug Command Reference, Release 12.3T at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios
123/123tcr/123dbr/index.htm
• Cisco IOS Voice Command Reference, Release 12.3T at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios
123/123tcr/123tvr/index.htm
• Cisco CallManager Express 3.3 Command Reference
Cisco IOS troubleshooting information Cisco IOS Voice Troubleshooting and Monitoring Guide at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/
123cgcr/vvfax_c/voipt_c/index.htm
Additional configuration guides • Cisco IOS Voice Configuration Library
• Tcl IVR API Version 2.0 Programmer’s Guide
• Cisco VoiceXML Programmer’s Guide

Related Websites
Related Topic Title and Location
Cisco IOS configuration examples Cisco Systems Technologies website at
http://cisco.com/en/US/tech/index.html
Note From the website, select a technology category and
subsequent hierarchy of subcategories, then click Technical
Documentation > Configuration Examples.

Standards

Standards Title
No new or modified standards are supported by this —
feature, and support for existing standards has not been
modified by this feature.

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Additional References

MIBs

MIBs MIBs Link

MIB CISCO-VOICE-DIAL-CONTROL-MIB To locate and download MIBs for selected platforms, Cisco IOS
releases, and feature sets, use Cisco MIB Locator found at the
following URL:
http://www.cisco.com/go/mibs

RFCs

RFCs Title
No new or modified RFCs are supported by this —
feature, and support for existing RFCs has not been
modified by this feature.

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Additional References

Cisco CallManager Express 3.3 System Administrator Guide

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 Setting Up a Cisco CME System

This chapter explains how to prepare a router for Cisco CallManager Express (Cisco CME).

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Information About Setting Up a Cisco CME System, page 37
• Prerequisites, page 38
• Setting Network Parameters for Cisco CME, page 38
• Downgrading the Cisco IOS Software Image to an Earlier Version, page 43
• Verifying the Configuration, page 44
• What to Do Next, page 44

Information About Setting Up a Cisco CME System

The purpose of the tasks in this chapter is to enable the Cisco CME system to communicate with your
VoIP network and to prepare the Cisco CME system to register IP phones.

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Prerequisites

Prerequisites
You must have the following capabilities in place before starting to set up a Cisco CME system:
• Network—Your VoIP network must be up and operational.
• Cisco IOS software—You must have downloaded and installed a version of Cisco IOS software that
supports this version of Cisco CME.

Note Cisco CME customers should use Cisco IOS Release 12.3(11)T on Cisco CME routers unless they
require support for the Cisco IP Phone 7970G or for the Basic ACD feature. Customers who require that
support should use Cisco IOS Release 12.3(11)XL. If you are not sure which Cisco IOS software release
you need, contact your sales representative.

• Cisco CME software—You must have downloaded the Cisco CME files and copied them to flash
memory on the Cisco CME router. For more information, see the “Software Prerequisites” section
in the “Cisco CallManager Express Overview” chapter.
• Hardware—You must have a router with sufficient memory for the Cisco CME system that you are
installing and the appropriate IP phones for end users.
For details, see the “Prerequisites” section in the “Cisco CallManager Express Overview” chapter.

Setting Network Parameters for Cisco CME

The following sections describe tasks that allow the Cisco CME system to communicate with your
network:
• Setting Up DHCP Service for Cisco CME, page 38 (not required if using the Cisco CME setup tool)
• Configuring Network Time Protocol, page 42 (required)

Setting Up DHCP Service for Cisco CME

Note If you plan to use the Cisco CME setup tool and you need just one DHCP IP address pool, the
Cisco CME setup tool will create the address pool for you. You do not have to perform the steps in this
section. Proceed to the “Configuring Network Time Protocol” section on page 42.

When a Cisco IP phone is connected to the Cisco CME system, it automatically queries for a Dynamic
Host Configuration Protocol (DHCP) server. The DHCP server responds by assigning an IP address to
the Cisco IP phone and providing the IP address of the TFTP server through DHCP option 150. Then the
phone registers with the Cisco CME server and attempts to get configuration and phone firmware files
from the TFTP server.
If you need to change the address of the TFTP server after you have initially entered an address, refer to
the instructions in the “Changing the TFTP Server Address” section in the “Setting Up Phones in a
Cisco CME System” chapter.

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Setting Network Parameters for Cisco CME

Choose one of the following tasks to set up DHCP service for your IP phones:
• If your Cisco CME router is the DHCP server and you can use a single shared address pool for all
your DHCP clients, use the steps in the “Defining a Single DHCP IP Address Pool” section on
page 39.
• If your Cisco CME router is the DHCP server and you need separate pools for non-IP-phone DHCP
clients, use the steps in the “Defining a Separate DHCP IP Address Pool for Each Cisco IP Phone”
section on page 40.
• If the Cisco CME router is not the DHCP server and you want to relay DHCP requests from IP
phones to a DHCP server on a different router, use the steps in the “Defining a DHCP Relay Server”
section on page 41.
For more information about DHCP, see the "DHCP" part of the Cisco IOS IP Addressing Services
Configuration Guide at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124cg/hiad_c/ch10/index.htm.

Defining a Single DHCP IP Address Pool

This task creates a shared pool of IP addresses, in which all DHCP clients receive the same information,
including the option 150 TFTP server IP address. The benefit of selecting this method to set up DHCP
is that you set up only one DHCP pool. However, this method will not be adequate if some
(non-IP-phone) clients need to use a different TFTP server address.

SUMMARY STEPS

1. ip dhcp pool pool-name

2. network ip-address [mask | /prefix-length]
3. option 150 ip ip-address
4. default-router ip-address
5. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ip dhcp pool pool-name Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.
Example:
Router(config)# ip dhcp pool mypool
Step 2 network ip-address [mask | /prefix-length] Specifies the IP address of the DHCP address pool
and the optional mask or number of bits in the
address prefix, preceded by a forward slash.
Example:
Router(config-dhcp)# network 10.0.0.0 255.255.0.0
Step 3 option 150 ip ip-address Specifies the TFTP server address from which the
Cisco IP phone downloads the image configuration
file. This is your Cisco CME router address.
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1

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Setting Network Parameters for Cisco CME

Command or Action Purpose

Step 4 default-router ip-address (Optional) Specifies the router that the IP phones
will use to send or receive IP traffic that is external
to their local subnet.
Example:
Router(config-dhcp)# default-router 10.0.0.1 If the Cisco CME router is the only router on the
network, this address should be the Cisco CME IP
source address. This command can be omitted if IP
phones need to send or receive IP traffic only to or
from devices on their local subnet.
The IP address that you specify for default router
will be used by the IP phones for fallback
(Survivable Remote Site Telephony or SRST)
purposes. If the Cisco CME IP source address
becomes unreachable, IP phones will attempt to
register to the address specified in this command.
Step 5 exit Exits DHCP pool configuration mode.

Example:
Router(config-dhcp)# exit

Defining a Separate DHCP IP Address Pool for Each Cisco IP Phone

This task creates a name for an DHCP server address pool and specifies IP and MAC addresses. This
method requires that you make an entry for every IP phone.

SUMMARY STEPS

1. ip dhcp pool pool-name

2. host ip-address subnet-mask
3. client-identifier mac-address
4. option 150 ip ip-address
5. default-router ip-address
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ip dhcp pool pool-name Creates a name for the DHCP server address pool
and enters DHCP pool configuration mode.
Example:
Router(config)# ip dhcp pool pool2
Step 2 host ip-address subnet-mask Specifies the IP address that you want the phone to
get.
Example:
Router(config-dhcp)# host 10.0.0.0 255.255.0.0

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Command or Action Purpose

Step 3 client-identifier mac-address Specifies the MAC address of the phone, which is
printed on a sticker on each Cisco IP phone.
Example: Note You must use a 01 prefix number before
Router(config-dhcp)# client-identifier 01238.380.3056 the MAC address.
Step 4 option 150 ip ip-address Specifies the TFTP server IP address from which
the Cisco IP phone downloads the image
configuration file, XmlDefault.cnf.xml. This is
Example:
Router(config-dhcp)# option 150 ip 10.0.0.1
your Cisco CME router IP address.
Step 5 default-router ip-address (Optional) Specifies the router that the IP phones
will use to send or receive IP traffic that is external
to their local subnet.
Example:
Router(config-dhcp)# default-router 10.0.0.1 If the Cisco CME router is the only router on the
network, this address should be the Cisco CME IP
source address. This command can be omitted if IP
phones need to send or receive IP traffic only to or
from devices on their local subnet.
The IP address that you specify for default router
will be used by the IP phones for fallback
(Survivable Remote Site Telephony or SRST)
purposes. If the Cisco CME IP source address
becomes unreachable, IP phones will attempt to
register to the address specified in this command.
Step 6 exit Exits DHCP pool configuration mode.

Example:
Router(config-dhcp)# exit

Defining a DHCP Relay Server

The Cisco IOS DHCP relay server feature is enabled on routers by default. If the DHCP relay server
becomes disabled for some reason and you need it, use the steps in this task to enable it.
This task sets up DHCP relay on the LAN interface where the Cisco IP phones are connected and enables
the Cisco IOS DHCP server feature to relay requests from DHCP clients (phones) to a DHCP server.
For more information about DHCP, see the "DHCP" part of the Cisco IOS IP Addressing Services
Configuration Guide at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124cg/hiad_c/ch10/index.htm.

SUMMARY STEPS

1. service dhcp
2. interface type number
3. ip helper-address ip-address
4. exit

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Setting Network Parameters for Cisco CME

DETAILED STEPS

Command or Action Purpose

Step 1 service dhcp Enables the Cisco IOS DHCP server feature on the
router.
Example:
Router(config)# service dhcp
Step 2 interface type number Enters interface configuration mode for the
specified interface.
Example:
Router(config)# interface vlan 10
Step 3 ip helper-address ip-address Specifies the helper address for any unrecognized
broadcast for TFTP server and DNS server
requests. For each server, a separate ip
Example:
Router(config-if)# ip helper-address 10.0.0.1
helper-address command is required if the
servers are on different hosts. You can also
configure multiple TFTP server targets by using
the ip helper-address commands for multiple
servers.
Step 4 exit Exits interface configuration mode.

Example:
Router(config-if)# exit

Configuring Network Time Protocol

Network Time Protocol (NTP) allows you to synchronize your Cisco CME router to a single clock on
the network, which is known as the clock master. NTP is disabled on all interfaces by default, but it is
essential for Cisco CME.
For more information about configuring NTP, refer to the “Performing Basic System Management”
chapter of Part 3: System and Network Management in the Cisco IOS Configuration Fundamentals and
Network Management Configuration Guide.

SUMMARY STEPS

1. clock timezone zone hours-offset [minutes-offset]

2. clock summer-time zone recurring [week day month hh:mm week day month hh:mm [offset]]
3. ntp server ip-address

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Downgrading the Cisco IOS Software Image to an Earlier Version

DETAILED STEPS

Command or Action Purpose

Step 1 clock timezone zone hours-offset [minutes-offset] Sets the local time zone.
• zone—Name of the time zone (typically a
Example: standard abbreviation).
Router(config)# clock timezone pst -8
• hours-offset—Number of hours that the
specified time zone differs from Coordinated
Universal Time (UTC).
• minutes-offset—(Optional) Number of
minutes that the time zone differs from UTC.
Step 2 clock summer-time zone recurring [week day month hh:mm (Optional) Specifies daylight savings time.
week day month hh:mm [offset]]
• zone—Name of the time zone (typically a
standard abbreviation).
Example:
Router(config)# clock summer-time pdt recurring
Default is that summer time is disabled. If the
clock summer-time zone recurring command is
specified without parameters, the summer time
rules default to United States rules. Default of the
offset argument is 60.
For more information, refer to the “Performing
Basic System Management” section in the
Cisco IOS Configuration Fundamentals and
Network Management Configuration Guide.
Step 3 ntp server ip-address Allows the clock on this router to be synchronized
with the specified NTP server.
Example: • ip-address—IP address of the time server that
Router(config)# ntp server 10.1.2.3 provides the clock synchronization.

Downgrading the Cisco IOS Software Image to an Earlier

Version
If it becomes necessary to downgrade the Cisco IOS image to an earlier version, you must use the
following sequence of tasks so that the phones boot correctly:
1. Downgrade the Cisco CME phone firmware files first.
2. Downgrade the Cisco CME image second.
3. (Optional) Downgrade the Cisco CME GUI files if you will be using the Cisco CME GUI.

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 Setting Up a Cisco CME System
Verifying the Configuration

Verifying the Configuration

Use the show running-config command to verify that DHCP is configured:
Router# show running-config

.
.
.
ip dhcp pool mypool
network 10.0.0.0 255.255.0.0
option 150 ip 10.0.0.1
default-router 10.0.0.1
.
.
.

What to Do Next
After establishing a DHCP server and configuring NTP, set up the phones as described in the “Setting
Up Phones in a Cisco CME System” chapter.

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 Setting Up Phones in a Cisco CME System

This chapter explains the basic tasks necessary to set up IP phones in an Cisco CallManager Express
(Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Information About Setting Up Phones in a Cisco CME System, page 45
• Specifying Cisco CME Parameters, page 46
• Setting Up Initial Extensions and Phones, page 50
• Specifying Phone-Related Parameters, page 64
• Resetting and Restarting Cisco CME Phones, page 75
• Verifying Cisco CME Phone Configuration, page 80
• Configuration Examples for Setting Up Phones in a Cisco CME System, page 83
• What to Do Next, page 86

Information About Setting Up Phones in a Cisco CME System

This chapter describes three basic tasks necessary to get phones in a Cisco CME system up and running:
1. Specifying Cisco CME Parameters, page 46
This task prepares the Cisco CME system for the setup of extensions and phones. If you are using
the Cisco CME setup tool to set up initial extensions (Task 2), you do not need to perform this task.
2. Setting Up Initial Extensions and Phones, page 50
This task sets up the initial extensions and phones that you want to start with. Later, you can use the
Cisco CME graphical user interface (GUI) or router command-line interface (CLI) to add or change
extensions and phones.

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Specifying Cisco CME Parameters

There are three ways to perform this task:

• Automated, using the Cisco CME setup tool.
• Partially automated, using the auto assign command.
• Manual, using the router CLI.
3. Specifying Phone-Related Parameters, page 64
This task allows you to define various required and optional parameters that should be in place when
you first set up your Cisco CME phones.
In addition, this chapter contains the following information:
• How to reboot phones after configuration changes in the “Resetting and Restarting Cisco CME
Phones” section on page 75.
• How to verify your configuration in the “Verifying Cisco CME Phone Configuration” section on
page 80.
• A configuration example is located in the “Configuration Examples for Setting Up Phones in a
Cisco CME System” section on page 83.

Specifying Cisco CME Parameters

Note If you are using the Cisco CME setup tool, you do not have to perform the tasks in this section. Proceed
to the “Automated Phone Setup Using the Cisco CME Setup Tool” section on page 50.

Note Cisco recommends that customers using Cisco CME 3.0 and later versions should configure the
transfer-system command using the full-consult or full-blind keyword, which allows IP phones to
perform consultative or blind transfers to local phones and phones across a WAN. Note that the default
for the transfer-system command is the blind keyword, so the transfer-system command must be
explicitly configured for the recommended full-consult or full-blind setting.
Customers running Cisco IOS Telephony Services (Cisco ITS) 2.1 or an earlier version should use the
local-consult or blind keyword with the transfer-system command to enable the Cisco proprietary
transfer method.
Customers using Cisco ITS 2.1 can use the full-consult or full-blind keyword to enable H.450.2 call
transfer by also configuring the router with a Tcl script that is contained in the file called
app-h450-transfer.x.x.x.x.zip. This file is posted on the Cisco CME software download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp. For configuration information, see the Cisco IOS
Telephony Services V2.1 guide.

The commands in this task identify and modify eXtensible Markup Language (XML) phone
configuration files so that IP phones can automatically find the defaults to configure themselves when
they come online or are rebooted. The last step in this task is to reset all phones, which causes them to
request the new firmware files.
Cisco ITS V2.1 introduced the use of XML configuration files for IP phones. There is one shared default
XML configuration file for each type of IP phone. When an IP phone comes online or is rebooted, it
automatically gets information about itself from the appropriate default configuration file. The phone
coming online uses a filename alias based on the phone type, which either is automatically detected by

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 Setting Up Phones in a Cisco CME System
Specifying Cisco CME Parameters

the Cisco CME router or is specified in the type command in ephone configuration mode. The type
command is mandatory only for ATA phones or for IP phones that are adding one or two Cisco IP Phone
7914 Expansion Modules.
In Cisco ITS V2.1, Cisco CME 3.0, and later versions, the XML configuration files have been moved to
system:/its/. The file named flash:SEPDEFAULT.cnf that was used with previous ITS versions is now
obsolete, but is retained as system:/its/SEPDEFAULT.cnf to support upgrades from older phone
firmware.
In a Cisco CME system, the IP phones receive their initial configuration information and phone firmware
from the TFTP server associated with the Cisco CME router. In most cases, the phones obtain the IP
address of their TFTP server using the Dynamic Host Configuration Protocol (DHCP) option 150
command. For Cisco CME operation, the TFTP server address obtained by the Cisco IP phones should
point to the Cisco CME router IP address. The Cisco IP phones attempt to transfer a configuration file
called XmlDefault.cnf.xml. This file is automatically generated by the Cisco CME router through the ip
source-address command and placed in router memory. The XmlDefault.cnf.xml file contains the IP
address that the phones use to register for service, using the Skinny Client Control Protocol (SCCP). This
IP address should correspond to a valid Cisco CME router IP address (and may be the same as the router
TFTP server address).
Similarly, when an analog telephone adaptor (ATA) such as the ATA-186 is attached to the Cisco CME
router, the ATA receives very basic configuration information and firmware from the TFTP server
XMLDefault.cnf.xml file. Access to the XML Default.cnf.xml file must be granted by using the
tftp-server command on the router. The XMLDefault.cnf.xml file is automatically generated by the
Cisco CME router with the ip source-address command and is placed in the router’s flash memory.

SUMMARY STEPS

1. tftp-server flash:filename
2. telephony-service
3. max-ephones max-phones
4. max-dn max-directory-numbers
5. load phone-type firmware-file
6. ip source-address ip-address [port port] [any-match | strict-match]
7. create cnf-files
8. keepalive seconds
9. reset all [time-interval]
10. exit
11. Verify that all phones have been upgraded.

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Specifying Cisco CME Parameters

DETAILED STEPS

Command or Action Purpose

Step 1 tftp-server flash:filename Permits the Cisco CME router to provide TFTP access to
the specified file by the IP phones served by the router.
Example: Note The phone version is derived from the phone
Router(config)# tftp-server firmware filename. For example, [03 02 02 09] is
flash:P00303020214.bin phone version 3.2(2.9).
Step 2 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 3 max-ephones max-phones Sets the maximum number of Cisco IP phones to be
supported by this router. The maximum number of
phones is platform- and version-dependent. Refer to CLI
Example:
Router(config-telephony)# max-ephones 24
help.
Step 4 max-dn max-directory-numbers Sets the maximum number of extensions (ephone-dns) to
be supported by this router. The maximum number of
extensions is platform- and version-dependent. Refer to
Example:
Router(config-telephony)# max-dn 48
CLI help.

Step 5 load phone-type firmware-file Identifies the Cisco IP phone firmware file to be used by
phones of the specified Cisco IP phone type when they
register.
Example:
Router(config-telephony)# load 7960-7940 • phone-type—Type of IP phone. Consult CLI help for
P00303020214 valid entries.
• firmware-file—Filename of the phone firmware,
without the filename suffix.
Note If a firmware file that you are loading for a
particular phone type is larger than 384KB, you
must first load a file on that phone type that is
smaller than 384KB, then load the larger file.
Step 6 ip source-address ip-address [port port] Identifies the IP address and port number that the
[any-match | strict-match] Cisco CME router uses for IP phone registration. The
default port is 2000.
Example: • any-match—(Optional) Disables strict IP address
Router(config-telephony)# ip source-address checking for registration. This is the default.
10.16.32.144
• strict-match—(Optional) Instructs the router to
reject IP phone registration attempts if the IP server
address used by the phone does not exactly match
the source address.
Step 7 create cnf-files Builds the XML configuration files that are required for
Cisco CME phones.
Example:
Router(config-telephony)# create cnf-files

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Specifying Cisco CME Parameters

Command or Action Purpose

Step 8 keepalive seconds (Optional) Sets the time interval, in seconds, between
keepalive messages that are sent to the router by Cisco IP
phones. The default is usually adequate. If the interval is
Example:
Router(config-telephony)# keepalive 45
set too large, it is possible that notification will be
delayed when a system goes down.
• seconds—Range is from 10 to 65535. Default is 30.
Step 9 reset {all [time-interval] | cancel | mac-address Performs a fast reboot on one or all IP phones associated
| sequence-all} with the Cisco CME router. A reset causes an IP phone
to ask for the phone firmware file associated with its
Example: phone type.
Router(config-telephony)# reset all • all—Resets all phones.
• time-interval—(Optional) Time interval, in seconds,
after each phone reset. Range is from 0 to 60.
Default is 15. This argument allows you to provide
an interval between resets so that all phones do not
attempt to access the TFTP resource at the same
time.
• cancel—Interrupts a sequential reset cycle.
• mac-address—MAC address of the phone to reset.
• sequence-all—Resets all phones in strict
one-at-a-time order by waiting for one phone to
reregister before starting the reset for the next phone.
The sequencing of resets prevents possible conflicts
between phones trying to access TFTP services
simultaneously.
Step 10 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 11 Verify that all phones have been upgraded. Check the version number displayed on an individual IP
phone when you press the Settings button.
From the router, check the firmware version loaded on a
specific phone by enabling the debug ephone register
command, resetting the phone, and looking at the Load
parameter that is displayed as part of the informational
StationAlarm message generated when the phone
registers.
If a phone fails to upgrade, reset the individual phone
again by using the reset mac-address command.

Troubleshooting Tips

• Check the download web page to determine the currently recommended phone firmware version
(http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp).
• Use the debug ephone register command to show status (alarm) messages at registration time. The
messages include the current IP phone firmware version.

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Setting Up Initial Extensions and Phones

Setting Up Initial Extensions and Phones

There are three ways to create an initial phone setup in an Cisco CME system:
• Automated Phone Setup Using the Cisco CME Setup Tool, page 50
• Partially Automated Setup Using the Router CLI, page 56
• Manual Setup Using the Router CLI, page 59
Automated phone setup with the Cisco CME setup tool is the easiest way to configure a group of phones.
The Cisco CME setup tool provides a question-and-answer interface from the Cisco CME router CLI
and quickly builds a configuration file after you complete the dialog. This method also sets up DHCP
service and the router IP address, so you do not have to perform most of the tasks in the “Setting Up a
Cisco CME System” chapter. This method assigns one phone number to each phone as it registers with
the Cisco CME router.
A partially automated setup method is available when you use the auto assign command to automatically
distribute ephone-dn tags (phone extension instance numbers) from a specified range to IP phones as
they register with your Cisco CME router. This method also assigns one phone number to each phone as
it registers, but you still make manual entries using the router command-line interface (CLI) for other
parameters. You must complete the tasks described in the “Setting Up a Cisco CME System” chapter,
prior to using the partially automated setup method.
A third way to set up IP phones is by using individual commands at the router CLI. Each phone and
ephone-dn is set up one at a time. This method is useful when you are not using the default values for
parameters or when you have a keyswitch-style operation in which each phone shares some lines. This
method provides the most control. You must perform the manual tasks described in the“Setting Up a
Cisco CME System” chapter, prior to using the manual setup method.
Typically you are going to use automated means for some part of your phone setup and finish the
configuration using router CLI. For example, you can manually preconfigure a series of ephone-dn
entries with certain parameter values and then use the auto assign command to instruct the Cisco CME
system to automatically assign the next available ephone-dn from the series to each new phone as it
registers with the router. Or you can use the Cisco CME setup tool to automatically build a series of
ephone-dn entries starting from a specified extension number and then use the auto assign command in
the same manner to assign lines to phones as they register. With either of these methods you can later
use the router CLI to add special features or custom parameters to specific ephone-dns or phones after
they are set up.
This chapter describes all three methods for setting up ephone-dns and phones.

Automated Phone Setup Using the Cisco CME Setup Tool

The Cisco CME setup tool provides a question-and-answer interface that allows you to set up an entire
Cisco CME system at one time.
The Cisco CME setup tool is started using the telephony-service setup command. If you do not use the
setup keyword, you can set up phones one at a time using router CLI, as described in the “Manual Setup
Using the Router CLI” section on page 59. The setup keyword is not stored in the router nonvolatile
random-access memory (NVRAM).

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Setting Up Initial Extensions and Phones

When using the Cisco CME setup tool, you provide information in response to a series of questions. The
actual configuration is created only when the entire dialog has been completed. You can interrupt the
process using Control-c at any point prior to the final question without having any configuration occur.

Note If you attempt to use the setup option for a system that has a nonempty telephony-service configuration,
an error message advises you to remove the existing configuration first by using the no
telephony-service command.

Note For Cisco IP Phones 7910G, 7940G, and 7960G, the setup tool automatically searches router flash for
firmware files and automatically loads the firmware files it finds. If you have other phone types in your
system, you need to use the show flash command to identify the firmware files in router flash, and then
use the load and tftp-server commands to manually load the firmware, as shown in the following
example. Note that the firmware filename suffix is not used with the load command.
tftp-server flash:CP7905060000SCCP050124A.sbin
telephony-service
load 7905 CP7905060000SCCP050124A

Prerequisites
• Phone firmware files must be installed in the Cisco CME router flash memory.
• You must have already configured NTP on the Cisco CME router, as described in the “Setting Up a
Cisco CME System” chapter.

Restrictions
• The Cisco CME setup tool uses the auto assign command to associate ephone-dn tags (sequence
numbers) with phones, so the same restrictions that are described in the “Partially Automated Setup
Using the Router CLI” section on page 56 apply here.
• When lines are automatically assigned using the keyswitch mode in the Cisco CME setup tool with
two ephone-dn entries for each extension number, the two ephone-dn entries are automatically
assigned to a single phone.

SUMMARY STEPS

1. telephony-service setup
2. exit
3. exit
4. show running-config

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Setting Up Initial Extensions and Phones

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service setup Enters telephony-service configuration mode with the setup
option, which automatically starts the Cisco CME setup
tool.
Example:
Router(config)# telephony-service setup Note Table 4 on page 52 provides information about the
questions in the setup dialog.
Step 2 exit Exits telephony-service configuration mode after you have
completed Cisco CME setup.
Example:
Router(config-telephony)# exit
Step 3 exit Exits global configuration mode.

Example:
Router(config)# exit
Step 4 show running-config Displays the running configuration so that you can verify
settings for DHCP server, telephony service, and
ephone-dns that were created with the Cisco CME setup
Example:
Router# show running-config
tool.

Cisco CME Setup Tool Dialog

After you enter the telephony-service setup command, the Cisco CME setup tool prompts you for the
information necessary to automatically configure a Cisco CME system. Table 4 shows the prompts and
describes the information that is requested of you.

Table 4 Cisco CME Setup Tool DIalog Prompts

Cisco CME Setup Tool Prompt Description

Do you want to setup DHCP service for • Yes configures the Cisco CME router to act as a
your IP phones? [yes/no]: DHCP server, automatically providing IP addresses
to your IP phones and provisioning the default
If you respond yes, you see the following
gateway and TFTP IP addresses to be used by the
prompts:
phones. This method creates a single pool of IP
IP network for telephony-service DHCP addresses. If you need a pool for non-IP phones or
Pool:
if the Cisco CME router cannot act as the DHCP
Subnet mask for DHCP network :
TFTP Server IP address (Option 150) : router, answer no and manually define the DHCP
Default Router for DHCP Pool : server as described in the “Setting Up a Cisco CME
System” chapter.
• No indicates that you have already configured
DHCP or static IP addresses for the IP phones.
Do you want to start telephony-service • Yes starts the Cisco CME setup for phones.
setup? [yes/no]:
• No terminates the Cisco CME setup tool.

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Table 4 Cisco CME Setup Tool DIalog Prompts (continued)

Cisco CME Setup Tool Prompt Description

Enter the IP source address for IP address on which the router provides Cisco CME
Cisco CallManager Express services: services, usually the default gateway for the IP subnet
Enter the Skinny Port for
that you are using for the IP phones, and the port for
Cisco CallManager Express services : Skinny Client Control Protocol (SCCP) messages.
[2000]:
How many IP phones do you want to Enter the maximum number of IP phones that this
configure : [0]: system will support. This number can be increased later,
to the maximum listed in the Cisco CallManager
Express 3.3 Specifications.
Note The Cisco CME setup tool associates one
number with each newly registering phone. If
you want additional numbers on a phone, add
them manually later.
Do you want dual-line extensions • Yes—Each newly registering IP phones is assigned
assigned to phones? [yes for dual-line a single number that is associated with a single
/ no for single-line]:
phone button. The system generates a dual-line
ephone-dn entry for each ephone-dn.
• No—IP phones are linked directly to one or more
PSTN trunk lines. Using keyswitch mode requires
manual configuration in addition to using the
Cisco CME setup tool. The system generates two
ephone-dn entries for each ephone-dn, and they are
both assigned to a single phone.
What language do you want on IP phones? Language for IP phone displays, selected from the list.
0 English The default is 0, English.
1 French
2 German Note Russian is not supported in Cisco IOS
3 Russian Release 12.2(15)ZJ.
4 Spanish
5 Italian
6 Dutch
7 Norwegian
8 Portuguese
9 Danish
10 Swedish
[0]:

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Table 4 Cisco CME Setup Tool DIalog Prompts (continued)

Cisco CME Setup Tool Prompt Description

Which Call Progress tone set do you Locale for the tone set used to indicate call status or
want on IP phones : progress, selected from the list. The default is 0, United
0 United States
1 France
States.
2 Germany
3 Russia
4 Spain
5 Italy
6 Netherlands
7 Norway
8 Portugal
9 UK
10 Denmark
11 Switzerland
12 Sweden
13 Austria
14 Canada
[0]:
What is the first extension number you Lowest number to use for extension numbers. Each
want to configure :[0]: additional extension that is created receives a number
that is incremented by one. Extension numbers must be
compatible with your telephone number plan and with
PSTN numbering requirements if you use DID service.
Do you have Direct-Inward-Dial service • Yes if you have trunk access to public telephone
for all your phones? [yes/no]: service by ISDN or VoIP for all extension numbers.
The system creates an appropriate dial plan.
• No if you have simple analog phone lines only (for
example, FXO interfaces) or if you have trunk
access for some lines but not all lines.
If you answer yes to the previous question, Complete ten-digit telephone number, including area
you see the following prompt: code, that corresponds to the first extension number.
Enter the full E.164 number for the
first phone:
Do you want to forward calls to a voice • Yes to forward calls to a single voice message
message service? [yes/no]: service number when an IP phone is busy or does
not answer. All phone extensions forward their calls
to the same voice message service pilot number.
• No to not forward calls to a single voice message
service number. Answer no if you do not have a
voice message system or if you want to customize
call forwarding behavior for each extension.
If you answer yes to the previous question, Voice message service pilot number. This step can be
you see the following prompt: ignored during the setup dialog and manually
Enter the extension or pilot number of configured later.
the voice message service:

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Table 4 Cisco CME Setup Tool DIalog Prompts (continued)

Cisco CME Setup Tool Prompt Description

Call forward No Answer Timeout: [18]: Timeout, in seconds, after which to forward calls to
voice mail if they are not answered. Default is 18.
Do you wish to change any of the above • Yes starts the dialog over again without
information? [yes/no]: implementing any of the answers that you
previously gave.
• No starts the automatic configuration process.

Configurations Resulting from the Cisco CME Setup Tool Dialog

The following configurations result from certain answers to the questions from the Cisco CME setup
tool:
• If DHCP configuration is requested, the following configuration CLI is generated according to the
parameters entered:
ip dhcp pool ITS
network x.x.x.x y.y.y.y
default-router s.s.s.s
option 150 ip s.s.s.s

If the address pool named ITS already exists, the DHCP configuration is not modified, and a warning
message is displayed.
• The following telephony-service configuration is generated:
telephony-service
max-ephones phones
max-dn phones
ip source-address s.s.s.s
voicemail vm-number

• The system also performs a directory check of the flash system on the Cisco CME router, searching
for phone firmware files and music-on-hold files. If a music-on-hold file called music-on-hold.au is
present in flash memory, the system adds the following line to the telephony-service configuration:
moh music-on-hold.au

• If phone firmware files with names of the form “P003........” and “P004........” are present in the flash
file system, the following configuration lines are added:
tftp-server flash:P003........
tftp-server flash:P004........

• If multiple phone-firmware files are present, a tftp-server command configuration will be created
for each file found. If multiple phone-firmware files are found for a particular phone type, the file
that was most recently stored in flash memory is chosen for use in the load command. The following
configuration lines are added:
telephony-service
load 7960 P003........
load 7910 P004........
create cnf-files

• If dual-line extensions are selected, the following commands are automatically entered into the
configuration for each ephone-dn instance. In the example below, the user specified an initial
extension number of 401, a voice-mail number of 501, and a timeout of 30 seconds.

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ephone-dn 1 dual-line
number 401
call-forward busy 501
call-forward noans 501 timeout 30

ephone-dn 2 dual-line
number 402
call-forward busy 501
call-forward noans 501 timeout 30

• If dual-line extensions are not selected, the following commands are automatically entered into the
configuration for each ephone-dn instance. In the example below, the user specified an initial
extension number of 401 and a voice-mail number of 501.
ephone-dn 1
number 401
call-forward busy 501
call-forward noans 501 timeout 30

ephone-dn 2
number 402
call-forward busy 501
call-forward noans 501 timeout 30

• If DID service is requested, the system automatically generates a dialplan-pattern command as

appropriate. At least the last (rightmost) digit of the first extension number and the last (rightmost)
digit of the full public telephone number must match. If these do not match, an error message is
generated and the user is asked answer the question about DID service again.
• Once all the ephone-dn entries have been generated, the system uses the auto assign command to
allow the specified ephone-dn tags to be assigned to IP phones as the phones register:
auto assign dn-tag to dn-tag

Partially Automated Setup Using the Router CLI

The partially automated method of setting up IP phones allows you to use the auto assign command to
specify a range of ephone-dn tags to which newly discovered IP phones are automatically assigned. This
method is useful when you have a phone setup in which each phone is assigned a separate, unique
extension number.
You can restrict a particular type of phone to a certain range of extension numbers by specifying in the
auto assign command a range of ephone-dn tags that have already been assigned those extension
numbers. For example, you may want extensions 6001 to 6025 assigned to Cisco IP Phone 7960G unitss
and extensions 5001 to 5025 assigned to Cisco IP Phone 7905G units. If no phone type is specified in
the auto assign command, numbers within the defined range are assigned to any phone, regardless of
type.
If all the ephone-dn tags in a range are to forward calls to the same voice-mail pilot number, call
forwarding can be set up automatically using the auto assign command. The remainder of the ephone-dn
setup can be handled automatically using the Cisco CME setup tool or manually using the router CLI.
The configuration resulting from automatic assignment is written to the Cisco CME router NVRAM at
preset intervals.

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Prerequisites
• The tasks described in the “Setting Up a Cisco CME System” chapter must be completed before you
start the task described in this section.
• The tasks described in the “Specifying Cisco CME Parameters” section on page 46 must be
completed before you start the task described in this section.

Restrictions
• Automatic assignment cannot be used to define ephone-dn tags for the Cisco IP Phone Expansion
Module 7914.
• Automatically assigned ephone-dn tags must belong to normal ephone-dns and cannot belong to
paging ephone-dns, intercom ephone-dns, music-on-hold (MOH) ephone-dns, or
message-waiting-indication (MWI) ephone-dns. Ephone-dn tags that are automatically assigned
must have at least a primary number defined for them.
• All of the ephone-dns in a single automatic assignment set must be of the same kind (either
single-line or dual-line).
• If an insufficient number of ephone-dns are available in the automatic assignment set, some phones
will not receive ephone-dns.
• Automatic assignment cannot create shared lines.
• Reversal or undoing of automatic assignment must be performed by manual CLI entry.

SUMMARY STEPS

1. telephony-service
2. auto assign dn-tag to dn-tag [type phone-type] [cfw extn-number timeout seconds]
3. restart all
or
reset {all [time-interval] | sequence-all}
4. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 auto assign dn-tag to dn-tag [type phone-type] Automatically assigns ephone-dn tags from the specified
[cfw extn-number timeout seconds] range to newly discovered IP phones.
• dn-tag to dn-tag—Range of ephone-dn tags (unique
Example: sequence numbers) to be automatically assigned to IP
Router(config-telephony)# auto assign 1 to 10 phones. The value of the dn-tag argument ranges from
type 7910 cfw 5001 timeout 30
1 to 288.
• type phone-type—(Optional) IP phone type to which
this range of ephone-dn tags is to be restricted. Valid
phone types are 7902, 7905, 7910, 7912, 7920, 7935,
7936, 7940, 7960, 7970, 7971, and ATA.
• cfw extn-number—(Optional) Extension number to
which calls should be forwarded on busy and no-answer
conditions.
• timeout seconds—(Optional; required if the cfw
keyword is used) Amount of time to wait when a call is
not answered before forwarding a call to the specified
number, in seconds. The value of the seconds argument
ranges from 3 to 60000.
Step 3 restart all The restart command performs a complete phone reboot
without contacting the DHCP and TFTP servers. Use the
or restart command unless you use the type keyword in the
reset {all [time-interval] | sequence-all} auto assign command.
The reset command performs a complete phone reboot that
Example: includes contacting the DHCP and TFTP servers for the
Router(config-telephony)# restart all latest configuration information. Use the reset command if
you use the type keyword in the auto assign command.
or
• all—This keyword causes the router to pause 15
seconds between the reset start for each successive
Example: phone.
Router(config-telephony)# reset sequence-all
• time-interval—(Optional) Time interval, in seconds,
between the start of each phone reset. Range is from
0 to 60. Default is 15.
• sequence-all—This keyword causes the router to wait
until one reset is complete before starting to reset the
next phone.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

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Example

The following example sets up automatic assignment for ephone-dns 1 through 10 for Cisco IP
Phone 7910s that are newly discovered by the Cisco CME router. Call forwarding is also configured for
these ephone-dns, with calls forwarded to extension 5001 after they are not answered for 30 seconds.
telephony-service
auto assign 1 to 10 type 7910 cfw 5001 timeout 30
reset all

Manual Setup Using the Router CLI

The manual process for setting up an extension involves defining an ephone-dn and the manual process
for setting up a Cisco CME phone involves defining an ephone. These terms are defined as follows:
• Ephone-dn—A software construct that represents a line that connects a voice channel to a phone
instrument where a user can receive and make calls. An ephone-dn represents a virtual voice port in
the Cisco CME system, so the maximum number of ephone-dns in a Cisco CME system is the
maximum number of simultaneous call connections that can occur. Note that this concept is different
from the maximum number of physical lines in a traditional telephony system and also is different
from the maximum number of telephone or extension numbers that can be assigned.
• Ephone—A software construct that represents a physical telephone instrument. This construct
provides the ability to configure the instrument itself using Cisco IOS software. The maximum
number of ephones in a Cisco CME system is the maximum number of physical instruments that can
be connected to the system.
To set up an extension (ephone-dn), you associate one or two numbers and a name with a unique
ephone-dn tag (identifier or sequence number) that identifies this particular ephone-dn for configuration
purposes. This creates the virtual line, or extension, with which call connections can be made.
Each ephone-dn tag is then associated with a line button on one or more ephones (the software constructs
that allow you to program extensions and features onto physical IP phones). Each IP phone must be
configured as an ephone on the Cisco CME router to receive support in the LAN environment.
Using the ephone-dn command and the dual-line keyword, you can create an ephone-dn in dual-line
mode. Each dual-line ephone-dn has one voice port and two channels to handle two independent calls.
This mode enables call waiting, call transfer, and conference functions on a single ephone-dn. Dual-line
mode works with all phone types, but is not appropriate for ephone-dns that are used for voice-mail
numbers, intercoms, message-waiting indicators, paging, or hunt groups. Overlays and hunt groups that
use dual-line ephone-dns are supported. Only one dual-line ephone-dn can be set on a Cisco IP Phone
7910.
The button command in ephone configuration mode allows you to customize ring characteristics for
each button or to designate buttons that should not produce an audible ring when they receive incoming
calls. The feature-ring and silent-ring features are supported on all phones but are most appropriate for
multiline IP phones, including the Cisco IP Phones 7940 and 7940G, Cisco IP Phones 7960 and 7960G,
and Cisco IP Phone Expansion Module 7914.
For more information about types of ephone-dns and about ephones, see the “Before You Start: Basic
Cisco CME Concepts” section in the “Cisco CallManager Express Overview” chapter.

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Prerequisites
• The tasks described in the “Setting Up a Cisco CME System” chapter must be completed before you
start the task described in this section. In particular, be sure that you have copied Cisco CME files
to the Cisco CME router flash memory.
• The tasks described in the “Specifying Cisco CME Parameters” section on page 46 must be
completed before you start the task described in this section.

SUMMARY STEPS

1. ephone-dn dn-tag [dual-line]

2. number number [secondary number] [no-reg [both | primary]]
3. name name
4. exit
5. ephone phone-tag
6. mac-address [mac-address]
7. type phone-type [addon 1 module-type [2 module-type]]
8. button button-number{separator}dn-tag [[button-number{separator}dn-tag] ...]
9. keepalive seconds
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 55 dual-line ephone-dn during configuration tasks. The maximum
number of ephone-dns for a particular Cisco CME
system is version- and platform-specific. Refer to CLI
help for the range of values.
• dual-line—(Optional) Enables an ephone-dn with one
voice port and two voice channels, which supports
features such as call waiting, call transfer, and
conferencing with a single ephone-dn.
Note To change an ephone-dn from dual-line to
single-line mode or the reverse, you must first
delete the ephone-dn and then recreate it.

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Command or Action Purpose

Step 2 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn
| primary]] instance.
• number—String of up to 16 digits that represents a
Example: telephone or extension number to be associated with
Router(config-ephone-dn)# number 2345 this ephone-dn.
• secondary—(Optional) Allows you to associate a
second telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should
not register with the H.323 gatekeeper. Unless you
specify one of the optional keywords (both or
primary) after the no-reg keyword, only the
secondary number is not registered.
Step 3 name name Associates a name with this ephone-dn instance. This name
is used for caller-ID displays and in the local directory
listings.
Example:
Router(config-ephone-dn)# name Smith, John You must follow the name order that is specified in the
directory command in telephony-service configuration
mode (either first-name-first or last-name-first).
Step 4 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 5 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: this ephone during configuration tasks. The maximum
Router(config)# ephone 6 number of ephones for a particular Cisco CME system
is version- and platform-specific. For the range of
values, refer to CLI help.
Step 6 mac-address [mac-address] Specifies the MAC address of the IP phone that is being
configured.
Example: • mac-address—(Optional) MAC address on the bottom
Router(config-ephone)# mac-address 2946.3f2.311 of an IP phone. If you choose to register phones before
configuring them, the mac-address command can be
used during configuration without entering the
mac-address argument. The Cisco CME system
detects MAC addresses and automatically populates
phone configurations with their corresponding MAC
addresses and phone types. This capability is
supported only for Cisco CME 3.0 and later versions,
and is not supported for voice-mail ports.

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Command or Action Purpose

Step 7 type phone-type [addon 1 module-type (Optional) Specifies the type of phone. This command is
[2 module-type]] required for phones with ATA devices and when you enter
an add-on module. For other phone types, this command is
Example: optional.
Router(config-ephone)# type 7960 addon 1 7914 • phone-type—Valid entries are the following:
– 7902—Cisco IP Phone 7902G.
– 7905—Cisco IP Phone 7905G.
– 7910—Cisco IP Phone 7910.
– 7912—Cisco IP Phone 7912G.
– 7920—Cisco Wireless IP Phone 7920.
– 7935—Cisco IP Conference Station 7935.
– 7936—Cisco IP Conference Station 7936.
– 7940—Cisco IP Phones 7940 and 7940G.
– 7960—Cisco IP Phones 7960 and 7960G.
– 7970—Cisco IP Phone 7970G.
– 7971—Cisco IP Phone 7971G-GE.
– ata—Cisco ATA-186 or Cisco ATA-188.
• module-type—Valid entry is the following:
– 7914—Cisco IP Phone 7914 Expansion Module.

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Command or Action Purpose

Step 8 button button-number{separator}dn-tag Associates a button number and line characteristics with an
[[button-number{separator}dn-tag] ...] extension (ephone-dn). Maximum number of buttons is
determined by phone type.
Example: Note The Cisco IP Phone 7910 has only one line button,
Router(config-ephone)# button 1:10 2:11 3b12 but can be given 2 ephone-dn tags.
4o13,14,15
• button-number—Number of a line button on an IP
phone, starting with 1 as the top button.
• separator—Single character that denotes the type of
characteristics to be associated with the extension.
Valid entries are the following:
– : (colon)—Normal ring. Incoming calls, single
pulse for internal calls and double pulse for
external calls. Visible cues are a flashing ((< icon
in the phone display and a flashing red light on the
handset.
– b—Beep is allowed. Audible ring is suppressed
for incoming calls, but call-waiting beeps are
allowed. Visible cues are the same as for a normal
ring.
– f—Feature ring. Triple-pulse cadence
differentiates incoming calls on one line from
incoming calls on other lines on the phone.
– m—Monitor mode for a shared line. Visible line
status indicates in-use or not. Line cannot be used
on this phone to make or receive calls.
– o—Overlay line. Multiple ephone-dns share a
single button, up to a maximum of ten on a button.
The dn-tag argument can contain up to ten dn-tags,
separated by commas.
– s—Silent ring. Audible ring and call-waiting beep
are suppressed for incoming calls. Visible cues are
the same as for a normal ring.
• dn-tag—Unique sequence number of the ephone-dn
that you want to appear on this button. For overlay
lines, this argument can contain up to ten sequence
numbers, separated by commas.
Step 9 keepalive seconds (Optional) Sets the length of the time interval between
successive keepalive messages from the
Cisco CallManager Express router to a particular IP phone.
Example:
Router(config-ephone)# keepalive 200 • seconds—Interval time, in seconds. Range is from 10
to 65535. Default is 30.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

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Specifying Phone-Related Parameters

Whether you use an automated or manual method to set up Cisco CME phones, you must also consider
the following phone-related parameters:
• DTMF Relay for H.323 Networks, page 64 (required)
• Dial-Plan Pattern, page 65 (required if you have DID)
• Translation Rules, page 68 (optional)
• Translation Profiles, page 69 (optional)
• International Language, Date Formats, and Tone Support, page 72 (optional)

DTMF Relay for H.323 Networks

IP phones connected to Cisco CME systems require the use of out-of-band DTMF relay to transport
DTMF (keypad) digits across VoIP connections. The reason for this is that the codecs used for in-band
transport may distort the tones and make them unrecognizable. DTMF relay solves the problem of
DTMF tone distortion by transporting DTMF tones out-of-band, or separate, from the encoded voice
stream.
For IP phones on H.323 networks, DTMF is relayed using the H.245 alphanumeric method, which is
defined by the ITU H.245 standard. This method separates DTMF digits from the voice stream and sends
them as ASCII characters in H.245 user input indication messages through the H.245 signaling channel
instead of the RTP channel. For more information, refer to the “Configuring H.323 Gateways” chapter
of the Cisco IOS H.323 Configuration Guide.

Note For DTMF relay on SIP networks, refer to the configuration instructions in “Appendix B: Providing
Cisco CME Support for SIP”.

SUMMARY STEPS

1. dial-peer voice tag voip

2. dtmf-relay h245-alphanumeric
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 2 voip
Step 2 dtmf-relay h245-alphanumeric Specifies the H.245 alphanumeric method for
relaying dual tone multifrequency (DTMF) tones
between telephony interfaces and an H.323 network.
Example:
Router(config-dial-peer)# dtmf-relay
h245-alphanumeric
Step 3 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Example

The following excerpt from the show running-config command output shows a dial peer configured to
use H.245 alphanumeric DTMF relay:
dial-peer voice 4000 voip
destination-pattern 4000
session target ipv4:10.0.0.25
codec g711ulaw
dtmf-relay h245-alphanumeric

Dial-Plan Pattern
The dialplan-pattern command creates a sequence of digits that specifies a global prefix for the
expansion of abbreviated extension numbers into fully qualified E.164 numbers. You can define up to
five dial-plan patterns.

Note In networks that have a single router, you do not need to use the dialplan-pattern command.

If multiple dial-plan patterns are defined, the system matches extension numbers against the patterns in
sequential order, starting with the pattern that was defined first. Once a pattern matches an extension
number, the pattern is used to generate an expanded number. If additional patterns subsequently match
the extension number, they are not used.
The dialplan-pattern command builds additional dial peers for the expanded numbers it creates. For
example, when the ephone-dn with the number 1001 was defined, the following POTS dial peer was
automatically created for it:
dial-peer voice 20001 pots
destination-pattern 1001
voice-port 50/0/2

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If you then define a dial-plan pattern that 1001 will match, such as 40855510.., a second dial peer is
created so that calls to both the 1001 and 4085551001 numbers will be completed. In our example, the
additional dial peer that is automatically created might look like the following:
dial-peer voice 20002 pots
destination-pattern 4085551001
voice-port 50/0/2

Both dial peers can be seen with the show telephony-service dial-peer command.
In networks with multiple routers, you may need to use the dialplan-pattern command to expand
extensions to E.164 numbers because local extension numbering schemes can overlap each other.
Networks with multiple routers have authorities such as gatekeepers that route calls through the network.
These authorities use require E.164 numbers so that all numbers in the network will be unique. The
dialplan-pattern command expands extension numbers into unique E.164 numbers for that use.
A dial-plan pattern is required to register the Cisco IP phone lines with a gatekeeper. Ephone-dn numbers
for the Cisco IP phones must match the number in the extension-length argument. For example, if the
extension length is 3, all extensions must be three numbers in length. Otherwise, the extension number
cannot be converted to a qualified E.164 number.
Using the dialplan-pattern command to expand extension numbers can sometimes result in the
improper matching of numbers with dial peers. For example, the expanded E.164 number 2035550134
matches dial-peer destination-pattern 203, not 134, which would be the correct destination pattern for
the desired extension.
If it is necessary for you to use the dialplan-pattern command and you know that the expanded numbers
might match destination patterns for dial peers, you can manually configure an extensions E.164
expanded number as its secondary number instead of using the dialplan-pattern command, as shown in
the following example.
ephone-dn 23
number 134 secondary 2035550134

The pattern created by the dialplan-pattern command is also used to enable distinctive ringing for
inbound calls. If a calling-party number matches a dial-plan pattern, the call is considered an internal
call and has a distinctive ring that identifies the call as internal. Any call with a calling-party number
that does not match a dial-plan pattern is considered an external call and has a distinctive ring that is
different from the internal ringing.
When the extension-pattern keyword and extension-length argument are used, the leading digits of an
extension pattern are stripped and replaced with the corresponding leading digits of the dial plan. For
example, the following command maps all 4xx extension numbers to the PSTN number 40855501xx, so
that extension 412 corresponds to 4085550112.
dialplan-pattern 1 4085550100 extension-length 3 extension-pattern 4..
The extension-pattern keyword allows additional manipulation of abbreviated extension-number prefix
digits. When this keyword and its argument are used, the leading digits of an extension pattern are
stripped and replaced by the corresponding leading digits of the dial-plan pattern. This command can be
used to avoid having Direct Inward Dialing (DID) numbers like 408-555-0101 result in four-digit
extensions such as 0101.

SUMMARY STEPS

1. telephony-service
2. dialplan-pattern tag pattern extension-length length [extension-pattern epattern] [no-reg]
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 dialplan-pattern tag pattern extension-length length Maps a digit pattern for an abbreviated
[extension-pattern epattern] [no-reg] extension-number prefix to the full E.164 telephone
number pattern.
Example: • tag—Dial-plan string tag used before a ten-digit
Router(config-telephony)# dialplan-pattern 1 telephone number. Range is from 1 to 5.
4085550100 extension-length 3 extension-pattern 4..
• pattern—Dial-plan pattern for full E.164
Note This example maps all extension numbers 4xx to the number.
PSTN number 40855501xx, so that extension 412
• extension-length length—Number of digits in
corresponds to 4085550112.
the epattern argument that is associated with the
extension-pattern keyword.
• extension-pattern epattern—(Optional)
Internal extension pattern to use. In addition to
digits, the following wildcards can be used:
– . (period)—Stands for a single character.
– T—Stands for timeout in the context of the
user entering digits. For example, four
periods and a T (....T) tells the system to
receive at least four digits from the user and
wait for the user to stop entering digits.
• no-reg—(Optional) Prevents the E.164 number
in the dial peer from registering with a
gatekeeper. By not registering some numbers,
you leave them available to be used for other
telephony services.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example maps the extension pattern 4.. to the last three digits of the dial-plan pattern
4085550155:
telephony-service
dialplan-pattern 1 4085550155 extension-length 3 extension-pattern 4..

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Translation Rules
Translation rules are a powerful, general-purpose number-manipulation mechanism supported by
Cisco IOS software that can be used to perform operations such as automatically adding telephone area
and prefix codes to dialed numbers. Translation rules are applied to virtual voice ports created by
ephone-dns.
The translation rule mechanism inserts a delay into the dialing process when digits are entered that do
not explicitly match any of the defined translation rules. This delay is set by the interdigit timeout. The
translation-rule mechanism uses the delay to ensure that it has acquired all of the digits from the phone
user before making a final decision that there is no translation-rule match available (and therefore no
translation operation to perform). To avoid this delay, it is recommended that you include a dummy
translation rule to act as a pass-through rule for digit strings that do not require translation. For example,
a rule like “^5 5” that maps a leading 5 digit into a 5 would be used to prevent applying the translation
rule delay to local extension numbers that started with a 5.

SUMMARY STEPS

1. translation-rule number
1. ephone-dn dn-tag
2. translate {called | calling} translation-rule-tag
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 translation-rule number Defines a translation rule for voice calls.
• number—Number that identifies the translation rule.
Example: Range is from 1 to 2147483647.
Router(config)# translation-rule 123
Step 2 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—The ephone-dn tag (sequence number) of the
Example: extension to which you want to apply the translation
Router(config)# ephone-dn 11 rules.
Step 3 translate {called | calling} Applies a translation rule to numbers dialed by
translation-rule-tag Cisco IP phone users. The called keyword translates the
called number, and the calling keyword translates the
Example: calling number.
Router(config-ephone-dn)# translate called 123 • translation-rule-tag—Sequence number of the
previously created translation rule. Valid entries are
1 to 2147483647.
Step 4 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

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Verifying Translation Rules

Step 1 show translation-rule

Use this command to verify the translation rules that you have defined for your translation profiles.
Router# show translation-rule 6
Translation-rule tag: 6
Rule 1:
Match pattern: 65088801..
Replace pattern: 6508880101
Match type: none Replace type: none
Match plan: none Replace plan: none

Step 2 test translation-rule number input-test-string [type match-type [plan match-type] ]

Use this command to test your translation profiles. For more information about this command, go to
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123tvr/vrht_t1.htm#wp1488
921.
Router(config)# translation-rule 5
Router(cfg-translation-rule)# rule 1 ^201 102
Router(cfg-translation-rule)# exit
Router# test translation-rule 5 2015550101
Matched with rule 5
Original number:2015550101 Translated number:1025550101
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none

Translation Profiles
Cisco CME 3.2 and later versions support translation profiles. Translation profiles allow you to group
translation rules together and to associate translation rules with the following:
• Called numbers
• Calling numbers
• Redirected called numbers
Translation profiles are created and named with the voice translation-profile command. In the
translation-profile configuration mode, the translate command is used to associate translation rules with
the translation profile and to associate the translation rules with called numbers, calling numbers, or
redirected called numbers. Finally, translation profiles are added to ephone-dn configurations.

Note Voice translation rules are a separate feature from the translation rules described in the “Translation
Rules” section on page 68. For information about the voice translation-rule command command, go to
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_reference_chapter0918
6a00801a7f63.html#wp1051601.

SUMMARY STEPS

1. voice translation-rule number

2. rule precedence /match-pattern/ /replace-pattern/
3. exit

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4. voice translation-profile name

5. translate {called | calling | redirect-called} voice-translation-rule-tag
6. exit
7. ephone-dn tag
8. translation-profile {incoming | outgoing} name
9. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice translation-rule number Defines a translation rule for voice calls and enters voice
translation-rule configuration mode.
• number—Number that identifies the translation rule.
Example:
Range is from1 to 2147483647.
Router(config)# voice translation-rule 1
Step 2 rule precedence /match-pattern/ Defines a translation rule.
/replace-pattern/
• precedence—Priority of the translation rule. Range is
from 1 to 15.
Example:
Router(cfg-translation-rule)# rule 1 /^9/ //
• match-pattern—Stream editor (SED) expression used
to match incoming call information. The slash (/) is a
delimiter in the pattern.
• replace-pattern—SED expression used to replace the
match pattern in the call information. The slash (/) is a
delimiter in the pattern.
Step 3 exit Exits voice translation-rule configuration mode.

Example:
Router(cfg-translation-rule)# exit
Step 4 voice translation-profile name Defines a translation profile for voice calls.
• name—Value of the translation profile. Maximum
length of the voice translation profile name is 31
Example:
alphanumeric characters.
Router(config)# voice translation-profile name1
Step 5 translate {called | calling | redirect-called} Associates a voice translation rule with a voice translation
voice-translation-rule-tag profile.
• called—Associates the translation rule with called
Example:
Router(cfg-translation-profile)# translate numbers.
called 1 • calling—Associates the translation rule with calling
numbers.
• redirect-called—Associates the translation rule with
redirected called numbers.
• translation-rule-tag—Reference number of the
translation rule. Range is from 1 to 2147483647.

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Command or Action Purpose

Step 6 exit Exits translation-profile configuration mode.

Example:
Router(cfg-translation-profile)# exit
Step 7 ephone-dn tag Enters ephone-dn configuration mode to create an
extension (ephone-dn) for a Cisco IP phone line, an
intercom line, a paging line, a voice-mail port, or a
Example:
Router(config)# ephone-dn 1
message-waiting indicator (MWI).
• tag—Unique sequence number that identifies this
ephone-dn during configuration tasks. Range is from 1
to the maximum number of ephone-dns allowed on the
router platform. Refer to CLI help for the maximum
value for this argument.
Step 8 translation-profile {incoming | outgoing} name Assigns a translation profile for incoming or outgoing call
legs to or from Cisco IP phones.
Example: • incoming—Applies the translation profile to incoming
Router(config-ephone-dn)# translation-profile calls.
outgoing name1
• outgoing—Applies the translation profile to outgoing
calls.
• name—The name of the translation profile.
Step 9 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

The following example shows the configuration where a translation profile called profile1 is created with
two voice translation rules. Rule1 consists of associated calling numbers, and rule2 redirects called
numbers. Ephone-dn 1 is configured with profile1.
voice translation-profile name1
translate calling rule1
translate redirect-called rule2

ephone-dn 1
number 1001
translation-profile incoming name1

Verifying Translation Profiles

Step 1 show voice translation-rule

Use this command to verify the translation rules that you have defined for your translation profiles.
Router# show voice translation-rule 6
Translation-rule tag: 6
Rule 1:
Match pattern: 65088801..
Replace pattern: 6508880101

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Match type: none Replace type: none

Match plan: none Replace plan: none

Step 2 test voice translation-rule number input-test-string [type match-type [plan match-type] ]
Use this command to test your translation profiles. For more information about this command, go to
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123tvr/vrht_t1.htm#wp1488
921.
Router(config)# voice translation-rule 5
Router(cfg-translation-rule)# rule 1 /201/ /102/
Router(cfg-translation-rule)# exit
Router# test voice translation-rule 5 2015550101
Matched with rule 5
Original number:2015550101 Translated number:1025550101
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none

International Language, Date Formats, and Tone Support

On the Cisco IP Phones 7940 and 7940G and Cisco IP Phones 7960 and 7960G, the language displayed
on the phone and the locale for call progress tones and cadences can be set to one of several ISO-3166
codes that indicate specific languages and geographic regions.
You can also select from several formats for the date and time display on Cisco IP phones.

Restrictions
Cisco IP Phone 7910, Cisco IP Phone Expansion Module 7914, Cisco IP Conference Station 7935, and
Cisco IP Conference Station 7936 support the United States ISO-3166 language code only.

SUMMARY STEPS

1. telephony-service
2. user-locale language-code
3. network-locale locale-code
4. date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | yy-mm-dd}
5. time-format {12 | 24}
6. reset all [time-interval]
7. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 user-locale language-code Specifies a language for display on the Cisco IP Phones
7940 and 7940G and Cisco IP Phones 7960 and 7960G.
Example: • language-code—Refer to CLI help for a list of
Router(config-telephony)# user-locale FR ISO-3166 codes that are supported. United States (US)
is the default.
Step 3 network-locale locale-code Specifies a set of call progress tones and cadences on the
Cisco IP Phones 7940 and 7940G and Cisco IP Phones 7960
and 7960G.
Example:
Router(config-telephony)# network-locale FR • locale-code—Refer to CLI help for a list of ISO-3166
codes that are supported. United States (US) is the
default.
Step 4 date-format {mm-dd-yy | dd-mm-yy | yy-dd-mm | Sets the date format for IP phone display. The choices are
yy-mm-dd} mm-dd-yy, dd-mm-yy, yy-dd-mm, and yy-mm-dd, where
• dd—day
Example:
Router(config-telephony)# date-format yy-dd-mm
• mm—month
• yy—year
• Default is mm-dd-yy.
Step 5 time-format {12 | 24} Selects a 12-hour clock or 24-hour clock for the time
display format on all Cisco IP phones attached to the router.
Example: • Default is 12.
Router(config-telephony)# time-format 24
Step 6 reset all [time-interval] Performs a complete reboot of all phones, including
contacting the DHCP and TFTP servers for the latest
configuration information.
Example:
Router(config-telephony)# reset all • all—Resets all phones associated with a Cisco CME
router.
• time-interval—(Optional) Time interval, in seconds,
between the start of each phone reset. Range is from
0 to 60. Default is 15.
Note The network-locale and user-locale commands
require a phone reset. If you are using only the
date-format or time-format commands, you can
use the restart command instead.
Step 7 exit Exits telephony-service mode.

Example:
Router(config-telephony)# exit

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Example

The following example sets the locale for display language and call progress tones to France:
telephony-service
user-locale FR
network-locale FR

Troubleshooting Tips

To display the current locale codes that are associated with dictionary, language, and call progress tone
files, use the show telephony-service tftp-bindings command.

Changing the TFTP Server Address

Phones get their TFTP server address from the DHCP server. If you need to change the TFTP server
address after it has been configured, you must change the TFTP IP address on the DHCP server and reset
the phones so that they recontact the DHCP server for the new TFTP address. Use one of the following
tasks:
• Changing the TFTP Server Address When the Cisco CME Router is the DHCP Server
• Changing the TFTP Server Address When the Cisco CME Router is Not the DHCP Server

Changing the TFTP Server Address When the Cisco CME Router is the DHCP Server
If the Cisco CME router is also the DHCP server,use the following steps:
1. Modify the DHCP pool using the option 150 ip command to change the TFTP IP address.
2. Reset the phones using the reset command.
For more information, see the “Setting Up DHCP Service for Cisco CME” section in the “Setting Up
Phones in a Cisco CME System” chapter.

Changing the TFTP Server Address When the Cisco CME Router is Not the DHCP Server
If the Cisco CME router is not the DHCP server, use the following steps:
1. Reconfigure the external DHCP server with the new TFTP address.
2. Reset the phones using the reset command.
For more information about DHCP, see the "DHCP" part of the Cisco IOS IP Addressing Services
Configuration Guide at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124cg/hiad_c/ch10/index.htm.

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Resetting and Restarting Cisco CME Phones

Cisco IP phones must be rebooted after configuration changes in order for the changes to take effect.
There are two ways to reboot a phone: using the reset command and using the restart command. The
differences between these commands are summarized in Table 5.

Table 5 reset and restart Commands

reset Command restart Command

Similar to power-off, power-on reboot. Quick restart.
Contacts the DHCP server and TFTP server for Does not contact the DHCP or TFTP server.
their Cisco CME system updates.
Takes longer to process when updating multiple Faster processing for multiple phones.
phones.
Must be used when updating the following: Can be used when updating the following:
• Phone firmware • Phone buttons
• User locale • Phone lines
• Network locale • Speed-dial numbers
• URL parameters

With either of these commands, you can reboot a single phone or you can reboot all phones in a
Cisco CME system. When you use the reset command to reboot multiple IP phones, it is possible for a
conflict to occur if too many phones attempt to access changed Cisco CME configuration information
via TFTP simultaneously. The sequence-all keyword has been provided to specify a sequential reset of
multiple IP phones to minimize the risk of that conflict.
The reset command takes significantly longer to process than the restart command when you are
updating multiple phones, but it must be used when you update phone firmware, user locale, network
locale, or URL parameters. For simple button, line, or speed-dial changes, use the restart command.
This section describes the following tasks:
• Using the reset Command in Ephone Configuration Mode, page 76
• Using the reset Command in Telephony-Service Configuration Mode, page 76
• Using the restart Command in Ephone Configuration Mode, page 78
• Using the restart Command in Telephony-Service Configuration Mode, page 79

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Using the reset Command in Ephone Configuration Mode

The reset command must be used to reboot IP phones after you update phone firmware, user locale,
network locale, or URL parameters. A phone reset takes longer than a phone restart because the DHCP
and TFTP servers are contacted for updates.

SUMMARY STEPS

1. ephone phone-tag
2. reset
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number of the ephone
Example: that you want to reset.
Router(config)# ephone 1
Step 2 reset Performs a complete reboot of the phone.

Example:
Router(config-ephone)# reset
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example performs a complete reboot of ephone 1:

ephone 1
reset

Using the reset Command in Telephony-Service Configuration Mode

The reset command must be used to reboot IP phones after you update phone firmware, user locale,
network locale, or URL parameters. A phone reset takes longer than a phone restart because the DHCP
and TFTP servers are contacted for updates. You can reset one phone or all phones from
telephony-service configuration mode.

SUMMARY STEPS

1. telephony-service
2. reset {all [time-interval] | cancel | mac-address mac-address | sequence-all}
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 reset {all [time-interval] | cancel | Performs a complete reboot of all phones or the phone with
mac-address mac-address | sequence-all} the specified MAC address, including contacting the DHCP
and TFTP servers for the latest configuration information.
Example: • all—Resets all phones associated with a Cisco CME
Router(config-telephony)# reset all router. This keyword causes the router to pause
15 seconds between the reset start for each successive
phone.
• time-interval—(Optional) Time interval, in seconds,
between the start of each phone reset. Range is from
0 to 60. Default is 15.
• cancel—Interrupts a sequential reset cycle.
• mac-address—Resets the phone that has the specified
MAC address.
• sequence-all—Resets all phones associated with this
Cisco CME router. This keyword causes the router to
wait until one reset is complete before starting to reset
the next phone. After the reset timeout of 4 minutes, the
router stops waiting for the currently registering phone
to complete registration and starts to reset the next
phone.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example performs a complete sequential reboot of all phones associated with the router
after the user locale code has been changed:
telephony-service
user-locale FR
reset sequence-all

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Using the restart Command in Ephone Configuration Mode

The restart command is used to reboot a single IP phone after you make simple button, line, or
speed-dial changes. A phone restart is faster than a phone reset because the DHCP and TFTP servers are
not contacted for updates.

SUMMARY STEPS

1. ephone phone-tag
2. restart
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number of the ephone
Example: that you want to restart.
Router(config)# ephone 1
Step 2 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-ephone)# restart
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example performs a fast reboot of ephone 1 after a change of button assignment:
ephone 1
button 1:32 2:33
restart

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Using the restart Command in Telephony-Service Configuration Mode

The restart command is used to reboot IP phones after you make simple button, line, or speed-dial
changes. A phone restart is faster than a phone reset because the DHCP and TFTP servers are not
contacted for updates. You can restart one phone or all phones from telephony-service configuration
mode.

SUMMARY STEPS

1. telephony-service
2. restart {all [time-interval] | mac-address}
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 restart {all [time-interval] | mac-address} Performs a fast reboot of the specified phone or all phones
associated with this Cisco CME router. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-telephony)# restart all • all—Restarts all phones associated with a Cisco CME
router.
• time-interval—(Optional) Time interval, in seconds,
between the beginning of each phone restart. Range is
from 0 to 60. Default is 15.
• mac-address mac-address—Restarts the phone that
has the specified MAC address.
Step 3 exit Exits telephony-service mode.

Example:
Router(config-telephony)# exit

Example

The following example performs a fast reboot of all phones associated with the router:
telephony-service
restart all

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Verifying Cisco CME Phone Configuration

The following steps can be used to verify the Cisco CME phone configurations.

SUMMARY STEPS

1. Use the show running-config command to verify ephone-dn configurations.

2. Verify the correct phone firmware installation by setting registration debugging with the debug
ephone register command.
3. Use the show telephony-service all command to verify that the Cisco CME router is enabled.
4. Use the show telephony-service tftp-bindings command to ensure that the locale-specific files are
correct.
5. Use the show ephone command to verify Cisco IP phone setup after phones have registered with the
Cisco CME router.
6. Use the show ephone-dn command to see settings related to ephone-dns.
7. Use the show dialplan number command to display the number resolutions of a particular phone
number, which allows you to detect whether calls are going to unexpected destinations.

DETAILED STEPS

Step 1 Use the show running-config command to verify ephone-dn configurations.

.
.
.
ephone-dn 1
number 1101
name user1
no huntstop
call-forward noan 4000 timeout 30
!
ephone-dn 2
number 1101
name user1
preference 1
call-forward busy 4000
call-forward noan 4000 timeout 30
!
ephone-dn 3
number 1102
name user2
no huntstop
call-forward noan 4000 timeout 30
.
.
.

Step 2 Verify the correct phone firmware installation by setting registration debugging with the debug ephone
register command. Then reset the phones and look at the StationAlarmMessage displayed during phone
re-registration. The “Load=” parameter should appear in the display, followed by an abbreviated version
name corresponding to the correct phone firmware filename.

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Step 3 Use the show telephony-service all command to verify that the Cisco CME router is enabled. This
command also displays ephone-dn configurations.
Router# show telephony-service all

CONFIG
======
ip source-address 10.0.0.1 port 2000
max-ephones 24
max-dn 24
dialplan-pattern 1 408734....
voicemail 11111
transfer-pattern 510734....
keepalive 30

ephone-dn 1
number 5001
huntstop

ephone-dn 2
number 5002
huntstop
call-forward noan 5001 timeout 8

Step 4 Use the show telephony-service tftp-bindings command to ensure that the locale-specific files are
correct.
Router# show telephony-service tftp-bindings

tftp-server system:/its/SEPDEFAULT.cnf
tftp-server system:/its/SEPDEFAULT.cnf alias SEPDefault.cnf
tftp-server system:/its/XMLDefault.cnf.xml alias XMLDefault.cnf.xml
tftp-server system:/its/ATADefault.cnf.xml
tftp-server system:/its/XMLDefault7960.cnf.xml alias SEP00036B54BB15.cnf.xml
tftp-server system:/its/germany/7960-font.xml alias German_Germany/7960-font.xml
tftp-server system:/its/germany/7960-dictionary.xml alias
German_Germany/7960-dictionary.xml
tftp-server system:/its/germany/7960-kate.xml alias German_Germany/7960-kate.xml
tftp-server system:/its/germany/SCCP-dictionary.xml alias
German_Germany/SCCP-dictionary.xml
tftp-server system:/its/germany/7960-tones.xml alias Germany/7960-tones.xml

Step 5 Use the show ephone [mac-address] command to verify Cisco IP phone setup after phones have
registered with the Cisco CME router.
Router# show ephone

ephone-1 Mac:0003.E3E7.F627 TCP socket:[2] activeLine:0 REGISTERED

mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:20.0.0.2 51671 Telecaster 7940 keepalive 28 max_line 2
button 1: dn 1 number 4444 CM Fallback IDLE
ephone-2 Mac:0030.94C3.F43A TCP socket:[1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:20.0.0.3 50094 Telecaster 7960 keepalive 28 max_line 6
button 1: dn 3 number 5555 CM Fallback IDLE
ephone-3 Mac:0003.6B40.99DA TCP socket:[3] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:1.2.168.200 51879 Telecaster 7960 keepalive 28 max_line 6
button 1: dn 2 number 3333 CM Fallback IDLE

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Verifying Cisco CME Phone Configuration

Step 6 Use the show ephone-dn command to see settings related to an ephone-dn.
Router# show ephone-dn 7

50/0/7 INVALID
EFXS 50/0/7 Slot is 50, Sub-unit is 0, Port is 7
Type of VoicePort is EFXS
Operation State is UP
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Playout-delay Minimum mode is set to default, value 4 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Ringing Time Out is set to 8 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for US
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Voice card specific Info Follows:
Digit Duration Timing is set to 100 ms

Step 7 Use the show dialplan number command to display the number resolutions of a particular phone
number, which allows you to detect whether calls are going to unexpected destinations. This command
is useful for troubleshooting cases in which you dial a number but the expected phone does not ring.

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Configuration Examples for Setting Up Phones in a Cisco CME System

Configuration Examples for Setting Up Phones in a Cisco CME

System
This section provides an example of the required Cisco CME configuration, with some of the additional
options that are discussed in other chapters.
Router# show running-config

version 12.2
no parser cache
no service single-slot-reload-enable
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname Router
!
clock timezone pst -8
!
logging rate-limit console 10 except errors
!
!
ip dhcp pool mypool
network 10.0.0.0 255.255.0.0
option 150 ip 10.0.0.1
default-router 10.0.0.1
!
ip subnet-zero
!
!
no ip domain-lookup
!
no ip dhcp-client network-discovery
lcp max-session-starts 0
!
!
interface FastEthernet0/0
ip address 10.0.0.1 255.255.0.0
duplex auto
speed auto
!
interface Serial0/0
no ip address
shutdown
no fair-queue
clockrate 2000000
!
interface Serial0/1
no ip address
shutdown
clockrate 2000000
!
ip classless
ip route 0.0.0.0 0.0.0.0 10.0.0.10
!
!
tftp-server flash:P00303020214.bin
tftp-server flash:P00403020214.bin
!
snmp-server packetsize 4096
snmp-server manager

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call rsvp-sync
!
voice-port 1/0/0
connection plar opx 1100
!
voice-port 1/0/1
connection plar opx 1100
!
!
mgcp profile default
!
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 101 pots
destination-pattern 9.T
port 1/0/1
!
dial-peer voice 4000 voip
destination-pattern 4000
session target ipv4:10.0.0.25
codec g711ulaw
dtmf-relay h245-alphanumeric
!
!
telephony-service
load 7910 P00403020214
load 7960-7940 P00303020214
max-ephones 12
max-dn 48
ip source-address 10.0.0.1 port 2000 strict-match
max-conferences 4
keepalive 30
dialplan-pattern 1 40855511.. extension-length 4
create cnf-files
voicemail 4000
moh music-on-hold.au
transfer-system full-consult
!
!
!
ephone-template 1 softkey
softkey idle Redial Newcall
softkey connected Endcall Hold Trnsfer

ephone-template 2 softkey
softkey idle Redial Newcall
softkey seized Redial Endcall Pickup
softkey alerting Redial Endcall
softkey connected Endcall Hold Trnsfer

ephone-dn 1
number 1101
name user1
no huntstop
call-forward noan 4000 timeout 30
!
ephone-dn 2
number 1101
name user1
preference 1
call-forward busy 4000
call-forward noan 4000 timeout 30

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!
ephone-dn 3
number 1102
name user2
no huntstop
call-forward noan 4000 timeout 30
translate
!
ephone-dn 4
number 1102
name user2
preference 1
call-forward busy 4000
call-forward noan 4000 timeout 30
!
ephone-dn 5
number 1103
name user3
no huntstop
call-forward noan 4000 timeout 30
!
ephone-dn 6
number 1103
name user3
preference 1
call-forward busy 4000
call-forward noan 4000 timeout 30
!
ephone-dn 7
number 1104
name user4
no huntstop
call-forward noan 4000 timeout 30
!
ephone-dn 8
number 1104
name user4
preference 1
call-forward busy 4000
call-forward noan 4000 timeout 30
!
ephone-dn 9
number 1199
name intercom1199
intercom 1198 label Intercom
!
ephone-dn 10
number 1198
name intercom1198
intercom 1199 label Intercom
!
ephone-dn 11
number 1100
name Line1
no huntstop
call-forward noan 4000 timeout 60
!
ephone-dn 12
number 1100
name Line2
preference 1
call-forward noan 4000 timeout 60
call-forward busy 4000
!

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What to Do Next

ephone-dn 20
number 1111
name paging1111
paging ip 239.1.1.112 port 2000
!
ephone 1
mac-address 0003.6B54.BB15
button 1:1 2:2 3:11 4:12
paging-dn 20
ephone-template 1
!
ephone 2
mac-address 0003.6B09.63CF
button 1:3 2:4 3:11 4:12
paging-dn 20
ephone-template 1
!
ephone 3
mac-address 0003.6B54.C20F
button 1:5 2:6 3:11 4:12 5:9
paging-dn 20
ephone-template 2
!
ephone 4
mac-address 0003.6B40.892A
button 1:7 2:8 3:11 4:12 5:10
paging-dn 20
ephone-template 2
!
line con 0
line aux 0
line vty 0 4
login
!
ntp server 10.0.0.26
!
end

What to Do Next
After setting up basic Cisco CME phone configurations, you are ready to set up call transfer and
forwarding. Refer to the “Configuring Call Transfer and Call Forwarding” chapter.

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This chapter describes strategies that you can combine in various ways to meet the basic call handling
needs of the various call-processing systems that interwork within your VoIP network.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Information About Call Transfer and Call Forwarding, page 87
• Configuring Call Transfer and Call Forwarding, page 103
• Configuration Examples for Call Transfer and Call Forwarding, page 119

Information About Call Transfer and Call Forwarding

Note Cisco CallManager Express 3.2 (Cisco CME 3.2) and later versions provide full call-transfer and
call-forwarding interoperability with call processing systems on the network that support H.450.2,
H.450.3, and H.450.12 standards. For call processing systems that do not support H.450 standards,
Cisco CME 3.2 and later versions provide VoIP-to-VoIP hairpin call routing without requiring the use of
the special Tool Command Language (Tcl) script that was needed in earlier releases of Cisco CME.

This section contains the following sections:

• Background, page 88
• Strategies for Call Transfer and Call Forwarding, page 89
• Typical Network Scenarios for Call Transfer and Call Forwarding, page 98

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Background
In a mixed network that involves two or more types of call agents or managers, there can be
communication protocol discrepancies and dependencies, and therefore the opportunity for
interoperability errors. These discrepancies show up most often when a call is being transferred or
forwarded. The following recent Cisco CME releases have introduced features to address these
discrepancies and enable transparent transferring and forwarding of calls across VoIP networks.

Cisco IOS Telephony Services V2.1 (Cisco ITS V2.1), Cisco IOS Release 12.2(15)T
Cisco ITS V2.1 was the predecessor to Cisco CME 3.0. Prior to Cisco ITS V2.1, call transfer was
performed using a Cisco-proprietary method. Cisco ITS V2.1 introduced support for call transfer using
the ITU-T H.450.2 standard, which was configured using the transfer-system command and a Tcl
application script. Similarly, call forwarding using the H.450.3 standard was supported in
Cisco ITS V2.1 and was configured using the call forward pattern command and the same Tcl script.
To configure Cisco ITS V2.1 systems for call transfer and call forwarding, refer to the Cisco IOS
Telephony Services V2.1 guide.

Cisco CME 3.0, Cisco IOS Release 12.2(15)ZJ

Cisco CME 3.0 added support for IP phones to initiate call transfer using the H.450.2 protocol and call
forwarding using the H.450.3 protocol by using the default session application. The built-in H.450.2 and
H.450.3 support that is provided by the default session application applied to call transfers and forwards
initiated by IP phones, regardless of public switched telephone network (PSTN) interface type. In order
to use this feature, however, all endpoints in the VoIP network were required to support the H.450.2 and
H.450.3 standards. There was no fallback for other types of endpoints and no way to detect which
endpoints supported the standards. To configure Cisco CME 3.0 systems for call transfer and call
forwarding, refer to the Cisco CallManager Express 3.0 System Administrator Guide.

Cisco CME 3.0, Cisco IOS Releases 12.2(15)ZJ3 and 12.3(4)T

To handle calls to endpoints that did not recognize the H.450 standards, Cisco CME 3.0 in Cisco IOS
Releases 12.2(15)ZJ3 and 12.3(4)T introduced a new Tcl script that enabled hairpin VoIP-to-VoIP call
routing for transfers and forwards on the Cisco CME router. Hairpin call routing uses the Cisco CME
router to reoriginate a terminated call and route it as appropriate to complete a transfer or forward
generated by a phone or other application attached to the router. There was still no way to automatically
identify which endpoints supported H.450 standards, and hairpin call routing has the disadvantage of
using two calls’ worth of bandwidth for the duration of the transferred or forwarded call. To configure
Cisco CME 3.0 systems for call transfer and call forwarding, refer to the Cisco CallManager Express
3.0 System Administrator Guide.

Cisco CME 3.1, Cisco IOS Release 12.3(7)T

Cisco CME 3.1 introduces support for H.450.12 supplementary services, which provide dynamic
detection of H.450.2 and H.450.3 capabilities on a call-by-call basis. Calls that can be transferred or
forwarded to an H.450 endpoint are handled using H.450 standard protocols, while other calls are
handled by a fallback method. Note that although Cisco CME 3.0 and Cisco ITS V2.1 both support
H450.2 and H.450.3 standards, they do not support H.450.12. Therefore, Cisco CME 3.0 and
Cisco ITS V2.1 systems cannot automatically detect whether endpoints are capable of using the H.450.2
and H.450.3 standards.
Cisco CME 3.1 supports two fallback methods for calls to endpoints that do not support H.450 standards:
hairpin call routing and H.450 tandem gateways. Hairpin call routing was first supported in Cisco CME
3.0 using a Tcl script. In Cisco CME 3.1, hairpin call routing is enabled using Cisco IOS command-line
interface (CLI) commands rather than a Tcl script, thus simplifying its implementation.

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H.450 tandem gateways address the limitations of hairpin call routing. An H.450 tandem gateway is an
additional voice gateway that serves as a “front-end” for a call processor that does not support the H.450
standards, such as Cisco CallManager, Cisco BTS Softswitch (Cisco BTS), or Cisco PSTN Gateway
(Cisco PGW). Transferred and forwarded calls that are intended for non-H.450 endpoints are terminated
instead on the H.450 tandem gateway and reoriginated there for delivery to the non-H.450 endpoints.
The H.450 tandem gateway can also serve as a PSTN gateway. To configure Cisco CME 3.0 systems for
call transfer and call forwarding, refer to the Cisco CallManager Express 3.1 System Administrator
Guide.

Cisco CME 3.2, Cisco IOS Release 12.3(11)T

Transcoding between G.711 and G.729 is supported when one leg of a Voice over IP (VoIP)-to-VoIP
hairpin call uses G.711 and the other leg uses G.729. For information about transcoding, refer to the
“Transcoding Between G.729 and G.711” chapter of this guide.
H.323-to-SIP connections are allowed only for Cisco CME systems running Cisco Unity Express. For
more information, refer to Integrating Cisco CallManager Express with Cisco Unity Express.

Strategies for Call Transfer and Call Forwarding

As mentioned in the “Background” section on page 88, the risk of discrepancies among dissimilar call
processing systems exists in the handling of transferred and forwarded calls. This section describes the
following strategies for handling transferred and forwarded calls:
• H.450.2 and H.450.3 Support, page 89
• H.450.12 Support, page 92
• Hairpin Call Routing, page 93
• H.450 Tandem Gateways, page 95

H.450.2 and H.450.3 Support

Cisco CME 3.1 and later continues the support for the H.450.2 call-transfer standards and the H.450.3
call-forwarding standards that was introduced in Cisco ITS V2.1. Use of the H.450.2 and H.450.3
standards is the best way to handle call transfer and forwarding in a VoIP network and provides the
following benefits:
• The final call path from the transferred party to the transfer destination is optimal, with no
hairpinned routes or excessive use of resources.
• Call parameters (for example, codec) can be different for the different call legs.
• This solution is scalable.
• There is no limit to the number of times a call can be transferred.
Considerations for using the H.450.2 and H.450.3 standards include the following:
• Cisco IOS Release 12.2(15)T or a later release is required on all voice gateways in the network.
• Support of H.450.2 and H.450.3 is required on all voice gateways in the network. H.450.2 and
H.450.3 are used even when the transfer-to or forward-to target is on the same Cisco CME system
as the transferring party or the forwarding party, so the transferred party must also support H.450.2
and the forwarded party must also support H.450.3. The exception to this rule is calls that can be
reoriginated through hairpin call routing or through the use of an H.450 tandem gateway.

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• H.450 standards are not supported by Cisco CallManager, Cisco BTS, or Cisco PGW, although
hairpin call routing or an H.450 tandem gateway can be set up to handle calls to and from those types
of systems.
The following series of figures depicts a call being transferred using H.450.2 standards. Figure 11 on
page 90 shows that a call has been made from A to B. Figure 12 on page 90 shows B consulting with C
and putting A on hold. Figure 13 on page 91 shows that B has connected A and C, and Figure 14 on
page 91 shows A and C directly connected, with B no longer involved in the call.

Figure 11 Call Transfer Using H.450.2: A Calls B

H.323
V
Media Termination
Cisco CME 1 Point (MTP)
IP
Phone A Phone C
Cisco CME 2

95844
IP
Phone B

Figure 12 Call Transfer Using H.450.2: B Consults with C

H.323 V
Non-H.450
Cisco CME 1 gateway
IP
Phone A Cisco CME 2 Phone C

H.450.2 connection H.450.2 connection

95847

IP
Phone B

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Figure 13 Call Transfer Using H.450.2: B Transfers A to C

H.323 V
Non-H.450
Cisco CME 1 gateway
IP
Phone A Cisco CME 2 Phone C

95848
IP
Phone B

Figure 14 Call Transfer Using H.450.2: A and C Are Connected

H.323
V

IP
A C

95849
IP
B

Use H.450 standards when a network meets the following conditions:

• The router that you are configuring uses Cisco CME 3.0 and later, or Cisco ITS V2.1.
• For Cisco CME 3.0 or Cisco ITS V2.1 systems, all endpoints in the network must support H.450.2
and H.450.3 standards. For Cisco CME 3.1 systems, if some of the endpoints do not support H.450
standards (for example, Cisco CallManager, Cisco BTS, or Cisco PGW), you can use hairpin call
routing or an H.450 tandem gateway to handle transfers and forwards with those endpoints. Also,
either you must explicitly disable H.450.2 and H.450.3 on the dial peers that handle those calls or
you must enable H.450.12 capability to automatically detect the calls that support H.450.2 and
H.450.3 and those calls that do not.
Support for the H.450.2 standard and the H.450.3 standard is enabled by default and can be disabled
globally or for individual dial peers using the supplementary-service h450.2 and
supplementary-service h450.3 commands. Settings made for individual dial peers override the global
setting. For configuration information, see the “Enabling or Disabling H.450.2 and H.450.3
Capabilities” section on page 103.

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H.450.12 Support
Cisco CME 3.1 and later versions support the H.450.12 call capabilities standard, which provides a
means to advertise and dynamically discover H.450.2 and H.450.3 capabilities in voice gateway
endpoints. Once discovered, the calls associated with non-H.450 endpoints can be directed to use
non-H.450 methods for transfer and forwarding, such as hairpin call routing or H.450 tandem gateway.
You can have either of the following situations in your network:
• All gateway endpoints support H.450.2 and H.450.3 standards. In this situation, no special
configuration is required because support for H.450.2 and H.450.3 standards is enabled on the
Cisco CME 3.1 or later router by default. H.450.12 capability is disabled by default, but it is not
required because all calls can use H.450.2 and H.450.3 standards.
• Some gateway endpoints support H.450.2 and H.450.3 standards, and other gateway endpoints do
not. In this situation, you need to specify how non-H.450 calls are to be handled by choosing one of
the following options:
– Use the H.450.12 capability in Cisco CME 3.1 and later to dynamically determine, on a
call-by-call basis, whether each call has H.450.2 and H.450.3 support. To use this option, you
must explicitly enable H.450.12 capability in the router configuration because it is disabled by
default. If H.450.12 is enabled and a call is determined to have H.450 support, the call is
transferred using H.450.2 standards or forwarded using H.450.3 standards. If the call does not
have H.450 support, it can be handled by a VoIP-to-VoIP connection that you set up using dial
peers and the allow connections command. The VoIP-to-VoIP connection can be used for
hairpin call routing or routing to an H.450 tandem gateway. The following commands enable
H.450.12 capability and enable H.323 VoIP-to-VoIP connections:
Router(config)# voice service voip
Router(config-voice-service)# supplementary-service h450.12
Router(config-voice-service)# allow connections h323 to h323

Note that you can enable the supplementary-service h450.12 command in dial-peer
configuration mode to target specific dial peers if you do not want to enable the capability
globally.
– Your second choice for handling non-H.450 calls is to explicitly disable H.450.2 and H.450.3
capability on a global basis or by individual dial peer, which forces all calls to be handled by
the VoIP-to-VoIP connection that you have set up using dial peers and the allow connections
command. This connection can be used for hairpin call routing or routing to an H.450 tandem
gateway. The following commands globally disable H.450.2 and H.450.3 standards and enable
H.323 VoIP-to-VoIP connections:
Router(config)# voice service voip
Router(config-voice-service)# no supplementary-service h450.2
Router(config-voice-service)# no supplementary-service h450.3
Router(config-voice-service)# allow connections h323 to h323

Support for the H.450.12 standard is disabled by default. It can be enabled or disabled globally and can
be enabled for individual dial peers if it is disabled globally. Settings made for individual dial peers
override the global setting. For configuration information, see the “Enabling or Disabling H.450.12
Capabilities” section on page 108.

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Hairpin Call Routing

Hairpin call routing uses the VoIP-to-VoIP connection mechanisms that were introduced in
Cisco CME 3.1 to transfer and forward calls that cannot use H.450 standards. When a call that originally
terminated on a voice gateway is transferred or forwarded by a phone or other application attached to the
gateway, the gateway reoriginates the call and routes the call as appropriate, making a VoIP-to-VoIP, or
hairpin, connection. This approach avoids any protocol dependency on the far-end transferred-party
endpoint or transfer-destination endpoint. Hairpin routing of transferred and forwarded calls also causes
the generation of separate billing records for each call leg, so that the transferred or forwarded call leg
is typically billed to the user who initiates the transfer or forward.
For Cisco CME 3.2 and later versions, transcoding between G.711 and G.729 is supported when one leg
of a VoIP-to-VoIP hairpin call uses G.711 and the other leg uses G.729. For information about
transcoding, refer to the “Transcoding Between G.729 and G.711” chapter of this guide.
Hairpin call routing provides the following benefits:
• Call transfer and forwarding is provided to non-H.450 endpoints, such as Cisco CallManager,
Cisco BTS, or Cisco PGW.
• The network can also contain Cisco CME 3.0 or Cisco ITS 2.1 systems.
Hairpin call routing has the following disadvantages:
• End-to-end signaling and media delay are increased significantly.
• A single hairpinned call uses as much WAN bandwidth as two directly connected calls.
VoIP-to-VoIP hairpin connections can be made using dial peers if the allow-connections h323 to h323
command is enabled and at least one of the following is true:
• H.450.12 is used to detect calls on which H.450.2 or H.450.3 is not supported by the remote system.
• H.450.2 or H.450.3 is explicitly disabled.
• Cisco CME automatically detects that the remote system is a Cisco CallManager.
Figure 15 on page 93 shows a call that is made from A to B. Figure 16 on page 94 shows that B has
forwarded all calls to C. Figure 17 on page 94 shows that A and C are connected by an H.323 hairpin.

Figure 15 Hairpin with H.323: A Calls B

H.323
V
Media Termination
Cisco CME 1 Point (MTP)
IP
Phone A Phone C
Cisco CME 2
95844

IP
Phone B

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Figure 16 Hairpin with H.323: Call is Forwarded to C

H.323 V
Non-H.450
Cisco CME 1 gateway
IP
Phone A Cisco CME 2 Phone C

95845
IP Calls are forwarded
Phone B to phone C

Figure 17 Hairpin with H.323: A is Connected to C via B

H.323 V
Non-H.450
Cisco CME 1 gateway
IP
Phone A Cisco CME 2 Phone C

IP 95846

Phone B

Use hairpin call routing when a network meets the following three conditions:
• The router that you are configuring uses Cisco CME 3.1 or a later release.
• Some or all calls require VoIP-to-VoIP routing because they cannot use H.450 standards, which can
happen for any of the following reasons:
– H.450 capabilities have been explicitly disabled on the router.
– H.450 capabilities do not exist in the network.
– H.450 capabilities are supported on some endpoints and not supported on other endpoints,
including those handled by Cisco CallManager, Cisco BTS, and Cisco PGW. When some
endpoints support H.450 and others do not, you must enable H.450.12 capabilities on the router
to detect which endpoints are H.450-capable or designate some dial peers as H.450-capable. For
more information about enabling H.450.12 capabilities, see the “Enabling or Disabling
H.450.12 Capabilities” section on page 108.
• No voice gateway is available to act as an H.450 tandem gateway.
Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally using the allow
connections h323 to h323 command. For configuration information, see the “Enabling H.323-to-H.323
Connection Capabilities” section on page 111.

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Restrictions

Only one codec type is supported in the VoIP network at a time, and there are only two codec choices:
G.711 (A-law or mu-law) or G.729.

H.450 Tandem Gateways

An H.450 tandem voice gateway terminates and reoriginates calls to be transferred and forwarded to
endpoints that do not support H.450 standards, in a manner similar to hairpin call routing but without
the double WAN link traversal created by hairpin connections. An H.450 tandem gateway is a separate
router that provides a proxy, or “front-end,” for a system that does not support H.450 standards, such as
a Cisco CallManager system. Figure 18 on page 97 shows a tandem voice gateway that has been placed
between the central hub of the network of a CPE-based Cisco CME 3.1 or later network and a
Cisco CallManager network. This topology would work equally well with a Cisco BTS or Cisco PGW
in place of the Cisco CallManager. An H.450 tandem gateway can also work as a PSTN gateway for
remote Cisco CME systems and for Cisco CallManager (or other non-H.450 system). Use different
inbound dial peers to separate Cisco-CallManager-to-PSTN G.711 calls from tandem
gateway-to-Cisco-CME G.729 calls.
An H.450 tandem gateway is configured with a dial peer that points to the Cisco CallManager or other
system for which the H.450 tandem gateway is serving as a front end. The H.450 tandem voice gateway
is also configured with dial peers that point to all the Cisco CME systems in the private H.450 network.
In this way, Cisco CME and the Cisco CallManager are not directly linked to each other, but are instead
both linked to an H.450 tandem gateway that provides H.450 services to the non-H.450 platform.

Note An H.450 tandem gateway that is used in a network to support non-H.450-capable call processing
systems requires the Integrated Voice and Video Services feature license. This feature license, which was
introduced in March 2004, includes functionality for H.323 gatekeeper, IP-to-IP Gateway, and H.450
tandem gateway. With Cisco IOS Release 12.3(7)T, an H.323 gatekeeper feature license is required with a
JSX IOS image on the selected router. Please consult your Cisco CME SE regarding the required feature
license. With Cisco IOS Release 12.3(7)T, you cannot use Cisco CME and H.450 tandem gateway
functionality on the same router.

VoIP-to-VoIP connections can be made for an H.450 tandem gateway if the allow-connections h323 to
h323 command is enabled and one or more of the following is true:
• H.450.12 is used to dynamically detect calls on which H.450.2 or H.450.3 is not supported by the
remote VoIP system.
• H.450.2 or H.450.3 is explicitly disabled.
• Cisco CME 3.1 or later automatically detects that the remote system is a Cisco CallManager.

Note For Cisco CME 3.1 and earlier, the only type of VoIP-to-VoIP connection supported by Cisco CME is
H.323-to-H.323. For Cisco CME 3.2 and later versions, H.323-to-SIP connections are allowed only for
Cisco CME systems running Cisco Unity Express. For more information, refer to Integrating
Cisco CallManager Express with Cisco Unity Express.

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In the network topology in Figure 18 on page 97, the following events occur (refer to the event numbers
on the illustration):
1. A call is generated from extension 4002 on phone 2, which is connected to a Cisco CallManager.
The H.450 tandem gateway receives the H.323 call and, acting as the H.323 endpoint, the H.450
tandem gateway handles the call connection to a Cisco IP phone in a CPE-based Cisco CME 3.1 or
later network.
2. The call is received by extension 1001 on phone 3, which is connected to Cisco CME 1.
Extension 1001 performs a consultation transfer to extension 2001 on phone 5, which is connected
to Cisco CME 2.
3. When extension 1001 transfers the call, the H.450 tandem gateway receives an H.450.2 message
from extension 1001.
4. The H.450 tandem gateway terminates the call leg from extension 1001 and reoriginates a call leg
to extension 2001, which is connected to Cisco CME 2.
5. Extension 4002 is connected with extension 2001.

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Figure 18 H.450 Tandem Gateway

IP-to-IP
Gateway

Public VoIP Cisco CallManager

323 323
1
H.323 Connection
IP IP in ICT mode using slow start
Phone 1 Phone 2
4001 4002 Media Termination Point (MTP)
V

Private H.450 Network H.450

H.450Tandem
Tandem
Gateway
Gateway
3 PSTN
V

H.450.2 Message
Private VoIP Telephone

Cisco CME 1 Cisco CME 2

V V
2 5
4
IP IP IP IP
Phone 3 Phone 4 Phone 5 Phone 6
1001 1002 3001 3002

103360
Use this method when a network meets the following conditions:
• The router that you are configuring uses Cisco CME 3.1 or a later version.
• Some endpoints in the network are not H.450-capable, including those handled by
Cisco CallManager, Cisco BTS, and Cisco PGW.
Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally using the allow
connections h323 to h323 command. For more information, see the “Enabling H.323-to-H.323
Connection Capabilities” section on page 111.
Use dial peers to set up an H.450 tandem gateway. See the “Setting Up Dial Peers” section on page 118.

Restrictions

• Cisco CallManager must use a media termination point (MTP), intercluster trunk (ICT) mode, and
slow start.
• Codecs on all the VoIP dial peers of the H.450 tandem gateway must be the same.
• Only one codec type is supported in the VoIP network at a time, and there are only two codec
choices: G.711 (A-law or mu-law) or G.729.

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• Transcoding is not supported.

• Codec renegotiation is not supported. For example, if an H.323 call that uses a G.729 codec is
received by a Cisco CME system and is forwarded to a voice-mail system that requires a G.711
codec, the codec cannot be renegotiated from G.729 to G.711.

Typical Network Scenarios for Call Transfer and Call Forwarding

This section provides descriptions of the specific mixed-network scenarios that you might encounter
when configuring a router running Cisco CME 3.1 or a later version. Each description points to the
configuration instructions necessary to ensure call transfer and forwarding capabilities throughout the
network. The following descriptions are provided:
• Cisco CME 3.1 or Later and Cisco IOS Gateways, page 98
• Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, and Cisco IOS Gateways, page 99
• Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways, page 99
• Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, Non-H.450 Gateways, and Cisco IOS
Gateways, page 100
• Cisco CME 3.1 or Later, Cisco CallManager, and Cisco IOS Gateways, page 101
• Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, Cisco CallManager, and Cisco IOS
Gateways, page 102

Cisco CME 3.1 or Later and Cisco IOS Gateways

A network with Cisco CME 3.1 and Cisco IOS gateways can contain the following types of systems:
• Cisco CME 3.1 or later
• Cisco IOS gateways
In this scenario, all systems that might participate in calls that involve call transfer and call forwarding
are capable of supporting the H.450.2, H.450.3, and H.450.12 standards. This is the simplest
environment for operating the Cisco CME 3.1or later features.
Configuration for this type of network consists of the following general steps:
1. Set up call-transfer and call-forwarding parameters for transfers and forwards that are initiated on
this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations, forwarded
parties, and forwarding destinations are enabled by default). See the “Enabling or Disabling H.450.2
and H.450.3 Capabilities” section on page 103.
2. Enable H.450.12 globally to detect any calls on which H.450.2 and H.450.3 standards are not
supported. Although this step is optional, it is recommended. See the “Enabling or Disabling
H.450.12 Capabilities” section on page 108.
3. Optionally set up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route
calls that do not support H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323
Connection Capabilities” section on page 111.
4. Set up dial peers to manage call legs within the network. See the “Setting Up Dial Peers” section on
page 118.

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Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, and Cisco IOS Gateways
A network with Cisco CME 3.1 or later, Cisco CME 3.0 or Cisco ITS V2.1, and Cisco IOS gateways can
contain a combination of the following types of systems:
• Cisco CME 3.1 or later
• Cisco CME 3.0
• Cisco ITS V2.1
• Cisco IOS gateways
You might have this type of network while you are in the process of upgrading a Cisco CME 3.0 network
to Cisco CME 3.1 or later. Both Cisco CME 3.1 or later and Cisco CME 3.0 routers assume that H.450.2
and H.450.3 standards are to be used for all calls. Note that Cisco CME 3.0 and Cisco ITS 2.1 do not
support the H.450.12 standard.
Configuration for this type of network consists of the following general steps:
1. Set up call-transfer and call-forwarding parameters for transfers and forwards that are initiated on
this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations, forwarded
parties, and forwarding destinations are enabled by default). See the “Enabling or Disabling H.450.2
and H.450.3 Capabilities” section on page 103.
2. Enable H.450.12 in advertise-only mode on Cisco CME 3.1 or later systems. As each
Cisco CME 3.0 system is upgraded to Cisco CME 3.1 or later, enable H.450.12 in advertise-only
mode. Note that no checking for H.450.2 or H.450.3 support is done in advertise-only mode. When
all Cisco CME 3.0 systems in the network have been upgraded to Cisco CME 3.1 or later, remove
the advertise-only restriction. See the “Enabling or Disabling H.450.12 Capabilities” section on
page 108.
3. Optionally set up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route
calls that cannot use H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 111.
4. Set up dial peers to manage call legs within the network. See the “Setting Up Dial Peers” section on
page 118.

Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways
A network with Cisco CME 3.1 or later, non-H.450 gateways, and Cisco IOS gateways can contain a
combination of the following types of systems:
• Cisco CME 3.1 or later
• Gateways that do not support H.450.2 and H.450.3 standards, such as Cisco CallManager,
Cisco BTS, or Cisco PGW systems
• Cisco IOS gateways
In this type of network, the H.450.2 and H.450.3 services are provided only to calling endpoints that use
H.450.12 to explicitly indicate that they are capable of H.450.2 and H.450.3 operations. Because the
current releases of Cisco BTS and Cisco PGW do not support the H.450.12 standard, calls to and from
these systems that involve call transfer or forwarding are handled with H.323-to-H.323 hairpin call
routing.
Configuration for this type of network consists of the following general steps:
1. Set up call-transfer and call-forwarding parameters for transfers and forwards that are initiated on
this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations, forwarded
parties, and forwarding destinations are enabled by default). Optionally disable H.450.2 and

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H.450.3 capabilities on dial peers that point to non-H.450-capable systems such as

Cisco CallManager, Cisco BTS, or Cisco PGW. See the “Enabling or Disabling H.450.2 and
H.450.3 Capabilities” section on page 103.
2. Enable H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or for specific dial peers. See the “Enabling or Disabling H.450.12 Capabilities”
section on page 108.
3. Set up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route calls that
do not support H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 111.
4. Set up dial peers to manage call legs within the network. See the “Setting Up Dial Peers” section on
page 118.

Note If your network contains a Cisco CallManager, also see the instructions in the “Enabling Interworking
with Cisco CallManager” section on page 112.

Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, Non-H.450 Gateways, and Cisco IOS
Gateways
A network with Cisco CME 3.1 or later, Cisco CME 3.0 or Cisco ITS V2.1, non-H.450 gateways, and
Cisco IOS gateways can contain a combination of the following types of systems:
• Cisco CME 3.1 or later
• Cisco CME 3.0
• Cisco ITS V2.1
• Gateways that do not support H.450.2 and H.450.3 standards, such as Cisco CallManager,
Cisco BTS, or Cisco PGW systems
• Cisco IOS gateways
This type of network contains a mix of Cisco CME versions and at least one non-H.450 gateway.
Cisco CME 3.0 and Cisco ITS V2.1 systems do not have H.450.12 capabilities. The simplest
configuration approach for this type of network is to globally disable all H.450.2 and H.450.3 services
and force H.323-to-H.323 hairpin call routing for all transferred and forwarded calls. In this case, you
would enable H.450.12 detection capabilities globally. Alternatively, you could select to enable
H.450.12 capability for specific dial peers. In this case, you would not configure H.450.12 capability
globally; you would leave it in its default disabled state.
Configuration for this type of network consists of the following general steps:
1. Set up call-transfer and call-forwarding parameters for transfers and forwards that are initiated on
this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations, forwarded
parties, and forwarding destinations are enabled by default). See the “Enabling or Disabling H.450.2
and H.450.3 Capabilities” section on page 103.
2. Enable H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or on specific dial peers. See the “Enabling or Disabling H.450.12 Capabilities”
section on page 108.
3. Set up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls. See the “Enabling H.323-to-H.323 Connection Capabilities”
section on page 111.

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4. Set up dial peers to manage call legs within the network. See the “Setting Up Dial Peers” section on
page 118.

Note If your network contains a Cisco CallManager, also see the instructions in the “Enabling Interworking
with Cisco CallManager” section on page 112.

Cisco CME 3.1 or Later, Cisco CallManager, and Cisco IOS Gateways
A network with Cisco CME 3.1 or later, Cisco CallManager, and Cisco IOS gateways can contain a
combination of the following types of systems:
• Cisco CME 3.1 or later
• Cisco CallManager
• Cisco IOS gateways
This type of network contains only Cisco CME 3.1 or later systems and Cisco CallManager systems. The
Cisco CME 3.1 or later release supports automatic detection of calls to and from Cisco CallManager
using proprietary signaling elements that are included with the standard H.323 message exchanges. The
Cisco CME 3.1 or later system uses these detection results to determine the H.450.2 and H.450.3
capabilities of calls rather than using H.450.12 supplementary services capabilities exchange, which
Cisco CallManager does not support. If a call is detected to be coming from or going to a Cisco
CallManager endpoint, the call is treated as a non-H.450 call. All other calls in this type of network are
treated as though they support H.450 standards. Therefore, this type of network should contain only
Cisco CME 3.1 or later and Cisco CallManager call-processing systems.
Configuration for this type of network consists of the following general steps:
1. Set up call-transfer and call-forwarding parameters for transfers and forwards that are initiated on
this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations, forwarded
parties, and forwarding destinations are enabled by default). See the “Enabling or Disabling H.450.2
and H.450.3 Capabilities” section on page 103.
2. Enable H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or on specific dial peers. See the “Enabling or Disabling H.450.12 Capabilities”
section on page 108.
3. Set up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls that are detected as being to or from Cisco CallManager. See the
“Enabling H.323-to-H.323 Connection Capabilities” section on page 111.
4. Set up specific parameters for Cisco CallManager. See the instructions in the “Enabling
Interworking with Cisco CallManager” section on page 112.
5. Set up dial peers to manage call legs within the network. See the “Setting Up Dial Peers” section on
page 118.

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Cisco CME 3.1 or Later, Cisco CME 3.0 or Cisco ITS V2.1, Cisco CallManager, and Cisco IOS
Gateways
A network with Cisco CME 3.1 or later, Cisco CME 3.0 or Cisco ITS V2.1, Cisco CallManager, and
Cisco IOS gateways cancontain a combination of the following types of systems:
• Cisco CME 3.1 or later
• Cisco CME 3.0
• Cisco ITS V2.1
• Cisco CallManager
• Cisco IOS gateways
Calls between the Cisco CallManager and the older Cisco CME 3.0 or Cisco ITS V2.1 networks need
special consideration. Because Cisco CME 3.0 and Cisco ITS V2.1 systems do not support automatic
Cisco CallManager detection and also do not natively support H.323-to-H.323 call routing, alternative
arrangements are required for these systems.
To configure call transfer and forwarding on the Cisco CME 3.0 router, you can select from the
following three options:
• Use a Tcl script to handle call transfer and forwarding by invoking Tcl-script-based H.323-to-H.323
hairpin call routing (app-h450-transfer.2.0.0.9.tcl or a later version). Enable this script on all VoIP
dial peers and also under telephony-service mode, and set the local-hairpin script parameter to 1.
Refer to the configuration instructions in the “Configuring Call Transfer” chapter of the Cisco
CallManager Express 3.0 System Administrator Guide.
• Use a loopback-dn mechanism. Refer to the instructions in the “Loopback Call Routing” appendix
of the Cisco CallManager Express 3.0 System Administrator Guide.
• Configure a loopback call path using router physical voice ports.
All three options force use of H.323-to-H.323 hairpin call routing for all calls irrespective of whether
the call is from a Cisco CallManager or other H.323 endpoint (including Cisco CME 3.1 or later).
In addition to the special considerations above, configuration of the Cisco CME 3.1 or later router for
this type of network consists of the following general steps:
1. Set up call-transfer and call-forwarding parameters for transfers and forwards that are initiated on
this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations, forwarded
parties, and forwarding destinations are enabled by default). See the “Enabling or Disabling H.450.2
and H.450.3 Capabilities” section on page 103.
2. Leave H.450.12 capability in its default disabled state. For more information, see the “Enabling or
Disabling H.450.12 Capabilities” section on page 108.
3. Set up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls that are detected as being to or from Cisco CallManager. See the
“Enabling H.323-to-H.323 Connection Capabilities” section on page 111.
4. Set up specific parameters for Cisco CallManager. See the instructions in the “Enabling
Interworking with Cisco CallManager” section on page 112.
5. Set up dial peers to manage call legs within the network. See the “Setting Up Dial Peers” section on
page 118.

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Configuring Call Transfer and Call Forwarding

Note Some of the commands in this section were introduced in Cisco CME 3.1 or later. To configure H.450
call transfer and forwarding on Cisco CME 3.0 systems, refer to the instructions in the
Cisco CallManager Express 3.0 System Administrator Guide. For Cisco ITS V2.1 systems, refer to the
Cisco IOS Telephony Services V2.1 guide.

Note Cisco recommends that customers using Cisco CME 3.0 and later versions should configure the
transfer-system command using the full-consult or full-blind keyword, which allows IP phones to
perform consultative or blind transfers to local phones and phones across a WAN. Note that the default
for the transfer-system command is the blind keyword, so the transfer-system command must be
explicitly configured for the recommended full-consult or full-blind setting.
Customers running Cisco IOS Telephony Services (Cisco ITS) 2.1 or an earlier version should use the
local-consult or blind keyword with the transfer-system command to enable the Cisco proprietary
transfer method.
Customers using Cisco ITS 2.1 can use the full-consult or full-blind keyword to enable H.450.2 call
transfer by also configuring the router with a Tcl script that is contained in the file called
app-h450-transfer.x.x.x.x.zip. This file is posted on the Cisco CME software download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp. For configuration information, see the Cisco IOS
Telephony Services V2.1 guide.

The following configuration instructions are contained in this section:

• Enabling or Disabling H.450.2 and H.450.3 Capabilities, page 103
• Enabling or Disabling H.450.12 Capabilities, page 108
• Enabling H.323-to-H.323 Connection Capabilities, page 111
• Enabling Interworking with Cisco CallManager, page 112
• Setting Up Dial Peers, page 118
• Verifying and Troubleshooting, page 118

Enabling or Disabling H.450.2 and H.450.3 Capabilities

H.450.2 is a standard protocol for exchanging call-transfer information across a network, whereas
H.450.3 is a standard protocol for exchanging call-forwarding information across a network. The
H.450.2 and H.450.3 standards are supported by Cisco CME 3.1 or later, Cisco CME 3.0, and
Cisco ITS V2.1. The H.450.2 and H.450.3 standards are not supported by Cisco CallManager,
Cisco BTS, or Cisco PGW.
H.450.2 and H.450.3 capabilities are enabled by default for transferred or forwarded parties and
transfer-destination or forward-destination parties. To enable H.450 call transfers and forwards for
transferring or forwarding parties (that is, to allow transfers and forwards to be initiated from a
Cisco CME 3.1 or later system), you must use the transfer-system, transfer-pattern, and call-forward
pattern commands on the Cisco CME 3.1 or later system.

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The remaining commands in this task (supplementary-service h450.2 and supplementary-service

h450.3) enable or disable H.450.2 and H.450.3 capabilities for transferred or forwarded parties and
transfer-destination or forward-destination parties. Because these capabilities are enabled by default,
these commands are normally required only if you want to explicitly disable H.450.2 or H.450.3
capabilities, either globally or on specific dial peers.

SUMMARY STEPS

1. telephony-service
2. transfer-system {blind | full-blind | full-consult | local-consult}
3. transfer-pattern transfer-pattern [blind]
4. call-forward pattern pattern
5. exit
6. voice service voip
7. supplementary-service h450.2
8. supplementary-service h450.3
9. exit
10. dial-peer voice tag voip
11. supplementary-service h450.2
12. supplementary-service h450.3
13. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 2 transfer-system {blind | full-blind | Defines the call transfer method to allow call transfer with
full-consult | local-consult} consultation for all lines served by the router.
Cisco CME 3.0 and later versions should use only the full-blind
Example: or full-consult keyword. Cisco ITS 2.1 and earlier versions
Router(config-telephony)# transfer-system should use the local-consult or blind keyword. (Cisco ITS 2.1
full-consult
can use the full-blind or full-consult keyword by also using the
Tcl script in the file called app-h450-transfer.x.x.x.x.zip.)
For SIP networks, use only the full-blind or full-consult
keyword. For more information about SIP, refer to “Appendix B:
Providing Cisco CME Support for SIP” in this guide and to the
Cisco IOS SIP Configuration Guide.
• blind—Calls are transferred without consultation with a
single phone line using the Cisco-proprietary method. This is
the default if this command is not used.
• full-blind—Calls are transferred without consultation using
H.450.2 standard methods.
• full-consult—Calls are transferred with consultation using
H.450.2 standard methods and a second phone line if
available. The calls fall back to full-blind if the second line is
unavailable.
• local-consult—Calls are transferred with local consultation
using a second phone line if available. The calls fall back to
blind for nonlocal consultation or nonlocal transfer target.
Step 3 transfer-pattern transfer-pattern [blind] Allows transfer of telephone calls by Cisco IP phones to specified
phone number patterns. If no transfer pattern is set, the default is
that transfers are permitted only to other local IP phones.
Example:
Router(config-telephony)# • transfer-pattern—String of digits for permitted call transfers.
transfer-pattern .T Wildcards are allowed. A pattern of .T transfers all calling
parties using the H.450.2 standard.
• blind—(Optional) When H.450.2 consultative call transfer is
configured, forces transfers that match the pattern specified
in this command to be executed as blind transfers. Overrides
settings made using the transfer-system and transfer-mode
commands.
Note When defining transfers to nonlocal numbers, it is
important to note that transfer-pattern digit matching is
performed before translation-rule operations. Therefore,
you should specify in this command the digits actually
entered by phone users before they are translated. For
more information, see the “Translation Rules” section in
the “Setting Up Phones in a Cisco CME System” chapter.

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Command or Action Purpose

Step 4 call-forward pattern pattern Specifies the H.450.3 standard for call forwarding. Calling-party
numbers that do not match the patterns defined with this
command are forwarded using Cisco-proprietary call forwarding
Example:
Router(config-telephony)# call-forward
for backward compatibility (as described in the “Configuring Call
pattern .T Forwarding” chapter in the Cisco IOS Telephony Services V2.1
guide).
• pattern—Digits to match for call forwarding using the
H.450.3 standard. If an incoming calling-party number
matches the pattern, it can be forwarded using the H.450.3
standard. A pattern of .T forwards all calling parties using the
H.450.3 standard.
Note When defining forwards to nonlocal numbers, it is
important to note that pattern digit matching is performed
before translation-rule operations. Therefore, you should
specify in this command the digits actually entered by
phone users before they are translated. For more
information, see the “Translation Rules” section in the
“Setting Up Phones in a Cisco CME System” chapter.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 voice service voip (Optional) Enters voice-service configuration mode to establish
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 7 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services capabilities
exchange globally. This is the default. Use the no form of this
command to disable H.450.2 capabilities globally. This command
Example:
Router(conf-voi-serv)#
is also used in dial-peer configuration mode to affect a single dial
supplementary-service h450.2 peer.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for the
dial peer.

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Command or Action Purpose

Step 8 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services capabilities
exchange globally. This is the default. Use the no form of this
command to disable H.450.3 capabilities globally. This command
Example:
Router(conf-voi-serv)#
is also used in dial-peer configuration mode to affect a single dial
supplementary-service h450.3 peer.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for the
dial peer.
Step 9 exit (Optional) Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit
Step 10 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 1 voip
Step 11 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services capabilities
exchange for an individual dial peer. This is the default. This
command is also used in voice-service configuration mode to
Example:
Router(config-dial-peer)# no
enable H.450.2 services globally.
supplementary-service h450.2 • If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for the
dial peer.
Step 12 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services capabilities
exchange for an individual dial peer. This is the default. This
command is also used in voice-service configuration mode to
Example:
Router(config-dial-peer)# no
enable H.450.3 services globally.
supplementary-service h450.3 • If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for the
dial peer.

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Command or Action Purpose

Step 13 exit (Optional) Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Example

The following example sets all transfers and forwards that are initiated by a Cisco CME 3.1 or later
system to use the H.450 standards, globally enables H.450.2 and H.450.3 capabilities, and disables those
capabilities for dial peer 37. The supplementary-service commands under voice-service configuration
mode are not necessary because these values are the default, but they are shown here for illustration.
telephony-service
transfer-system full-consult
transfer-pattern .T
call-forward pattern .T
!
voice service voip
supplementary-service h450.2
supplementary-service h450.3
!
dial-peer voice 37 voip
destination-pattern 555....
session target ipv4:10.5.6.7
no supplementary-service h450.2
no supplementary-service h450.3

What to Do Next

If you are using H.450.12 capabilities in your network, see the instructions in the “Enabling or Disabling
H.450.12 Capabilities” section on page 108.
If you are configuring hairpin call routing or routing to an H.450 tandem gateway, see the instructions
in the “Enabling H.323-to-H.323 Connection Capabilities” section on page 111.
If you are setting up a network that includes a Cisco CallManager, see the instructions in the “Enabling
Interworking with Cisco CallManager” section on page 112.
Set up dial peers using the instructions in the Dial Peer Configuration on Voice Gateway Routers guide.

Enabling or Disabling H.450.12 Capabilities

The H.450.12 call capabilities standard provides a means to advertise and discover H.450.2 and H.450.3
capabilities in voice gateway endpoints on a call-by-call basis. When H.450.12 is enabled, H.450.2 and
H.450.3 services are disabled for call transfers and call forwards unless a positive H.450.12 indication
is received from all the other VoIP endpoints involved in the call. If positive H.450.12 indications are
received, the router uses the H.450.2 standard for call transfers and the H.450.3 standard for call
forwarding. If a positive H.450.12 indication is not received, the router uses the alternative method that
you have configured for call transfers and forwards, either hairpin call routing or an H.450 tandem
gateway.
H.450.12 capabilities are disabled by default to minimize the risk of compatibility issues with other
types of H.323 systems. This optional task allows you to enable H.450.12 capabilities globally or by
individual dial peer.

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Note that Cisco CME 3.0 does not provide H.450.12 indications for calls even though it supports the
H.450.2 and H.450.3 standards. The supplementary-service h450.12 command with the advertise-only
keyword is intended for use on Cisco CME 3.1 or later systems that are mixed in a network with
Cisco CME 3.0 systems. This scenario is usually found when you are upgrading a network from
Cisco CME 3.0 systems to Cisco CME 3.1 or later. When you use the advertise-only keyword, the
Cisco CME 3.1 or later router sends out H.450.12 indications for the benefit of remote VoIP endpoints,
but does not require H.450.12 responses and has H.450.2 and H.450.3 enabled for all calls (the default).
When in advertise-only mode, Cisco CME 3.1 or later is still able to automatically detect
Cisco CallManager systems.

SUMMARY STEPS

1. voice service voip

2. supplementary-service h450.12 [advertise-only]
3. exit
4. dial-peer voice tag voip
5. supplementary-service h450.12
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice service voip (Optional) Enters voice service configuration mode to establish
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 2 supplementary-service h450.12 (Optional) Enables H.450.12 supplementary services capabilities
[advertise-only] exchange globally. Use this command for call-by-call detection of
H.450 capabilities when some endpoints in your mixed network
Example: are H.450-capable and other endpoints are not. This command is
Router(conf-voi-serv)# disabled by default.
supplementary-service h450.12
• advertise-only—(Optional) Advertises H.450 capabilities to
the remote end but does not require H.450.12 responses. Use
this keyword when you have only Cisco CME 3.0 systems in
your network in addition to Cisco CME 3.1or later systems.
This command is also used in dial-peer configuration mode to
affect an individual dial peer.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer.
• If this command is enabled globally and disabled on a dial
peer, the functionality is enabled for the dial peer.
• If this command is disabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer.
• If this command is disabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer. This is the
default.

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Command or Action Purpose

Step 3 exit (Optional) Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit
Step 4 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode. Use this
command to set up individual dial peers to override global
settings.
Example:
Router(config)# dial-peer voice 1 voip
Step 5 supplementary-service h450.12 (Optional) Enables H.450.12 supplementary services capabilities
exchange for an individual dial peer. This command is disabled by
default.
Example:
Router(config-dial-peer)# This command is also used in voice-service configuration mode
supplementary-service h450.12 to enable H.450.12 services globally.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer.
• If this command is enabled globally and disabled on a dial
peer, the functionality is enabled for the dial peer.
• If this command is disabled globally and enabled on a dial
peer, the functionality is enabled for the dial peer.
• If this command is disabled globally and disabled on a dial
peer, the functionality is disabled for the dial peer. This is the
default.
Step 6 exit (Optional) Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Example

The following example globally disables H.450.12 capabilities and then enables them only on dial
peer 24.
voice service voip
no supplementary-service h450.12
!
dial-peer voice 24 voip
destination-pattern 555....
session target ipv4:10.5.6.7
supplementary-service h450.12

What to Do Next

If you are configuring hairpin call routing or routing to an H.450 tandem gateway, see the instructions
in the “Enabling H.323-to-H.323 Connection Capabilities” section on page 111.
If you are setting up a network that includes a Cisco CallManager, see the instructions in the “Enabling
Interworking with Cisco CallManager” section on page 112.
Set up dial peers using the instructions in the Dial Peer Configuration on Voice Gateway Routers guide.

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Enabling H.323-to-H.323 Connection Capabilities

VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls
over the VoIP network. VoIP-to-VoIP connections are used for hairpin call routing and for H.450 tandem
gateways. The only type of VoIP-to-VoIP connection that is supported by Cisco CME 3.1 or later is
H.323-to-H.323 connection.
VoIP-to-VoIP connections are disabled on the router by default, and they must be explicitly enabled to
make use of hairpin call routing or an H.450 tandem gateway. In addition, you must configure a
mechanism to direct transferred or forwarded calls to the hairpin or the H.450 tandem gateway, using
one of the following methods:
• Enable H.450.12 capabilities globally or on the routes that your transfers and forwards take. See the
“Enabling or Disabling H.450.12 Capabilities” section on page 108.
• Explicitly disable H.450.2 and H.450.3 capabilities globally or on the routes that your transfers and
forwards take. See the “Enabling or Disabling H.450.2 and H.450.3 Capabilities” section on
page 103.

Restrictions
H.323-to-SIP hairpin call routing supported only for Cisco Unity Express. For more information, refer
to Integrating Cisco CallManager Express with Cisco Unity Express.

SUMMARY STEPS

1. voice service voip

2. allow-connections h323 to h323
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice service voip Enters voice service configuration mode to establish global call
transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 2 allow-connections h323 to h323 Enables VoIP-to-VoIP call connections. Use the no form of the
command to disable VoIP-to-VoIP connections; this is the default.
Example:
Router(conf-voi-serv)# allow-connections
h323 to h323
Step 3 exit Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit

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Example

The following example globally enables H.323-to-H.323 connections:

voice service voip
allow-connections h323 to h323

What to Do Next

If you are setting up a network that includes a Cisco CallManager, see the instructions in the “Enabling
Interworking with Cisco CallManager” section on page 112.
Set up dial peers to establish hairpin call routing or routing to an H.450 tandem gateway using the
instructions in the Dial Peer Configuration on Voice Gateway Routers guide.

Enabling Interworking with Cisco CallManager

When Cisco CME 3.1 or later and Cisco CallManager are used in the same network, some additional
configuration is necessary, as described in the following sections:
• Configuring Cisco CME 3.1 or Later to Interwork with Cisco CallManager, page 113
• Configuring Cisco CallManager to Interwork with Cisco CME 3.1 or Later, page 116
Figure 19 shows a network containing Cisco CME and Cisco CallManager systems.

Figure 19 Network with Cisco CME and Cisco CallManager Systems

Cisco CallManager 1 Cisco CallManager 2

IP IP
Cisco CallManager 3
Phone 1 Phone 2
4001 4002 H.323 Connection
in ICT mode using slow start

Media Termination Point (MTP)

V

Cisco CallManager Network VoIP

Cisco CME Network

PSTN

Cisco CME 1 Cisco CME 2 Cisco CME 3

V V V
Telephone

IP IP IP IP IP IP
103359

Phone 3 Phone 4 Phone 5 Phone 6 Phone 7 Phone 8

1001 1002 2001 2002 3001 3002

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Configuring Cisco CME 3.1 or Later to Interwork with Cisco CallManager

All of the Cisco IOS commands in this section are optional because they are set by default to work with
Cisco CallManager. They are included here only to explain how to implement optional capabilities or
return nondefault settings to their defaults.

SUMMARY STEPS

1. voice service voip

2. h323
3. telephony-service ccm-compatible
4. h225 h245-address on-connect
5. exit
6. supplementary-service h225-notify cid-update
7. exit
8. voice class h323 tag
9. telephony-service ccm-compatible
10. h225 h245-address on-connect
11. exit
12. dial-peer voice tag voip
13. supplementary-service h225-notify cid-update
14. voice-class h323 tag
15. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice service voip Enters voice-service configuration mode to establish global
parameters.
Example:
Router(config)# voice service voip
Step 2 h323 Enters H.323 voice-service configuration mode.

Example:
Router(conf-voi-serv)# h323
Step 3 telephony-service ccm-compatible (Optional) Globally enables a Cisco CME 3.1 or later system to
detect a Cisco CallManager and exchange calls with it. This is the
default.
Example:
Router(conf-serv-h323)# telephony-service • Use the no form of the command to disable
ccm-compatible Cisco CallManager detection and exchange. Using the no
form of the command is not recommended.
• Using this command in an H.323 voice class definition allows
you to specify this behavior for an individual dial peer.

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Command or Action Purpose

Step 4 h225 h245-address on-connect (Optional) Globally enables a delay for the H.225 message
exchange of an H.245 transport address until a call is connected.
The delay allows the Cisco CallManager to generate local
Example:
Router(conf-serv-h323)# h225 h245-address
ringback for calls to Cisco CME phones. This is the default.
on-connect • The no form of this command disables the delay. Using the
no form of the command is not recommended.
• Using this command in an H.323 voice class definition allows
you to specify this behavior for an individual dial peer.
Step 5 exit Exits H.323 voice-service configuration mode.

Example:
Router(conf-serv-h323)# exit
Step 6 supplementary-service h225-notify (Optional) Globally enables H.225 messages with caller-ID
cid-update updates to be sent to Cisco CallManager. This is the default.
• The no form of the command disables caller-ID update.
Example: Using the no form of the command is not recommended.
Router(conf-voi-serv)#
supplementary-service h225-notify This command is also used in dial-peer configuration mode to
cid-update affect a single dial peer.
• If this command is enabled globally and enabled on a dial
peer, the functionality is enabled for that dial peer. This is the
default.
• If this command is enabled globally and disabled on a dial
peer, the functionality is disabled for that dial peer.
• If this command is disabled globally and either enabled or
disabled on a dial peer, the functionality is disabled for that
dial peer.
Step 7 exit Exits voice-service configuration mode.

Example:
Router(config-voice-service)# exit
Step 8 voice class h323 tag (Optional) Creates a voice class that contains commands to be
applied to one or more dial peers.
Example:
Router(config)# voice class h323 48
Step 9 telephony-service ccm-compatible (Optional) When this voice class is applied to a dial peer, enables
the dial peer to exchange calls with a Cisco CallManager system.
This is the default.
Example:
Router(config-voice-class)# • The no form of the command disables call exchange with
telephony-service ccm-compatible Cisco CallManager. Using the no form of the command is not
recommended.

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Command or Action Purpose

Step 10 h225 h245-address on-connect (Optional) When this voice class is applied to a dial peer, enables
the calls that use this dial peer to delay the exchange of H.225
messages that contain the H.245 transport address until calls are
Example:
Router(config-voice-class)# h225
connected. The delay allows the playing of local ringback for
h245-address on-connect calls from Cisco CallManager. This is the default.
• The no form of this command disables the delay. Using the
no form of the command is not recommended.
Step 11 exit Exits voice-class configuration mode.

Example:
Router(config-voice-class)# exit
Step 12 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode to set parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 28 voip
Step 13 supplementary-service h225-notify (Optional) Enables H.225 messages with caller-ID updates to
cid-update Cisco CallManager for a specific dial peer. This is the default.
• The no form of the command disables caller-ID updates.
Example: Using the no form of the command is not recommended.
Router(config-dial-peer)# no
supplementary-service h225-notify
cid-update
Step 14 voice-class h323 tag (Optional) Applies the previously defined voice class with the
specified tag number to this dial peer.
Example:
Router(config-dial-peer)# voice-class
h323 48
Step 15 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

What to Do Next

Set up Cisco CallManager using the steps in the “Configuring Cisco CallManager to Interwork with
Cisco CME 3.1 or Later” section on page 116.

Configuring Cisco CallManager to Interwork with Cisco CME 3.1 or Later

Set up a Cisco CallManager that is intended to interwork with a Cisco CME system using the following
special steps in addition to the normal Cisco CallManager configuration.

SUMMARY STEPS

1. Set Cisco CallManager service parameters.

2. Configure the Cisco CME 3.1or later system as an ICT in the Cisco CallManager network.
3. Ensure that the Cisco CallManager network uses an MTP.

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DETAILED STEPS

Step 1 Set Cisco CallManager service parameters. From Cisco CallManager Administration, choose Service
Parameters. Choose the Cisco CallManager service, and make the following settings:
• Set the H323 FastStart Inbound service parameter to False, as shown in Figure 20.
• Set the Send H225 User Info Message service parameter to H225 Info for Ring Back, as shown in
Figure 21.
For details, refer to the “Service Parameters Configuration” chapter of the Cisco CallManager
Administration Guide, Release 3.3(3).

Figure 20 Setting the H.323 Fast Start Inbound Parameter

Figure 21 Setting the Send H225 User Info Message

Step 2 Configure the Cisco CME 3.1 system as an ICT in the Cisco CallManager network, as shown in
Figure 22 on page 117. For information about different intercluster trunk types and configuration
instructions, refer to the Cisco CallManager Administration Guide, Release 3.3(3).

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Figure 22 Configuring an ICT

Step 3 Ensure that the Cisco CallManager network uses an MTP. The MTP is required to provide DSP resources
for transcoding and for sending and receiving G.729 calls to the Cisco CME 3.1 or later system. All
media streams between Cisco CallManager and Cisco CME 3.1 or later must pass through the MTP
because Cisco CME 3.1 does not support transcoding. For more information, refer to the
Cisco CallManager System Guide, Release 3.3(3).

For more information about Cisco CallManager, refer to the Cisco CallManager Administration, System,
and Features and Services Guides, Release 3.3(3).

What to Do Next

Set up dial peers to establish routing using the instructions in the Dial Peer Configuration on Voice
Gateway Routers guide.

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Setting Up Dial Peers

Dial peers describe the virtual interfaces to or from which a call is established. All voice technologies
use dial peers to define the characteristics associated with a call leg. Attributes applied to a call leg
include specific quality of service (QoS) features, compression/decompression (codec), voice activity
detection (VAD), and fax rate. Dial peers are also used to establish the routing paths in your network,
including special routing paths such as hairpins and H.450 tandem gateways. Set up dial peers using the
instructions in the Dial Peer Configuration on Voice Gateway Routers guide.

Example

The following example shows dial peer 1001, which points to a Cisco CallManager connection, and dial
peer 1002, which is on the Cisco CME 3.1 or later router itself:
dial-peer voice 1001 voip
description points-to-CCM
destination-pattern 1.T
codec g711ulaw
session target ipv4:172.26.82.10
!
dial-peer voice 1002 voip
description points to router
destination-pattern 4...
codec g711ulaw
session target ipv4:172.25.82.2

What to Do Next

After setting up dial peers and using the other appropriate commands in this chapter, you should be able
to transfer and forward calls across your mixed network. Verify and troubleshoot the configuration as
needed.

Verifying and Troubleshooting

To verify the configuration, use the show running-config command. Output samples are located in the
“Configuration Examples for Call Transfer and Call Forwarding” section on page 119.
If you encounter lack of ringback on direct calls from a Cisco CallManager phone to an IP phone on a
Cisco CME system, check the show running-config command output to make sure that the following
two commands do not appear: no h225 h245-address on-connect and no telephony-service
ccm-compatible. Both of these commands should be enabled, which is their default state.
The debug h225 asn1 command can be used to look at the H.323 messages that are being sent from the
Cisco CME system to the Cisco CallManager system to see if the H.245 address is being sent too early.
The following show commands are useful to examine network operation:
• show voip rtp connections detail
For calls that are routed using VoIP-to-VoIP connections, this command displays the call
identification number, IP addresses, and port numbers involved for all VoIP call legs. This command
includes VoIP-to-POTS and VoIP-to-VoIP call legs. The following is sample output for this
command:

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Router# show voip rtp connections detail

VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 7 8 16586 22346 172.27.82.2 172.29.82.2
2 8 7 17010 16590 172.27.82.2 200.1.1.29
Found 2 active RTP connections

• show call prompt-mem-usage detail

This command shows information on ringback tone generation that uses the interactive voice
response (IVR) prompt playback mechanism. This ringback is needed for hairpin transfers that are
committed during the alerting-of-the-transfer-destination phase of the call and for calls to
destinations that do not provide in-band ringback tone, such as IP phones (FXS analog ports do
provide in-band ringback tone). Ringback tone is played to the transferred party by the Cisco CME
system that performs the transfer (the system attached to the transferring party). The system
automatically generates tone prompts as needed based on the network-locale setting for the
Cisco CME system.
If you are not getting ringback tone when you should, use the show call prompt-mem-usage
command to make sure that the correct prompt is loaded and playing. The following sample output
indicates that a prompt is playing (“Number of prompts playing”) and indicates the country code
used for the prompt (GB for Great Britain) and the codec.
Router# show call prompt-mem-usage detail

Prompt memory usage:

config'd wait active free mc total ms total
file(s) 0200 0001 -001 00200 00001 00002
memory 02097152 00003000 00000000 02094152 00003000
Prompt load counts: (counters reset 0)
success 0(1st try) 0(2nd try), failure 0
Other mem block usage:
mcDynamic mcReader
gauge 00001 00001
Number of prompts playing: 1
Number of start delays : 0
MCs in the ivr MC sharing table
===============================
Media Content: NoPrompt (0x83C64554)
URL:
cid=0, status=MC_READY size=24184 coding=g711ulaw refCount=0
Media Content: tone://GB_g729_tone_ringback (0x83266EC8)
URL: tone://GB_g729_tone_ringback

Configuration Examples for Call Transfer and Call Forwarding

The following configuration examples are included in this section:
• Cisco CME 3.1 or Later and Cisco CallManager in the Same Network: Example, page 120
• H.450 Tandem Gateway Working with Cisco CME 3.1 or Later and Cisco CallManager: Example,
page 122

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Cisco CME 3.1 or Later and Cisco CallManager in the Same Network: Example
The following example shows a running configuration for a Cisco CME 3.1 or later router that has a
Cisco CallManager in its network.
Router# show running-config

version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
enable password cisco
!
aaa new-model
!
!
aaa session-id common
no ip subnet-zero
!
ip dhcp pool phone1
host 172.24.82.3 255.255.255.0
client-identifier 0100.07eb.4629.9e
default-router 172.24.82.2
option 150 ip 172.24.82.2
!
ip dhcp pool phone2
host 172.24.82.4 255.255.255.0
client-identifier 0100.0b5f.f932.58
default-router 172.24.82.2
option 150 ip 172.24.82.2
!
ip cef
no ip domain lookup
no mpls ldp logging neighbor-changes
no ftp-server write-enable
!
voice service voip
allow-connections h323 to h323
!
voice class codec 1
codec preference 1 g711ulaw
!
no voice hpi capture buffer
no voice hpi capture destination
!
interface FastEthernet0/0
ip address 172.24.82.2 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.24.82.2
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.24.82.1
ip route 192.168.254.254 255.255.255.255 172.24.82.1
!
ip http server
!
tftp-server flash:P00303020700.bin
!

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voice-port 1/0/0
!
voice-port 1/0/1
!
dial-peer cor custom
!
dial-peer voice 1001 voip
description points-to-CCM
destination-pattern 1.T
voice-class codec 1
session target ipv4:172.26.82.10
!
dial-peer voice 1002 voip
description points to router
destination-pattern 4...
voice-class codec 1
session target ipv4:172.25.82.2
!
dial-peer voice 1 pots
destination-pattern 3000
port 1/0/0
!
dial-peer voice 1003 voip
destination-pattern 26..
session target ipv4:22.22.22.38
!
!
telephony-service
load 7960-7940 P00303020700
max-ephones 48
max-dn 15
ip source-address 172.24.82.2 port 2000
create cnf-files version-stamp Jan 01 2002 00:00:00
keepalive 10
max-conferences 4
moh minuet.au
transfer-system full-consult
transfer-pattern ....
!
ephone-dn 1
number 3001
name abcde-1
call-forward busy 4001
!
ephone-dn 2
number 3002
name abcde-2
!
ephone-dn 3
number 3003
name abcde-3
!
ephone-dn 4
number 3004
name abcde-4
!
ephone 1
mac-address 0003.EB27.289E
button 1:1 2:2
!
ephone 2
mac-address 000D.39F9.3A58
button 1:3 2:4
!

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line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
password cisco
!
end

H.450 Tandem Gateway Working with Cisco CME 3.1 or Later and Cisco CallManager: Example
The following example shows a sample configuration for a Cisco CME 3.1 or later system that is linked
to an H.450 tandem gateway that serves as a proxy for a Cisco CallManager system.
Router# show running-config

Building configuration...

Current configuration : 1938 bytes

!
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable password cisco
!
aaa new-model
!
aaa session-id common
no ip subnet-zero
!
ip cef
no ip domain lookup
no ftp-server write-enable
no scripting tcl init
no scripting tcl encdir
!
voice call send-alert
!
voice service voip
allow-connections h323 to h323
supplementary-service h450.12
h323
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
!
interface FastEthernet0/0
ip address 172.27.82.2 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip h323-id host24
!

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ip classless
ip route 0.0.0.0 0.0.0.0 172.26.82.1
ip route 0.0.0.0 0.0.0.0 172.27.82.1
ip http server
!
dial-peer cor custom
!
dial-peer voice 1001 voip
description points-to-CCM
destination-pattern 4...
session target ipv4:172.24.89.150
!
dial-peer voice 1002 voip
description points to CCME1
destination-pattern 28..
session target ipv4:172.24.22.38
!
dial-peer voice 1003 voip
description points to CCME3
destination-pattern 9...
session target ipv4:192.168.1.29
!
dial-peer voice 1004 voip
description points to CCME2
destination-pattern 29..
session target ipv4:172.24.22.42
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
password cisco
!
end

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Configuration Examples for Call Transfer and Call Forwarding

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 Transcoding Between G.729 and G.711

This chapter describes how to configure Cisco CallManager Express (Cisco CME) to transcode G.729
voice signals to G.711, and vice versa, for various Cisco CME features.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Prerequisites, page 125
• Restrictions, page 125
• Information About Transcoding Between G.729 and G.711, page 126
• Configuring Transcoding Between G.711 and G.729, page 127
• Configuration Examples for Transcoding Between G.711 and G.729, page 143

Prerequisites
• Cisco CME routers and external voice routers on the same LAN must be configured with digital
signal processors (DSPs) that support transcoding.
• For Cisco CME 3.2 and later versions, DSPs on the NM-HDV, NM-HDV2, NM-HD-1V, NM-HD-2V
and NM-HD-2VE can be configured for transcoding. PVDM2-xx on the Cisco 2800 series and the
Cisco 3800 series motherboards can also be configured for transcoding.

Restrictions
Transcoding between G.711 and G.729 does not support the following:
• Meet-me conferencing
• Multiple-party conferencing
• Transcoding security

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Information About Transcoding Between G.729 and G.711

Information About Transcoding Between G.729 and G.711

Versions of Cisco CME prior to Cisco CME 3.2 supported G.729 compressed voice calls for two-party
calls only. Cisco CME V3.2 and later versions support transcoding between G.711 and G.729 for the
following features:
• Ad hoc conferencing—One or more remote conferencing parties use G.729.
• Call transfer and forward—One leg of a Voice over IP (VoIP)-to-VoIP hairpin call uses G.711 and
the other leg uses G.729. A hairpin call is an incoming call that is transferred or forwarded over the
same interface from which it arrived. As shown in Figure 23, the interface could be a Cisco 2800
router.
• Cisco Unity Express—An H.323 or SIP call using G.729 is forwarded to Cisco Unity Express. Note
that Cisco Unity Express supports only G.711, so G.729 must be transcoded.
• Music on hold (MOH)—The phone receiving MOH is part of a system that uses G.729. The G.711
MOH is translated to G.729. Because of compression, the MOH sent using G.729 loses the fidelity
that it has with G.711.
Figure 23 provides an example of each of the four call situations described.

Figure 23 Three-Way Conferencing, Call Transfer and Forward, Cisco Unity Express, and MOH
Between G.711 and G.729

Conferencing
Phone A calls phone B.
PSTN C
Phone B conferences phone C.
Call Transfer and Forward
Phone A calls phone B.
Phone B transfers or forwards Branch office
to phone C.
PSTN gateway IP
V
A IP

G.711
IP Central Office Branch office B
IP

IP WAN IP
G.729 G.729
Cisco 3745 Cisco 2800
with PVDM2, CME, IP
CUE
IP MOH, and CUE 50 phones
Phone A calls phone B using H.323 or SIP.
120 phones Phone B is busy and phone A is sent to voice mail.
MOH
103375

Phone A calls phone B.

Phone B answers and places phone A on hold.

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Configuring Transcoding Between G.711 and G.729

Configuring Transcoding Between G.711 and G.729

Regardless of whether you are planning to use transcoding for all or some of the conferencing, call
transfer, call forward, Cisco Unity Express, and MOH functionalities, one configuration is required for
all. The configuration tasks are as follows:
• Determining Digital Signal Processor Resources, page 127
• Determining the Correct DSP Allocation for Transcoding, page 130
• Provisioning NMs or NM Farms for Transcoding, page 130
• Setting Up DSP Farms for NMs, page 130
• Changing the Number of Transcoding Sessions, page 136 (Optional)
• Configuring Cisco CME to Act as the DSP Farm Host, page 136
• Verifying That the DSP Farm Is Registered and Running, page 139
• Tuning Performance, page 142

Determining Digital Signal Processor Resources

Transcoding is facilitated through DSPs, which are located in network modules (NMs). All NMs have
single in-line memory module (SIMM) sockets or packet voice/data modules (PVDM) slots that each
hold a Packet Voice DSP Module (PVDM). Each PVDM holds DSPs. As shown in Figure 24, the
NM-HDV has five SIMM sockets or PVDM slots that each hold a 12-Channel PVDM (PVDM-12). Each
PVDM-12 holds three TI 549 DSPs. Each DSP supports four channels.

Figure 24 NM-HDV Supports Up to Five PVDMs

Physical top view of NM-HDV

4 3 2 1 0

0
1 DSP DSP DSP
2 DSP DSP DSP
3 DSP DSP DSP
4 DSP DSP DSP
DSP DSP DSP

Logical view of PDVM

103376

PVDM slots
or SIMM socket

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A router can have multiple NMs. A group of NMs is called an NM farm. Figure 25 shows an NM farm
for the NM-HDV.

Figure 25 NM-HVD Farms

NM-HDVs
NM-HDV

V0

BANK 4 BANK 3 BANK 2 BANK 1 BANK 0 EH

V
NM-HDV

V0
NM-HDV Farm
BANK 4 BANK 3 BANK 2 BANK 1 BANK 0 EH

NM-HDVs
NM-HDV

V0

BANK 4 BANK 3 BANK 2 BANK 1 BANK 0 EH

NM-HDV
V
V0

BANK 4 BANK 3 BANK 2 BANK 1 BANK 0 EH

NM-HDV Farm
NM-HDV

V0

BANK 4 BANK 3 BANK 2 BANK 1 BANK 0 EH

NM-HDV
V

V0

BANK 4 BANK 3 BANK 2 BANK 1 BANK 0 EH

103377

DSP resources can be used to provide voice termination of the digital voice trunk group or resources for
the DSP farm. The collection of DSP resources available for transcoding and not used for voice
termination is called a DSP farm. See Figure 26. The DSP farm is managed by Cisco CME.

Note Transcoding of G.729 calls to G.711 allows G.729 calls to participate in existing G.711 software-based,
three-party conferencing, thus eliminating the need to divide DSPs between transcoding and
conferencing.

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Figure 26 DSP Farm

DSP = Transcoding
DSP DSP DSP
DSP = Voice termination

DSP DSP DSP

DSP farm

DSP DSP DSP

DSP DSP DSP

DSP DSP DSP

103378

To determine how many DSP voice resources are on your Cisco CME router, use the show voice dsp
command.
To determine how many DSP farms have been configured, use the show sdspfarm sessions and show
sdspfarm units commands.

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Determining the Correct DSP Allocation for Transcoding

You must allocate the DSP resources that are used by NMs or NM farms to either a DSP farm or the
system’s digital voice trunk group that handles standard voice termination. For information about
determining if your router has the correct DSP allocation for transcoding, refer to the “Allocation of DSP
Resources” section in the “Configuring Enhanced Conferencing and Transcoding for Voice Gateway
Routers” chapter of the Cisco CallManager and Cisco IOS Interoperability Guide.

Provisioning NMs or NM Farms for Transcoding

To provision NMs or NM farms for transcoding, you must determine the required number of PVDMs
and install them in either NMs or NM farms. A single NM holds up to five PVDMs. On routers capable
of holding multiple devices, NMs or NM farms can be allocated to support different functionalities.

Step 1 Determine performance requirements.

Determine the number of transcoding sessions that your router must support.
Step 2 Determine the number of DSPs that are required.
From Table 8 or Table 9 in the “Allocation of DSP Resources” section of the “Configuring Enhanced
Conferencing and Transcoding for Voice Gateway Routers” chapter of the Cisco CallManager and
Cisco IOS Interoperability Guide, determine the number of DSPs that are required to support the
transcoding sessions. Note that Cisco CME does not support DSP-farm conferencing, so only the
transcoding portion of this discussion applies to Cisco CME. If voice termination is required in addition,
determine the additional number of required DSPs from the tables. For example, 16 transcoding sessions
(30-ms packetization) and 4 G.711 voice calls require two DSPs.
Step 3 Determine the number of DSPs that are supportable.
From Table 4 in the “Allocation of DSP Resources” section of the “Configuring Enhanced Conferencing
and Transcoding for Voice Gateway Routers” chapter of the Cisco CallManager and Cisco IOS
Interoperability Guide, determine the maximum number of NMs or NM farms that your router can
support.
Step 4 Verify your solution.
Ensure that your requirements fall within router capabilities, taking into account whether your router
supports multiple NMs or NM farms. If necessary, reassess performance requirements.
Step 5 Install hardware to prepare your system for DSP-farm configuration.
Install PVDMs, NMs, and NM farms as needed.

Setting Up DSP Farms for NMs

DSP farm configuration instructions for the various NMs are described in the following sections:
• Setting Up DSP Farms for NM-HDVs, page 131
• Setting Up DSP Farms for NM-HDs and NM-HDV2s, page 132

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Setting Up DSP Farms for NM-HDVs

Setting up DSP farms on NM-HDVs involves enabling DSP farms and Skinny Client Control Protocol
(SCCP) on the Cisco CME routers. You must also specify the number of transcoding sessions per DSP
farm and select a local SCCP interface.

SUMMARY STEPS

1. voice-card slot
2. dsp services dspfarm
3. exit
4. sccp local interface-type interface-number
5. sccp ccm ip-address priority priority-number
6. sccp
7. dspfarm transcoder maximum sessions number
8. dspfarm

DETAILED STEPS

Command or Action Purpose

Step 1 voice-card slot Enters voice-card configuration mode and identifies the slot
in the chassis in which the NM-HDV or NM-HDV farm is
located.
Example:
Router(config)# voice-card 1
Step 2 dsp services dspfarm Enables DSP-farm services on the NM-HDV or NM-HDV
farm.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 3 exit Returns to global configuration mode.

Example:
Router(config-voicecard)# exit
Step 4 sccp local interface-type interface-number Selects the local interface that the SCCP applications
(transcoding and conferencing) should use to register with
Cisco CME.
Example:
Router(config)# sccp local FastEthernet 0/0 • interface-type—Interface type that the SCCP
application uses to register with Cisco CME. The type
can be an interface address or a virtual-interface
address such as Ethernet.
• interface-number—Interface number that the SCCP
application uses to register with Cisco CME.

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Command or Action Purpose

Step 5 sccp ccm ip-address priority priority-number Specifies the Cisco CME address.
• ip-address—IP address of the Cisco CME server.
Example: • priority priority—Priority of the Cisco CME server
Router(config)# sccp ccm 10.10.10.1 priority 1
relative to other connected servers. Range is from 1
(highest) to 4 (lowest).
Step 6 sccp Enables SCCP and its associated transcoding and
conferencing applications.
Example:
Router(config)# sccp
Step 7 dspfarm transcoder maximum sessions number Specifies the maximum number of transcoding sessions to
be supported by the DSP farm. A DSP can support up to
four transcoding sessions.
Example:
Router(config)# dspfarm transcoder maximum Note When you assign this value, take into account the
sessions 12 number of DSPs allocated for conferencing
services.
Step 8 dspfarm Enables the DSP farm.

Example:
Router(config)# dspfarm

Setting Up DSP Farms for NM-HDs and NM-HDV2s

Setting up DSP farms on NM-HDs and NM-HDV2s involves enabling DSP farms and SCCP on routers.
You must also select a local SCCP interface and configure a DSP farm profile and a Cisco CME group.
The DSP farm profile declares codec usage and the maximum number of transcoding sessions and
associates SCCP with the DSP farm profile.
A Cisco CME group is a naming device under which data for the DSP farms is declared. Only one group
is required. For the Cisco CME group you must assign a priority to the group, associate the group with
a DSP farm profile, and set the keepalive, switchback, and switchover parameters.

SUMMARY STEPS

1. voice-card slot
2. dsp services dspfarm
3. exit
4. sccp local interface-type interface-number
5. sccp ccm ip-address identifier identifier-number
6. sccp
7. sccp ccm group group-number
8. associate ccm identifier-number priority
9. associate profile profile-identifier register device-name
10. keepalive retries number
11. switchover method {graceful | immediate}

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12. switchback method {graceful | guard timeout-guard-value | immediate | uptime

uptime-timeout-value}
13. switchback interval seconds
14. exit
15. dspfarm profile profile-identifier transcode
16. codec codec-type
17. maximum sessions number
18. associate application sccp
19. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice-card slot Enters voice-card configuration mode and identifies the slot
in the chassis in which the NM-HDV or NM-HDV farm is
located.
Example:
Router(config)# voice-card 1
Step 2 dsp services dspfarm Enables DSP-farm services on the NM-HDV or NM-HDV
farm.
Example:
Router(config-voicecard)# dsp services dspfarm
Step 3 exit Exits voice card configuration mode.

Example:
Router(config-voicecard)# exit
Step 4 sccp local interface-type interface-number Selects the local interface that the SCCP applications
(transcoding and conferencing) should use to register with
Cisco CME.
Example:
Router(config)# sccp local FastEthernet 0/0 • interface-type—Interface type that the SCCP
application uses to register with Cisco CME. The type
can be an interface address or a virtual-interface
address such as Ethernet.
• interface-number—Interface number that the SCCP
application uses to register with Cisco CME.
Step 5 sccp ccm ip-address identifier Specifies the Cisco CME address.
identifier-number
• ip-address—IP address of the Cisco CME server.
• identifier identifier-number—Identifier used to
Example:
Router(config)# sccp ccm 10.10.10.1 priority 2
associate the SCCP Cisco CME IP address with a
Cisco CME group. See the associate ccm command in
Step 8.
Step 6 sccp Enables SCCP and its associated transcoding and
conferencing applications.
Example:
Router(config)# sccp

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Command or Action Purpose

Step 7 sccp ccm group group-number Creates a Cisco CME group and enters the SCCP
configuration mode for Cisco CME.
Example: • group-number—Number that identifies the Cisco CME
Router(config)# sccp ccm group 1 group. Range is 1 to 65535. There is no default value.
Step 8 associate ccm identifier-number priority Associates a Cisco CME with a Cisco CME group and
establishes its priority within the group.
Example: • identifier-number—Number that identifies
Router(config-sccp-ccm)# associate ccm 1 Cisco CME. Range is 1 to 65535. There is no default
priority value.
• priority—The priority of the Cisco CME router in the
Cisco CME group. The default is 1 because only one
Cisco CME group is possible.
Step 9 associate profile profile-identifier register Associates a DSP farm profile with a Cisco CME group.
device-name
• profile-identifier—Number that identifies the DSP farm
profile. Range is 1 to 65535. There is no default value.
Example:
Router(config-sccp-ccm)# associate profile 1
• register device-name—User-specified device name in
register mtp000a8eaca80 Cisco CME. The device-name must use the format of
mtpmac-address, where the mac-address is the burnt-in
address (bia) of the physical interface that is used to
register as the SCCP device.
Step 10 keepalive retries number Sets the number of keepalive retries from SCCP to
Cisco CME.
Example: • number—Number of keepalive attempts. Range is 1 to
Router(config-sccp-ccm)# keepalive retries 5 32. The default is 3.
Step 11 switchover method [graceful | immediate] Sets the switchover method that the SCCP client uses when
the communication link between the active Cisco CME and
the SCCP client goes down.
Example:
Router(config-sccp-ccm)# keepalive retries 5 • graceful—Switchover happens only after all the active
sessions have been terminated gracefully.
• immediate—Switches over to any one of the secondary
Cisco CME systems immediately.

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Command or Action Purpose

Step 12 switchback method {graceful | guard Sets the switchover method that the SCCP client uses when
timeout-guard-value | immediate | uptime the communication link between the active Cisco CME and
uptime-timeout-value}
the SCCP client goes down.
• graceful—Switchback happens only after all the active
Example: sessions have been terminated gracefully.
Router(config-sccp-ccm)# switchback method
immediate • guard timeout-guard-value—Switchback happens
either when the active sessions have been terminated
gracefully or when the guard timer expires, whichever
happens first. Timeout value is in seconds. Range is
from 60 to 172800. Default is 7200.
• immediate—Switches back to the higher order
Cisco CME immediately as soon as the timer expires,
whether there is an active connection or not.
• uptime uptime-timeout-value—Initiates the uptime
timer when the higher-order Cisco CME comes alive.
Timeout value is in seconds. Range is from 60 to
172800. Default is 7200.
Step 13 switchback interval seconds Sets the amount of time that the DSP farm waits before
polling the primary Cisco CME when the current
Cisco CME switchback connection fails.
Example:
Router(config-sccp-ccm)# switchback interval 5 • seconds—Timer value, in seconds. Range is 1 to 3600.
Default is 60.
Step 14 exit Exits voice card configuration mode.

Example:
Router(config-voicecard)# exit
Step 15 dspfarm profile profile-identifier transcode Enters DSP farm profile configuration mode and defines a
profile for DSP farm services.
Example: • profile-identifier—Number that uniquely identifies a
Router(config)# dspfarm profile 1 transcode profile. Range is 1 to 65535. There is no default.
• transcode—Enables profile for transcoding.
Step 16 codec codec-type Specifies the codecs supported by a DSP farm profile.
• codec-type—Specifies the codec preferred.
Example: • Use CLI help to locate a list of codecs.
Router(config-dspfarm-profile)# codec g711ulaw
Step 17 maximum sessions number Specifies the maximum number of sessions that are
supported by the profile.
Example: • number—Number of sessions supported by the profile.
Router(config-dspfarm-profile)# maximum Range is 0 to X. Default is 0. The X value is determined
sessions 5 at run time depending on the number of resources
available with the resource provider.

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Command or Action Purpose

Step 18 associate application sccp Associates SCCP with the DSP farm profile.

Example:
Router(config-dspfarm-profile)# associate
application sccp
Step 19 exit Exits DSP farm profile configuration mode.

Example:
Router(config-dspfarm-profile)# exit

Changing the Number of Transcoding Sessions

Perform this task if you must change the maximum number of DSP farm transcoding sessions.

SUMMARY STEPS

1. no dspfarm
2. dspfarm transcoder maximum sessions number
3. dspfarm

DETAILED STEPS

Command or Action Purpose

Step 1 no dspfarm Disables the DSP farm.

Example:
Router(config)# no dspfarm
Step 2 dspfarm transcoder maximum sessions number Specifies the maximum number of transcoding sessions to
be supported by the DSP farm.
Example:
Router(config)# dspfarm transcoder maximum
sessions 12
Step 3 dspfarm Enables the DSP farm.

Example:
Router(config)# dspfarm

Configuring Cisco CME to Act as the DSP Farm Host

Configuring the Cisco CME router to act as the DSP farm host involves the following:
• Setting the Cisco CME router to receive IP phone messages over the Cisco CME router’s IP address.
• Setting the SCCP server to use a maximum number of DSP farms.

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• Setting the Cisco CME router to allow for a maximum number of G.711 and G.729 transcode
sessions.
• Tagging and defining DSP farm units for Cisco CME router registry.
To determine the maximum number of transcode sessions that can take place at one time, multiply the
maximum number of transcoder sessions you have configured using the dspfarm transcoder maximum
sessions command by the number of DSP farms in your NM or NM farms. To determine how many DSP
farms have been configured, use the show sdspfarm sessions and show sdspfarm units commands.
The DSP farm units are tagged as 1 through 5 and are defined using the Cisco CME interface’s MAC
address. For example, if you have the following configuration:
interface FastEthernet 0/0
ip address 10.5.49.160 255.255.0.0
.
.
.
sccp local FastEthernet 0/0
sccp

the show interface FastEthernet 0/0 command will yield a MAC address as shown in the following
output:
Router# show interface FastEthernet 0/0
.
.
.
FastEthernet0/0 is up, line protocol is up
Hardware is AmdFE, address is 000a.8aea.ca80 (bia 000a.8aea.ca80)
.
.
.

The MAC address of the Fast Ethernet interface is 000a.8aea.ca80.

Note You can unregister all active calls’ transcoding streams with the sdspfarm unregister force command.

SUMMARY STEPS

1. telephony-service
2. ip source-address ip-address [port port] [any-match | strict-match]
3. sdspfarm units number
4. sdspfarm transcode sessions number
5. sdspfarm tag number device-number
6. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 ip source-address ip-address [port port] Enables a router to receive messages from Cisco IP phones
[any-match | strict-match] through specified IP addresses and ports.
• address—The range is 0 through 5. The default is 0.
Example:
Router(config-telephony)# ip source address
• port port—(Optional) TCP/IP port used for Skinny
10.10.10.1 port 3000 Protocol. The default is 2000.
• any-match—(Optional) Disables strict IP address
checking for registration. This is the default.
• strict-match—(Optional) Requires strict IP address
checking for registration.
Step 3 sdspfarm units number Specifies the maximum number of DSP farms that are
allowed to be registered to the SCCP server.
Example: • number—The range is 0 through 5. The default is 0.
Router(config-telephony)# sdspfarm units 4
Step 4 sdspfarm transcode sessions number Specifies the maximum number of transcode sessions for
G.729 allowed by the Cisco CME router.
Example: • One transcode session consists of two transcode
Router(config-telephony)# sdspfarm transcode streams between callers using transcode. Use the
sessions 40 maximum number of transcoding sessions and
conference calls that you want your router to support at
one time.
• number—The range is 0 through 128. The default is 0.
Step 5 sdspfarm tag number device-name Permits a DSP farm unit to be registered to Cisco CME and
associates it with an SCCP client interface’s MAC address .
Example: • number—The tag number. The range is 1 through 5.
Router(config-telephony)# sdspfarm tag 1
mtp000a8eaca80
• device-name—The MAC address of the SCCP client
interface, with the “mtp” prefix added.
Step 6 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Examples

The following example configures Cisco CME router address 10.100.10.11 port 2000 to act as the farm
host using the DSP farm at mtp000a8eaca80 to allow for a maximum of 1 DSP farm and 16 transcode
sessions:
telephony-service
ip source address 10.100.10.11 port 2000
sdspfarm units 1

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sdspfarm transcode sessions 16

sdspfarm tag 1 mtp000a8eaca80
exit

Verifying That the DSP Farm Is Registered and Running

Ensure that Cisco CME is properly configured to provision transcoding and conferencing resources.
You can clear any of the following features by disabling the DSP farm or SCCP:
• Active calls
• DSPs
• Active (primary) Cisco CME
• Active connection to a Cisco CME
If your primary Cisco CME fails and hands control over to a secondary Cisco CME, you can switch back
to the primary system in either of two ways:
• Graceful switchback: Switches back when active calls on the currently primary system have been
terminated.
• Immediate switchback: Switches back immediately.
To verify that the DSP farm is registered and running, perform the following steps in any order.

SUMMARY STEPS

1. show sccp [statistics | connections]

2. show sdspfarm units
3. show sdspfarm sessions
4. show sdspfarm sessions summary
5. show sdspfarm sessions active
6. show sccp connections details
7. debug sccp {all | errors | events | packets | parser}
8. debug dspfarm {all | errors | events | packets}
9. debug ephone mtp

DETAILED STEPS

Step 1 show sccp [statistics | connections]

Use this command to display the SCCP configuration information and current status.
Router# show sccp statistics

SCCP Application Service(s) Statistics:

Profile ID:1, Service Type:Transcoding

TCP packets rx 7, tx 7
Unsupported pkts rx 1, Unrecognized pkts rx 0
Register tx 1, successful 1, rejected 0, failed 0
KeepAlive tx 0, successful 0, failed 0

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OpenReceiveChannel rx 2, successful 2, failed 0

CloseReceiveChannel rx 0, successful 0, failed 0
StartMediaTransmission rx 2, successful 2, failed 0
StopMediaTransmission rx 0, successful 0, failed 0
Reset rx 0, successful 0, failed 0
MediaStreamingFailure rx 0
Switchover 0, Switchback 0

Step 2 show dspfarm units

Use this command to display the configured and registered DSP farms.
Router# show sdspfarm units

mtp-1 Device:MTP003080218a31 TCP socket:[2] REGISTERED

actual_stream:8 max_stream 8 IP:10.10.10.3 11470 MTP YOKO keepalive 1
Supported codec:G711Ulaw
G711Alaw
G729a
G729ab

max-mtps:1, max-streams:40, alloc-streams:8, act-streams:2

Step 3 show dspfarm sessions

Use this command to display the transcoding streams.
Router# show sdspfarm sessions
Stream-ID:1 mtp:1 10.10.10.3 18404 Local:2000 START
usage:Ip-Ip
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:2

Stream-ID:2 mtp:1 10.10.10.3 17502 Local:2000 START

usage:Ip-Ip
codec:G729AnnexA duration:20 vad:0 peer Stream-ID:1

Stream-ID:3 mtp:1 0.0.0.0 0 Local:0 IDLE

usage:
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0

Stream-ID:4 mtp:1 0.0.0.0 0 Local:0 IDLE

usage:
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0

Stream-ID:5 mtp:1 0.0.0.0 0 Local:0 IDLE

usage:
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0

Stream-ID:6 mtp:1 0.0.0.0 0 Local:0 IDLE

usage:
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0

Stream-ID:7 mtp:1 0.0.0.0 0 Local:0 IDLE

usage:
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0

Stream-ID:8 mtp:1 0.0.0.0 0 Local:0 IDLE

usage:
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:0

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Configuring Transcoding Between G.711 and G.729

Step 4 show dspfarm sessions summary

Use this command to display a summary view the transcoding streams.
Router# show sdspfarm sessions summary

max-mtps:2, max-streams:240, alloc-streams:40, act-streams:2

ID MTP State CallID confID Usage Codec/Duration
==== ===== ====== =========== ====== ============================= ==============
1 2 IDLE -1 0 G711Ulaw64k /20ms
2 2 IDLE -1 0 G711Ulaw64k /20ms
3 2 START -1 3 MoH (DN=3 , CH=1) FE=TRUE G729 /20ms
4 2 START -1 3 MoH (DN=3 , CH=1) FE=FALSE G711Ulaw64k /20ms
5 2 IDLE -1 0 G711Ulaw64k /20ms
6 2 IDLE -1 0 G711Ulaw64k /20ms
7 2 IDLE -1 0 G711Ulaw64k /20ms
8 2 IDLE -1 0 G711Ulaw64k /20ms
9 2 IDLE -1 0 G711Ulaw64k /20ms
10 2 IDLE -1 0 G711Ulaw64k /20ms
11 2 IDLE -1 0 G711Ulaw64k /20ms
12 2 IDLE -1 0 G711Ulaw64k /20ms
13 2 IDLE -1 0 G711Ulaw64k /20ms
14 2 IDLE -1 0 G711Ulaw64k /20ms
15 2 IDLE -1 0 G711Ulaw64k /20ms
16 2 IDLE -1 0 G711Ulaw64k /20ms
17 2 IDLE -1 0 G711Ulaw64k /20ms
18 2 IDLE -1 0 G711Ulaw64k /20ms
19 2 IDLE -1 0 G711Ulaw64k /20ms
20 2 IDLE -1 0 G711Ulaw64k /20ms
21 2 IDLE -1 0 G711Ulaw64k /20ms
22 2 IDLE -1 0 G711Ulaw64k /20ms
23 2 IDLE -1 0 G711Ulaw64k /20ms
24 2 IDLE -1 0 G711Ulaw64k /20ms
25 2 IDLE -1 0 G711Ulaw64k /20ms
26 2 IDLE -1 0 G711Ulaw64k /20ms
27 2 IDLE -1 0 G711Ulaw64k /20ms
28 2 IDLE -1 0 G711Ulaw64k /20ms
29 2 IDLE -1 0 G711Ulaw64k /20ms
30 2 IDLE -1 0 G711Ulaw64k /20ms
31 2 IDLE -1 0 G711Ulaw64k /20ms
32 2 IDLE -1 0 G711Ulaw64k /20ms
33 2 IDLE -1 0 G711Ulaw64k /20ms
34 2 IDLE -1 0 G711Ulaw64k /20ms
35 2 IDLE -1 0 G711Ulaw64k /20ms
36 2 IDLE -1 0 G711Ulaw64k /20ms

Step 5 show dspfarm sessions active

Use this command to display the transcoding streams for all active sessions.
Router# show sdspfarm sessions active
Stream-ID:1 mtp:1 10.10.10.3 18404 Local:2000 START
usage:Ip-Ip
codec:G711Ulaw64k duration:20 vad:0 peer Stream-ID:2

Stream-ID:2 mtp:1 10.10.10.3 17502 Local:2000 START

usage:Ip-Ip
codec:G729AnnexA duration:20 vad:0 peer Stream-ID:1

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Configuring Transcoding Between G.711 and G.729

Step 6 show sccp connections details

Use this command to display the SCCP connections details such as call-leg details.
Router# show sccp connections details

bridge-info(bid, cid) - Normal bridge information(Bridge id, Calleg id)

mmbridge-info(bid, cid) - Mixed mode bridge information(Bridge id, Calleg id)

sess_id conn_id call-id codec pkt-period type bridge-info(bid, cid)

mmbridge-info(bid, cid)

1 - 14 N/A N/A transmsp All RTPSPI Callegs N/A

1 2 15 g729a 20 rtpspi (4,14) N/A

1 1 13 g711u 20 rtpspi (3,14) N/A

Total number of active session(s) 1, connection(s) 2, and callegs 3

Step 7 debug sccp {all | errors | events | packets | parser}

Use this command to set debugging levels for SCCP and its applications.
Step 8 debug dspfarm {all | errors | events | packets}
Use this command to set debugging levels for DSP-farm service.
Step 9 debug ephone mtp
Use this command to enable Message Transfer Part (MTP) debugging. Use this debug command with
the debug ephone mtp, debug ephone register, debug ephone state, and debug ephone pak
commands.

Tuning Performance
Use the following commands to tune performance.

SUMMARY STEPS

1. sccp ip precedence value

2. dspfarm rtp timeout seconds
3. dspfarm connection interval seconds

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Configuration Examples for Transcoding Between G.711 and G.729

DETAILED STEPS

Command or Action Purpose

Step 1 sccp ip precedence value (Optional) Sets the IP precedence value.
• Doing so allows you to increase the priority of voice
Example: packets over connections controlled by SCCP.
Router(config)# sccp ip precedence 5
Step 2 dspfarm rtp timeout seconds (Optional) Configures the Real-Time Transport Protocol
(RTP) timeout interval for when the error condition “RTP
port unreachable” occurs.
Example:
Router(config)# dspfarm rtp timeout 60
Step 3 dspfarm connection interval seconds (Optional) Specifies how long to monitor RTP inactivity
before deleting an RTP stream.
Example:
Router(config)# dspfarm connection interval 60

Configuration Examples for Transcoding Between G.711 and

G.729
This section provides the following configuration examples:
• NM-HDV: Example, page 143
• NM-HD and NM-HDV2: Example, page 144

NM-HDV: Example
The following configuration example sets up a DSP farm of 4 DSPs to handle up to 16 sessions (4
sessions per DSP) on a router with an IP address of 10.5.49.160 and a priority of 1 among other servers:
voice-card 1
dsp services dspfarm
exit
sccp local FastEthernet 0/0
sccp
sccp ccm 10.5.49.160 priority 1
dspfarm transcoder maximum sessions 16
dspfarm

telephony-service
ip source-address 10.5.49.200 port 2000
sdspfarm units 4
sdspfarm transcode sessions 40
sdspfarm tag 1 mtp000a8eaca80
sdspfarm tag 2 mtp123445672012

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NM-HD and NM-HDV2: Example

The following example shows a transcoding configuration for a Cisco CME router with either an
NM-HD or NM-HDV2. The configuration example sets up six transcoding sessions on a router with one
DSP farm, an IP address of 10.5.49.160, and a priority of 1 among servers.
voice-card 1
dsp services dspfarm

sccp local FastEthernet 0/1

sccp
sccp ccm 10.5.49.160 identifier 1

sccp ccm group 123

associate ccm 1 priority
associate profile 1 register mtp123456792012
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 5

dspfarm profile 1 transcode

codec g711ulaw
codec g711alaw
codec g729ar8
codec g719abr8
maximum sessions 6
associate application sccp

telephony-service
ip source-address 10.5.49.200 port 2000
sdspfarm units 1
sdspfarm transcode sessions 40
sdspfarm tag 1 mtp000a8eaca80
sdspfarm tag 2 mtp123445672012

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 Setting Up the Cisco CME GUI

This chapter describes the Cisco CallManager Express (Cisco CME) graphical user interface (GUI) and
explains how to set it up for three different classes of user.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Information About Setting Up the Cisco CME GUI, page 145
• Setting Up GUI Access for the System Administrator, page 147
• Accessing the Cisco CME GUI, page 150
• Setting Up GUI Access for Customer Administrators and Phone Users, page 152
• What to Do Next, page 160

Information About Setting Up the Cisco CME GUI

Note If you are downgrading or upgrading Cisco CME and use the Cisco CME GUI, you must downgrade or
upgrade your GUI files. For more information, see the “Software Prerequisites” section on page 12.

The Cisco CME GUI provides a web-based interface to manage most systemwide and phone-based
features. In particular, the GUI facilitates the routine adds and changes associated with employee
turnover, allowing these changes to be performed by nontechnical staff. Cisco CME GUI
The GUI provides three levels of access to support the following user classes:
• System administrator—Able to configure all systemwide and phone-based features. This person is
familiar with Cisco IOS software and VoIP network configuration.
• Customer administrator—Able to perform routine phone adds and changes without having access to
systemwide features. This person does not have to be trained in Cisco IOS software.

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Information About Setting Up the Cisco CME GUI

• Phone user—Able to program a small set of features on his or her own phone and search the
Cisco CME directory.
The Cisco CME GUI uses HTTP to transfer information from the router to the PC of an administrator or
phone user. The router must be configured as an HTTP server, and an initial system administrator
username and password must be defined from the router command-line interface (CLI). Additional
customer administrators and phone users can be added from the Cisco CME router using CLI commands
or from a PC using GUI screens.
Cisco CME provides support for eXtensible Markup Language (XML) cascading style sheets (files with
a .css suffix) that can be used to customize the browser GUI display.
The GUI supports authentication, authorization, and accounting (AAA) authentication for system
administrators through a remote server when this capability is enabled with the ip http authentication
command. If authentication through the server fails, the local router is searched.
The sequence of tasks to set up the Cisco CME GUI is as follows:
1. Setting Up GUI Access for the System Administrator—Define the HTTP server on the Cisco CME
router and create an account for the system administrator to log on to the GUI.
2. Accessing the Cisco CME GUI—Log on to the GUI as the system administrator to verify its
installation.
3. Setting Up GUI Access for Customer Administrators and Phone Users—Optionally create accounts
for customer administrators and phone users to use to log on to the GUI. You can create additional
accounts from the GUI itself or with router CLI.

Prerequisites
Files required for the operation of the GUI must have been be copied into flash memory on the router.
For a complete list of the GUI files, see the “Software Prerequisites” section in the “Cisco CallManager
Express Overview” chapter.

Note Cisco CME GUI files are version-specific; GUI files for one version of Cisco CME are not compatible
with any other version of Cisco CME. When Cisco CME is downgraded or upgraded, the GUI files for
the old version must be overwritten with GUI files that match the Cisco CME version that is being
installed.

Restrictions
• The web browser that you use to access the GUI must be Microsoft Internet Explorer Version 5.5 or
a later version. No other type of browser can be used to access the GUI.
• If you use an XML configuration file to create a customer administrator login, the size of that XML
file must be 4000 bytes or smaller.
• The password of the system administrator cannot be changed through the GUI. Only the password
of a customer administrator or a phone user can be changed through the GUI.
• If more than 100 phones are configured, choosing to display all phones will result in a long delay
before results are shown.

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Setting Up GUI Access for the System Administrator

To set up GUI access for the Cisco CME system administrator, complete the following tasks:
• Setting Up the HTTP Server, page 147 (required)
• Setting Up GUI Access for the System Administrator, page 148 (required)

Setting Up the HTTP Server

By default the HTTP server on a router is disabled. This task enables the HTTP server, specifies the path
to files that are used for the GUI, and specifies a method of user authentication for security.
Use of the ip http authentication command is critical to prevent unauthorized users from accessing the
Cisco CME router. The default if this command is not used is that the enable password for the Cisco
CME router is used to authenticate user access to the GUI. Because a username and password pair is
more secure than using only a password for authentication, use of only the enable password for
authentication is strongly discouraged. Instead, Cisco recommends use of the local or TACACS
authentication options, configured as part of a global AAA framework and specified by the ip http
authentication command.
By explicitly using the ip http authentication command, you designate alternative authentication
methods, such as by a local login account or by the method that is specified in the AAA configuration
on the Cisco CME router. If you select the AAA authentication method, you must also define an
authentication method in your AAA configuration, using commands similar to the ones in this example
but appropriate for your situation:
aaa new-model
aaa authentication login default group tacacs+ local
tacacs-server host 10.1.2.3

For more information about AAA authentication, refer to the “Configuring Authentication” chapter of
the Cisco IOS Security Configuration Guide.

SUMMARY STEPS

1. ip http server
2. ip http path flash:
3. ip http authentication {aaa | enable | local | tacacs}

DETAILED STEPS

Command or Action Purpose

Step 1 ip http server Enables the Cisco web browser user interface on the
local Cisco CME router.
Example:
Router(config)# ip http server

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Command or Action Purpose

Step 2 ip http path flash: Sets the base HTTP path for HTML files to flash
memory on the router.
Example:
Router(config)# ip http path flash:
Step 3 ip http authentication {aaa | enable | local | Specifies method of authentication to use for the
tacacs} HTTP server. Default is the enable keyword.
• aaa—Indicates that the authentication method
Example: used for the AAA login service should be used
Router(config)# ip http authentication aaa for authentication. The AAA login service
method is specified by the aaa authentication
login command.
• enable—Uses the enable password. This is the
default if this command is not used.
• local—Uses login user name, password, and
privilege level access combination specified in
the local system configuration (by the username
global configuration command).
• tacacs—Uses TACACS (or XTACACS) server.

Setting Up GUI Access for the System Administrator

This task defines an initial username and password for a system administrator to use to access the GUI
and enables the GUI to be used to set the time and to add directory listings.
Once you have created this account, you can log in to the GUI and create additional login accounts using
the GUI itself. Alternatively, you can continue to use router CLI to create additional accounts. Both
methods are explained in the sections following this section.

SUMMARY STEPS

1. telephony-service
2. web admin system name username {password string | secret {0 | 5} string}
3. dn-webedit
4. time-webedit
5. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 web admin system name username {password string | Defines a username and password for a system
secret {0 | 5} string} administrator. The default username is Admin. There
is no default password.
Example: • name username—System administrator
Router(config-telephony)# web admin system name pwa3 username.
secret 0 wp78pw
• password string—String to verify system
administrator’s identity. Default is empty string.
• secret {0 | 5} string—Password should be
encrypted. The digit specifies state of encryption
of the string that follows, as explained here:
– 0—Password that follows is not yet
encrypted.
– 5—Password that follows is encrypted using
Message Digest 5 (MD5).
Note The secret 5 keyword pair is used in the
output of show commands when encrypted
passwords are displayed. It indicates that the
password that follows is encrypted.
Step 3 dn-webedit (Optional) Enables the ability to add directory
numbers through the web interface.
Example: The no form of this command disables the ability to
Router(config-telephony)# dn-webedit create IP phone extension telephone numbers. That
ability could disrupt the network-wide management
of telephone numbers.
If this command is not used, the ability to create
directory numbers is disabled by default.

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Command or Action Purpose

Step 4 time-webedit (Optional) Enables the ability to set the phone time
for the Cisco CME system through the web interface.
Example: Note Cisco discourages this method for setting
Router(config-telephony)# time-webedit network time. The router should be set up to
automatically synchronize its router clock
from a network-based clock source using
Network Time Protocol (NTP). In the rare
case that a network NTP clock source is not
available, the time-webedit command can be
used to allow manual setting and resetting of
the router clock through the GUI.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Troubleshooting Tips
If you are having trouble starting the Cisco CME GUI, try the following actions:
• Make sure you are using Microsoft Internet Explorer (IE) Version 5.5 or a later release. No other
type of browser can be used to access the GUI.
• Clear your browser cache or history.
• Make sure that you have in router flash memory the correct version of the GUI files for the version
of Cisco CME that you have. Compare the filenames in flash memory with the list in the
“Prerequisites” section in the “Cisco CallManager Express Overview” chapter. Compare the sizes
of files in flash memory with the sizes of the files in the tar archive called cme-3.2.0-gui.tar (or a
later version of the file) to be sure that you have the most recent files installed in flash memory. The
latest version can be downloaded from the Cisco CME Software Download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.

Accessing the Cisco CME GUI

Note The Cisco CME GUI requires use of Microsoft Internet Explorer (IE) Version 5.5 or a later release. No
other type of browser can be used to access the GUI.

To access the Cisco CME router through the web to make configuration changes, point your IE 5.5
browser (or a later version) to the following URL:
http://router_ipaddr/ccme.html

where router_ipaddr is the IP address of your Cisco CME router. For example, if the IP address of your
Cisco CME router is 10.10.10.176, enter the following:
http://10.10.10.176/ccme.html

You are presented with a login screen. Enter your username and password.

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Accessing the Cisco CME GUI

The Cisco CME system evaluates your privilege level and presents the appropriate screen. Note that
users with Cisco IOS software privilege level 15 also have system-administrator-level privileges in the
Cisco CME GUI once they have been authenticated locally or remotely through AAA. The ip http
authentication command that has been configured on the Cisco CME router determines where
authentication occurs.
After you have been logged in and have been authenticated, you see one of the following home screens,
based on your user class:
• The system administrator home screen is shown in Figure 27 on page 151.
• The customer administrator sees a reduced version of the options available on the system
administrator screen, according to the XML configuration file that the system administrator created.
• The phone user home screen is shown in Figure 28 on page 152.
After you log in successfully, online help is available from the Help menu.

Figure 27 Cisco CME GUI System Administrator Home Screen

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Figure 28 Cisco CME GUI Phone User Home Screen

Setting Up GUI Access for Customer Administrators and

Phone Users
After creating a system administrator account for GUI access, you can create accounts for customer
administrators or for individual phone users so that they can log in to the GUI from their PCs. You can
create these accounts using router CLI or by using the GUI itself. These procedures are explained in the
following sections:
• Setting Up GUI Access for Customer Administrators, page 152
• Setting Up GUI Access for Phone Users, page 158

Setting Up GUI Access for Customer Administrators

The following tasks are required to set up customer administrator GUI access for a Cisco CME system:
• Creating and Loading an XML Configuration File, page 153 (required)
• Defining a Customer Administrator, page 156 (required)

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Creating and Loading an XML Configuration File

The XML configuration file specifies the characteristics and features that you want to make available to
customer administrators and the characteristics and features that should be restricted. The file follows a
template that conforms to the Cisco XML Document Type Definition (DTD), which is documented in
Cisco IP Phone Services Application Development Notes. The template is named xml.template and is
one of the Cisco CME files that you download from the Cisco Software Center when you first set up a
Cisco CME system. The XML template is shown in the “XML Configuration File Template Example”
section on page 154 and a sample XML file is shown in “XML Configuration File Example” section on
page 155.
Edit and load the XML configuration file by using the following steps.

SUMMARY STEPS

1. Make a copy of the XML template and open it in any text editor.
2. Edit the XML template.
3. Copy the file to a TFTP or FTP server that can be accessed by the Cisco CME router.
4. Copy your file to flash memory on the Cisco CME router.
5. Load the XML file from router flash memory.

DETAILED STEPS

Step 1 Make a copy of the XML template that you downloaded from the Cisco Software Center (shown in the
“XML Configuration File Template Example” section on page 154) and open it in any text editor. Give
the file a name that is meaningful to you and that uses “xml” as its suffix. For example, you could name
the file “custadm.xml.”
Step 2 Edit the XML template. Within the template, each line that starts with a title enclosed in angle brackets
describes an XML object and matches an entity name in the CME GUI. For example, “<AddExtension>”
refers to the Add Extension capability, and “<Type>” refers to the Type field on the Add Extension
screen. For each object in the template, you have a choice of actions. Your choices appear within
brackets; for example, “[Hide | Show]” indicates that you have a choice between whether this object is
hidden or visible when a customer administrator logs into the GUI. Delete the action that you do not
want and the vertical bar and brackets around the actions.
For example, to hide the Sequence Number field, change the following text in the template file:
<SequenceNumber> [Hide | Show] </SequenceNumber>

to the following text in your configuration file:

<SequenceNumber> Hide </SequenceNumber>

Edit every line in the template until you have changed each choice in brackets to a single action and you
have removed the vertical bars and brackets. A sample XML file is shown in the “XML Configuration
File Example” section on page 155.
Step 3 Copy the file to a TFTP (or FTP) server on your network that can be accessed by the Cisco CME router.
Step 4 Copy your file to flash memory on the Cisco CME router.
Router# copy tftp flash

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Step 5 Load the XML file from router flash memory.

Router(config)# telephony-service
Router(config-telephony)# web customize load filename
Router(config-telephony)# exit

XML Configuration File Template Example

<Presentation>
<MainMenu>
<!-- Take Higher Precedence over CLI "dn-web-edit" -->
<AddExtension> [Hide | Show] </AddExtension>
<DeleteExtension> [Hide | Show] </DeleteExtension>
<AddPhone> [Hide | Show] </AddPhone>
<DeletePhone> [Hide | Show] </DeletePhone>
</MainMenu>

<Extension>
<!-- Control both view and change, and possible add or delete -->
<SequenceNumber> [Hide | Show] </SequenceNumber>
<Type> [Hide | Show] </Type>
<Huntstop> [Hide | Show] </Huntstop>
<Preference> [Hide | Show] </Preference>
<HoldAlert> [Hide | Show] </HoldAlert>
<TranslationRules> [Hide | Show] </TranslationRules>
<Paging> [Hide | Show] </Paging>
<Intercom> [Hide | Show] </Intercom>
<MWI> [Hide | Show] </MWI>
<MoH> [Hide | Show] </MoH>
<LBDN> [Hide | Show] </LBDN>
<DualLine> [Hide | Show] </DualLine>
<Reg> [Hide | Show] </Reg>
<PGroup> [Hide | Show] </PGroup>
</Extension>

<Phone>
<!-- control both view and change, and possible add and delete --->
<SequenceNumber> [Hide | Show] </SequenceNumber>
</Phone>

<System>
<!-- Control View Only -->
<PhoneURL> [Hide | Show] </PhoneURL>
<PhoneLoad> [Hide | Show]</PhoneLoad>
<CallHistory> [Hide | Show] </CallHistory>
<MWIServer> [Hide | Show] </MWIServer>
<!-- Control Either View and Change or Change Only -->
<TransferPattern attr=[Both | Change]> [Hide | Show] </TransferPattern>
<VoiceMailNumber attr=[Both | Change]> [Hide | Show] </VoiceMailNumber>
<MaxNumberPhone attr=[Both | Change]> [Hide | Show] </MaxNumberPhone>
<DialplanPattern attr=[Both | Change]> [Hide | Show] </DialplanPattern>
<SecDialTone attr=[Both | Change]> [Hide | Show] </SecDialTone>
<Timeouts attr=[Both | Change]> [Hide | Show] </Timeouts>
<CIDBlock attr=[Both | Change]> [Hide | Show] </CIDBlock>
<HuntGroup attr=[Both | Change]> [Hide | Show] </HuntGroup>
<NightSerBell attr=[Both | Change]> [Hide | Show] </NightSerBell>
<!-- Control Change Only -->
<!-- Take Higher Precedence over CLI "time-web-edit" -->
<Time> [Hide | Show] </Time>
</System>

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<Function>
<AddLineToPhone> [No | Yes] </AddLineToPhone>
<DeleteLineFromPhone> [No | Yes] </DeleteLineFromPhone>
<NewDnDpCheck> [No | Yes] </DpDnCrossCheck>
<MaxLinePerPhone> [1-6] </MaxLinePerPhone>
</Function>
</Presentation>

XML Configuration File Example

sample.xml
<Presentation>
<MainMenu>
<AddExtension> Hide </AddExtension>
<DeleteExtension> Hide </DeleteExtension>
<AddPhone> Hide </AddPhone>
<DeletePhone> Hide </DeletePhone>
</MainMenu>

<Extension>
<SequenceNumber> Hide </SequenceNumber>
<Type> Hide </Type>
<Huntstop> Hide </Huntstop>
<Preference> Hide </Preference>
<HoldAlert> Hide </HoldAlert>
<TranslationRule> Hide </TranslationRule>
<Paging> Show </Paging>
<Intercom> Hide </Intercom>
<MWI> Hide </MWI>
<MoH> Hide </MoH>
<LBDN> Hide </LBDN>
<DualLine> Hide </DualLine>
<Reg> Hide </Reg>
<PGroup> Show </PGroup>
</Extension>

<Phone>
<SequenceNumber> Hide </SequenceNumber>
</Phone>

<System>
<PhoneURL> Hide </PhoneURL>
<PhoneLoad> Hide </PhoneLoad>
<CallHistory> Hide </CallHistory>
<MWIServer> Hide </MWIServer>
<TransferPattern attr=Both> Hide </TransferPattern>
<VoiceMailNumber attr=Both> Hide </VoiceMailNumber>
<MaxNumberPhone attr=Both> Hide </MaxNumberPhone>
<DialplanPattern attr=Change> Hide </DialplanPattern>
<SecDialTone attr=Both> Hide </SecDialTone>
<Timeouts attr=Both> Hide </Timeouts>
<CIDBlock attr=Both> Hide </CIDBlock>
<HuntGroup attr=Change> Hide </HuntGroup>
<NightSerBell attr=Change> Hide </NightSerBell>
<Time> Hide </Time>
</System>

<Function>
<AddLineToPhone> No </AddLineToPhone>
<DeleteLineFromPhone> No </DeleteLineFromPhone>
<MaxLinePerPhone> 4 </MaxLinePerPhone>

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</Function>
</Presentation>

Defining a Customer Administrator

You can define a login account for a customer administrator in either of the following two ways:
• Method 1: Using the Cisco CME GUI to Define a Customer Administrator, page 156
• Method 2: Using the Cisco IOS CLI to Define a Customer Administrator, page 157

Method 1: Using the Cisco CME GUI to Define a Customer Administrator

This method allows the system administrator to use the GUI itself to create a customer administrator
login account for the GUI.

SUMMARY STEPS

1. From the Configure System Parameters menu, choose Administrator’s Login Account.
2. Complete the Admin User Name (username), Admin User Type (Customer), and New Password
fields.
3. Click Change.

DETAILED STEPS

Step 1 From the Configure System Parameters menu, choose Administrator’s Login Account. You see the
screen shown in Figure 29 on page 157.
Step 2 Complete the Admin User Name (username), Admin User Type (Customer), and New Password fields
for the user that you are defining as a customer administrator. Type the password again to confirm it.
Step 3 Click Change for your changes to become effective.

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Figure 29 Using the Cisco CME GUI to Define a Customer Administrator

Method 2: Using the Cisco IOS CLI to Define a Customer Administrator

This method allows the system administrator to create a customer administrator account for the
Cisco CME GUI by using the Cisco IOS CLI.

SUMMARY STEPS

1. telephony-service
2. web admin customer name username password string
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 web admin customer name username password string Defines a username and password for a customer
administrator. The default username is Customer.
There is no default password.
Example:
Router(config-telephony)# web admin customer name • name username—Username of customer
user44 password pw10293847 administrator.
• password string—String to verify customer
administrator identity.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Setting Up GUI Access for Phone Users

You can enable GUI access for a phone user in either of the following two ways:
• Method 1: Using the Cisco CME GUI to Define a GUI Account for a Phone User, page 158
• Method 2: Using the Cisco IOS CLI to Define a GUI Account for a Phone User, page 160

Method 1: Using the Cisco CME GUI to Define a GUI Account for a Phone User
This method uses the Cisco CME GUI itself to create a GUI login account for a phone user.

SUMMARY STEPS

1. From the Configure Phones menu, choose Add Phone or Change Phone.
2. Enter a username and password in the Login Account area of the screen.
3. Click Change.

DETAILED STEPS

Step 1 From the Configure Phones menu, choose Add Phone to add GUI access for a user with a new phone or
Change Phone to add GUI access for a user with an existing phone. You see the Add Phone screen or
the Change Phone screen as shown in Figure 30 on page 159.
Step 2 Enter a username and password in the Login Account area of the screen. If you are adding a new phone,
complete the other fields as appropriate.

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Step 3 Click Change for your edits to become effective.

Figure 30 Using the Cisco CME GUI to Define a Phone User GUI Account

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What to Do Next

Method 2: Using the Cisco IOS CLI to Define a GUI Account for a Phone User
This method uses the Cisco IOS CLI to create a Cisco CME GUI login account for a phone user.

SUMMARY STEPS

1. ephone phone-tag
2. username username password password
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone tag Enters ephone configuration mode to register
Cisco IP phones.
Example:
Router(config)# ephone 2
Step 2 username username password password Assigns a phone user login account name and
password.
Example: • This allows individual phone users to log in to the
Router(config-ephone)# username prx password pk59wq Cisco CME router through a web interface to
change a limited number of personal settings.
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-telephony)# exit

What to Do Next
Once you have set up your Cisco CME system and phones, call transfer, call forwarding, and the
Cisco CME GUI, you can use the GUI to complete your specification of optional features. These tasks
are documented in online help in the GUI itself.
The remaining chapters in this guide explain how to configure the same optional features using the
Cisco IOS CLI on the Cisco CME router.

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This chapter describes features that are set up for all phone users systemwide in a Cisco CallManager
Express (Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Call Park, page 161
• Secondary Dial Tone, page 165
• Busy Timeout, page 166
• Interdigit Timeout, page 167
• Ringing Timeout, page 168
• Music on Hold, page 169

Call Park
Call park allows a phone user to place a call on hold at a special ephone-dn that is used as a temporary
parking spot from which the call can be retrieved by anyone on the system. In contrast, a call that is
placed on hold using the Hold button or Hold soft key can be retrieved only from the extension that
placed the call on hold. The special ephone-dn at which a call is parked is known as a call-park slot. A
call-park slot is a floating extension, or ephone-dn that is not bound to a physical phone, to which calls
are sent to be held.
Each call-park slot occupies one ephone-dn. During configuration, any number of ephone-dns can be
designated as call-park slots using the park-slot command, as long as the total number of park slots and
normal extensions does not exceed the maximum number of allowable ephone-dns for a system. After
an administrator defines at least one call-park slot and restarts phones, the Park soft key is displayed on
all the IP phones that are able to display soft keys.
Each call-park slot can hold one call at a time, so the number of simultaneous calls that can be parked is
equal to the number of slots that have been created in the Cisco CME system. In Cisco CME 3.2.1 and

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later releases, call-park slots can also be monitored. If a call-park slot is assigned to a monitor button
using the button m command, the line status shows "in use" when a call is parked in the monitored slot.
You can create a call-park slot that is reserved for use by one extension by assigning that slot a number
whose last two digits are the same as the last two digits of the extension. When an extension starts to
park a call, the system searches first for a call-park slot that has the same final two digits as the extension.
If no such call-park slot exists, the system chooses an available call-park slot.
Multiple call-park slots can be created with the same extension number so that more than one call can
be parked for a particular department or group of people at a known extension number. For example, at
a hardware store, calls for the plumbing department can be parked at extension 101, calls for lighting
can be parked at 102, and so forth. Everyone in the plumbing department knows that calls parked at 101
are for them and can pick up calls from extension 101. When multiple calls are parked at the same
call-park slot number, they are picked up in the order in which they were parked; that is, the call that has
been parked the longest is the first call picked up from that call-park slot number.
For cases where multiple call-park slots use the same extension number, you must configure each
ephone-dn that uses the extension number with the no huntstop command, except for the last ephone-dn
to which calls are sent. In addition, each ephone-dn must be configured with the preference command.
The preference numeric values must increase to match the order of the ephone-dns. That is, the lowest
ephone-dn tag park-slot must have the lowest numeric preference number, and so forth.
Without these configuration, all calls that are parked after a second call has been parked will generate a busy
signal. The caller who is being transferred will hear a busy signal, while the Cisco CME user who parked the
call will receive no indication that the call was lost.After at least one call-park slot has been defined and
phones have been restarted, phone users are able to park calls using the Park soft key. Users who attempt
to park a call at a busy slot hear a busy tone.
A phone user who parked a call can retrieve that call using the PickUp soft key and an asterisk (*). Phone
users other than the one who parked the call can retrieve the call by pressing the PickUp soft key and the
extension number of the call-park slot, which is available on their phone displays.
Directed call park allows calls to be transferred to a call-park-slot extension number using the Transfer
key; a transfer to a call-park slot is always a blind transfer. Calls can also be forwarded to a call-park
slot number, and callers can directly dial call-park slot numbers.
When a call that uses a G.711 codec is parked, the caller hears the music-on-hold (MOH) audio stream;
otherwise, callers hear tone on hold.
A reminder ring can be sent to the extension that parked the call by using the timeout keyword with the
park-slot command. The timeout keyword and argument set the interval length during which the
call-park reminder ring is timed out or inactive. If the timeout keyword is not used, no reminder ring is
sent to the extension that parked the call. The number of timeout intervals and reminder rings are
configured with the limit keyword and argument. For example, a limit of 3 timeout intervals sends 2
reminder rings (interval 1, ring 1, interval 2, ring 2, interval 3). The timeout and limit keywords and
arguments also set the maximum time that calls stay parked. For example, a timeout interval of 10
seconds and a limit of 5 timeout intervals (park-slot timeout 10 limit 5) will park calls for
approximately 50 seconds.
The reminder ring is sent only to the extension that parked the call unless the notify keyword is also used
to specify an additional extension number to receive a reminder ring. When an additional extension
number is specified using the notify keyword, the phone user at that extension can retrieve a call from
this slot by pressing the PickUp soft key and the asterisk (*) key.

SUMMARY STEPS

1. ephone-dn dn-tag
2. number number [secondary number] [no-reg [both | primary]]

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3. park-slot [timeout seconds limit count [notify extension-number [only]]]

4. exit
5. telephony-service
6. restart all
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 55
Step 2 number number [secondary number] [no-reg Configures a valid extension number for this ephone-dn instance.
[both | primary]]
• number—String of up to 16 digits that represents a telephone
or extension number to be associated with this ephone-dn.
Example:
Router(config-ephone-dn)# number 2345
• secondary—(Optional) Allows you to associate a second
telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should not
register with the H.323 gatekeeper. Unless you specify one of
the optional keywords (both or primary) after the no-reg
keyword, only the secondary number is not registered.
Step 3 park-slot [timeout seconds limit count Creates a floating extension (ephone-dn) at which calls can be
[notify extension-number [only]]] temporarily held (parked).
• timeout seconds—(Optional) Sets the call-park reminder
Example: timeout interval, in seconds. Range is from 0 to 65535. When
Router(config-ephone-dn)# park-slot the interval expires, the call-park reminder sends a 1-second
timeout 10 limit 10
ring and displays a message on the LCD panel of the Cisco IP
phone that parked the call and to any extension that may be
specified with the notify keyword. By default, the reminder
ring is sent only to the phone that parked the call.
• limit count—(Optional) Sets a limit for the number of
reminder time-out intervals and reminder rings for a parked
call. For example, a limit of 3 sends 2 reminder rings
(interval 1, ring 1, interval 2, ring 2, interval 3). When a limit
is set, a call parked at this slot is disconnected after the limit
has been reached. Range is from 1 to 65535.
• notify extension-number—(Optional) Sends a reminder ring
to the specified extension in addition to the reminder ring that
is sent to the phone that parked the call.
• only—(Optional) Sends a reminder ring only to the extension
specified with the notify keyword and does not send a
reminder ring to the phone that parked the call. This option
allows all reminder rings for parked calls to be sent to a
receptionist’s phone or an attendant’s phone, for example.

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Command or Action Purpose

Step 4 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 5 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 6 restart all Performs a fast reboot of all phones associated with this
Cisco CME router. Does not contact the DHCP or TFTP server
for updated information.
Example:
Router(config-telephony)# restart all Note The first time that call-park slots are defined, IP phones
must be rebooted before the Park soft key is displayed on
phones. This command is not required after subsequent
call-park slot definitions.
Step 7 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example creates a call-park slot with the number 1560. After a call is parked at this
number, the system provides 10 reminder rings at intervals of 30 seconds to the extension that parked
the call.
ephone-dn 50
number 1560
park-slot timeout 30 limit 10

Troubleshooting Tips

Use the show running-config command to verify the configuration.

Use the show ephone-dn park command to display configured call-park slots and their status, as shown
in the following example:
Router# show ephone-dn park

DN 50 (1560) park-slot state IDLE

Notify to () timeout 30 limit 10

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Secondary Dial Tone

Secondary Dial Tone

A secondary dial tone is available for Cisco IP phones that are running Cisco CME. The secondary dial
tone is generated when a phone user dials a predefined PSTN access prefix and terminates when
additional digits are dialed. An example is when a secondary dial tone is heard after the number 9 is
dialed to reach an outside line.

SUMMARY STEPS

1. telephony-service
2. secondary-dialtone digit-string
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 secondary-dialtone digit-string Activates a secondary dial tone when digit-string is dialed.
• digit-string—String of up to 32 digits that, when dialed,
Example: activates a secondary dial tone.Typical usage is that
Router(config-telephony)# secondary-dialtone 9 digit-string contains a single digit.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example sets the number 8 to trigger a secondary dial tone:
telephony-service
secondary-dialtone 8

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Busy Timeout

Busy Timeout
This task sets the timeout value for call transfers to busy destinations. The busy timeout value is the
amount of time that can elapse after a transferred call reaches a busy signal before the call is
disconnected.

SUMMARY STEPS

1. telephony-service
2. timeouts busy seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 timeouts busy seconds Sets the amount of time after which calls are disconnected
when they are transferred to busy destinations.
Example: • seconds—Number of seconds. Range is from 0 to 30.
Router(config-telephony)# timeouts busy 20 Default is 10.
Note This command sets the busy timeout only for calls
that are transferred to busy destinations and not for
calls that directly dial busy destinations.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example sets a timeout of 20 seconds for calls that are transferred to busy destinations:
telephony-service
timeouts busy 20

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Interdigit Timeout

Interdigit Timeout
This task configures the interdigit timeout value for all Cisco IP phones. The interdigit timeout is the
amount of time that can elapse between the dialing of digits before the dialing process times out and is
terminated.

SUMMARY STEPS

1. telephony-service
2. timeouts interdigit seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 timeouts interdigit seconds Configures the interdigit timeout value for all
Cisco IP phones attached to the router.
Example: The interdigit timeout specifies the number of seconds that
Router(config-telephony)# timeouts interdigit the system waits after the caller has entered the initial digit
30 or a subsequent digit of the dialed string. If the timeout ends
before the destination is identified, a tone sounds and the
call ends. This value is important when using
variable-length dial-peer destination patterns (dial plans).
For more information, refer to Dial Peer Configuration on
Voice Gateway Routers.
• seconds—Number of seconds before the interdigit
timer expires. Range is from 2 to 120. Default is 10.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony-service)# exit

Example

The following example sets an interdigit timeout of 30 seconds:

telephony-service
timeouts interdigit 30

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Ringing Timeout
The ringing timeout is the amount of time a phone can ring with no answer before returning a disconnect
code to the caller. This timeout is used only for extensions that do not have no-answer call forwarding
enabled. The ringing timeout prevents hung calls received over interfaces such as FXO that do not have
forward-disconnect supervision.

SUMMARY STEPS

1. telephony-service
2. timeouts ringing seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 timeouts ringing seconds Sets the duration, in seconds, for which the Cisco CME
system allows ringing to continue if a call is not answered.
Range is from 5 to 60000. Default is 180.
Example:
Router(config-telephony)# timeouts ringing 30
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example sets the ringing timeout default to 30 seconds:

telephony-service
timeouts ringing 30

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Music on Hold

Music on Hold
Music on hold (MOH) is an audio stream that is played to PSTN and VoIP G.711 or G.729 callers who
are placed on hold by phones in a Cisco CME system. This audio stream is intended to reassure callers
that they are still connected to their calls. MOH is not played to local Cisco CME phones that are on hold
with other Cisco CME phones; these parties hear a periodic repeating tone instead.
The audio stream that is used for MOH can derive from one of two sources: an audio file or a live feed.
If both are configured concurrently on the Cisco CME router, the router seeks the live feed first. If the
live feed is found, it displaces the audio file source. If the live feed is not found or fails at any time, the
router falls back to the audio file source that was specified for MOH during configuration.
If the MOH audio stream is also identified as a multicast source, the Cisco CME router additionally
transmits the stream on the physical IP interfaces of the Cisco CME router that you specify during
configuration, which permits external devices to have access to it.
A MOH audio stream from an audio file is supplied from an .au or .wav file held in router flash memory.
A MOH audio stream from a live feed is supplied from a standard line-level audio connection that is
directly connected to the router through an FXO or “ear and mouth” (E&M) analog voice port. The
live-feed feature is typically used to connect to a CD jukebox player. Only one live MOH feed is
supported per system.
When the phone receiving MOH is part of a system that uses G.729, transcoding is required between
G.711 and G.729. The G.711 MOH must be translated to G.729. Note that because of compression, MOH
using G.729 is of significantly lower fidelity than MOH using G.711. For information about transcoding,
refer to the “Transcoding Between G.729 and G.711” chapter of this guide.
Configuration of MOH is explained in the following sections:
• Configuring Music on Hold from an Audio File, page 169
• Configuring Music on Hold from a Live Feed, page 172

Configuring Music on Hold from an Audio File

This section explains how to configure music on hold when you are using a file to supply the audio
stream. You can also optionally specify that this audio stream should be multicast on router interfaces.
If MOH from an audio file and MOH from a live feed are both configured on the Cisco CME router, the
router seeks the live feed first. If a live feed is found, it displaces an audio file source. If the live feed is
not found or fails at any time, the router falls back to the audio file source.
To change the audio file to a different file, you must remove the first file using the no moh command
before specifying a second file, as shown in the following example:
Router(config-telephony)# no moh file1
Router(config-telephony)# moh file2

If you configure a second file without removing the first file, the MOH mechanism stops working and
may require a router reboot to clear the problem.

Note If the phones receiving MOH are part of a system that uses G.729, transcoding is required between G.711
and G.729. For information about transcoding, refer to the “Transcoding Between G.729 and G.711”
chapter of this guide.

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Prerequisites
A music file must be in stored in the router’s flash memory. This file should be in G.711 format. The file
can be in .au or .wav file format, but the file format must contain 8-bit 8-kHz data; for example, ITU-T
A-law or mu-law data format.

Restrictions
• MOH is supplied only to PSTN and VoIP G.711 or G.729 calls. Local IP phone callers hear a
repeating tone on hold for reassurance that they are still connected.
• IP phones do not support multicast at 224.x.x.x addresses.
• The volume level of a MOH file cannot be adjusted through the Cisco IOS software, so it cannot be
changed once the file is loaded into the flash memory of the router. To adjust the volume level of a
MOH file, edit the file in an audio editor prior to downloading the file to router flash memory.

SUMMARY STEPS

1. telephony-service
2. moh filename
3. multicast moh ip-address port port-number [route ip-address-list]
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 moh filename Configures music on hold using the specified file.
• filename—Source of the audio stream for MOH.
Example: Note If you specify a filename with this command
Router(config-telephony)# moh minuet.au
and later want to use a different file, you must
disable use of the first file with the no moh
command before configuring the second file.

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Command or Action Purpose

Step 3 multicast moh ip-address port port-number [route (Optional) Specifies that the MOH audio stream
ip-address-list] should also be multicast as specified.
• ip-address—Specifies that this audio stream is to
Example: be used for multicast as well as for MOH, and
Router(config-telephony)# multicast moh 239.10.16.4 specifies the destination IP address for multicast.
port 2123 route 10.10.29.17 10.10.29.33
• port port-number—Media port for multicast.
Range is from 2000 to 65535. Port 2000 is
recommended because it is already used for
normal RTP media transmissions between IP
phones and the router.
• route—(Optional) Specifies a list of explicit
router interfaces on which to transmit the IP
multicast packets.
• ip-address-list—(Optional) List of up to four
explicit routes for multicast MOH. The default is
that the MOH multicast stream is automatically
output on the interfaces that correspond to the
address that was configured with the ip
source-address command.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Examples

The following example enables music on hold and specifies the music file to use:
telephony-service
moh minuet.wav

The following example enables MOH and additionally specifies a multicast address for the audio stream:
telephony-service
moh minuet.wav
multicast moh 239.23.4.10 port 2000

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Configuring Music on Hold from a Live Feed

To configure MOH from a live feed, you establish a voice port and dial peer for the call and also create
a “dummy” ephone-dn. The ephone-dn must have a phone or extension number assigned to it so that it
can make and receive calls, but the number is never assigned to a physical phone.
The recommended interface for live-feed MOH is an analog E&M port because it requires the minimum
number of external components. You connect a line-level audio feed (standard audio jack) directly to
pins 3 and 6 of an E&M RJ-45 connector. The E&M voice interface card (VIC) has a built-in audio
transformer that provides appropriate electrical isolation for the external audio source. (An audio
connection on an E&M port does not require loop-current). The signal immediate and
auto-cut-through commands disable E&M signaling on this voice port. A G.711 audio packet stream is
generated by a digital signal processor (DSP) on the E&M port.
If you are using an FXO voice port for live-feed MOH instead of an E&M port, connect the MOH source
to the FXO voice port. This connection requires an external adapter to supply normal telephone company
(telco) battery voltage with the correct polarity to the tip and ring leads of the FXO port. The adapter
must also provide transformer-based isolation between the external audio source and the tip and ring
leads of the FXO port.
Music from a live feed is continuously fed into the MOH playout buffer instead of being read from a
flash file, so there is typically a 2-second delay. An outbound call to a MOH live-feed source is attempted
(or reattempted) every 30 seconds until the connection is made by the directory number that has been
configured for MOH. If the live-feed source is shut down for any reason, the flash memory source will
be automatically activated.
A live-feed MOH connection is established as an automatically connected voice call that is made by the
Cisco CME MOH system itself or by an external source directly calling in to the live-feed MOH port.
An MOH call can be from or to the PSTN or can proceed via VoIP with voice activity detection (VAD)
disabled. The call is assumed to be an incoming call unless the optional out-call keyword is used with
the moh command during configuration.
The Cisco CME router uses the audio stream from the call as the source for the MOH stream, displacing
any audio stream that is available from a flash file. An example of an MOH stream received over an
incoming call is an external H.323-based server device that calls the ephone-dn to deliver an audio
stream to the Cisco CME router.

Note If the phones receiving MOH are part of a system that uses G.729, transcoding is required between G.711
and G.729. For information about transcoding, refer to the “Transcoding Between G.729 and G.711”
chapter of this guide.

Restrictions
• An FXO port can be used for a live feed if the port is supplied with an external third-party adapter
to provide a battery feed.
• An foreign exchange station (FXS) port cannot be used for a live feed.
• For a live feed from VoIP, VAD must be disabled.
• MOH is supplied only to PSTN and VoIP G.711 or G.729 calls. Local IP phone callers hear a
repeating tone on hold for reassurance that they are still connected.

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SUMMARY STEPS

1. voice-port port
2. input gain decibels
3. auto-cut-through (E&M only)
4. operation 4-wire (E&M only)
5. signal immediate (E&M only)
6. exit
7. dial peer voice tag pots
8. destination-pattern string
9. port port
10. exit
11. ephone-dn dn-tag
12. number number
13. moh [out-call outcall-number] [ip ip-address port port-number [route ip-address-list]]
14. exit

DETAILED STEPS

Command or Action Purpose

Step 1 voice-port port Enters voice-port configuration mode. To find the correct
definition of the port argument for your router, refer to the
Cisco IOS Voice Command Reference.
Example:
Router(config)# voice-port 1/1/0
Step 2 input gain decibels Specifies, in decibels, the amount of gain to be inserted at
the receiver side of the interface. Acceptable values are
integers from –6 to 14.
Example:
Router(config-voice-port)# input gain 0
Step 3 auto-cut-through (E&M ports only) Enables call completion when a PBX
does not provide an M-lead response. MOH requires that
you use this command with E&M ports.
Example:
Router(config-voice-port)# auto-cut-through
Step 4 operation 4-wire (E&M ports only) Selects the 4-wire cabling scheme. MOH
requires that you specify 4-wire operation with this
command for E&M ports.
Example:
Router(config-voice-port)# operation 4-wire
Step 5 signal immediate (E&M ports only) For E&M tie trunk interfaces, directs the
calling side to seize a line by going off-hook on its E-lead
and to send address information as dual tone multifrequency
Example:
Router(config-voice-port)# signal immediate
(DTMF) digits.

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Command or Action Purpose

Step 6 exit Exits voice-port configuration mode.

Example:
Router(config-voice-port)# exit
Step 7 dial peer voice tag pots Enters dial-peer configuration mode.

Example:
Router(config)# dial peer voice 7777 pots
Step 8 destination-pattern string Specifies either the prefix or the full E.164 telephone
number to be used for a dial peer.
Example:
Router(config-dial-peer)# destination-pattern
7777
Step 9 port port Associates the dial peer with the voice port that was
specified in Step 1.
Example:
Router(config-dial-peer)# port 1/1/0
Step 10 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 11 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies this
Example: ephone-dn during configuration tasks. Range is from 1
Router(config)# ephone-dn 55 to 288.
Step 12 number number Configures a valid extension number for this ephone-dn
instance. This number is not assigned to any phone; it is
only used to make and receive calls that contain an audio
Example:
Router(config-ephone-dn)# number 5555
stream to be used for MOH.
• number—String of up to 16 digits that represents a
telephone or extension number to be associated with
this ephone-dn.

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Command or Action Purpose

Step 13 moh [out-call outcall-number] [ip ip-address Specifies that this ephone-dn is to be used for an incoming
port port-number [route ip-address-list]] or outgoing call that is to be the source for an MOH stream.
If this command is used without the out-call keyword, the
Example: MOH stream is received from an incoming call.
Router(config-ephone-dn)# moh out-call 7777 ip • out-call outcall-number—(Optional) Indicates that the
239.10.16.8 port 2311 route 10.10.29.3
router is calling out for a live feed that is to be used for
10.10.29.45
MOH and specifies the number to be called. Forces a
connection to the local router voice port that was
specified in Step 1.
• ip ip-address—(Optional) Indicates that this audio
stream is to be used as a multicast source as well as for
MOH, and specifies the destination IP address for
multicast.
Note If you specify a multicast address with this
command and a different multicast address with the
multicast moh command under telephony-service
configuration mode, you can send the MOH audio
stream to two multicast addresses.

• port port-number—(Optional) Media port for

multicast. Range is from 2000 to 65535. Port 2000 is
recommended because it is already used for RTP media
transmissions between IP phones and the router.
• route ip-address-list—(Optional) Indicates specific
router interfaces on which to transmit the IP multicast
packets. Up to four IP addresses can be listed. The
default is that the MOH multicast stream is
automatically output on the interfaces that correspond
to the address that was configured with the ip
source-address command.
Step 14 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

The following example enables MOH from an outgoing call on voice port 1/1/0 and dial peer 7777:
voice-port 1/1/0
auto-cut-through
operation 4-wire
signal immediate
!
dial-peer voice 7777 pots
destination-pattern 7777
port 1/1/0
!
ephone-dn 55
number 5555
moh out-call 7777

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Troubleshooting Tips

The following commands can help troubleshoot live-feed MOH:

• debug ephone moh
• show ephone summary—Note that this command provides extended output if the debug ephone
moh command is enabled.

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This chapter describes optional features that affect individual IP phone use in a Cisco CallManager
Express (Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Related Features, page 177
• Dial Features, page 178
• Speed-Dial Features, page 179
• Automatic Line Selection, page 187
• Call-Transfer and Call-Forward Features, page 189
• Do Not Disturb Features, page 191
• Conference Call Features, page 194
• Configurable Phone Displays and Sounds, page 197
• Direct FXO Trunk Lines, page 227
• Cisco IP Phone 7970G and 7971G-GE Settings, page 232

Related Features
• A configurable label supplies text that is displayed next to a line button for an ephone-dn. See the
“Label Support” section in the “Configuring an Attendant for Primary Call Coverage” chapter.
• Call pickup and call pickup groups are described in the “Configuring Secondary Call Coverage”
chapter.
• Paging and intercom features are described in the “Configuring Productivity Tools” chapter.

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Dial Features
This section describes the following dial features:
• Callback Busy Subscriber, page 178
• On-Hook Dialing, page 178

Callback Busy Subscriber

This feature allows callers who dial a busy extension number to request a callback from the system when
a called number that was busy is free. Callers can also request callbacks for extensions that do not
answer, and the system will notify them after the called phone is next used.
This feature is available only on the Cisco Wireless IP Phone 7920, Cisco IP Phones 7940 and 7940G,
and Cisco IP Phones 7960 and 7960G.
To activate the callback feature after dialing an extension number and hearing a busy or no-answer tone,
press the CallBack soft key. When the system calls back to notify you that the called line is free, the
phone rings once, briefly, and displays “CallBack Redial” and the extension number. To call the number,
press the Redial soft key.
There can be only one callback request pending against a particular extension number, although one
caller may initiate a number of callbacks to different numbers. If a caller attempts to place a callback
request on a number that already has a pending callback request, the caller hears a fast-busy tone.
If the called number has call forwarding enabled, the callback request is placed against the final
destination number.
No configuration is required for this feature. The show ephone-dn callback command lists phones that
have pending callback requests against them.

On-Hook Dialing
On-hook dialing allows you to enter dialed digits with the phone on hook and the handset still in its
cradle. Digits appear in the phone display as they are dialed, and a Backspace soft key (<<) allows you
to erase digits that are entered incorrectly. When you have finished entering the digits and want the phone
to dial the number, use one of the following methods:
• Press a line button or the Dial soft key if you are using the speakerphone or a headset.
• Pick up the handset.
No configuration is required to activate this feature.

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Speed-Dial Features
Cisco CME provides a number of mechanisms to implement speed dialing on IP and analog phones.
Table 6 provides a summary and comparison of the different types of speed dial that are available in
Cisco CME systems.

Table 6 Types of Speed Dial Available in Cisco CME

Type of Speed Dial Availability of Numbers Description How Configured

Monitor-Line Speed-dial entries are local to a IP phone buttons that are Monitor lines must be set up by
Button Speed Dial specific IP phone. configured as monitor lines can administrators.
be used to speed-dial the line that
There can be up to the number of
is being monitored.
monitor lines on a phone
Local Speed Dial Speed-dial entries are Users invoke entries from the Numbers are set up by
Menu systemwide. Directories > Local Speed Dial administrators using an XML file
menu on IP phones. called speeddial.xml.
There can be up to 32 numbers.
Personal Speed Speed-dial entries are local to a Users invoke entries from the Numbers are set up by
Dial Menu specific IP phone. Directories > Local Services > administrators using the fastdial
Personal Speed Dials menu on IP command in ephone
There can be up to 24 numbers
phones. configuration mode.
per phone.
Speed-Dial There can be a total of up to 99 For IP phones, the first entries Local speed-dial codes from 1 to
Buttons and speed-dial codes per phone. A that are set up occupy any unused 33 are defined by administrators
Abbreviated maximum of 33 codes can be line buttons and are invoked when using the speed-dial command in
Dialing local to a particular phone. The a user presses one of these line ephone mode.
remainder are used for buttons. Subsequent entries are Systemwide codes from 1 to 99
systemwide speed-dial codes. invoked when a phone user dials are defined by administrators
the speed-dial code (tag) and the
Note Entries from 34 to 99 are using the directory-entry
Abbr soft key.
available only in command in telephony-service
Cisco IOS Analog phone users invoke speed configuration mode.
Release 12.3(11)XL, dial by entering an asterisk and
If the same code has been defined
12.3(14)T, and later the code (tag) number of the for both local and systemwide
releases . desired entry. speed dial, the local definition
takes precedence. To prevent this
conflict, use only codes 34
through 99 for systemwide
speed-dial numbers.

Monitor-Line Button Speed Dial

For Cisco CME 3.2 and later versions, a monitor-line button can be used to speed-dial the monitor line’s
number. A monitor line is a line that is shared by two people. Only one person can make and receive calls
on the shared line at a time, while the other person, whose line is in monitor mode, is able to see that the
line is in use. Speed dialing is available when monitor lines’ lamps are off, indicating that the line is not
in use. For example, an assistant who wants to talk with a manager can press an unlit monitor-line button
to speed-dial the manager’s number.
A monitor-line lamp can be off or unlit only when its line is in the idle call state. The idle state occurs
before a call is made and after a call has been completed. For all other call states, the monitor line lamp
is on or lit.

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The following example shows a monitor-line configuration. Extension 2311 is the manager’s line, and
ephone 1 is the manager’s phone. The manager’s assistant monitors extension 2311 on button 2 of
ephone 2. When the manager is on the line, the lamp is lit on the assistant’s phone. If the lamp is not lit,
the assistant can speed-dial the manager by pressing button 2.
ephone-dn 11
number 2311

ephone-dn 22
number 2322

ephone 1
button 1:11

ephone 2
button 1:22 2m11

Local Speed Dial Menu

The local speed-dial feature provides a systemwide list of frequently called numbers that can be
programmed by an administrator on all phones, up to a maximum of 32 entries. Local speed-dial
numbers are available from the Directories button on a phone.
To define local speed-dial numbers, a system administrator first creates an XML file called
speeddial.xml, similar to the example shown in the “speeddial.xml File Example” section on page 180.
After the administrator places the speeddial.xml file in the Cisco CME router’s flash memory, the local
speed-dial menu appears when Local Speed Dial is selected from the Directories menu.

Prerequisites
• In any text editor, create a file called speeddial.xml in the Cisco-specified directory DTD format.
Use the keywords and format shown in the “speeddial.xml File Example” section on page 180 to
specify names and numbers for a local speed-dial list. For more information about Cisco DTD
formats, refer to Cisco IP Phone Services Application Development Notes.
• Copy the file to the TFTP server application on the Cisco CME router.

speeddial.xml File Example

<CiscoIPPhoneDirectory>
<Title>Local Speed Dial</Title>
<Prompt>Record 1 to 1 of 1 </Prompt>

<DirectoryEntry>
<Name>Security</Name>
<Telephone>71111</Telephone>
</DirectoryEntry>

<DirectoryEntry>
<Name>Marketing</Name>
<Telephone>71234</Telephone>
</DirectoryEntry>

<DirectoryEntry>
<Name>Tech Support</Name>
<Telephone>71432</Telephone>
</DirectoryEntry>

</CiscoIPPhoneDirectory>

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SUMMARY STEPS

1. copy tftp flash

2. configure terminal
3. ip http server
4. ip http path flash:

DETAILED STEPS

Command or Action Purpose

Step 1 copy tftp flash Copies the file from the TFTP server to the router flash
memory.
Example: • At the first prompt, enter the IP address or the DNS
Router# copy tftp flash name of the remote host.
Address or name of remote host []? 172.24.59.11 • At both filename prompts, enter speeddial.xml.
Source filename []? speeddial.xml
• At the prompt to erase flash, enter no.
Destination filename [speeddial.xml]?
Accessing tftp://172.24.59.11/speeddial.xml...
Erase flash:before copying? [confirm]n
Loading speeddial.xml from 172.24.59.11 (via
FastEthernet0/0):!
[OK - 329 bytes]

Verifying checksum... OK (0xF5DB)

329 bytes copied in 0.044 secs (7477 bytes/sec)
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ip http server Enables the Cisco web-browser user interface on the router.

Example:
Router(config)# ip http server
Step 4 ip http path flash: Sets the base HTTP path to flash memory.

Example:
Router(config)# ip http path flash:

Example

The following example enables the Cisco web browser and sets the HTTP path to flash memory so that
the speeddial.xml file in flash memory is accessible to IP phones:
ip http server
ip http path flash:

Troubleshooting Tips

Use the debug ephone detail command to diagnose problems with local speed-dial numbers.

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Personal Speed Dial Menu

The personal speed-dial menu feature allows you to program up to 24 personal speed-dial numbers per
IP phone. This feature is available only on Cisco IP Phones 7940, 7960, 7960G, 7970G and 7971G-GE.
Personal speed-dial numbers appear as menu entries displayed from the Directories > Local Services >
Personal Speed Dials option on the phone. Personal speed-dial entries are displayed in the order in which
they were entered.

SUMMARY STEPS

1. ephone phone-tag
2. fastdial dial-tag number name name-string
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number for the phone for
Example: which you want to program personal speed-dial
Router(config)# ephone 1 numbers.
Step 2 fastdial dial-tag number name name-string Creates an entry for a personal speed-dial number on this IP
phone.
Example: • dial-tag—Unique identifier to identify this entry during
Router(config-ephone)# fastdial 1 5552 name configuration. Range is from 1 to 24.
Sales
• number—Telephone number or extension to be dialed.
• name name-string—Label to appear in the Personal
Speed Dial menu containing a string of up to 24
alphanumeric characters.
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example creates a directory of three personal speed-dial listings for one IP phone:
ephone 1
fastdial 1 5489 name Marketing
fastdial 2 12125550155 name Sales-NY
fastdial 3 12135550112 name Sales-LA

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Speed-Dial Buttons and Abbreviated Dialing

In a Cisco CME system, each phone can have up to 33 local speed-dial numbers (codes 1 through 33),
up to 99 systemwide speed-dial numbers (codes 1 through 99), or a combination of the two. If you
program both a local and a systemwide speed-dial number with the same code (tag number), the local
number takes precedence. In most cases you will want to reserve codes 1 through 33 for local, per-phone
speed-dial numbers and use codes 34 through 99 for systemwide speed-dial numbers so that there is no
conflict.
On an IP phone, a speed-dial number can be on a speed-dial button or it can be an abbreviated code that
is dialed from the keypad. On an analog phone, a speed-dial number can only be an abbreviated code.
Each speed-dial definition contains a speed-dial code (an identifier also known as a tag) and usually the
full extension or telephone number that should be dialed when the code is used at the phone. However,
local speed-dial entries on IP phone buttons can be defined without telephone numbers so that phone
users can enter whatever numbers they want from the phone. Local speed-dial definitions can also be
created with locked numbers that cannot be changed from the phone. Systemwide speed-dial definitions
cannot be changed from the phone.

Using Speed-Dial Buttons and Abbreviated Dialing

Speed-dial buttons and abbreviated dialing are accessed in different ways from IP phones and from
analog phones, as described in the following sections:
• IP Phones
• Analog Phones

Note On-hook abbreviated dialing is limited to Cisco IP Phones 7905G, 7912G, 7920G, 7970G, and
7971G-GE.

IP Phones
IP phone buttons that are not used for extensions are automatically populated with local speed-dial
definitions when they exist for the phone on which the buttons appear. Speed-dial definitions are
assigned to phone buttons in the order of their code (tag) numbers. For example, if you define
speed-dial 1, it is assigned to the first phone button that is available. If you have used two buttons for
extensions on this phone, speed-dial 1 is assigned to the third physical button. When you define
speed-dial 2, it is assigned to the fourth physical button on the phone, and so on.
If more speed-dial definitions are created than are supported by the IP phone setup, the extra speed-dial
definitions can be dialed from IP phones using the following procedure for abbreviated dialing:
1. Press the one- or two-digit speed-dial code (tag number) and the Abbr soft key. The phone dials the
full telephone number associated with the speed-dial tag in speakerphone (hands-free) mode.
2. Pick up the handset or activate the headset to transition to handset mode.
Note that prior to Cisco IOS Releases 12.3(11)XL and 12.3(14)T, speed-dial entries that were in excess
of the number of physical phone buttons available were ignored by IP phones.

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Analog Phones
Analog phone users who use a Cisco ATA-186, Cisco ATA-188, or Cisco VG 224 to connect to a
Cisco CME system use a different method to access speed-dial numbers. To dial a speed-dial number
from an analog phone, use the following procedure for abbreviated dialing:
• Press the asterisk (*) key and the two-digit speed-dial code (tag number) of the desired speed-dial
number. For instance, press *01 to speed-dial the number that has been programmed as speed-dial 1
on that ephone.
Note that prior to Cisco IOS Releases 12.3(11)XL and 12.3(14)T, analog phones were limited to nine
speed-dial numbers.

Configuring Speed-Dial Buttons and Abbreviated Dialing

The following tasks are contained in this section:

• Defining Speed-Dial Buttons and Abbreviated Dialing Codes from the Router CLI, page 184
• Changing Existing Speed-Dial Button Information from an IP Phone, page 186

Defining Speed-Dial Buttons and Abbreviated Dialing Codes from the Router CLI
Speed-dial definitions can be added or modified by an administrator from the Cisco CME router CLI.
Local speed dial numbers on IP phones can be locked so that they cannot be changed from the phone or
they can be defined as empty so that the phone user can enter a number to be dialed. Changes that are
made to speed-dial button definitions are saved into the router nonvolatile random-access memory
(NVRAM) configuration after a timer-based delay.
A speed-dial definition consists of a unique identifier (speed-tag), a number to dial, and an optional
label. Local speed-dial definitions are automatically assigned to any IP phone buttons that remain unused
after all the assigned extensions have been associated with buttons. Definitions are assigned in the order
of their speed-dial identifier (speed-tag) numbers. Note that these identifier numbers are not related to
the physical button layout of the phone.

Note The speed-dial command is used to enter speed-dial definitions that are local to a particular phone, and
the directory entry command is used to enter systemwide speed-dial definitions. If the same identifier
(tag) is used in a local definition and in a systemwide definition, the local definition takes precedence.

Restrictions

On-hook abbreviated dialing using the Abbr soft key is supported only on the following phone types:
• Cisco IP Phone 7905G
• Cisco IP Phone 7912G
• Cisco IP Phone 7920G
• Cisco IP Phone 7970G
• Cisco IP Phone 7971G-GE

SUMMARY STEPS

1. ephone phone-tag
2. speed-dial speed-tag digit-string [label label-text]
3. Repeat Step 2 to make additional speed-dial definitions on this phone.

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4. restart
5. exit
6. telephony-service
7. directory entry {directory-tag number name name | clear}

DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone on which you are adding speed-dial
Router(config)# ephone 55 capability.
Step 2 speed-dial speed-tag digit-string [label Defines a unique speed-dial identifier, a digit string to dial,
label-text] and an optional label to display next to the button.
• speed-tag—Identifier for a speed-dial definition. Range
Example: is from 1 to 33.
Router(config-ephone)# speed-dial 1 +5001 label
“Head Office” • digit-string—Digits to be dialed when the speed-dial
button is pressed. If the first character of this string is
the plus sign (+), this speed-dial number is locked and
cannot be changed at the phone. If the only character in
this string is a pound sign (#), an empty speed-dial
definition is defined with no speed-dial number
attached to it. An empty speed-dial definition can be
programmed at the phone.
• label label-text—Keyword and string that contains
identifying text to be displayed next to the speed-dial
button. Enclose the string in quotation marks if it
includes spaces. Maximum length is 32 characters, but
the phone is only able to display 16 characters.
Step 3 Repeat Step 2 to make additional speed-dial
definitions on this phone.
Step 4 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-ephone)# restart
Step 5 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone)# exit
Step 6 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 7 directory entry {{directory-tag number name Adds a systemwide directory and speed-dial definition.
name} | clear}
• directory-tag—Digit string that provides a unique
identifier for this entry. Range is from 1 to 99. Tags 1
Example: through 33 can be overridden by tags with the same
Router(config-telephony)# directory entry 45 number that are configured using the speed-dial
8185550143 name Corp Acctg
command.
• number—String of up to 32 digits that provides the full
telephone number for this entry.
• name name—String of up to 24 characters that
provides a name for this entry.
• clear—Removes all directory entries that were made
with this command.

Example

The following example defines two locked speed-dial numbers with labels to appear next to the
speed-dial buttons on ephone 1. These speed-dial definitions are assigned to the next empty buttons after
all extensions have been assigned. For instance, if two extensions are assigned on the Cisco IP
Phones 7960 and 7960G, these speed-dial definitions appear on the third and fourth buttons.
The example also defines two systemwide speed-dial numbers with the directory entry command. One
is a local extension and the other is a ten-digit telephone number.
ephone 1
mac-address 1234.5678.ABCD
button 1:24 2:25
speed-dial 1 +5002 label Receptionist
speed-dial 2 +5001 label Security

telephony-service
directory entry 34 5003 name Accounting
directory entry 45 8185550143 name Corp Acctg

Changing Existing Speed-Dial Button Information from an IP Phone

Once an administrator has created one or more unlocked speed-dial definitions that appear on buttons on
an IP phone, a phone user can reprogram the buttons by using the following steps.

SUMMARY STEPS

1. Select an available phone line.

2. Press the pound key (#).
3. Press the speed-dial button that you want to program.
4. Enter the full telephone number to be dialed when the button is pressed.
5. Press the speed-dial button that you are programming a second time.
6. Hang up the receiver or press a new speed-dial button.

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Automatic Line Selection

DETAILED STEPS

Step 1 Select an available phone line. Lift the handset, press the NewCall soft key, or press a button. Listen for
the dial tone. Note that the Newcall soft key must not be disabled for the Cisco IP Phone 7905G and the
Cisco IP Phone 7912G.
Step 2 Press the pound key (#).
Step 3 Press the speed-dial button that you want to program. A short beep confirms that you are starting to
program this button.
Step 4 Enter the full telephone number to be dialed when the button is pressed. The digits are output to the
phone display. When speed-dial numbers are entered on a Cisco IP Phones 7940 and 7940G or Cisco IP
Phones 7960 and 7960G, the Backspace soft key (<<) is available to let you correct digits that were typed
incorrectly. To remove a speed-dial number without replacing it with a new one, press the pound key (#).
Step 5 Press the speed-dial button that you are programming a second time to indicate that you have finished
entering the speed-dial digits and to store the new speed-dial number.
Step 6 Hang up the receiver or press a new speed-dial button, and repeat the process.

Automatic Line Selection

For a multiline phone, it is often useful to set up the phone so that lifting the handset automatically
selects the first ringing line on the phone or, if no line is ringing, selects the first available idle line on
the phone. This is the behavior of all multiline IP phones before Cisco CME 3.0.
Under some circumstances, however, you might want to specify that a line button must be explicitly
pressed to select an outgoing line or to answer an incoming call. Starting in Cisco CME 3.0, you have
the flexibility to assign the type of line selection that each IP phone employs.
Any of the following behaviors can be assigned on a per-phone basis:
• Automatic line selection—Picking up the handset answers the first ringing line or, if no line is
ringing, selects the first idle line. Use the auto-line command with no keyword or argument.
• Manual line selection (no automatic line selection)—Pressing the Answer soft key answers the first
ringing line, and pressing a line button selects a line for an outgoing call. Picking up the handset
does not answer calls or provide dial tone. Use the no auto-line command.
• Automatic line selection for incoming calls only—Picking up the handset answers the first ringing
line, but if no line is ringing, it cannot select an idle line for an outgoing call. Pressing a line button
selects a line for an outgoing call. Use the auto-line incoming command.
• Automatic line selection for outgoing calls only—Picking up the handset selects the line that has
been configured for this purpose with the auto-line command and the button-number argument in
Cisco CME 3.1 and later versions. If a button number has been specified and the line associated with
that button is unavailable (because it is a shared line in use on another phone), no dial tone is heard
when the handset is lifted and you must press an available line button to make an outgoing call. This
feature does not apply to incoming calls, which must be answered by pressing the Answer soft key
or a ringing line button.

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Automatic Line Selection

SUMMARY STEPS

1. ephone phone-tag
2. [no] auto-line [incoming] [button-number]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number for the phone on
Example: which you want to configure automatic line selection.
Router(config)# ephone 24
Step 2 [no] auto-line [incoming] [button-number] Assigns a type of line selection behavior to this phone.
• auto-line—Picking up a handset answers the first
Example: ringing line or, if no line is ringing, selects the first idle
Router(config-ephone)# no auto-line line. This is also the default if this command is not
used.
• no auto-line—Pressing the Answer soft key answers
the first ringing line, and pressing a line button selects
a line for an outgoing call. Picking up the handset does
not answer calls or provide dial tone.
• auto-line incoming—Picking up the handset answers
the first ringing line but, if no line is ringing, does not
select an idle line for an outgoing call. Pressing a line
button selects a line for an outgoing call.
• auto-line button-number—For Cisco CME 3.1 and
later versions, picking up the handset selects the line
associated with the that has been specified with this
command during configuration. The default if this
argument is not used is the topmost available line.
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example assigns no automatic line selection to phones 1 and 2 and assigns automatic line
selection for incoming calls only to phone 3:
ephone 1
mac-address 00e0.8646.9242
button 1:1 2:4 3:16
no auto-line

ephone 2
mac-address 01c0.4612.7142
button 1:5 2:4 3:16
no auto-line

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Call-Transfer and Call-Forward Features

ephone 3
mac-address 10b8.8945.3251
button 1:6 2:4 3:16
auto-line incoming

Call-Transfer and Call-Forward Features

This section describes the following call-transfer and call-forward features:
• Consult Transfer Support for Direct Station Select, page 189
• CFwdAll Soft-Key Restriction Control, page 190
• Do Not Disturb Features, page 191

Consult Transfer Support for Direct Station Select

For CME 3.2 and later versions, consult transfers can take place during direct station select. The default
behavior is a blind transfer. Direct station select is a functionality that allows phone users to transfer calls
to idle monitor lines. A monitor line is a line that is shared by two people. Only one person can make
and receive calls on the shared line at a time, while the other person, whose line is in monitor mode, is
able to see that the line is in use.
Consult transfers to monitor lines are performed with the following steps:

Step 1 Answer the incoming call.

Step 2 Press the Trnsfer (transfer) soft key.
Step 3 If the monitor lamp is off, press the monitor-line button.
Step 4 Announce the call.
Step 5 Place the handset on hook or press the Trnsfer soft key a second time to transfer the call.

If the person sharing the monitor line does not want to accept the call, the person announcing the call
can reconnect to the incoming call by pressing the EndCall soft key to terminate the announcement call
and pressing the Resume soft key to reconnect to the original caller.
Direct station select consult transfer is enabled with the addition of the dss keyword to the
transfer-system full-consult command, which defines the call transfer method for all lines served by
the router.
Note that the transfer-system full-consult dss command also supports the keep-conference command.
See the “Conference Initiator Drop-Off Control” section on page 196.

SUMMARY STEPS

1. telephony-service
2. transfer-system full-consult dss
3. exit

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Call-Transfer and Call-Forward Features

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 transfer-system full-consult dss Specifies the call transfer method for IP phone extensions that use
the International Telecommunications Union (ITU-T) ITU-T
H.450.2 standard and Session Interface Protocol (SIP) to use
Example:
Router(config-telephony)# transfer-system
consultation and to all the transfer of calls to idle monitor lines.
full-consult dss • The full-consult mode is required for Session Interface
Protocol (SIP).
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

CFwdAll Soft-Key Restriction Control

In Cisco CME 3.1 and later versions, the call-forward max-length command can be used to prevent
someone from using the CFwdAll soft key on an IP phone to forward calls to numbers that will incur toll
charges. The length argument in this command can be set to a number of digits that will allow the
CFwdAll soft key to set only a local or internal number as the destination for forwarded calls.
If the length argument is set to 0, the CFwdAll soft key is completely disabled.
The restriction created by this command does not apply to call-forward destinations that are entered
using the Cisco IOS command-line interface (CLI) or the Cisco CME GUI.

SUMMARY STEPS

1. ephone-dn dn-tag
2. number number [secondary number] [no-reg [both | primary]]
3. call-forward max-length length
4. exit

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Do Not Disturb Features

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 55
Step 2 number number [secondary number] [no-reg Configures a valid extension number for this ephone-dn.
[both | primary]]
• number—String of up to 16 digits that represents a telephone
or extension number to be associated with this ephone-dn.
Example:
Router(config-ephone-dn)# number 2345
• secondary—(Optional) Allows you to associate a second
telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should not
register with the H.323 gatekeeper. Unless you specify one of
the optional keywords (both or primary) after the no-reg
keyword, only the secondary number is not registered.
Step 3 call-forward max-length length Restricts the number of digits that can be entered using the
CFwdAll soft key on an IP phone.
Example: • length—Number of digits allowed to be entered using the
Router(config-ephone-dn)# call-forward CFwdAll soft key on an IP phone.
max-length 4
Step 4 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

The following example restricts the number of digits that a phone user can enter using the CFwdAll soft
key to four. In this example, extensions in the phone user’s Cisco CME system have four digits, so that
means that the user can use the IP phone to forward all calls to any extension in the system, but not to
any number outside the system.
ephone-dn 1
number 5001
call-forward max-length 4

Do Not Disturb Features

This section describes the following do not distub (DND) features:
• DND Call Forward, page 192
• DND Disable for Feature Ring, page 193

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Do Not Disturb Features

DND Call Forward

For Cisco CME 3.2 and later versions, the do not disturb (DND) soft key can either forward calls to a
target number, such as voice mail, or mute calls. To forward calls, ephone-dns must be configured with
a call forward no answer (CFNA) number. To mute calls, ephone-dns must be configured without a
CFNA number.

Note You can use the DND soft key to switch on or off the DND functionality in all call states except
connected. That is, you can enable or disable DND when an incoming call is ringing or when you are
not connected to a call. You cannot enable or disable DND when you are connected to an incoming call.

SUMMARY STEPS

1. ephone-dn dn-tag
2. call-forward noan directory-number timeout seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line, an intercom line, a
paging line, a voice-mail port, or a message-waiting
Example:
Router(config)# ephone-dn 1
indicator (MWI).
Step 2 call-foward noan directory-number timeout Configures call forwarding so that incoming calls to an
seconds extension (ephone-dn) that does not answer are forwarded
to another extension. During the alerting stage of a call,
Example: allows phone user to press the DND soft key to transfer the
Router(config-ephone-dn)# call-forward noan call to another extension.
5555 timeout 30
• directory-number—Telephone number to which calls
are forwarded. Represents a fully qualified E.164
number.
• timeout seconds—Sets how long a call can ring with
no answer before the call is forwarded to another
extension. Range is from 3 to 60000. There is no default
value.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

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DND Disable for Feature Ring

For Cisco CME 3.2.1 and later versions, do not disturb (DND) can be blocked from phones configured
with the feature-ring function, which is configured with the f keyword in the button command. A feature
ring is a triple-pulse ring, a type of ring cadence in addition to internal call and external call ring
cadences. For example, an internal call in the United States rings for 2 seconds on and 4 seconds off
(single-pulse ring), and an external call rings for 0.4 seconds on, 0.2 seconds off, 0.4 seconds on, and
0.2 seconds off (double-pulse ring).
The triple-pulse ring is used as an audio identifier for phone users. For example, each salesperson in a
sales department could have an IP phone with a button sharing the same set of ephone-dns with the sales
staff and another button for their private line for preferred customers. To help a salesperson identify an
incoming call to his or her private line, the private line can be configured with the feature-hring function.
The no dnd feature-ring command disables the DND function on feature-ring lines. If this command
were used for the configuration in the above example, the salespeople could activate DND on their
phones and still hear calls to their private lines.
This feature can be used with the Cisco CME B-ACD. For more information, see the “Cisco CME Basic
Automatic Call Distribution and Auto-Attendant Service” chapter in Cisco CME B-ACD and Tcl
Call-Handling Applications. In addition, the no dnd feature-ring command allows phone uses to accept
incoming calls while they are logged out (in a DND state) from a hunt group.

SUMMARY STEPS

1. ephone phone-tag
2. no dnd feature-ring
3. exit

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line.
Example: • phone-tag—Unique sequence number that identifies an
Router(config)# ephone 10 ephone-dn during configuration tasks. Range is from 1
to the maximum number of ephone-dns allowed on the
router platform. Refer to CLI help for the maximum
value for this argument.
Step 2 no dnd feature-ring Allows phone buttons configured with the feature-ring
option to ring when their IP phones are in do-not-disturb
(DND) mode.
Example:
Router(config-ephone)# no dnd feature-ring
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

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Conference Call Features

Example

For the following configuration example, when DND is activated on ephone 1 and ephone 2, button 1
will ring, but button 2 will not.
ephone-dn 1
number 1001

ephone-dn 2
number 1002

ephone-dn 10
number 1110
preference 0
no huntstop

ephone-dn 11
number 1111
preference 1
no huntstop

ephone 1
button 1f1
button 2o10,11
no dnd feature-ring

ephone 1
button 1f1
button 2o10,11
no dnd feature-ring

Conference Call Features

This section describes the following conference features:
• Conference Initiator Drop-Off Control, page 196
• Three-Party G.711 Conference Calls, page 194

Three-Party G.711 Conference Calls

Cisco CME supports three-party conference for local and on-net calls. This service supports conversion
between G.711 mu-law and A-law and between G.711 and G.729. For information about transcoding,
refer to the “Transcoding Between G.729 and G.711” chapter of this guide.
The maximum number of simultaneous conferences is platform-specific to the type of Cisco CME
router, and each individual Cisco IP phone can host a maximum of one conference at a time. You cannot
create a second conference on a phone if you already have an existing conference on hold.
In Cisco 3.3 and later releases, you can adjust the gain level of an external call to provide more adequate
volume. This functionality is applied to inbound audio packets so that conference participants can more
clearly hear a remote PSTN or VoIP caller joining their call. Note that this functionality cannot
discriminate between a remote VoIP/foreign exchange office (FXO) source, which requires a volume
gain, and a remote VoIP/IP phone, which does not require a volume gain and may therefore incur some
sound distortions.

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Conference Call Features

SUMMARY STEPS

1. telephony-service
2. max-conferences max-conference-number [gain -6 | 0 | 3 | 6]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 max-conferences max-conference-number [gain -6 | 0 | Sets the maximum number of simultaneous
3 | 6] three-party conferences supported by the router.
• max-conference-number—Maximum number of
Example: simultaneous three-party conferences supported
Router(config-telephony)# max-conferences 6 by a router, which is platform-dependent. The
default is half of the maximum number. The
maximum number of conferences per platform is
as follows:
– Cisco 2600 series, Cisco 2801—8
– Cisco 2811, Cisco 2821, Cisco 2851,
Cisco 3600 series, Cisco 3700 series—16
– Cisco 3800 series—24 (requires Cisco IOS
Release 12.3(11)XL or higher)
• gain—(Optional) Amount to increase the sound
volume of VoIP and PSTN calls joining a
conference call, in decibels. Valid values are -6,
0, 3, and 6. The default is -6.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Examples
The following example sets the maximum number of conferences for a Cisco IP phone to 4 and
configures a gain of 6 db for inbound audio packets from remote PSTN or VoIP calls joining a
conference:
telephony-service
max-conferences 4 gain 6

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Conference Call Features

Conference Initiator Drop-Off Control

For Cisco CME 3.2 and later versions, a person who initiates a conference call and hangs up can either
keep the remaining parties connected or disconnect them.
The keep-conference and keep-conference endcall commands configure IP phones to keep the
remaining conference parties connected when the conference initiator hangs up (places the handset back
in the on-hook position). Conference originators can disconnect from their conference calls by pressing
the Confrn (conference) soft key. When an initiator uses the Confrn key to disconnect from the
conference call, the oldest call leg will be put on hold, leaving the initiator connected to the most recent
call leg. The conference initiator can then navigate between the two parties by pressing either the Hold
soft key or the line buttons to select the desired call.
The keep-conference command causes the remote conference parties to remain connected when the
conference initiator hangs up the phone and to disconnect the conference parties if the initiator presses
the EndCall soft key. The keep-conference endcall command causes the remote conference parties to
remain connected when the conference initiator hangs up or presses the EndCall soft key.
Conference initiator dropoff can be configured per ephone.

Prerequisites

When the conference initiator hangs up, Cisco CME executes a call transfer to connect the two remaining
lines. To facilitate call transfer, the transfer-system command is required using the full-blind,
full-consult or full-consult dss keyword. The drop-off control behavior is the same for all three
keywords. When initiator hangs up, the remaining calls are transferred without consultation.

SUMMARY STEPS

1. telephony-service
2. transfer-system {full-blind | full-consult [dss]}
3. exit
4. ephone ephone-tag
5. keep-conference [endcall]
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 transfer-system {full-blind | full-consult Specifies the call transfer method for IP phone extensions
[dss]} that use the International Telecommunications Union
(ITU-T) H.450.2 standard.
Example: • You must use either the full-consult or full-blind
Router(config-telephony)# transfer-system keyword for the keep-conference command
full-blind
functionality to work correctly.

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Configurable Phone Displays and Sounds

Command or Action Purpose

Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 4 ephone ephone-tag Enters Ethernet phone (ephone) configuration mode for an
IP phone.
Example:
Router(config)# ephone 10
Step 5 keep-conference [endcall] Allows IP phone conference originators to drop off a
conference and to either end or maintain the conference for
the remaining parties.
Example:
Router(config-ephone)# keep-conference endcall • endcall—Allows phone conference originators to exit
their conference and keep the remaining parties
connected either by hanging up the call or by pressing
the EndCall soft key.
Step 6 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Examples
The following example configures ephone 1 so a conference initiator can hang up and keep the remaining
conference legs connected. Calls are transferred without consultation.
telephony-service
transfer-system full-blind

ephone 1
keep-conference

The following example configures ephone 10 so a conference initiator can either hang up or press the
EndCall soft key to leave the conference call and keep the remaining conference legs connected. Calls
are transferred without consultation.
telephony-service
transfer-system full-consult

ephone 10
keep-conference endcall

Configurable Phone Displays and Sounds

The following types of displays on IP phones can be customized for phone users in a Cisco CME system:
• Call-Waiting Beep, page 198
• Call-Waiting Ring, page 200
• Called Name and Ephone-dn Name Display, page 201

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• Caller ID Blocking on Outbound Calls, page 213

• Caller ID Blocking per Call, page 215
• IP Phone Header Bar, page 216
• Soft-Key Selection and Order, page 218
• System Text Message for the Cisco IP Phone 7905G, Cisco IP Phone 7912G, Cisco IP Phones 7940
and 7940G, and Cisco IP Phones 7960 and 7960G, page 222
• System Display Message (Idle URL) for the Cisco IP Phones 7940 and 7940G and Cisco IP Phones
7960 and 7960G, page 223
• URL Provisioning for Customized Function Buttons, page 225

Call-Waiting Beep
For Cisco CME 3.2 and later versions, call-waiting beeps can be switched on or off for individual
ephone-dns. You can choose to enable or disable the call-waiting beeps generated from and accepted by
an ephone-dn.
Call-waiting beeps are enabled by default. The command for disabling an ephone-dn’s beep generation
is no call-waiting beep generate. The command for disabling an ephone’s acceptance of call-waiting
beeps is no call-waiting beep accept.
If an ephone-dn’s beep generation is disabled, incoming calls to the ephone-dn do not generate
call-waiting beeps. If an ephone dn’s beep acceptance is disabled, the ephone-dn user will not hear beep
sounds when using the ephone-dn for an active call.
Table 7 shows the possible beep behaviors of one ephone-dn calling another ephone-dn that is connected
to another caller.

Table 7 Call-Waiting Beep Behavior

Incoming
Active Call Call Expected
Ephone-dn 1 Configuration Ephone-dn 2 Configuration on DN on DN Behavior
— no call-waiting beep DN 1 DN 2 No beep
no call-waiting beep — DN 1 DN 2 No beep
— no call-waiting beep generate DN 1 DN 2 No beep
— no call-waiting beep accept DN 1 DN 2 Beep
— no call-waiting beep accept DN 1 DN 2 No beep
no call-waiting beep generate
no call-waiting beep — DN 1 DN 1 No beep
no call-waiting beep generate — DN 1 DN 1 No beep
no call-waiting beep accept — DN 1 DN 1 No beep
no call-waiting beep accept no — DN 1 DN 1 No beep
call-waiting beep generate
no call-waiting beep generate — DN 1 DN 2 Beep
no call-waiting beep accept — DN 1 DN 2 No beep
— no call-waiting beep DN 1 DN 1 Beep

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To display how call-waiting beeps are configured, use the show running-config command. If the
no call-waiting beep generate and the no call-waiting beep accept commands are configured, the
show running-config command output will display the no call-waiting beep command.

Restrictions
The call-waiting beep volume cannot be adjusted through Cisco CME for the Cisco IP Phone 7902G,
Cisco IP Phone 7905G, Cisco IP Phone 7912G, Cisco ATA-186, and Cisco ATA-188.

SUMMARY STEPS

1. ephone-dn dn-tag
2. no call-waiting beep [accept | generate]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 12
Step 2 no call-waiting beep [accept | generate] Allows an ephone-dn to generate call-waiting beeps that
may be received by another ephone-dn. The beep will be
heard only if the other ephone-dn is configured to accept
Example:
Router(config-ephone-dn)# no call-waiting beep
call-waiting beeps (default).
accept • accept—Allows incoming call-waiting beeps.
• generate—Allows an ephone-dn to generate
call-waiting beeps.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Examples
In the following example, ephone-dn 10 neither accepts nor generates a beep, ephone-dn 11 does not
accept a beep, and ephone-dn 12 does not generate a beep.
ephone-dn 10
no call-waiting beep
number 4410

ephone-dn 11
no call-waiting beep accept
number 4411

ephone-dn 12
no call-waiting beep generate
number 4412

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Call-Waiting Ring
For Cisco CME 3.2.1 and later versions, you can use either a standard call waiting beep sound through
the handset or a short ring for call-waiting notification. Selection is made through an ephone-dn’s
configuration. The default is for ephone-dns to accept call interruptions, such as call waiting, and to issue
a beeping sound for notification. To use a ring sound, you must ensure that your ephone-dns will accept
call waiting. To be sure that this is the case, verify that the ephone-dn has not been configured with the
no call-waiting beep accept command. If an ephone-dn has been configured in this way, remove this
command.
After you have ensured that the ephone-dn will accept call waiting, you can configure it to issue ringing
notification with the call-waiting ring command.

Restrictions
The call-waiting ring option cannot be used on the Cisco IP Phone 7902, Cisco IP Phone 7905, or
Cisco IP Phone 7912. Do not use the call-waiting ring command for ephone-dns associated with these
types of phones.

SUMMARY STEPS

1. ephone-dn phone-tag [dual-line]

2. call-waiting ring
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn phone-tag [dual-line] Enters ephone-configuration mode for an IP phone for the
purposes of creating and configuring an ephone.
Example: • phone-tag—Unique sequence number that identifies
Router(config)# ephone-dn 10 dual-line this ephone-dn during configuration tasks. Range is
from 1 to the maximum number of ephone-dns allowed
on the router platform. Refer to CLI help for the
maximum value for this argument.
• dual-line—(Optional) Enables dual-line mode for the
ephone-dn.
Step 2 call-waiting ring Allows an ephone-dn to use a ring sound for call-waiting
notification.
Example:
Router(config-ephone-dn)# call-waiting ring
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

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Example

The following example shows how to configure ephone-dn 1 to use ringing as its method of call-waiting
notification:
ephone-dn 1 dual-line
number 1001
call-waiting ring

ephone-dn 2 dual-line
number 1002

Called Name and Ephone-dn Name Display

For Cisco 3.2 and later versions, either the called number or ephone-dn name of an incoming call can be
displayed on IP phones that use overlay ephone-dns. Called names can also be displayed on IP phones
that use hunt groups and ephone-dns that are not overlaid. For more information about overlaid
ephone-dns, refer to the “Overlaid Ephone-dns” section on page 262.
The display of a called name or ephone-dn name is useful for situations in which calls from different
numbers are configured to ring on a single line or lines, and the phone user must identify who the caller
is and what the caller has dialed. For example, if order-entry personnel are servicing three catalogs with
individual 800 numbers configured on one overlay ephone-dn, they need to know which catalog is being
called to give the correct greeting, such as “Thank you for calling catalog N. May I take your order?”
The operators of a doctors’ answering service must know the name of the doctor the patient is calling.
For called names, the service dnis dir-lookup command and the name-lookup service are used to display
the called number’s name, which is obtained from a Cisco CME system’s list of directory names created
under the directory entry command in the telephony-service configuration mode. For ephone-dn names,
the service dnis overlay and the ephone-dn’s name configured in the name command in ephone-dn
configuration mode are used to display an ephone-dn name.

Note If the service dnis overlay and service dnis dir-lookup commands are both used in one configuration
under telephony-service mode, the service dnis dir-lookup will take precedence.

Configurations for displaying called names and ephone-dn names are given in the following sections:
• Configuring for the Display of Called Names for Overlaid Ephone-dns, page 201
• Configuring for the Display of Called Names for Nonoverlaid Ephone-dns, page 206
• Configuring for the Display of Ephone-dn Names for Overlaid Ephone-dns, page 210

Configuring for the Display of Called Names for Overlaid Ephone-dns

The display of called names from overlaid ephone-dns is configured using the service dnis dir-lookup
command and the name value declared in the directory entry command. Up to 100 names can be set
using the directory entry command. To allow individual ephone-dns to receive calls from called
numbers that use these directory-entry names, the ephone-dns’ names must contain wildcard (.)
characters. Ephone-dn names are configured with the name command in ephone-dn configuration mode.

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Overlaid ephone-dns can be configured with hunt groups. To use the service dnis dir-lookup command
in conjunction with the ephone-hunt command, you must configure the ephone-hunt group to use a pilot
number that contains wildcard characters. This configuation allows the ephone-hunt group to receive
calls from different numbers. Individual ephone-dns that are configured as members of the hunt group
with the list command must not have wildcard characters in their number fields.

Restrictions
Call waiting is not supported for overlaid ephone-dns.

SUMMARY STEPS

1. telephony-service
2. service dnis dir-lookup
3. directory entry {directory-tag number name name | clear}
4. exit
5. ephone-dn dn-tag
6. number number
7. exit
8. ephone phone-tag
9. button button-numberodn-tag[[,dn-tag][,dn-tag] ...]
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 service dnis dir-lookup Allows the display of names associated with called numbers
for incoming calls on IP phones.
Example:
Router(config-telephony)# service dnis overlay

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Command or Action Purpose

Step 3 directory entry {directory-tag number name name Adds an entry to a local phone directory that can be displayed
| clear} on IP phones.
• directory-tag—Digit string that provides a unique
Example:
Router(config-telephony)# directory entry 1 identifier for this particular directory entry. Range is
5550001 name user1 from 1 to 100.
• number—Telephone number or extension for the
directory entry, up to 32 characters.
• name name—Name that appears in the directory
associated with the number. String of up to 24
characters.
• clear—Removes all directory entries that were made
with this command.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 5 ephone-dn dn-tag Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 1 ephone-dn during configuration tasks. Range is from 1 to
the maximum number of ephone-dns allowed on the
router platform. Refer to CLI help for the maximum
value for this argument.
Step 6 number number Associates a telephone or extension number with an
extension (ephone-dn) in a Cisco CME.system.
Example:
• number—String of up to 16 characters that represents an
Router(config-ephone-dn)# number 555000.
E.164 telephone number. Normally the string is
composed of digits, but the string may contain
alphabetic characters when the number is dialed only by
the router, as with an intercom number.
Step 7 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 8 ephone phone-tag Enters Ethernet phone (ephone) configuration mode for an IP
phone.
Example: • phone-tag—Unique sequence number that identifies an
Router(config)# ephone 1 ephone during configuration tasks. The maximum
number is platform-dependent; refer to Cisco IOS
command-line interface (CLI) help.

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Command or Action Purpose

Step 9 button button-numberodn-tag[[,dn-tag][,dn-tag] Associates ephone-dns with individual buttons on a Cisco IP
...] phone and specifies the ringing behavior for those buttons.
• button-number—Number of a line button on a Cisco IP
Example:
Router(config-ephone)# button 1o1,2,3 phone to be associated with an extension (ephone-dn).
The maximum number of button-ephone-dn pairs is
determined by phone type.
• o—Overlay line. Multiple ephone-dns share a single
button, up to a maximum of ten on a button. The dn-tag
argument can contain up to ten individual dn-tags,
separated by commas.
• dn-tag—Overlay line. Multiple ephone-dns share a
single button, up to a maximum of ten on a button. The
dn-tag argument can contain up to ten individual dn-tags,
separated by commas.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following is an example of a configuration of overlaid ephone-dns that uses wildcards in the
secondary numbers for the ephone-dns. The wildcards allow you to control the display according to the
number that was dialed. The example is for a medical answering service with three IP phones that accept
calls for nine doctors on one button. When a call to 5550001 rings on button 1 on ephone 1 through
ephone 3, “doctor1” is displayed on all three ephones.
telephony-service
service dnis dir-lookup

directory entry 1 5550001 name doctor1

directory entry 2 5550002 name doctor2
directory entry 3 5550003 name doctor3

directory entry 4 5550010 name doctor4

directory entry 5 5550011 name doctor5
directory entry 6 5550012 name doctor6

directory entry 7 5550020 name doctor7

directory entry 8 5550021 name doctor8
directory entry 9 5550022 name doctor9

ephone-dn 1
number 5500 secondary 555000.

ephone-dn 2
number 5501 secondary 555001.

ephone-dn 3
number 5502 secondary 555002.

ephone 1
button 1o1,2,3
mac-address 1111.1111.1111

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ephone 2
button 1o1,2,3
mac-address 2222.2222.2222

ephone 3
button 1o1,2,3
mac-address 3333.3333.3333

The following example shows a hunt-group configuration for a medical answering service with two
phones and four doctors. Each phone has two buttons, and each button is assigned two doctors’ numbers.
When a patient calls 5550341, Cisco CME matches the hunt-group pilot secondary number (555....),
rings button 1 on one of the two phones, and displays “doctor1.” For more information about hunt-group
behavior, refer to the “Ephone Hunt Groups” section on page 273. Note that wildcards are used only in
secondary numbers and cannot be used with primary numbers.
telephony-service
service dnis dir-lookup
max-redirect 20
directory entry 1 5550341 name doctor1
directory entry 2 5550772 name doctor1
directory entry 3 5550263 name doctor3
directory entry 4 5550150 name doctor4

ephone-dn 1
number 1001
ephone-dn 2
number 1002
ephone-dn 3
number 1003
ephone-dn 4
number 104

ephone 1
button 1o1,2
button 2o3,4
mac-address 1111.1111.1111

ephone 2
button 1o1,2
button 2o3,4
mac-address 2222.2222.2222

ephone-hunt 1 peer
pilot number 5100 secondary 555....
list 1001, 1002, 1003, 1004
final number 5556000
hops 5
preference 1
timeout 20
no-reg

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Configuring for the Display of Called Names for Nonoverlaid Ephone-dns

The display of called names from ephone-dns that are not overlaid is configured using the service dnis
dir-lookup command and the name value declared in the directory entry command. Up to 100 names
can be set with the directory entry command. To allow individual ephone-dns to receive calls from
called numbers that use these directory-entry names, the ephone-dns’ names must contain wildcard (.)
characters. Ephone-dn names are configured with the name command in ephone-dn configuration mode.
Secondary line numbers also can be configured, so that the secondary line displays called names.

SUMMARY STEPS

1. telephony-service
2. service dnis dir-lookup
3. directory entry {directory-tag number name name | clear}
4. exit
5. ephone-dn dn-tag
6. number number [secondary number]
7. exit
8. ephone phone-tag
9. button button-number:dn-tag[[,dn-tag][,dn-tag] ...]
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 service dnis dir-lookup Allows the display of names associated with called numbers
for incoming calls on IP phones.
Example:
Router(config-telephony)# service dnis overlay

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Command or Action Purpose

Step 3 directory entry {directory-tag number name name Adds an entry to a local phone directory that can be displayed
| clear} on IP phones.
• directory-tag—Digit string that provides a unique
Example:
Router(config-telephony)# directory entry 1 identifier for a particular directory entry. Range is from
5550001 name user1 1 to 100.
• number—Telephone number or extension for the
directory entry, up to 32 characters.
• name name—Name that appears in the directory
associated with the number. String of up to 24
characters.
• clear—Removes all directory entries that were made
with this command.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 5 ephone-dn dn-tag Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line.
Example: • dn-tag—Unique sequence number that identifies an
Router(config)# ephone-dn 1 ephone-dn during configuration tasks. Range is from 1 to
the maximum number of ephone-dns allowed on the
router platform. Refer to CLI help for the maximum
value for this argument.
Step 6 number number [secondary number] Associates a telephone or extension number with an
extension (ephone-dn) in a Cisco CME system.
Example:
• Required if using the secondary-line method of
Router(config-ephone-dn)# number 1001 secondary
555000. displaying called names.
• number—String of up to 16 characters that represents an
E.164 telephone number. Normally the string is
composed of digits, but the string may contain
alphabetic characters when the number is dialed only by
the router, as with an intercom number or is not intended
to be dialed at alls. Secondary numbers can contain
wildcards in the string. For details, see the “Usage
Guidelines” for this command in the Cisco CallManager
Express 3.3 Command Reference.
• (Optional) secondary—Associates the number that
follows with this extension (ephone-dn) as an additional
number.
Step 7 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

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Command or Action Purpose

Step 8 ephone phone-tag Enters Ethernet phone (ephone) configuration mode for an IP
phone.
Example: • phone-tag—Unique sequence number that identifies an
Router(config)# ephone 1 ephone during configuration tasks. The maximum
number is platform-dependent; refer to Cisco IOS
command-line interface (CLI) help.
Step 9 button button-number:dn-tag[[,dn-tag][,dn-tag] Associates ephone-dns with individual buttons on a Cisco IP
...] phone and specifies the ringing behavior for those buttons.
• button-number—Number of a line button on a Cisco IP
Example:
Router(config-ephone)# button 1o1,2,3 phone to be associated with an extension (ephone-dn).
The maximum number of button-ephone-dn pairs is
determined by phone type.
• : (colon)—Normal ring. For incoming calls on this
extension, the phone produces audible ringing, a
flashing ((< icon in the phone display, and a flashing red
light on the handset. On the Cisco IP Phone 7914
Expansion Module, a flashing yellow light also
accompanies incoming calls.
• dn-tag—Overlay line. Multiple ephone-dns share a
single button, up to a maximum of ten on a button. The
dn-tag argument can contain up to ten individual dn-tags,
separated by commas.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following is a configuration for three IP phones, each with two buttons. Button 1 receives calls from
doctor1, doctor2, and doctor3, and button 2 receives calls from doctor4, doctor5, and doctor5.
telephony-service
service dnis dir-lookup
directory entry 1 5550001 name doctor1
directory entry 2 5550002 name doctor2
directory entry 3 5550003 name doctor3
directory entry 4 5550010 name doctor4
directory entry 5 5550011 name doctor5
directory entry 6 5550012 name doctor6

ephone-dn 1
number 1001 secondary 555000.

ephone-dn 2
number 1002 secondary 555001.

ephone 1
button 1:1
button 2:2
mac-address 1111.1111.1111

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ephone 2
button 1:1
button 2:2
mac-address 2222.2222.2222

ephone 3
button 1:1
button 2:2
mac-address 3333.3333.3333

Configuring for the Display of Ephone-dn Names for Overlaid Ephone-dns

To display ephone-dn names on IP phones, you must use the service dnis overlay command and provide
each ephone-dn with a name using the name command in ephone-dn configuration mode. Up to ten
ephone-dns per overlay set and ten name displays are allowed per button.
The number of the first ephone-dn listed in the button command is the default display for all phones
using the same set of overlaid ephone-dns. Calls to the first ephone-dn display caller IDs. Calls to the
remaining ephone-dns display ephone-dn names. The following is a configuration for three phones that
use the same set of overlaid ephone-dns for each phone’s button 1.
telephony-service
service dnis overlay

ephone-dn 1
number 18005550000
ephone-dn 2
name department1
number 18005550001
ephone-dn 3
name department2
number 18005550002

ephone 1
button 1o1,2,3

ephone 2
button 1o1,2,3

ephone 3
button 1o1,2,3

The default display for all three phones is the number of the first ephone-dn listed in the overlay set
(18005550000). A call is made to the first ephone-dn (18005550000), and the caller ID (for example,
4085550123) is displayed on all three phones. The user for phone 1 answers the call. The caller ID
(4085550123) remains displayed on phone 1, and the displays on phone 2 and phone 3 return to the
default display (18005550000). A call to the next ephone-dn is made. The default display on phone 2
and phone 3 is replaced with the called ephone-dn’s name (18005550001).

Restrictions
Regardless of the call-waiting setting, call waiting for overlaid ephone-dns is disabled automatically.

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SUMMARY STEPS

1. telephony-service
2. service dnis overlay
3. exit
4. ephone-dn dn-tag
5. name name
6. number number
7. exit
8. ephone phone-tag
9. button button-numberodn-tag[[,dn-tag][,dn-tag] ...]
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 service dnis overlay Allows incoming calls to an ephone-dn overlay to display the
individual called ephone-dn names.
Example:
Router(config-telephony)# service dnis overlay
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 4 ephone-dn dn-tag Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 1 ephone-dn during configuration tasks. Range is from 1 to
the maximum number of ephone-dns allowed on the
router platform. Refer to CLI help for the maximum
value for this argument.
Step 5 name name Associates a name with a Cisco CME extension (ephone-dn).
• name—Name that will appear on IP phone displays.
Example:
Router(config-ephone-dn)# name Main Office
Number
Step 6 number number Associates a telephone number or extension with a
Cisco CME extension (ephone-dn).
Example: • name—String of up to 16 characters that represents an
Router(config-ephone-dn)# name 18005550000 E.164 telephone number.

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Command or Action Purpose

Step 7 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 8 ephone phone-tag Enters Ethernet phone (ephone) configuration mode for an IP
phone.
Example: • phone-tag—Unique sequence number that identifies an
Router(config)# ephone 1 ephone during configuration tasks. The maximum
number is platform-dependent; refer to Cisco IOS
command-line interface (CLI) help.
Step 9 button button-numberodn-tag[[,dn-tag][,dn-tag] Associates ephone-dns with individual buttons on a Cisco IP
...] phone and specifies the ringing behavior for those buttons.
• button-number—Number of a line button on a Cisco IP
Example:
Router(config-ephone)# button 1o1,2,3 phone to be associated with an extension (ephone-dn).
The maximum number of button-ephone-dn pairs is
determined by phone type.
• o—Overlay line. Multiple ephone-dns share a single
button, up to a maximum of ten on a button. The dn-tag
argument can contain up to ten individual dn-tags,
separated by commas.
• dn-tag—Overlay line. Multiple ephone-dns share a
single button, up to a maximum of ten on a button. The
dn-tag argument can contain up to ten individual dn-tags,
separated by commas.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

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Example

The following example shows three phones that have button 1 assigned to pick up three 800 numbers for
three different catalogs.
The default display for all four phones is the number of the first ephone-dn listed in the overlay set
(18005550000). A call is made to the first ephone-dn (18005550000), and the caller ID (for example,
4085550123) is displayed on all phones. The user for phone 1 answers the call. The caller ID
(4085550123) remains displayed on phone 1, and the displays on phone 2 and phone 3 return to the
default display (18005550000). A call to the second ephone-dn (18005550001) is made. The default
display on phone 2 and phone 3 is replaced with the called ephone-dn's name (catalog1) and number
(18005550001).
telephony-service
service dnis overlay

ephone-dn 1
number 18005550000

ephone-dn 2
name catalog1
number 18005550001

ephone-dn 3
name catalog2
number 18005550002

ephone-dn 4
name catalog3
number 18005550003

ephone 1
button 1o1,2,3,4

ephone 2
button 1o1,2,3,4

ephone 3
button 1o1,2,3,4

Caller ID Blocking on Outbound Calls

Using the commands in this section, you can block all caller-ID (CLID) displays on all calls from a
particular ephone-dn, or you can selectively choose to block name, number, or both on outbound VoIP
calls only.
To block all CLID displays for calls from a particular extension, use the caller-id block command in
ephone-dn configuration mode. This command asks the far-end gateway device to block display of
calling-party information for the calls received from this ephone-dn.
Alternatively, you can allow the local display of CLID information by not using the caller-id block
command and independently block CLID name, CLID number, or both on outbound VoIP calls using the
clid strip command in dial-peer configuration mode. This arrangement has the benefit of allowing
caller-ID display for local calls while preventing caller-ID display for external calls going to VoIP.
This feature can also be used for public switched telephony network (PSTN) calls that go out over ISDN.

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Restrictions
Caller ID blocking does not apply to PSTN calls that are made through foreign exchange office (FXO)
ports. Caller ID features on FXO-connected subscriber lines are under the control of the PSTN service
provider, who may require that you subscribe to their caller ID blocking service.

SUMMARY STEPS

1. ephone-dn dn-tag
2. caller-id block
3. exit
4. dial-peer voice tag voip
or
dial-peer voice tag pots
5. clid strip
6. clid strip name
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.

Example:
Router(config)# ephone-dn 3
Step 2 caller-id block (Optional) Blocks display of all caller-ID information for
outbound calls that originate from this ephone-dn.
Example: • By default, caller ID is not blocked on calls that originate
Router(config-ephone-dn)# caller-id block from a Cisco IP phone.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 4 dial-peer voice tag voip Enters dial-peer configuration mode.
or
Note You can configure caller-ID blocking on POTS dial
dial-peer voice tag pots peers if the POTS interface is ISDN. This feature is not
available on FXO/CAS lines.
Example:
Router(config)# dial-peer voice 3 voip
or
Router(config)# dial-peer voice 5 pots
Step 5 clid strip (Optional) Removes the calling-party number from the CLID
information being sent with VoIP calls.
Example:
Router(config-dial-peer)# clid strip

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Command or Action Purpose

Step 6 clid strip name (Optional) Removes the calling-party name from the CLID
information being sent with VoIP calls.
Example:
Router(config-dial-peer)# clid strip name
Step 7 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit

Examples

The following example sets CLID blocking for the ephone-dn with tag 3.
ephone-dn 3
number 2345
caller-id block

The following example blocks the display of CLID name and number on VoIP calls but allows CLID
display for local calls:
ephone-dn 3
number 2345
dial-peer voice 2 voip
clid strip
clid strip name

Caller ID Blocking per Call

The display of caller ID can be blocked for outgoing calls on a per-call basis, allowing users to maintain
their privacy when necessary. The system administrator first defines a code for caller ID blocking in the
system. Users can then dial the code prior to making any call on which they do not want their number
displayed on the called-party phone. The caller ID is still sent, but its presentation parameter is set to
“restricted” so that the caller ID is not displayed.
To enable caller ID blocking on a per-call basis, a system administrator must first define a code. Users
then enter the code prior to making calls on which they do not want their number sent.

SUMMARY STEPS

1. telephony-service
2. caller-id block code code-string
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 caller-id block code code-string Defines a code to be entered before making calls on which
the caller ID should not be sent.
Example: • code-string—Digit string of up to 16 characters. The
Router(config-telephony)# caller-id block code first character must be an asterisk (*).
*1234
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example defines a code of *1234 for users to enter to block caller ID on their outgoing
calls:
telephony-service
caller-id block code *1234

IP Phone Header Bar

Normally the IP phone header bar, or top line, of a Cisco IP Phones 7940 and 7940G or Cisco IP Phones
7960 and 7960G replicates the text that appears next to the first line button. The header bar is shown in
Figure 31. The header bar can, however, contain a user-definable message instead of the extension
number. For example, the header bar can be used to display a name or the full E.164 number of the phone.
If no description is specified, the header bar replicates the extension number that appears next to the first
button on the phone.

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Figure 31 Cisco IP Phone Display

13:09 06/08/01 3270 Header bar

Title line

Content lines
Service window

Prompt and status area

82878
Softkey 1 Softkey 2 Softkey 3 Softkey 4

SUMMARY STEPS

1. ephone-dn dn-tag
2. description display-text
3. exit
4. ephone phone-tag
5. restart
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—The unique sequence number that identifies
Example: the ephone-dn for which the description should be in
Router(config)# ephone-dn 55 effect.
Step 2 description display-text Defines a description for the header bar of a Cisco IP
Phones 7940 and 7940G or Cisco IP Phones 7960 and
7960G that has the specified ephone-dn associated with its
Example:
Router(config-ephone-dn)# description
first line button.
408-555-0134 • display-text—Alphanumeric character string, up to 40
characters. String is truncated to 14 characters in the
display.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

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Command or Action Purpose

Step 4 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number of the ephone on
Example: which the new description appears.
Router(config)# ephone 1
Step 5 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-ephone)# restart
Step 6 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example provides the full E.164 number for a phone line in the phone header bar:
ephone-dn 55
description 408-555-0149

Soft-Key Selection and Order

For Cisco CME 3.2 and later versions, you can disable and enable IP phone soft keys and change the
order in which they appear in the displays of individual ephones using ephone templates that are then
applied to ephones. This feature is available on the Cisco IP Phone 7905G, Cisco IP Phone 7912G, Cisco
IP Phone 7940G, and Cisco IP Phone 7960G.
You can create up to five ephone templates but you cannot apply more than one template to an ephone.
If you try to apply a second ephone template to an ephone that already has an ephone-template applied
to it, the second ephone-template will overwrite the first ephone template information, but the new
information will only take affect after you use the restart command to reboot the phone. If you do not
reboot the phone, the previously configured ephone-template will remain in effect.
Each ephone template can include soft-key settings for all or some of the following call stages:
• Alerting—When the remote point is being notified of an incoming call, and the status of the remote
point is being relayed to the caller as either ringback or busy. The soft-key options and their default
order at this calling stage are as follows:
– Acct—Short for “account code.” Provides access to configured accounts.
– Callback—Requests callback notification when a busy called line becomes free.
– Endcall—Ends the current call.
• Connected—When the connection to a remote point has been established. The soft-key options and
their default order at this calling stage are as follows:
– Acct—Short for “account code.” Provides access to configured accounts.
– Confrn—Short for “conference.” Connects callers to a conference call.
– Endcall—Ends the current call.

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– Flash—Short for “hookflash.” Provides hookflash functionality for public switched telephone
network (PSTN) services on calls connected to the PSTN via a foreign exchange office (FXO)
port.
– Hold—Places an active call on hold and resumes the call.
– Park—Places an active call on hold so it can be retrieved from another phone in the system.
– Trnsfer—Short for “call transfer.” Transfers active calls to another extension.
• Idle—Before a call is made and after a call is complete. The soft-key options and their default order
at this calling stage are as follows:
– Cfwdall—Short for “call forward all.” Forwards all calls.
– Dnd—Short for “do not disturb.” Enables the do-not-disturb features.
– Gpickup—Short for “group call pickup.” Selectively picks up calls coming into a phone number
that is a member of a pickup group.
– Login—Provides personal identification number (PIN) access to restricted phone features.
– Newcall—Opens a line on a speakerphone to place a new call. Note that the Newcall soft key
must not be disabled for the Cisco IP Phone 7905G and the Cisco IP Phone 7912G.
– Pickup—Selectively picks up calls coming into another extension.
– Redial—Redials the last number dialed.
• Seized—When a caller is attempting a call but has not yet been connected. The soft-key options and
their default order at this calling stage are as follows:
– Cfwdall—Short for “call forward all.” Forwards all calls.
– Endcall—Ends the current call.
– Gpickup—Short for “group call pickup.” Selectively picks up calls coming into a phone number
that is a member of a pickup group.
– Pickup—Selectively picks up calls coming into another extension.
– Redial—Redials the last number dialed.
The default soft-key display configuration is as follows:
• Alerting—Acct, Callback, and Endcall
• Connected—Acct, Confrn, Endcall, Flash, Hold, and Trnsfr
• Idle—Cfwdall, Dnd, Gpickup, Login, Newcall, Pickup, and Redial
• Seized—Cfwdall, Endcall, Gpickup, Pickup, and Redial
Configuring ephone soft-key templates involves the following tasks:
• Declaring and defining a template:
ephone-template 1
softkeys idle Newcall Redial
softkeys seized Endcall
softkeys connected Endcall Hold Trnsfer

• Applying the templates to individual ephones:

ephone 15
ephone-template 1
ephone 20
ephone-template 5

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To view your ephone-template configurations, use the show telephony-service ephone-template

command.

Note The third soft-key button selection on the Cisco IP Phone 7905G and Cisco IP Phone 7912G is reserved
for the Message soft key. For these phones’ templates, the third soft-key defaults to the Message soft key.
For example, the softkeys idle Redial Dnd Pickup Login Gpickup command configuration displays,
in order, the soft keys Redial, Dnd, Message, Pickup, Login, and Gpickup.

SUMMARY STEPS

1. ephone-template tag
2. softkeys alerting [Acct] [Callback] [Endcall]
3. softkeys connected [Acct] [Confrn] [Endcall] [Flash] [Hold] [Trnsfer]
4. softkeys idle [Cfwdall] [Dnd] [Gpickup] [Login] [Newcall] [Pickup] [Redial]
5. softkeys seized [Cfwdall] [Endcall] [Gpickup] [Pickup] [Redial]
6. exit
7. ephone phone-tag
8. ephone-template tag
9. restart
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-template tag Declares and names an ephone template to configure the
display of soft keys on IP phones and enters the
ephone-template configuration mode.
Example:
Router(config)# ephone-template 1 • tag—Number 1 through 5.
Step 2 softkeys alerting [Acct] [Callback] [Endcall] (Optional) Configures an ephone template for soft-key
display during the alerting call stage. You can enter any of
the keywords in any order. Soft keys that are not explicitly
Example:
Router(config-ephone-template)# softkeys
defined will be disabled. The default soft keys for the alerting
alerting Callback Endcall call stage and the order in which they appear are, from first
to last: Acct, Callback, and Endcall.
Step 3 softkeys connected [Acct] [Confrn] [Endcall] (Optional) Configures an ephone template for soft-key
[Flash] [Hold] [Trnsfer] display during the connected call stage. You can enter any of
the keywords in any order. Soft keys that are not explicitly
Example: defined will be disabled. The default soft keys for the
Router(config-ephone-template)# softkeys connected call stage and the order in which they appear are,
connected Hold Trnsfer Confrn Endcall from first to last: Acct, Confrn, Endcall, Flash, Hold, Park,
and Trnsfer.

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Command or Action Purpose

Step 4 softkeys idle [Cfwdall] [Dnd] [Gpickup] [Login] (Optional) Configures an ephone template for soft-key
[Newcall] [Pickup] [Redial] display during the idle call stage. You can enter any of the
keywords in any order. Soft keys that are not explicitly
Example: defined will be disabled. The default soft keys for the idle
Router(config-ephone-template)# softkeys idle call stage and the order in which they appear are, from first
Dnd Pickup Redial Cfwdall to last: Cfwdall, Dnd, Gpickup, Login, Newcall, Pickup, and
Redial.
Step 5 softkeys seized [Cfwdall] [Endcall] [Gpickup] (Optional) Configures an ephone template for soft-key
[Pickup] [Redial] display during the seized call stage. You can enter any of the
keywords in any order. Soft keys that are not explicitly
Example: defined will be disabled. The default soft keys for the idle
Router(config-ephone-template)# softkeys seized call stage and the order in which they appear are, from first
Pickup Redial Cfwdall Endcall Gpickup to last: Cfwdall, Endcall, Gpickup, Pickup, and Redial.
Step 6 exit Exits ephone template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 7 ephone phone-tag Enters ephone configuration mode for an IP phone.

Example:
Router(config)# ephone 10
Step 8 ephone-template tag Applies an ephone template to an ephone.
• Ephones without an ephone template will use the default
Example: soft-key order.
Router(config-ephone)# ephone-template 1
Step 9 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example: Note If you are applying the template to more than one
Router(config-ephone)# restart ephone, you can use the restart all command in
telephony-service configuration mode to reboot all
the phones so they have the new template
information.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following are configurations for ephone templates 1 and 2. Template 1 is applied to ephone 11, 13,
and 15. Template 2 is applied to ephone 34.
ephone-template 1
softkeys idle Redial Newcall
softkeys connected Endcall Hold Trnsfer

ephone-template 2
softkeys idle Redial Newcall
softkeys seized Redial Endcall Pickup

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softkeys alerting Redial Endcall

softkeys connected Endcall Hold Trnsfer

ephone 10
ephone-template 2
ephone 13
ephone-template 1
ephone 15
ephone-template 1
ephone 34
ephone-template 2

System Text Message for the Cisco IP Phone 7905G, Cisco IP Phone 7912G,
Cisco IP Phones 7940 and 7940G, and Cisco IP Phones 7960 and 7960G
For the Cisco IP Phone 7905G, Cisco IP Phone 7912G, Cisco IP Phones 7940 and 7940G, and Cisco IP
Phones 7960 and 7960G, the system text message feature allows you to set the default static text display
for phones that are idle. If no message is set, the default message “Cisco CallManager Express” is
displayed.
The number of characters that can be displayed is not fixed because IP phones typically use a
proportional (as opposed to fixed-width) font. There is room for approximately 30 alphanumeric
characters.
The display message is refreshed with a new message after one of the following events occurs:
• Busy phone goes back on-hook.
• Idle phone receives a keepalive message.
• Phone is restarted.
A similar feature allows you to name a URL that contains a file that you want displayed on idle phones.
See the “System Display Message (Idle URL) for the Cisco IP Phones 7940 and 7940G and Cisco IP
Phones 7960 and 7960G” section on page 223.

SUMMARY STEPS

1. telephony-service
2. system message text-message
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 system message text-message Sets the message to display when a phone is idle.
• text-message—Alphanumeric string to display. Display
Example: uses proportional-width font, so the number of
Router(config-telephony)# system message ABC characters that are displayed varies based on the width
Company of the characters that are used. The maximum number
of displayed characters is approximately 30.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example specifies text that should be displayed on IP phones when they are not being
used:
telephony-service
system message ABC Company

System Display Message (Idle URL) for the Cisco IP Phones 7940 and 7940G and
Cisco IP Phones 7960 and 7960G
The system display message feature allows you to specify a file to display on Cisco IP Phones 7940 and
7940G and Cisco IP Phone 7960 and 7960G units when they are not in use. You can use this feature to
provide the phone display with a system message that is refreshed at configurable intervals, similar to
the way that the system text message feature provides a message. The difference between the two is that
the system text message feature displays a single line of text at the bottom of the phone display, whereas
the system display message feature can use the entire display area and contain graphic images. For more
information about the system text message feature, see the “System Text Message for the Cisco IP Phone
7905G, Cisco IP Phone 7912G, Cisco IP Phones 7940 and 7940G, and Cisco IP Phones 7960 and 7960G”
section on page 222.

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The system display message feature requires a back-end web server to serve the browser page to the
phone display, as the Cisco CME system only provisions the URL. The system display message display
can also provide soft keys for the phone and thereby take input from the phone user for interactive
services.
To specify a system display message for idle phones, designate a URL that contains an XML file to be
displayed on the Cisco IP phones that are not in use. You also specify the refresh interval for the display,
in seconds. The file can contain text, icons, or images, and must conform to the Cisco XML DTD that is
described in Cisco IP Phone Services Application Development Notes.

SUMMARY STEPS

1. telephony-service
2. url idle url idle-timeout seconds
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 url idle url idle-timeout seconds Defines a URL that contains a file to display on the phone
when the phone is not in use, and specifies the interval
between refreshes of the display, in seconds.
Example:
Router(config-telephony)# url idle • url—Any URL that conforms to RFC 2396.
http://www.abcwrecking.com/public/logo
idle-timeout 35 • seconds—Range is from 0 to 300.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example specifies that a file called logo.htm should be displayed on IP phones when they
are not being used:
telephony-service
url idle http://www.abcwrecking.com/public/logo.htm idle-timeout 35

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URL Provisioning for Customized Function Buttons

The Cisco IP Phones 7940 and 7940G and the Cisco IP Phones 7960 and 7960G have customized
function buttons that show the phone call status and activities on the display panels. These customized
function buttons also invoke programmable noncall-related services. The four buttons—Services,
Directories, Messages, and Information (the i button)—are linked to appropriate feature operations
through programmable URLs. You cannot customize the Settings button.
Specific URLs are provisioned on the Cisco IP phone to populate these buttons. The URLs point to
XML-based web pages formatted with XML tags that the Cisco IP phone understands and uses. When
you press a function button, the Cisco IP phone uses the configured URL to access the appropriate XML
web page for instructions. The web page sends instructions to the Cisco IP phone to display information
on the screen for you to navigate. You can select options and enter information by using soft keys and
the scroll button.
The Cisco IP Phones 7940 and 7940G and Cisco IP Phones 7960 and 7960G can support four URLs in
association with the four programmable feature buttons on IP phones. The four feature buttons on an IP
phone are configured using the url command keywords. The fifth button—Settings—is managed entirely
by the phone. Operation of these services is determined by the IP phone capabilities and the content of
the referenced URL.

SUMMARY STEPS

1. telephony-service
2. url {directory | information | messages | services} url
3. reset all [time-interval]
4. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 url {directory | information | messages | services} Provisions URLs for use by Cisco IP phones. The
url four keywords (directory, information, messages,
and services) correspond to the four feature buttons
Example: on an IP phone: Directories, Information, Messages,
Router(config-telephony)# url information and Services. The purpose of the url command is to
http://10.4.212.4/CCMUser/ provision the URLs through the SEPDEFAULT.cnf
GetTelecasterHelpText.asp configuration file supplied by the Cisco CME router
to the Cisco IP phones during phone registration. The
maximum character length for the URL is 128.
You can disable the local directory by entering the url
directories none command. You must reset the
Cisco IP phones before the url command can take
effect.
Note By default, the router automatically uses the
local directory service. Provisioning the
directory URL to select an external directory
resource disables the Cisco CME local
directory service.
Step 3 reset all [time-interval] Performs a complete reboot of all phones, including
contacting the DHCP and TFTP servers for the latest
configuration information.
Example:
Router(config-telephony)# reset all • all—Resets all phones associated with the
Cisco CME router.
• time-interval—Time interval, in seconds,
between the starts of succesive phone resets.
Range is 0 to 60. Default is 15.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example provisions the Information, Directories, Services, and Messages buttons.
telephony-service
url information http://10.4.212.4/CCMUser/GetTelecasterHelpText.asp
url directories http://10.4.212.11/localdirectory
url services http://10.4.212.4/CCMUser/123456/urltest.html
url messages http://10.4.212.4/Voicemail/MessageSummary.asp

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Direct FXO Trunk Lines

Direct FXO Trunk Lines

For Cisco CME3.2 and later versions, IP phones can be configured to have buttons for dedicated FXO
trunk lines, hereinafter referred to as FXO lines. Direct FXO trunk lines may used by companies whose
employees require private PSTN numbers. For example, a salesperson may need a special number that
customers can call without having to go through a main number. Whena call comes in to the direct
number, the salesperson knows that the caller is a customer. In the salesperson’s absence the customer
can leave voice mail. Dedicated lines can use PSTN service provider voice mail: when the line button is
pressed, the PSTN-FXO line is seized, allowing the user to hear the stutter dial tone provided by the
PSTN to indicate that voice messages are available.
Because dedicated FXO lines behave as private lines, users do not have to dial a prefix, such as 9 or 8,
to reach an outside line. To reach phone users within the company, FXO-line users must dial numbers
that use the company's PSTN number. For calls to non-PSTN destinations, such as local IP phones, a
second ephone-dn must be provisioned.
Calls placed to or received on an FXO line have restricted Cisco CME services (see the “Restrictions”
section on page 228) and cannot be transferred by Cisco CME. However, phone users are able to access
hookflash-controlled PSTN services using the flash soft key. Refer to the the fxo hook-flash command
in the Cisco CallManager Express 3.3 Command Reference.
From a high level, configuration of a direct FXO trunk line consists of the following:
• Configuring the FXO port for a private line automatic ringdown (PLAR) off-premises extension
(OPX) connection and declaring a private line’s number; for example:
voice-port 1/1/0
connection plar-opx 1020

• Configuring dial peers for FXO port and declaring a trunk tag to bind the FXO port and its dial peer
to an ephone-dn; for example:
dial-peer voice 111 pots
destination-pattern 82
port 1/1/0

• Configuring the ephone-dn and ephone; for example:

ephone-dn 12
number 1020

ephone-dn 1
mac-address 1111.1111.1111
button 1:12

• Binding the ephone-dn to the FXO port with the trunk command; for example:
ephone-dn 12
number 1020
trunk 82 timeout 30

The trunk command’s timeout seconds keyword and argument control the amount of time that
Cisco CME waits to collect digits for the dialed number, for the purpose of inclusion of the digits in
the redial buffer and the Placed Calls directory of the phone. Digits that are entered after the timeout
period are not included in the redial buffer or in the Placed Calls directory on the phone. The timeout
parameter does not affect the time used to cut through the connection from the phone’s trunk button
to the FXO port.
The phone user also has the option to use the phone’s on-hook dialing feature so that the phone itself
performs complete dial-string digit collection before signaling offhook to the Cisco CME. In this
case all digits will be included in the Redial and Placed Calls Directory.

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Restrictions
• An ephone-dn with a trunk line cannot be configured for call forward, busy, or no answer.
• An ephone-dn with a trunk line can be configured only as a single-line ephone-dn.
• Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed
on IP phones.
• Numbers entered after trunk line is seized will not appear in call history or call detail records
(CDRs) of a Cisco CME router. Only the trunk tag will be logged for calls made from trunk lines.
• FXO trunk lines do not support the CFwdAll, Transfer, Pickup, GPickUp, Park, CallBack, and
NewCall soft keys.
• FXO trunk lines do not support conference initiator dropoff.
• FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk
line before pressing the Redial button.
• FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference
an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP
phone connected. The conference initiator is unable to participate in the conference, but can place
calls on other lines.

SUMMARY STEPS

1. voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}

2. connection plar-opx digits
3. exit
4. dial-peer voice tag pots
5. destination-pattern [+] string [T]
6. port {slot-number/subunit-number/port | slot/port:ds0-group-number}
7. exit
8. ephone-dn dn-tag
9. number number
10. trunk digit-string timeout seconds
11. exit
12. ephone phone-tag
13. mac-address tag
14. button button-number{separator}dn-tag [[button-number{separator}dn-tag] ...]
15. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 voice-port {slot-number/subunit-number/port | Enters voice-port configuration mode.
slot/port:ds0-group-number}
Note The example shows a voice-port configuration for the
Cisco 2600, Cisco 3600 series, and Cisco 7200
Example: series. The syntax options for other platforms may
Router(config)# voice-port 0/0/0 vary. For more information, refer to the Cisco IOS
Voice Command Reference.
Step 2 connection plar-opx digits Specifies a PLAR OPX connection.
• Using this option, the local voice port provides a local
Example: response before the remote voice port receives an
Router(config-voice-port)# connection plar-opx answer. On FXO interfaces, the voice port does not
5550111 answer until the remote side has answered.
• digits—Specifies the destination telephone number.
Valid entries are any series of digits that specify the
E.164 telephone number.
Step 3 exit Exits voice-port configuration mode.

Example:
Router(config-voice-port)# exit
Step 4 dial-peer voice tag pots Enters dial-peer configuration mode for POTS.
• tag—Digits that define a particular dial peer. Range is
Example: from 1 to 2147483647.
Router(config)# dial-peer voice 53 pots
Step 5 destination-pattern [+] string [T] Declares a prefix, access code, or full E.164 telephone
number (depending on your dial plan) to be used for a dial
peer.
Example:
Router(config-dial-peer)# destination-pattern 20
Step 6 port {slot-number/subunit-number/port | Associates a dial peer with a specific voice port.
slot/port:ds0-group-number}
Note The example shows a voice-port configuration for the
Cisco 2600, Cisco 3600 series, and Cisco 7200
Example: series. The syntax options for other platforms may
Router(config-dial-peer)# port 0/0/0 vary. For more information, refer to the Cisco IOS
Voice Command Reference.

• Use the PLAR connection voice port configured by the

connection command.
Step 7 exit Exits dial-peer configuration mode.

Example:
Router(config-ephone-template)# exit

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Command or Action Purpose

Step 8 ephone-dn tag Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line, an intercom line, a
paging line, a voice-mail port, or an MWI.
Example:
Router(config)# ephone-dn 1 • tag—Unique sequence number that identifies an
ephone-dn during configuration tasks. Range is from 1 to
the maximum number of ephone-dns allowed on the
router platform. Refer to the command-line interface
(CLI) help for the maximum value for this argument.
Step 9 number number Associates a telephone or extension number with an
extension (ephone-dn).
Example: • number—String of up to 16 characters that represents an
Router(config-ephone-dn)# number 5550111 E.164 telephone number. Enter the PLAR number
configured by the connection command.
Step 10 trunk digit-string timeout seconds Associates an ephone-dn with an FXO port’s trunk number
so the ephone-dn can support a direct FXO line.
Example: • digit-string—Declares the number of the trunk line. Use
Router(config-ephone-dn)# trunk 20 timeout 30 the string argument specified in the destination-pattern
command.
• timeout seconds—Sets the timeout between digits for
dialing. The phone user has to either enter the pound (#)
key or wait for the interdigit timeout to complete the
digit dialing. Range is from 3 to 30. The default is 3.
Step 11 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 12 ephone phone-tag Enters ephone configuration mode for an IP phone.

Example:
Router(config)# ephone 1
Step 13 mac-address mac-address Associates the MAC address of a Cisco IP phone with an
ephone configuration in a Cisco CME system.
Example: • mac-address—The MAC address of an IP phone, which
Router(config-ephone)# mac-address is found on a sticker located on the bottom of the phone.
CFBA.321B.96FA

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Command or Action Purpose

Step 14 button button-number{separator}dn-tag Associates ephone-dns with individual buttons on Cisco IP
[[button-number{separator}dn-tag] ...] phones and specifies ring behavior per button.
• button-number—Number of a line button on a Cisco IP
Example: phone to be associated with an extension (ephone-dn).
Router(config-ephone)# button 1:1 The maximum number of button-ephone-dn pairs is
determined by phone type.
• separator—Single character that denotes the
characteristics to be associated with this button and
extension or extensions. For a list of options, see the
button command description in the Cisco CallManager
Express 3.3 Command Reference.
Note When o is used for the separator, the dn-tag
argument can contain up to ten individual DN tags,
separated by commas.

• dn-tag—Ephone-dn tag previously defined using the

ephone-dn command.
Step 15 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Examples
The following example shows the configuration for one phone that has 2 buttons: the first button is for
making calls to local extensions and for receiving calls, and the second button is for a private line that
goes out an FXO port as a direct trunk.
voice-port 1/0/0
connection plar opx 1001

dial-peer voice 100 pots

destination-pattern 81
voice-port 1/0/0

ephone-dn 1
name MainExtension
number 1001

ephone-dn 2
name PrivateTrunkLine
trunk 81 timeout 5

ephone 1
mac-address 1111.1111.1110
button 1:1 2:2

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Cisco IP Phone 7970G and 7971G-GE Settings

Cisco IP Phone 7970G and 7971G-GE Settings

Note For information about installing Cisco IP Phone 7970G or 7971G-GE firmware, see the “Software
Prerequisites” section on page 12.

The clocks in the Cisco IP Phone 7970G and 7971G-GE units obtain Coordinated Universal Time (UTC)
from their Cisco CME router’s clocks. To display the correct local time, nearly all Cisco IP Phone 7970G and
7971G-GE units’ time must be offset with the time-zone command.
In addition, the following display settings can be configured with the service phone command:
• The days of the week on which the phone displays will be inactive or blank.
• The length of time for which the phones displays will be active.
• The length of time for which the IP phones’ displays will remain active, starting from the last time
that the phone was used.
• The time at which the phones’ displays are activated, using a 24-hour time format. The default is
07:30.
The service phone command also enables and disables the following on Cisco IP Phone 7970G and
7971G-GE units:
• Speakerphone
• Headset
• Settings button
• Ethernet switch port on the IP phone for access to an Ethernet connection
• Ability to respond to gratuitous Address Resolution Protocol (ARP) messages from their Ethernet
interface
• Ability to permit data traffic from a PC attached to the IP phones’ ethernet port to access the voice
LAN (VLAN) that the IP phone uses for voice calls
• Ability to automatically delay activation of the IP phone’s PC Ethernet switch port when a Cisco IP
Phone 7970G or 7971G-GE unit is booted.
Note that the list of settings is not comprehensive. The service phone command configures all Cisco IP
Phone 7970G and 7971G-GE units in a Cisco CME system. The command works with the vendorConfig
section of the Sep*.conf.xml configuration file, which is read by the phone firmware when a Cisco IP
Phone 7970G or 7971G-GE is booted. The following is an example of an entry created in a
Sep*.conf.xml file:
<vendorConfig>
<parameter-name>parameter-value</parameter-name>
</vendorConfig>

Only the vendorConfig parameters that are supported by the currently loaded firmware are available. The
number and type of parameters may vary from one Cisco IP Phone 7970G or 7971G-GE firmware
version to the next.
For changes to the time-zone and service-phone settings take effect, the Sep*.conf.xml file must be
updated with the create cnf-files command and the Cisco IP Phone 7970G or 7971G-GE units must
rebooted with the reset command.
The show telephony-service tftp-binding command allows you to view the SEP*.cnf.xml files that are
associated with individual phones.

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Restrictions
The Cisco IP Phone 7970G and 7971G-GE running with Cisco CME do not support user and network
localization. The the user-locale and network-locale command configurations must be set to their
default, United States (US).

SUMMARY STEPS

1. telephony-service
2. time-zone number
3. service phone parameter-name parameter-value
4. create cnf-files
5. reset {all [time-interval] | cancel | mac-address mac-address | sequence-all}
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 time-zone number Sets the time zone for Cisco IP Phone 7970G and
7971G-GE clocks in a Cisco CallManager Express
(Cisco CME) system.
Example:
Router(config-telephony)# time-zone 2 • number—Numeric time zone name. Refer to CLI
help or the Cisco CallManager Express 3.3
Command Reference for a list of the time-zone
numbers.
• The default is time-zone 5, Pacific
Standard/Daylight Time (-480).
Step 3 service phone parameter-name parameter-value Sets display and phone functionality for the Cisco IP
Phone 7970G and 7971G-GE units using the
vendorConfig parameters of the downloaded
Example:
Router(config-telephony)# service phone garp 0
firmware’s Sep*.conf.xml configuration file.
• Refer to the Cisco CallManager Express 3.3
Command Reference for a complete description
of this command.
Step 4 create cnf-files Builds the eXtensible Markup Language (XML)
configuration files that are required for IP phones
used with Cisco CME and refreshes the time stamp
Example:
Router(config-telephony)# create cnf-files
that is applied to the Sep*.conf.xml configuration
file.
• You must issue the create cnf-files command for
service phone and time-zone command
configurations to take effect.

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Command or Action Purpose

Step 5 reset {all [time-interval] | cancel | Performs a complete reboot of all phones or the
mac-address mac-address | sequence-all} phone with the specified MAC address, including
contacting the DHCP and TFTP servers for the latest
Example: configuration information.
Router(config-telephony)# reset all • all—Resets all phones associated with a
Cisco CME router. This keyword causes the
router to pause 15 seconds between the reset start
for each successive phone.
• time-interval—(Optional) Time interval, in
seconds, between the start of each phone reset.
Range is from 0 to 60. Default is 15.
• cancel—Interrupts a sequential reset cycle.
• mac-address—Resets the phone that has the
specified MAC address.
• sequence-all—Resets all phones associated with
this Cisco CME router. This keyword causes the
router to wait until one reset is complete before
starting to reset the next phone. After the reset
timeout of 4 minutes, the router stops waiting for
the currently registering phone to complete
registration and starts to reset the next phone.
Step 6 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Examples

The following example sets the idle timeout parameter to 5 minutes and the time zone to Pacific Time:
Router(config)# telephony-service
Router(config-telephony)# service phone displayIdleTimeout 00:05
Router(config-telephony)# time-zone 5
Router(config-telephony)# create cnf-files
Router(config-telephony)# reset all

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 Integrating Voice Mail with Cisco CME

This chapter describes Cisco CallManager Express (Cisco CME) integration with Cisco Unity and how
to specify appropriate in-band dual tone multifrequency (DTMF) integration for other types of
voice-mail systems.

Note For extensions associated with analog telephone adaptors (ATAs), the message-waiting indication (MWI) is
a lit function button on the ATA and a stutter dial tone on the connected analog phone.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Cisco CME Integration with Cisco Unity, page 235
• Cisco CME Integration with Cisco Unity Express, page 236
• DTMF Integration for Legacy Voice-Mail Devices, page 236

Cisco CME Integration with Cisco Unity

Cisco Unity is a Windows 2000-based communications solution that brings you voice mail and unified
messaging and integrates them with the desktop applications you use every day. Cisco Unity gives you
the ability to access all of your messages—voice, fax, and e-mail—by using your desktop PC, a
touchtone phone, or the Internet. The Cisco Unity Voice Mail system supports voice-mail integration
with Cisco CME. This integration requires configuration of both the Cisco CME router and Cisco Unity
software to get voice-mail service.
For configuration instructions for the Cisco CME router and for Cisco Unity software on a PC, refer to
the Cisco CallManager Express 3.0 Integration Guide for Cisco Unity 4.0 at
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/integuid.

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Cisco CME Integration with Cisco Unity Express

Cisco CME Integration with Cisco Unity Express

Cisco Unity Express offers easy, one-touch access to messages and commonly used voice-mail features
that enable users to reply, forward, and save messages. To improve message management, users can
create alternate greetings, access envelope information, and mark or play messages based on privacy or
urgency. For instructions on how to configure Cisco Unity Express, refer to the administrator guides in
the Cisco Unity Express index at
http://www.cisco.com/univercd/cc/td/doc/product/voice/unityexp/index.htm.
For instructions on how to integrate Cisco CME with Cisco Unity Express, refer to Integrating
Cisco CallManager Express with Cisco Unity Express at
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/cmeinter/cue_cme1.pdf.

Note Cisco CME and Cisco Unity Express must both be configured before they can be integrated.

DTMF Integration for Legacy Voice-Mail Devices

For dual tone multifrequency (DTMF) integrations, information on how to route incoming or forwarded
calls is sent by a telephone system in the form of DTMF digits. The DTMF digits are sent in a pattern
that is based on the integration file in the voice-mail system connected to the Cisco CME router. These
patterns are required for the DTMF integration of Cisco CME with most voice-mail systems. DTMF
integration configuration on the Cisco CME router works with any analog voice-mail system. Voice-mail
systems are designed to respond to DTMF after the system has answered the incoming calls. The
following tasks are required:
• Configuring DTMF Patterns on the Router, page 236 (required)
• Configuring Integration Files on Legacy Voice-Mail Systems, page 238 (required)

Configuring DTMF Patterns on the Router

The Cisco CME router provides the flexibility to integrate with any legacy voice-mail system. You can
configure multiple tags and tokens for each pattern, depending on the voice-mail system and type of
access. The tag argument used in the configuration pattern must match the number defined in the
voice-mail system’s integration file to identify the type of call. The keywords—CGN (calling number),
CDN (called number), and FDN (forwarding number)—define the type of call information that is sent
to the voice-mail system.

Note Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a
single command line, it is permissible to do so.

SUMMARY STEPS

1. vm-integration
2. pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
3. pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]

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DTMF Integration for Legacy Voice-Mail Devices

4. pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
5. pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
6. pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
7. exit

DETAILED STEPS

Command or Action Purpose

Step 1 vm-integration Enters voice-mail integration configuration mode and
enables voice-mail integration with DTMF and an analog
voice-mail system.
Example:
Router(config) vm-integration
Step 2 pattern direct tag1 {CGN | CDN | FDN} [tag2 Configures the DTMF digit pattern forwarding necessary to
{CGN | CDN | FDN}] [tag3 {CGN | CDN | FDN}] activate the voice-mail system when the user presses the
[last-tag]
messages button on the phone.
• The tag attribute is an alphanumeric string fewer than
Example: four DTMF digits in length. The alphanumeric string
Router(config-vm-integration) pattern direct
consists of a combination of four letters (A, B, C, and D),
2 CGN *
two symbols (* and #), and ten digits (0 to 9). The tag
numbers match the numbers defined in the voice-mail
system’s integration file, immediately preceding either
the number of the calling party, the number of the called
party, or a forwarding number. The Cisco SRS Telephony
router supports a maximum of four tags.
• The keywords—CGN, CDN, and FDN—configure the
type of call information sent to the voice-mail system,
such as calling number (CGN), called number (CDN), or
forwarding number (FDN).
Step 3 pattern ext-to-ext busy tag1 {CGN | CDN | Configures the DTMF digit pattern forwarding necessary to
FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | activate the voice-mail system once an internal extension
CDN | FDN}] [last-tag]
attempts to connect to a busy extension and the call is
forwarded to voice mail.
Example:
Router(config-vm-integration) pattern
ext-to-ext busy 7 FDN * CGN *
Step 4 pattern ext-to-ext no-answer tag1 {CGN | CDN Configures the DTMF digit pattern forwarding necessary to
| FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | activate the voice-mail system once an internal extension
CDN | FDN}] [last-tag]
fails to connect to an extension and the call is forwarded to
voice mail.
Example:
Router(config-vm-integration) pattern
ext-to-ext no-answer 5 FDN * CGN *

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Command or Action Purpose

Step 5 pattern trunk-to-ext busy tag1 {CGN | CDN | Configures the DTMF digit pattern forwarding necessary to
FDN} [tag2 {CGN | CDN | FDN}] [tag3 {CGN | activate the voice-mail system once an external trunk call
CDN | FDN}] [last-tag]
reaches a busy extension and the call is forwarded to voice
mail.
Example:
Router(config-vm-integration) pattern
trunk-to-ext busy 6 FDN * CGN *
Step 6 pattern trunk-to-ext no-answer tag1 {CGN | Configures the DTMF digit pattern forwarding necessary to
CDN | FDN} [tag2 {CGN | CDN | FDN}] [tag3 activate the voice-mail system when an external trunk call
{CGN | CDN | FDN}] [last-tag]
reaches an unanswered extension and the call is forwarded to
voice mail.
Example:
Router(config-vm-integration)# pattern
trunk-to-ext no-answer 4 FDN * CGN *
Step 7 exit Exits voice-mail integration configuration mode.

Example:
Router(config-vm-integration)# exit

Configuring Integration Files on Legacy Voice-Mail Systems

To configure the integration files on a third-party legacy voice-mail system, follow the instructions in
the documents that accompany the voice-mail system. You must design the DTMF integration patterns
appropriately so that the voice-mail system and the Cisco CME router work with each other.

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 Configuring an Attendant for Primary Call
Coverage

This chapter describes features that assist you in setting up an attendant to be the single initial source of
incoming call coverage in a Cisco CallManager Express (Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Related Features, page 239
• Label Support, page 240
• Monitor Lamp and Direct Station Select, page 241
• Silent Ring, page 243
• On-Hold Call Notification, page 244
• Night Service, page 246

Related Features
The following related features in the “Configuring Secondary Call Coverage” chapter provide additional
call-coverage functionality:
• Ephone-dn Overlays
• Ephone-dn Dial-Peer Preference
• Huntstop
• Ephone Hunt Groups
• Call-Pickup Groups

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Label Support

Label Support
The label support feature controls the display adjacent to an ephone-dn line button. By default, an IP
phone displays the extension number that is set using the number command under an ephone-dn. The
label support feature allows you to enter a meaningful text string for each ephone-dn so that a phone user
with multiple lines can select a line by name instead of by phone number, thus eliminating the need to
consult in-house phone directories.

SUMMARY STEPS

1. ephone-dn dn-tag
2. label label-string
3. exit
4. ephone phone-tag
5. restart
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies the
Example: ephone-dn to which the label is to be associated.
Router(config)# ephone-dn 1
Step 2 label label-string Creates a custom label that is displayed on the phone next
to the line button that is associated with this ephone-dn. The
custom label replaces the default label, which is the number
Example:
Router(config-ephone-dn)# label user1
that was assigned to this ephone-dn.
• label-string—String of up to 30 alphanumeric
characters that provides the label text.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 4 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique identifier (sequence number) of
Example: the ephone on which you are defining the new label.
Router(config)# ephone 1

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Monitor Lamp and Direct Station Select

Command or Action Purpose

Step 5 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-ephone)# restart
Step 6 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example creates text labels for two ephone-dns:

ephone-dn 1
number 2001
label Sales

ephone-dn 2
number 2002
label Engineering

Monitor Lamp and Direct Station Select

For multibutton phones and expansion modules, the buttons for lines that are shared with other phones
can be designated as monitor buttons. A monitor button is created primarily to show the status of a line
that also appears on another phone. When the shared line is in use on the other phone, the console display
for the line’s monitor button shows an icon of a phone covered by an X. An idle line is represented by
an icon of a phone with the handset in the on-hook position. Monitor buttons cannot be used to make
outgoing calls, as opposed to line buttons for ordinary shared lines. This functionality is known on
private branch exchange (PBXs) as a monitor lamp.
When a monitored line is idle, pressing the monitor button will speed-dial the monitored line. This
functionality is sometimes known as fast transfer or direct station select.
The monitor lamp feature is configured using the m keyword of the button command. This feature can
be configured on any multibutton IP phone, such as a Cisco IP Phone 7960G with one or two Cisco IP
Phone 7914 modules. Phones are configured with a shared line for each extension that uses the monitor
function.

SUMMARY STEPS

1. ephone phone-tag
2. button button-number{m}dn-tag [[button-number{m}dn-tag]...]
3. restart
4. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone (Ethernet phone) configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone on which you are adding a button in monitor
Router(config)# ephone 1 mode.
Step 2 button button-number{m}dn-tag This command with the m keyword as a separator assigns an
[[button-number{m}dn-tag]...] ephone-dn to a line button in monitor mode.
• button-number—Number of a phone button, starting
Example: with 1 as the topmost button on the phone.
Router(config-ephone)# button 1m22 2m23
• m—Separator that indicates that the extension is a
monitor extension.
• dn-tag—Ephone-dn tag (sequence number) of the
extension that you want to monitor on this phone. This
ephone-dn tag must also be assigned to a button on
another phone. This is a shared ephone-dn.
For a description of other arguments and keywords used in
this command, refer to the Cisco CallManager Express 3.3
Command Reference.
Step 3 restart Performs a fast reboot of this ephone. Does not contact the
DHCP or TFTP server for updated information.
Example:
Router(config-ephone)# restart
Step 4 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

In this example, lines 1 through 4 represent incoming PSTN lines, and lines 5 and 6 are monitored
extension numbers.
ephone 1
button 1:10 2:11 3:12 4:13 5m21 6m22

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Silent Ring

Silent Ring
The silent ring feature allows you to designate phone buttons that do not emit an audible ring when they
receive incoming calls. Although this feature is supported by all phone types, it is most useful on phone
buttons that are used to display the activity of shared lines, which are typically found on the
Cisco IP Phones 7960 and 7960G and Cisco IP Phone Expansion Module 7914.
When the silent ring feature is enabled, the only visible cue that signals an incoming call is a flashing
((< icon in the phone display. On lines that have silent ring, you can specify whether you want to hear
call-waiting beeps or whether you want them suppressed if a second call comes in.
The silent ring feature is enabled when a line button is configured with ringer option s or b.

SUMMARY STEPS

1. ephone phone-tag
2. button button-number{b | s}dn-tag [[button-number{b | s}dn-tag]...]
3. restart
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone (Ethernet phone) configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone on which you are adding a button in silent
Router(config)# ephone 4 mode.
Step 2 button button-number{b | s}dn-tag This command with a b or s keyword assigns an ephone-dn
[[button-number{b | s}dn-tag]...] to a line button in silent mode.
• button-number—Number of a phone button, starting
Example: with 1 as the topmost button on the phone.
Router(config-ephone)# button 1b10 2b11 3s12
• b—Beep allowed, but no ring. Audible ring is
suppressed for incoming calls, but call-waiting beeps
are allowed. Visible cues are the same as for normal
ring.
• s—Silent ring. Audible ring and call-waiting beeps are
suppressed for incoming calls. The only visible cue is a
flashing ((< icon in the phone display.
• dn-tag—Ephone-dn tag (sequence number) of the
extension to which you want to assign the silent ring on
this phone.
For a description of other arguments and keywords used in
this command, refer to the Cisco CallManager Express 3.3
Command Reference.

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On-Hold Call Notification

Command or Action Purpose

Step 3 restart Performs a fast reboot of this ephone. Does not contact the
Dynamic Host Configuration Protocol (DHCP) or TFTP
server for updated information.
Example:
Router(config-ephone)# restart
Step 4 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example sets buttons 1 and 2 for silent ring with call-waiting beeps allowed and sets
button 3 for silent ring:
ephone 4
button 1b10 2b11 3s12

On-Hold Call Notification

This service adds an audible alert as a reminder to an IP phone user that a call is waiting on hold. This
feature is primarily intended for use by an attendant but may be used on any IP phone.

SUMMARY STEPS

1. ephone-dn dn-tag
2. hold-alert timeout {idle | originator | shared}
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—The unique sequence number that identifies
Example: the ephone-dn to which you want to assign the
Router(config)# ephone-dn 11 hold-alert feature.
Step 2 hold-alert timeout {idle | originator | shared} Sets audible alert notification on the Cisco IP phone for
alerting the user about on-hold calls. The timeout argument
specifies the time interval in seconds from the time the call
Example:
Router(config-ephone-dn)# hold-alert 15 idle
is placed on hold to the time the on-hold audible alert is
generated. The alert is repeated at the end of the set timeout
value.
• idle—Generates a 1-second burst of ringing on the IP
phone that placed the call into the hold state if that
phone is in the idle state. If the phone is in active use,
no on-hold alert is generated.
• originator—Generates a 1-second burst of ringing on
the phone that placed the call into the hold state if the
phone is in the idle state. If the phone is in use on
another call, an audible beep is generated (call-waiting
tone).
Note From the perspective of the originator of the call on
hold, the originator and shared keywords provide
the same functionality.

• shared—Generates a 1-second burst of ringing for all

the idle phones that share the same line appearance. If
the phones are in use, they do not hear an audible beep.
Only the phone that initiated the call, if it is in use on
another call, hears call-waiting beeps.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

The following example generates a burst of audible ringing every 10 seconds that a call is on hold on an
idle phone:
ephone-dn 1
hold-alert 10 idle

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 Configuring an Attendant for Primary Call Coverage
Night Service

Night Service
The night-service feature allows you to provide coverage for unstaffed extensions during hours that you
designate as “night-service” hours. During the night-service hours, calls to the designated extensions
send a special “burst” ring to other phones that have been specified to receive this special ring. Phone
users at the other phones can then use the call-pickup feature to answer the incoming calls.
For example, the night-service feature allows an employee working after hours to intercept calls that are
presented to an unattended receptionist’s phone and redirect them to be answered at the employee’s own
phone. This feature is useful for sites at which all incoming public switched telephone network (PSTN)
calls have to be transferred by a receptionist because the PSTN connection to the Cisco CME system
does not support Direct Inward Dialing (DID). When a call arrives at the unattended receptionist’s phone
during hours that are specified as night service, a ring burst notifies a specified set of phones of the
incoming call. Any phone user at one of the specified phones can intercept the call using the call-pickup
feature. Night-service call notification is sent every 12 seconds until the call is either answered or
aborted.
If optionally configured, night service can be manually toggled on and off from any phone that has a line
that is designated as a night-service line. When night service is active, a message is displayed on the
night-service phones.
Night service requires that you define the following parameters:
1. Night-service time period—Day or date and hours during which night service is active. Steps 2 and
3 in the following procedure define the night-service period.
2. Night-service extensions (ephone-dns)—When a night-service extension receives an incoming call
during the night-service period, night-service notification is triggered. Steps 6 through 8 in the
following procedure define an ephone-dn as having night service and specify the ephone on which
the ephone-dn appears.
3. Night-service notification phones (ephones)—Night-service notification phones are alerted with a
distinctive ring when incoming calls are received on night-service lines during the night-service
time period. The night-service notification phone user can answer the call using call pickup or group
call pickup. Steps 9 through 11 in the following procedure assign night-service notification to a
phone. This phone receives a distinctive alerting ring and notification display when a night-service
extension receives an incoming call.
4. (Optional) Night-service toggle code—A code to allow night-service treatment to be manually
toggled off and on from any phone that has a line assigned to night service. Prior to Cisco CME 3.3,
using the night-service code turned night service on or off only for ephone-dns on the phone at
which the code was entered. In Cisco CME 3.3 and later versions, using the night-service code at
any phone with a night-service ephone-dn turns night service on or off for all phones with
night-service ephone-dns. Step 4 in the following procedure defines a night-service toggle code.
Figure 32 illustrates night service.

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Night Service

Figure 32 Night Service

Phone 5
1 Extension 1000 has been designated as a night-service Button 1 is extension 1000
IP
extension (ephone-dn). When extension 1000 receives an Extension 1000 is a night-
incoming call during a night-service period, phone 5 rings service extension
and notification is made to the night-service phones.

2 Phones 14 and 15 have been designated as night-

service phones. When phone 5 starts ringing,
phones 14 and 15 ring once and display “Night Service
1000.” The incoming call on extension 1000 can be V
answered from phone 14 or phone 15 using call pickup.

telephony-service IP
night-service day fri 17:01 17:00
Phone 14
night-service day sat 17:01 17:00
Button 1 is extension 1010
night-service day sun 17:01 07:59
Phone 14 is a night-service phone
night-service date jan 1 00:00 00:00
night-service code *1234
!
ephone-dn 1 IP
number 1000
night-service bell Phone 15
! Button 1 is extension 1011
ephone-dn 10 Phone 15 is a night-service phone
number 1010
!
ephone-dn 11
number 1011
!
ephone 5
mac-address 1111.2222.0001
button 1:1
!
ephone 14
mac-address 1111.2222.0002
button 1:10
night-service bell
!
ephone 15
mac-address 1111.2222.0003

88951
button 1:11
night-service bell

SUMMARY STEPS

1. telephony-service
2. night-service day day start-time stop-time
3. night-service date month date start-time stop-time
4. night-service code digit-string
5. exit
6. ephone-dn dn-tag
7. night-service bell

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Night Service

8. exit
9. ephone phone-tag
10. night-service bell
11. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 night-service day day start-time stop-time Defines a recurring time period associated with a day of the
week during which night service is active.
Example: • day—Day of the week abbreviation. The following are
Router(config-telephony)# night-service day mon valid day abbreviations: sun, mon, tue, wed, thu, fri,
19:00 07:00 sat.
• start-time stop-time—Beginning and ending times for
night service, in an HH:MM format using a 24-hour
clock. If the stop time is a smaller value than the start
time, the stop time occurs the day following the start
time. For example, “mon 19:00 07:00” means “from
Monday at 7 p.m. until Tuesday at 7 a.m.”
Step 3 night-service date month date start-time Defines a recurring time period associated with a month and
stop-time date during which night service is active.
• month—Month abbreviation. The following are valid
Example: month abbreviations: jan, feb, mar, apr, may, jun, jul,
Router(config-telephony)# night-service date aug, sep, oct, nov, dec.
jan 1 00:00 00:00
• date—Date of the month. Range is from 1 to 31.
• start-time stop-time—Beginning and ending times for
night service, in an HH:MM format using a 24-hour
clock. The stop time must be greater than the start time.
The value 24:00 is not valid. If 00:00 is entered as an
stop time, it is changed to 23:59. If 00:00 is entered for
both start time and stop time, calls are blocked for the
entire 24-hour period on the specified date.
Step 4 night-service code digit-string Designates a code that can be dialed from any night-service
line (ephone-dn) to toggle night service on and off for all
lines that have been assigned to night service in the system.
Example:
Router(config-telephony)# night-service code
The night-service state is indicated in a display message on
*6483 phones that have active night-service lines.
• digit-string—String of up to 16 keypad digits. The code
must begin with an asterisk (*).

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Command or Action Purpose

Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 ephone-dn dn-tag Enters ephone-dn configuration mode to define an
ephone-dn to receive night-service treatment.
Example: • dn-tag—Unique sequence number that identifies the
Router(config)# ephone-dn 55 ephone-dn to receive night-service treatment.
Step 7 night-service bell Marks this ephone-dn for night-service treatment. Incoming
calls to this ephone-dn during the night-service time period
send an alert notification to all IP phones that are marked to
Example:
Router(config-ephone-dn)# night-service bell
receive night-service bell notification.
Step 8 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 9 ephone phone-tag Enters ephone configuration mode. This is a phone that will
be notified when an incoming call is received by a
night-service ephone-dn during a night-service period.
Example:
Router(config)# ephone 12 • phone-tag—The unique sequence number of the phone
that you are designating as a night-service phone.
Step 10 night-service bell Marks this phone to receive night-service bell notification
when incoming calls are received on ephone-dns marked for
night service during the night-service time period. The alert
Example:
Router(config-ephone)# night-service bell
notification is a splash ring that is not associated with any
of the individual lines on the IP phone and a visual display
of the ephone-dn line number. The phone user can pick up
the call by executing a PickUp or GPickUp.
Step 11 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example provides night service before 8 a.m. and after 5 p.m. Monday through Friday,
before 8 a.m. and after 1 p.m. on Saturday, and all day Sunday. Extension 1000 is designated as a
night-service extension, which means that incoming calls to extension 1000 during the night-service
period will ring on extension 1000 and provide night-service notification to phones that are designated
as night-service phones. In this example, the night-service phones are ephone 14 and ephone 15. The
night-service notification consists of a single ring on the phone and a display of “Night Service 1000.”
A night-service toggle code has been configured, *6483 (*NITE), by which a phone user can activate or
deactivate night-service conditions during the hours of night service.
telephony-service
night-service day mon 17:00 08:00
night-service day tue 17:00 08:00
night-service day wed 17:00 08:00

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night-service day thu 17:00 08:00

night-service day fri 17:00 08:00
night-service day sat 13:00 12:00
night-service day sun 12:00 08:00
night-service code *6483
!
ephone-dn 1
number 1000
night-service bell
!
ephone-dn 2
number 1001
night-service bell
!
ephone-dn 10
number 2222
!
ephone-dn 11
number 3333
!
ephone 5
mac-address 1111.2222.0001
button 1:1 2:2
!
ephone 14
mac-address 1111.2222.0002
button 1:10
night-service bell
!
ephone 15
mac-address 1111.2222.0003
button 1:11
night-service bell

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 Configuring Call Blocking

This chapter describes features that can be used to restrict calls in a Cisco CallManager Express
(Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date, page 251
• Call-Blocking (Toll Bar) Override, page 254
• Do-Not-Disturb Service, page 256
• Class of Restriction, page 257

Call Blocking (Toll Bar) Based on Time of Day and Day of Week
or Date
Call blocking to prevent unauthorized use of phones is implemented by matching a pattern of specified
digits during a specified time of day and day of week or date. Up to 32 patterns of digits can be specified.
Call blocking is supported on IP phones only and not on analog foreign exchange station (FXS) phones.
When a user attempts to place a call to digits that match a pattern that has been specified for call blocking
during a time period that has been defined for call blocking, a fast busy signal is played for
approximately 10 seconds. The call is then terminated, and the line is placed back in on-hook status.
Call blocking applies to all IP phones in a Cisco CME system, although individual IP phones can be
exempted from all call blocking.

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Call Blocking (Toll Bar) Based on Time of Day and Day of Week or Date

Individual phone users can be allowed to override call blocking associated with designated time periods
by entering personal identification numbers (PINs) that have been assigned to their phones. For more
information, see the “Call-Blocking (Toll Bar) Override” section on page 254.

SUMMARY STEPS

1. telephony-service
2. after-hours block pattern tag pattern [7-24]
3. after-hours day day start-time stop-time
4. after-hours date month date start-time stop-time
5. exit
6. ephone phone-tag
7. after-hour exempt
8. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 after-hours block pattern tag pattern [7-24] Defines a pattern of outgoing digits to be blocked. Up to 32
patterns can be defined, using individual commands.
Example: • If the 7-24 keyword is specified, the pattern is always
Router(config-telephony)# after-hours block blocked, 7 days a week, 24 hours a day.
pattern 1 91900
• If the 7-24 keyword is not specified, the pattern is
blocked during the days and dates that are defined using
the after-hours day and after-hours date commands.
Step 3 after-hours day day start-time stop-time Defines a recurring time period based on the day of the
week during which calls are blocked to outgoing dial
patterns that are defined using the after-hours block
Example:
Router(config-telephony)# after-hours day mon
pattern command.
19:00 7:00 • day—Day of the week abbreviation. The following are
valid day abbreviations: sun, mon, tue, wed, thu, fri,
sat.
• start-time stop-time—Beginning and ending times for
call blocking, in an HH:MM format using a 24-hour
clock. If the stop time is a smaller value than the start
time, the stop time occurs on the day following the start
time. For example, “mon 19:00 07:00” means “from
Monday at 7 p.m. until Tuesday at 7 a.m.”

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Command or Action Purpose

Step 4 after-hours date month date start-time Defines a recurring time period based on month and date
stop-time during which calls are blocked to outgoing dial patterns that
are defined using the after-hours block pattern command.
Example: • month—Month abbreviation. The following are valid
Router(config-telephony)# after-hours date jan month abbreviations: jan, feb, mar, apr, may, jun, jul,
1 0:00 0:00
aug, sep, oct, nov, dec.
• date—Date of the month. Range is from 1 to 31.
• start-time stop-time—Beginning and ending times for
call blocking, in an HH:MM format using a 24-hour
clock. The stop time must be larger than the start time.
The value 24:00 is not valid. If 00:00 is entered as an
stop time, it is changed to 23:59. If 00:00 is entered for
both start time and stop time, calls are blocked for the
entire 24-hour period on the specified date.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 ephone phone-tag Enters ephone configuration mode.
• phone-tag—The unique sequence number for the phone
Example: that is to be exempt from call blocking.
Router(config)# ephone 4
Step 7 after-hour exempt Specifies that this phone is exempt from call blocking. Note
that phones exempted in this manner are not restricted from
any call-blocking patterns and that no authentication of the
Example:
Router(config-ephone)# after-hour exempt
phone user is required.
For a different method of allowing phone users to call
blocked patterns, see the “Call-Blocking (Toll Bar)
Override” section on page 254.
Step 8 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example defines several patterns of digits for which outgoing calls are blocked. Patterns 1
and 2, which block calls to external numbers that begin with “1” and “011,” are blocked on Monday
through Friday before 7 a.m. and after 7 p.m., on Saturday before 7 a.m. and after 1 p.m., and all day
Sunday. Pattern 3 blocks calls to 900 numbers 7 days a week, 24 hours a day. The IP phone with sequence
number (phone-tag) 23 and MAC address 00e0.8646.9242 is not restricted from calling any of the
blocked patterns.
telephony-service
after-hours block pattern 1 91
after-hours block pattern 2 9011
after-hours block pattern 3 91900 7-24
after-hours block day mon 19:00 07:00
after-hours block day tue 19:00 07:00

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after-hours block day wed 19:00 07:00

after-hours block day thu 19:00 07:00
after-hours block day fri 19:00 07:00
after-hours block day sat 13:00 12:00
after-hours block day sun 12:00 07:00
!
ephone 23
mac 00e0.8646.9242
button 1:33
after-hour exempt

ephone 24
mac 2234.1543.6352
button 1:34

Call-Blocking (Toll Bar) Override

For IP phones that support soft keys, such as the Cisco IP Phone 7940G and the Cisco IP Phone 7960G,
the call-blocking override feature allows individual phone users to override the call blocking that has
been defined for designated time periods. The system administrator must first assign a personal
identification number (PIN) to any phone that will be allowed to override call blocking.
Then, to override call blocking, the phone user presses the Login soft key on the phone and enters the
PIN that is associated with the phone. Note that logging in to a phone with a PIN only allows the user to
override call blocking that is associated with particular time periods. Blocking patterns that are created
with the 7-24 keyword in the after-hours block pattern command are in effect 7 days a week, 24 hours
a day, and they cannot be overridden by using a PIN.
When PINs are configured for call-blocking override, they are cleared at a specific time of day or after
phones have been idle for a specific amount of time. The time of day and amount of time can be set by
the system administrator, or the defaults can be accepted.

Restrictions
Call-blocking override is available only on IP phones that have soft-key support, such as the Cisco IP
Phone 7940G and the Cisco IP Phone 7960G.

SUMMARY STEPS

1. ephone phone-tag
2. pin pin-number
3. exit
4. telephony-service
5. login [timeout [minutes]] [clear time]
6. restart all
7. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 ephone phone-tag Enters ephone configuration mode.
• phone-tag—The unique sequence number of the phone
Example: to which you are assigning a PIN.
Router(config)# ephone 2
Step 2 pin pin-number Declares a personal identification number (PIN) for a
designated ephone.
Example: • pin-number—Number from four to eight digits in
Router(config-ephone)# pin 5555 length.
Step 3 exit Exits ephone configuration mode.

Example:
Router(config-ephone) # exit
Step 4 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 5 login [timeout [minutes]] [clear time] Specifies that the Cisco CME system should deactivate all
user logins at a specific time or after a designated period of
idle time on a phone.
Example:
Router(config-telephony)# login timeout 120 • timeout—(Optional) Deactivates logins a given
clear 23:00 number of minutes after a phone becomes idle.
• minutes—(Optional) Number from 5 to 1440. Default
is 60.
• clear—(Optional) Deactivates all logins at a specified
time.
• time—(Optional) Time of day using 00:00 to 24:00 on
a 24-hour clock. For example, 10:30 p.m. is 22:30.
Default is 24:00 (midnight).
Step 6 restart all Performs a fast reboot of all phones associated with this
Cisco CME router. Does not contact the DHCP or TFTP
server for updated information.
Example:
Router(config-telephony)# restart all
Step 7 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

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Do-Not-Disturb Service

Example

The following example deactivates a phone’s login after three hours of idle time and clears all logins at
10 p.m.:
ephone 1
pin 1000
!
telephony-service
login timeout 180 clear 2200

Troubleshooting Tips

Use the show ephone login command to display the login status of all phones.

Do-Not-Disturb Service
Do-not-disturb (DND) service can be enabled using a soft key on a Cisco IP Phone 7902G, Cisco IP
Phone 7905G, Cisco IP Phone 7940G, or Cisco IP Phone 7960G. When DND is enabled, incoming calls
do not ring on the phone, but they do provide visual alerting and call information and can be answered
if desired. A display message indicates that DND is in effect.
For Cisco CME 3.2 and later versions, when a local IP phone calls another local IP phone that is in the
DND state, the message “Ring out DND” is displayed on the calling phone indicating that the target
phone is in the DND state. Pressing DND during an incoming call diverts the call to a call-forward
no-answer destination. If call-forward no-answer is not configured, the ringer is disabled.
For Cisco CME 3.2 and later versions, enabling DND automatically removes an IP phone from its hunt
group or groups, as illustrated in Figure 33. The ability to log out of hunt groups allows phone users to
leave their desks and have their calls routed to other phones. Unnecessary routing is avoided, and callers
are sent to phones that will be answered. Disabling DND automatically logs an IP phone back into its
hunt group or hunt groups.
For information about hunt groups, see the “Ephone Hunt Groups” section on page 273.
No configuration is required for DND. The show ephone dnd command displays all phones that have
DND enabled. The show ephone-hunt command can be used to determine what phones are logged in
and logged out.

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Class of Restriction

Figure 33 Phone 1004 Presses DND to Logs Out of Its Hunt Groups Temporarily

Any phone dials the pilot number.

IP

Pilot number ephone hunt 1 longest-idle

5000 pilot 5000
list 1001, 1002, 1003, 1004, 1005, 1006
final 6000
hops 3
hunt group 1 preference 1
V timeout 30
IP no-reg
6000 IP
Voice-mail server IP
IP

103588
IP
IP

Class of Restriction
Class of restriction (COR) is used to specify which incoming dial peer can use which outgoing dial peer
to make a call. Each dial peer can be provisioned with an incoming and an outgoing COR list. The cor
command sets the dial-peer COR parameter for dial peers and the directory numbers that are created for
Cisco IP phones associated with the Cisco CME router. COR functionality provides the ability to deny
certain call attempts on the basis of the incoming and outgoing class of restrictions that are provisioned
on the dial peers. This functionality provides flexibility in network design, allows users to block calls
(for example, calls to 900 numbers), and applies different restrictions to call attempts from different
originators.
For more information on setting COR, refer to the “Class of Restrictions” section in the “Dial Peer
Configuration on Voice Gateway Routers” chapter in the Cisco Voice Configuration Library,
Release 12.3.

SUMMARY STEPS

1. ephone-dn dn-tag
2. cor {incoming | outgoing} cor-list-name
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—The unique sequence number for the ephone-dn to
Example: which you want to apply a COR.
Router(config)# ephone-dn 12
Step 2 cor {incoming | outgoing} cor-list-name Configures a COR on the dial peers associated with an ephone-dn.

Example:
Router(config-ephone-dn)# cor outgoing
corlist2
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

In the example are three dial peers for dialing local destinations, long distance, and 911. COR list user1
can access the dial peers used to call 911 and local destinations. COR list user2 can access all three dial
peers. Ephone-dn 1 is assigned COR list user1 to call local destinations and 911, and ephone-dn 2 is
assigned COR list user2 to call 911, local destinations, and long distance.
dial-peer cor custom
name local
name longdistance
name 911

dial-peer cor list call-local

member local

dial-peer cor list call-longdistance

member longdistance

dial-peer cor list call-911

member 911

dial-peer cor list user1

member 911
member local

dial-peer cor list user2

member 911
member local
member longdistance

dial-peer voice 1 pots

corlist outgoing call-longdistance
destination-pattern 91..........
port 2/0/0

dial-peer voice 2 pots

corlist outgoing call-local
destination-pattern 9[2-9]......
port 2/0/0

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dial-peer voice 3 pots

corlist outgoing call-911
destination-pattern 9911
port 2/0/0

ephone-dn 1
cor incoming user1

ephone-dn 2
cor incoming user2

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 Configuring Secondary Call Coverage

This chapter describes features that allow you to selectively control the ways in which calls are routed
to IP phone users to ensure incoming call coverage in the absence of, or in addition to, a primary
attendant in a Cisco CallManager Express (Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Related Features, page 261
• Information About Configuring Call Coverage, page 262
• Overlaid Ephone-dns, page 262
• Call Waiting for Overlaid Ephone-dns, page 267
• Ephone-dn Dial-Peer Preference, page 270
• Huntstop, page 271
• Ephone Hunt Groups, page 273
• Automatic Hunt Group Logout, page 284
• Call-Pickup Groups, page 286

Related Features
See the features in the “Configuring an Attendant for Primary Call Coverage” chapter.

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Information About Configuring Call Coverage

Information About Configuring Call Coverage

Call coverage can be defined as the ability to ensure that all incoming calls are answered by someone,
even if the number that was originally dialed is busy or does not answer. Call coverage means that you
ensure an adequate number of ephone-dns (virtual voice ports) for incoming calls as well as the ability
for incoming calls to search among available ephone-dns until they are answered. This chapter explains
some of the Cisco CME options that you can use to provide call coverage.
Ephone-dn overlays, for example, are a means of placing more lines (ephone-dns) on a phone than there
are line buttons to accommodate them. The ephone-dns that are overlaid on a button can also be shared
among other phones to increase the likelihood that someone will be available to answer calls on them.
See the “Ephone-dns” section in the “Cisco CallManager Express Overview” chapter for information
about the types of ephone-dn that can be used for various types of call coverage.
Another feature, ephone dial-peer preference, allows you to specify the order in which you would like
ephone-dns hunted, and the huntstop command allows you to control whether calls continue to hunt if
the number that was dialed is busy or does not answer. Specific ephone hunt groups can be created that
hunt only through a designated set of ephone-dns in a specified order.
Call-pickup groups enable phone users to easily answer calls that are ringing on other phones. Any or
all of these features can be combined with the use of shared lines and secondary numbers to design the
call coverage plan that is best suited to your needs.

Overlaid Ephone-dns
Overlaid ephone-dns allow more than one ephone-dn to share the same physical line button on an IP
phone. Overlaid ephone-dns can be used to receive incoming calls and place outgoing calls. Up to 25
ephone-dns can be assigned to a single phone button.
Overlaid ephone-dns can use ephone-dns with the same number or different numbers. The order in which
ephone-dns are used by incoming calls can be determined by the ephone-dns’ preference configurations.
For example, ephone-dn 1 through ephone-dn 4 have the same number of 1001, and three phones are
configured with the button 1o1,2,3,4 command. A call to 1001 will ring on the ephone-dn with the
highest preference and display the caller ID on all phones that are on hook. If another incoming call to
1001 is placed while the first call is active (and the first ephone-dn with the highest preference is
configured with the no huntstop command), the call will roll over to the ephone-dn with the next-highest
preference, and so forth. If the ephone-dns in an ephone-dn overlay use different numbers, incoming
calls will go to these numbers’ ephone-dns with highest preferences. If no preferences are configured,
the dial-peer hunt command setting will be used to determine which ephone-dns are used for incoming
calls. The default setting for the dial-peer hunt command is to randomly select an ephone-dn that
matches the called number.

Note To continue or to stop the search for ephone-dns, you must use, respectively, the no huntstop and
huntstop commands under the individual ephone-dns. The huntstop setting is applied only to the dial
peers affected by the ephone-dn command in telephony-service mode. Dial peers configured in global
configuration mode comply with the global configuration huntstop setting.

When a call on an ephone-dn is answered, that ephone-dn is no longer accessible to other phones that
share the ephone-dn in overlay mode. For example, if extension 1001 is answered by phone 1, caller ID
for extension 1001 remains on phone 1 and is removed from the displays of phone 2 through phone 5.
All actions pertaining to the call to extension 1001 (ephone-dn 1) are visible from the phone 1 display
only. If phone 1 puts extension 1001 on hold, the other phones will not be able to directly pick up the

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Overlaid Ephone-dns

on-hold call using a simple shared-line pickup. In addition, none of the other four phones will be able to
make outgoing calls from the ephone-dn while it is in use. When they press the button 1, they will be
connected to the next available ephone-dn listed in the button command. For example, if phone 1 and
phone 2 are using ephone-dn 1 and ephone-dn 2, respectively, phone 3 would be designated ephone-dn
3 for an outgoing call.
If there are more phones than ephone-dns associated with an ephone-dn overlay set, it is possible for
some phones to find that all the ephone-dns within their overlay set are in use by other phones. For
example, if five phones have a line button configured with the button 1o1, 2, 3 command, there may be
times when all three of the ephone-dns in the overlay set are in use. When that occurs, the other two
phones will not be able to use an ephone-dn in the overlay set. When all ephone-dns in an overlay set
are in use, phones with this overlay set will display the remote-line-in-use icon (a picture of a phone with
a flashing X through it) for the corresponding line button. When at least one ephone-dn becomes
available within the overlay set (that is, an ephone-dn is either idle or ringing), the phone display reverts
to showing the status of the available ephone-dn (idle or ringing).
Dual-line ephone-dns can use overlays. The configuration parameters are the same as for single-line
ephone-dns, except that the huntstop channel command must be used to keep calls from hunting to the
ephone-dn’s second channel.

Restrictions
Ephone-dn overlays disable call waiting. If a phone is using an overlaid ephone-dn on an active call, call
waiting will be disabled for any incoming calls to any ephone-dn in the overlay set.

SUMMARY STEPS

1. ephone-dn phone-tag [dual-line]

2. number number
3. preference preference-value
4. no huntstop
5. huntstop channel
6. exit
7. ephone phone-tag
8. mac-address mac-address
9. button button-number{o}dn-tag, dn-tag... [[button-number{o}dn-tag, dn-tag...]...]
10. exit

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Overlaid Ephone-dns

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn phone-tag [dual-line] Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line
Example: • phone-tag—Unique sequence number that identifies
Router(config)# ephone-dn 10 dual-line this ephone-dn during configuration tasks. Range is
from 1 to the maximum number of ephone-dns allowed
on the router platform. Refer to CLI help for the
maximum value for this argument.
• dual-line—(Optional) Enables dual-line mode for the
ephone-dn.
Step 2 number number Associates a telephone or extension number with an
ephone-dn.
Example: • number—String of up to 16 characters that represents
Router(config-ephone-dn)# number 1001 an E.164 telephone number.
Step 3 preference preference-order Sets dial-peer preference order for an ephone-dn.
• preference-order—Preference order for the primary
Example: number associated with an extension (ephone-dn).
Router(config-ephone-dn)# preference 1 Refer to CLI help for a range of numeric options, where
0 is the highest preference. Default is 0.
Step 4 no huntstop Continues call hunting behavior for an ephone-dn.

Example:
Router(config-ephone-dn)# no huntstop
Step 5 huntstop channel (Recommended for dual-line ephone-dns) Keeps incoming
calls from hunting to the second channel if the first channel
is busy or does not answer.
Example:
Router(config-ephone-dn)# huntstop channel
Step 6 exit Exits ephone-dn configuration mode

Example:
Router(config-ephone-dn)# exit
Step 7 ephone phone-tag Enters ephone (Ethernet phone) configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone to which you are adding an overlay set.
Router(config)# ephone 4
Step 8 mac-address mac-address Specifies the MAC address of the registering phone.

Example:
Router(config-ephone)# mac-address
1234.5678.abcd

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Command or Action Purpose

Step 9 button button-number{o}dn-tag, dn-tag... Creates a set of ephone-dns overlaid on a single button.
[[button-number{o}dn-tag, dn-tag...]...]
• o—Overlay button. Multiple ephone-dns share this
button. A maximum of 25 ephone-dns can be specified
Example: for a single button, separated by commas.
Router(config-ephone)# button 1o14,15,16
• dn-tag—Unique identifier previously defined with the
ephone-dn command for the ephone-dn to be added to
this overlay set.
Note For other keywords, see the “Setting Up Individual
Directory Numbers and Phones” section in
the“Setting Up Phones in a Cisco CME System”
chapter.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

Example

The following example creates three lines (ephone-dns) that are shared across three IP phones to handle
three simultaneous calls to the same telephone number. Three instances of a shared line with the
extension number 1001 are overlaid onto a single button on each of three phones. A typical call flow is
as follows. The first call goes to ephone 1 (highest preference) and rings button 1 on all three phones
(huntstop is off). The call is answered on ephone 1. A second call to extension 1001 hunts onto
ephone-dn 2 and rings on the two remaining ephones, 11 and 12. The second call is answered by
ephone 12. A third simultaneous call to extension 1001 hunts onto ephone-dn 3 and rings on ephone 11,
where it is answered. Note that the no huntstop command is used to allow hunting for the first two
ephone-dns, and the huntstop command is used on the final ephone-dn to stop call-hunting behavior.
The preference command is used to create different selection preferences for each ephone-dn.
ephone-dn 1
number 1001
no huntstop
preference 0

ephone-dn 2
number 1001
no huntstop
preference 1

ephone-dn 3
number 1001
huntstop
preference 2

ephone 10
button 1o1,2,3

ephone 11
button 1o1,2,3

ephone 12
button 1o1,2,3

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Overlaid Ephone-dns

The following example shows how to overlay dual-line ephone-dns. In addition to using the huntstop
and preference commands, you must use the huntstop channel command to prevent calls from hunting
to the second channel of an ephone-dn. This example overlays five ephone-dns on button 1 on five
different ephones. This allows five separate calls to the same number to be connected simultaneously,
while occupying only one button on each phone.
ephone-dn 10 dual-line
number 1001
no huntstop
huntstop channel
preference 0

ephone-dn 11 dual-line
number 1001
no huntstop
huntstop channel
preference 1

ephone-dn 12 dual-line
number 1001
no huntstop
huntstop channel
preference 2

ephone-dn 13 dual-line
number 1001
preference 3
no huntstop
huntstop channel

ephone-dn 14 dual-line
number 1001
preference 4
huntstop
huntstop channel

ephone 33
mac 00e4.5377.2a33
button 1o10,11,12,13,14

ephone 34
mac 9c33.0033.4d34
button 1o10,11,12,13,14

ephone 35
mac 1100.8c11.3865
button 1o10,11,12,13,14

ephone 36
mac 0111.9c87.3586
button 1o10,11,12,13,14

ephone 37
mac 01a4.8222.3911
button 1o10,11,12,13,14

Troubleshooting Tips

The show dialplan number command displays all the number resolutions of a particular phone number,
which allows you to detect whether calls are going to unexpected destinations. This command is useful
for troubleshooting cases in which you dial a number but the expected phone does not ring.

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Call Waiting for Overlaid Ephone-dns

Call waiting allows phone users to know that another person is calling them while they are talking on
the phone. Phone users hear a call-waiting tone indicating that another party is trying to reach them.
Calls to IP phones with soft keys can be answered with the Answer soft key. Calls to analog phones
configured on Cisco CME systems are answered using hookflash. When phone users answer a
call-waiting call, their original call will be put on hold automatically. If phone users do not respond, the
caller can be forwarded to voice mail.
For Cisco CME 3.2.1 and later versions, call waiting is available for overlaid ephone-dns. Overlaid
ephone-dns are described in the “Overlaid Ephone-dns” section on page 262. The difference in
configuration between overlaid ephone-dns with call waiting and overlaid ephone-dns without call
waiting is that overlaid ephone-dns with call waiting use the c: keyword in the button command and
overlaid ephone-dn without call waiting use the o: keyword.
The behavior of overlaid ephone-dns with call waiting and overlaid ephone-dns without call waiting is
the same, except for the following:
• Calls to numbers included in overlaid ephone-dns with call waiting will cause inactive phones to
ring and active phones connected to other parties to generate auditory call-waiting notification. The
default sound is beeping, but you can configure an ephone-dn to use a ringing sound. (See the
“Call-Waiting Ring” section on page 200.) Visual call-waiting notification includes the blinking of
handset indicator lights and the display of caller IDs.
For example, if three of four phones are engaged in calls to numbers from the same overlaid
ephone-dn with call-waiting and another call comes in, the one inactive phone will ring, and the
three active phones will issue auditory and visual call-waiting notification.
• Two calls to numbers in an overlaid ephone-dn set can be announced. For the first call, the phone
user will hear a ring; for the second, call-waiting notification. Subsequent calls must wait in line,
remaining invisible until one of the two original calls has ended. The callers who are waiting in the
line will hear a ringback tone.
For example, a phone has a button configured with a set of overlaid ephone-dns with call waiting
(button 1c1,2,3,4). A call to ephone-dn 1 is answered. A call to ephone-dn 2 comes in and
call-waiting notification is issued. Calls to ephone-dn 3 and ephone-dn 4 will wait in line and remain
invisible to the phone user until one of the two original calls ends. When the call to ephone-dn 1
ends, the phone user will then talk to the person who called ephone-dn 2. The call to ephone-dn 3
will issue call-waiting notification while the call to ephone-dn 4 waits in line.
Note that if an overlaid ephone-dn has call forward no answer configured, calls to the ephone-dn
that are unanswered before the no answer timeout expires will be forwarded to voice mail. If call
forward no answer is not configured, incoming calls will receive ringback tones until the calls are
answered.
More than one phone may be using the same set overlaid ephone-dns; for example:
ephone 1
button 1c1,2,3,4
ephone 2
button 1c1,2,3,4

In that case, the behavior is slightly different. A call to ephone-dn 1 rings on ephone 1 and ephone
2. Ephone 1 answers, and the call is no longer visible to ephone 2. A call to ephone-dn 2 issues a
call-waiting notification to ephone 1 and rings on ephone 2, which answers. The second call is no
longer visible to ephone 1.

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A call to ephone-dn 3 issues a call-waiting notification to ephone 1 and ephone 2. Ephone 1 puts the
call to ephone-dn 1 on hold and answers the call to ephone-dn 3. The call to ephone-dn 3 is no longer
visible to ephone 2.
A call to ephone-dn 4 is issues a call-waiting notification on ephone 2. The call is not visible on
ephone 1 because it has met the two-call maximum by handling the calls to ephone-dn 1 and
ephone-dn 3.
• Phones configured for call waiting will not generate call-waiting notification when they are
transferring calls or hosting conference calls.

Note Ephone-dns accept call interruptions, such as call waiting, by default. For call waiting to work, the
default must be active. To ensure that this is the case, remove the no call-waiting beep accept command
from the configurations of ephone-dns for which you want to use call waiting. For more information,
refer to the Cisco CallManager Express 3.3 Command Reference.

SUMMARY STEPS

1. ephone-dn phone-tag [dual-line]

2. number number
3. preference preference-order
4. no huntstop
5. huntstop channel
6. exit
7. ephone phone-tag
8. mac-address mac-address
9. button button-number{c}dn-tag, dn-tag... [[button-number{c}dn-tag, dn-tag...]...]
10. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn phone-tag [dual-line] Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco IP phone line.
Example: • phone-tag—Unique sequence number that identifies an
Router(config)# ephone-dn 10 dual-line ephone-dn during configuration tasks. Range is from 1
to the maximum number of ephone-dns allowed on the
router platform. Refer to CLI help for the maximum
value for this argument.
• dual-line—(Optional) Enables dual-line mode for the
ephone-dn.
Step 2 number number Associates a telephone or extension number with an
ephone-dn.
Example: • number—String of up to 16 characters that represents
Router(config-ephone-dn)# number 1001 an extension or E.164 telephone number.

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Command or Action Purpose

Step 3 preference preference-order Sets dial-peer preference order for an ephone-dn.
• preference-order—Order of preference for the primary
Example: number associated with an extension (ephone-dn).
Router(config-ephone-dn)# preference 1 Refer to CLI help for a range of numeric options, where
0 is the highest preference. Default is 0.
Step 4 no huntstop Continues call-hunting behavior for an ephone-dn.

Example:
Router(config-ephone-dn)# no huntstop
Step 5 huntstop channel (Recommended for dual-line ephone-dns that are shared or
overlaid) Keeps incoming calls from hunting to the second
channel if the first channel is busy or does not answer.
Example:
Router(config-ephone-dn)# huntstop channel
Step 6 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 7 ephone phone-tag Enters ephone (Ethernet phone) configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the phone to which you are adding an overlay set.
Router(config)# ephone 4
Step 8 mac-address mac-address Specifies the MAC address of the registering phone.

Example:
Router(config-ephone)# mac-address
1234.5678.abcd
Step 9 button button-number{c}dn-tag, dn-tag... Creates a set of ephone-dns overlaid on a single button.
[[button-number{c}dn-tag,dn-tag...]...]
• c—Overlaid call-waiting button. Multiple ephone-dns
share this button. A maximum of ten ephone-dns can be
Example: specified for a single button, separated by commas.
Router(config-ephone)# button 1o10,11,12
• dn-tag—Unique identifier previously defined with the
ephone-dn command for the ephone-dn that is to be
added to this overlay set.
• For other keywords, see the button command in the
Cisco CallManager Express 3.3 Command Reference.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

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Ephone-dn Dial-Peer Preference

Example

In following example, button 1 on ephone 1 though ephone 3 uses the same set of overlaid ephone-dns
with call waiting that share the number 1111. The button also accept calls to each ephone’s unique
(nonshared) ephone-dn number. Note that if ephone-dn 10 and ephone-dn 11 are busy, the call will go to
ephone-dn 12. If ephone-dn 12 is busy, the call will go to voice mail.
ephone-dn 1 dual-line
number 1001

ephone-dn 2 dual-line
number 1001

ephone-dn 3 dual-line
number 1001

ephone-dn 10 dual-line
number 1111
no huntstop
huntstop channel
call-forward noans 7000 timeout 30

ephone-dn 11 dual-line
number 1111
preference 1
no huntstop
huntstop channel
call-forward noans 7000 timeout 30

ephone-dn 12 dual-line
number 1111
preference 2
huntstop channel
call-forward noans 7000 timeout 30
call-forward busy 7000

ephone 1
button 1c1,10,11,12

ephone 2
button 1c2,10,11,12

ephone 3
button 1c3,10,11,12

Ephone-dn Dial-Peer Preference

Dial peers in the Cisco CME router use certain call-matching criteria to route calls. One call-matching
criterion for incoming calls is the destination pattern, which associates a dialed number with an
ephone-dn. When more than one ephone-dn matches the destination pattern (which can include
wildcards), you can set the ephone-dn dial-peer preference for each of the matching ephone-dns to
designate the order in which incoming calls should be routed to the matching ephone-dns. Use the
preference (ephone-dn) command to assign each matching ephone-dn a preference value from 0 to 10,
with 0 as the highest preference. The default is 0 if no preference is assigned.
For example, you have two ephone-dns with the same extension number, 2680. These two ephone-dns
both match the same destination pattern because they both have the same extension number. One of the
ephone-dns is on button 1 of your phone, and the other is on button 2. The ephone-dn on button 1 has

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Huntstop

been assigned a preference value of 0 (the highest value and also the default if this command is not used),
and the ephone-dn on button 2 has been assigned a preference value of 1. The first call to extension 2680
is routed to button 1 because the ephone-dn on button 1 has the higher preference value. Incoming calls
will always be routed to button 1 if it is free. If the ephone-dn on button 1 is occupied when a second
call to extension 2680 is received by the Cisco CME system, the second call is routed to button 2.

SUMMARY STEPS

1. ephone-dn dn-tag
2. preference preference-order [secondary secondary-order]
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies the
Example: ephone-dn for which you are setting dial peer preference.
Router(config)# ephone-dn 33
Step 2 preference preference-order [secondary Sets the preference value for an ephone-dn with the associated
secondary-order] dial peer.
• preference-order—Preference value for the primary number
Example: of an ephone-dn. Refer to CLI help for a range of numeric
Router(config-ephone-dn)# preference 2 options, where 0 is the highest preference. Default is 0.
• secondary secondary-order—(Optional) Preference value
for the secondary number of an ephone-dn. Refer to CLI help
for a range of numeric options, where 0 is the highest
preference. Default is 0.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

The following example sets an ephone-dn preference number of 2 for the primary number of the
ephone-dn with dn-tag 3:
ephone-dn 3
preference 2

Huntstop
Huntstop prevents an incoming call from rolling over to another ephone-dn if the called ephone-dn is
busy or does not answer. This allows you to prevent hunt-on-busy from redirecting a call to a busy phone
into a dial-peer setup with a catch-all default destination.

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Huntstop

In ephone-dn configuration mode, huntstop is set by default. The no huntstop command disables
huntstop to allow hunting to a nonbusy ephone-dn.
Channel huntstop works in a similar way for the two channels of a dual-line ephone-dn. If it is enabled,
channel huntstop keeps incoming calls from hunting to the second channel if the first channel is busy or
does not answer. This keeps the second channel free for call transfer, call waiting, or three-way
conferencing. Channel huntstop also prevents situations in which a call can ring for 30 seconds on the
first channel of a line with no person available to answer and then ring for another 30 seconds on the
second channel before rolling over to another line.
No-huntstop call redirection is based on standard Cisco IOS voice gateway routing mechanisms.

SUMMARY STEPS

1. ephone-dn dn-tag
2. no huntstop
3. huntstop channel
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies
Example: the ephone-dn for which you want to change
Router(config)# ephone-dn 4 huntstop status.
Step 2 no huntstop (Optional) Disables huntstop, allowing calls to hunt
to the next ephone-dn if this ephone-dn is busy or
does not answer. Default is that huntstop is enabled.
Example:
Router(config-ephone-dn)# no huntstop
To reenable huntstop, use the huntstop command.
Step 3 huntstop channel (Optional) For dual-line ephone-dns, enables channel
huntstop, which keeps a call from hunting to the next
channel of an ephone-dn if the first channel is busy or
Example:
Router(config-ephone-dn)# huntstop channel
does not answer. Default is no huntstop channel.
Step 4 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Examples

The following example shows an instance in which huntstop is not desired and is explicitly disabled. In
this example, ephone 4 is configured with two lines, each with the same extension number 5001. This is
done to allow the second line to provide call waiting notification for extension number 5001 when the
first line is in use. Setting no huntstop on the first line (ephone-dn 1) allows incoming calls to hunt to
the second line (ephone-dn 2) on the same phone when the ephone-dn 1 line is busy.

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Ephone-dn 2 has call forwarding set to extension 6000, which corresponds to a locally attached
answering machine connected to a foreign exchange station (FXS) voice port. The plain old telephone
service (POTS) dial peer for extension 6000 also has the dial-peer huntstop attribute explicitly set to
prevent further hunting.
ephone-dn 1
number 5001
no huntstop
preference 1
call-forward noan 6000

ephone-dn 2
number 5001
preference 2
call-forward busy 6000
call-forward noan 6000

ephone 4
button 1:1 2:2
mac-address 0030.94c3.8724

dial-peer voice 6000 pots

destination-pattern 6000
huntstop port 1/0/0
description answering-machine

The following is an example that uses the huntstop channel command. It shows a dual-line ephone-dn
configuration in which calls do not hunt to the second channel of any ephone-dn, but they do hunt
through each ephone-dn’s channel 1 in this order: ephone-dn 10, ephone-dn 11, ephone-dn 12.
ephone-dn 10 dual-line
number 1001
no huntstop
huntstop channel

ephone-dn 11 dual-line
number 1001
no huntstop
huntstop channel
preference 1

ephone-dn 12 dual-line
number 1001
no huntstop
huntstop channel
preference 2

Ephone Hunt Groups

Ephone hunt groups provide the ability to direct incoming calls for a specific number (the ephone
hunt-group pilot number) to a defined group of ephone-dns. Each ephone hunt group can consist of up
to 20 member ephone-dns.
Incoming calls are redirected from a hunt group pilot number to the first ephone-dn as defined by the
configuration. If the first ephone-dn is busy or does not answer, the call is redirected to the next phone
in the list. A call continues to be redirected on busy or no answer from ephone-dn to ephone-dn in the
list until it is answered or until the call reaches the number that was defined as the final number.

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The redirect from one ephone-dn to the next in the list is also known as a hop. The maximum number of
redirects can be configured globally using the max-redirect command and can be configured for specific
peer or longest-idle hunt groups using the hops command. If the maximum number of redirects or hops
is reached, the call is dropped.
There are three different kinds of ephone hunt groups. Each type of group uses a different strategy to
determine the first ephone-dn that will ring for successive calls to the pilot number. Hunt group types
include the following:
• Sequential ephone hunt groups—Ephone-dns always ring in the left-to-right order in which they are
listed when the hunt group is defined. The first number in the list is always the first number to be
tried when the pilot number is called. Maximum number of hops is not a configurable parameter for
sequential ephone hunt groups.
• Peer ephone hunt groups—The first ephone-dn to ring is the number to the right of the ephone-dn that
was the last to ring when the pilot number was last called. Ringing proceeds in a circular manner, left
to right, for the number of hops specified when the ephone hunt group was defined.
• Longest-idle ephone hunt group—Calls go first to the ephone-dn that has been idle the longest for the
number of hops specified when the ephone hunt group was defined. The longest-idle is determined
from the last time that a phone registered, reregistered, or went on-hook.
The number that is defined as the final number for a hunt group may also be the pilot number for another
hunt group (with suitable protection to avoid infinite loops). If a final number is assigned as the pilot
number of a second hunt group, the pilot number of the first hunt group cannot be configured as a final
number in any hunt group. If there is a third hunt group, the second hunt group cannot be configured as
a final number, and so forth.
Hunt-group chains can be configured in any length, but the actual number of hops that can be reached in
the chain is determined by the max-redirect command configuration. In the following example, a
maximum redirect number 15 or greater must be configured for callers to reach the final 5000 number.
If a lower number is configured, the call will disconnect.
ephone-hunt 1 sequential
pilot 8000
list 8001, 8002, 8003, 8004
final 9000
ephone-hunt 2 sequential
pilot 9000
list 9001, 9002, 9003, 9004
final 7000
ephone-hunt 3 sequential
pilot 7000
list 7001, 7002, 7003, 7004
final 5000

Figure 34 on page 275 illustrates a sequential ephone hunt group, Figure 35 on page 276 illustrates a
peer ephone hunt group, and Figure 36 on page 277 illustrates a longest-idle hunt group.
The show ephone-hunt command is used to display information about ephone-hunt behavior, such as
the amount of time that a hunt list member of a hunt group has been idle.
To create ephone hunt groups, complete the tasks in the following sections:
• Configuring Sequential Ephone Hunt Groups, page 277
• Configuring Peer Ephone Hunt Groups, page 280
• Configuring Longest-Idle Ephone Hunt Groups, page 282

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Ephone Hunt Groups

Figure 34 Sequential Ephone Hunt Group

ephone-dn 88
1 Any phone dials the pilot number, 5601.
number 5001

2 Extension 5001, the leftmost number in the hunt group list, rings first ephone-dn 89
on phone 1. If extension 5001 is busy or does not answer, the call is number 5002
redirected to extension 5002 on phone 2.
3 If extension 5002 on phone 2 is busy or does not answer, the call is ephone-dn 90
redirected to extension 5017 on phone 3. number 5017
4 If phone 3 is busy or does not answer, the call is redirected to the final
number, extension 6000, which is associated with a voice-mail server. ephone 1
mac-address 1111.1111.1111
button 1:88
Any phone dials the pilot number.
IP ephone 2
mac-address 2222.2222.2222
6000 Voice-mail server button 1:89
5601
Pilot number
ephone 3
mac-address 3333.3333.3333
V button 1:90

Phone 1 ephone-hunt 1 sequential

Button 1 is extension 5001
IP
pilot 5601
list 5001, 5002, 5017
Phone 2 final 6000
Button 1 is extension 5002
IP preference 1
timeout 30

88955
Phone 3
Button 1 is extension 5017
IP

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Figure 35 Peer Ephone Hunt Group

1 Any phone dials the pilot number, 5601, which is not associated with a
physical phone instrument.
ephone-dn 88
2 Extension 5017 on phone 3 is selected to ring first because extension
5002 was the last number to ring the last time that the pilot number number 5001
was called.
ephone-dn 89
3 If extension 5017 is busy or does not answer, the call is redirected to number 5002
extension 5044 on phone 4 (first hop).

4 If extension 5044 is busy or does not answer, the call is redirected to ephone-dn 90
extension 5001 on phone 1 (second hop). number 5017

5 If extension 5001 is busy or does not answer, the call has reached the ephone-dn 91
maximum number of hops (3), and it is redirected to the final number, number 5044
extension 6000, which is associated with a voice-mail server.
ephone 1
Any phone dials the pilot number. mac-address 1111.1111.1111
IP button 1:88
Voice-mail server
Pilot number ephone 2
6000
5601 mac-address 2222.2222.2222
button 1:89

V ephone 3
mac-address 3333.3333.3333
Phone 1 button 1:90
Button 1 is extension 5001 IP
ephone 4
mac-address 4444.4444.4444
Phone 2
Button 1 is extension 5002 IP button 1:91

ephone-hunt 1 peer
Phone 3 pilot 5601
Button 1 is extension 5017
IP
list 5001, 5002, 5017, 5044
final 6000
Phone 4 hops 3
Button 1 is extension 5044 IP preference 1

88956
timeout 30
no-reg

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Figure 36 Longest-Idle Ephone Hunt Group

1 Any phone dials the pilot number, 5601, which is not associated with a
physical phone instrument. ephone-dn 88
number 5001
2 Extension 5001 on phone 1 is selected to ring first because it has
been idle the longest. ephone-dn 89
3 If extension 5001 does not answer, the call is redirected to extension number 5002
5002 on phone 2 because it has been idle the longest (first hop).
ephone-dn 90
4 If extension 5002 does not answer, the call is redirected to extension
number 5017
5044 on phone 4 because it has been idle the longest (second hop).
5 If extension 5044 does not answer, the call has reached the maximum ephone-dn 91
number of hops (3), and it is redirected to the final number, extension 6000, number 5044
which is associated with a voice-mail server
ephone 1
Any phone dials the pilot number. mac-address 1111.1111.1111
IP button 1:88
Voice-mail server
Pilot number ephone 2
6000
5601 mac-address 2222.2222.2222
button 1:89

V ephone 3
mac-address 3333.3333.3333
Phone 1 button 1:90
Button 1 is extension 5001 IP
ephone 4
mac-address 4444.4444.4444
Phone 2
Button 1 is extension 5002 IP button 1:91

ephone-hunt 1 longest-idle
Phone 3 pilot 5601
Button 1 is extension 5017
IP
list 5001, 5002, 5017, 5044
final 6000
Phone 4 hops 3
Button 1 is extension 5044 IP preference 1

103299
timeout 30
no-reg

Configuring Sequential Ephone Hunt Groups

In a sequential ephone hunt group, ephone-dns ring in the left-to-right order in which they were listed
when the hunt group was defined. The first number to ring is always the leftmost number in the list.

SUMMARY STEPS

1. ephone-hunt hunt-tag sequential

2. pilot number [secondary number]
3. list dn-number, dn-number[, dn-number...]
4. final final-number
5. preference preference-order [secondary secondary-order]
6. timeout seconds
7. no-reg [both | pilot]
8. exit

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9. telephony-service
10. max-redirect number
11. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-hunt hunt-tag sequential Enters ephone-hunt configuration mode to define a
sequential ephone hunt group.
Example: • hunt-tag—Unique sequence number that identifies this
Router(config)# ephone-hunt 5 sequential hunt group during all configuration tasks. Range is
from 1 to 20.
Step 2 pilot number [secondary number] Defines the pilot number, which is the number that callers
dial to reach the hunt group.
Example: • number—E.164 number with a maximum length of 27
Router(config-ephone-hunt)# pilot 5601 characters. The dialplan pattern can be applied to the
pilot number.
• secondary—(Optional) Defines the number that
follows as an additional pilot number for the ephone
hunt group.
Step 3 list dn-number, dn-number[, dn-number...] Defines the list of numbers to which the ephone hunt group
redirects the incoming calls. There must be between two
and twenty numbers in the list.
Example:
Router(config-ephone-hunt)# list 5001, 5002, • dn-number—An ephone-dn primary or secondary
5017, 5028 number.
Step 4 final final-number Defines the last number in the ephone hunt group, after
which the call is no longer redirected. This number can be
an ephone-dn primary or secondary number, a voice-mail
Example:
Router(config-ephone-hunt)# final 6000
pilot number, a pilot number of another hunt group, or an
FXS number.
Note Once a final number is defined as a pilot number of
another hunt group, the pilot number of the first
hunt group cannot be configured as a final number
in any other hunt group.
Step 5 preference preference-order [secondary Sets a preference order for the ephone-dn associated with
secondary-order] the hunt-group pilot number.
• preference-order—Refer to CLI help for a range of
Example: numeric options, where 0 is the highest preference.
Router(config-ephone-hunt)# preference 1 Default is 0.
• secondary secondary-order—(Optional) Preference
order for the secondary pilot number. Refer to CLI help
for a range of numeric options, where 0 is the highest
preference. Default is 9.

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Command or Action Purpose

Step 6 timeout seconds Sets the number of seconds after which a call that is not
answered at one number is redirected to the next number in
the hunt-group list.
Example:
Router(config-ephone-hunt)# timeout 30 • seconds—Number of seconds. Range is from 3 to
60000. Default is 180.
Step 7 no-reg [both | pilot] Prevents the hunt-group pilot number from registering with
an H.323 gatekeeper. The default if this command is not
used is that the pilot number registers with the H.323
Example:
Router(config-ephone-hunt)# no-reg
gatekeeper. In Cisco CME 3.1 and later versions, if this
command is used but neither the both nor the pilot keyword
is used, only the secondary number is not registered.
• both—(Optional) Both the primary and secondary pilot
numbers are not registered.
• pilot—(Optional) Only the primary pilot number is not
registered.
Step 8 exit Exits ephone-hunt configuration mode.

Example:
Router(config-ephone-hunt)# exit
Step 9 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 10 max-redirect number Sets the number of times that a call can be redirected within
a Cisco CME system.
Example: • number—Range is from 5 to 20. Default is 5.
Router(config-telephony)# max-redirect 8
Step 11 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example defines a sequential ephone hunt group with the pilot number 5601 and the final
number 6000, with four numbers in the list of phones that answer for the pilot number.
ephone-hunt 1 sequential
pilot 5601
list 5001, 5002, 5017, 5028
final 6000
preference 1
timeout 30

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Configuring Peer Ephone Hunt Groups

In a peer ephone hunt group, the first ephone-dn to ring is the number to the right of the ephone-dn that was
the last to ring when the pilot number was last called. Ringing proceeds in a circular manner, left to right,
for the number of hops specified when the ephone hunt group was defined.

SUMMARY STEPS

1. ephone-hunt hunt-tag peer

2. pilot number [secondary number]
3. list dn-number, dn-number[, dn-number...]
4. final final-number
5. preference preference-order [secondary secondary-order]
6. hops number
7. timeout seconds
8. no-reg [both | pilot]
9. exit
10. telephony-service
11. max-redirect number
12. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-hunt hunt-tag peer Enters ephone-hunt configuration mode to define a peer
ephone hunt group.
Example: • hunt-tag—Unique sequence number that identifies this
Router(config)# ephone-hunt 5 peer hunt group during all configuration tasks. Range is
from 1 to 20.
Step 2 pilot number [secondary number] Defines the pilot number, which is the number that callers
dial to reach the hunt group.
Example: • number—E.164 number with a maximum length of 27
Router(config-ephone-hunt)# pilot 5601 characters. The dial-plan pattern can be applied to the
secondary 2223 pilot number.
• secondary—(Optional) Defines the number that
follows as an additional pilot number for the ephone
hunt group.
Step 3 list dn-number, dn-number[, dn-number...] Defines the list of numbers to which the ephone hunt group
redirects the incoming calls that are made to the pilot
number. There must be from two to ten numbers in the list.
Example:
Router(config-ephone-hunt)# list 5001, 5002, • dn-number—An ephone-dn primary or secondary
5017, 5028 number.

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Command or Action Purpose

Step 4 final final-number Defines the last number in the ephone hunt group, after
which the call is no longer redirected. This number can be
an ephone-dn primary or secondary number, a voice-mail
Example:
Router(config-ephone-hunt)# final 6000
pilot number, a pilot number of another hunt group, or an
FXS caller ID number.
Note Once a final number is defined as a pilot number of
a second hunt group, the pilot number of the first
hunt group cannot be configured as a final number
in any other hunt group.
Step 5 preference preference-order [secondary Sets a preference order for the ephone-dn associated with
secondary-order] the hunt-group pilot number.
• preference-order—Refer to CLI help for a range of
Example: numeric options, where 0 is the highest preference.
Router(config-ephone-hunt)# preference 1 Default is 0.
• secondary secondary-order—(Optional) Preference
order for the secondary pilot number. Refer to CLI help
for a range of numeric options, where 0 is the highest
preference. Default is 9.
Step 6 hops number Sets the number of hops before a call proceeds to the final
number.
Example: • number—Number of hops. Range is from 2 to 20, but
Router(config-ephone-hunt)# hops 7 must be less than or equal to the number of extensions
that are specified in the list command. Default is 2.
Step 7 timeout seconds Sets the number of seconds after which a call that is not
answered at one number is redirected to the next number in
the hunt-group list.
Example:
Router(config-ephone-hunt)# timeout 30 • seconds—Number of seconds. Range is from 3 to
60000. Default is 180.
Step 8 no-reg [both | pilot] Prevents the hunt-group pilot number from registering with
an H.323 gatekeeper. The default if this command is not
used is that the pilot number registers with the H.323
Example:
Router(config-ephone-hunt)# no-reg
gatekeeper. In Cisco CME 3.1 and later releases, if this
command is used but neither the both nor the pilot keyword
is used, only the secondary number is not registered.
• both—(Optional) Both the primary and secondary pilot
numbers are not registered.
• pilot—(Optional) Only the primary pilot number is not
registered.
Step 9 exit Exits ephone-hunt configuration mode.

Example:
Router(config-ephone-hunt)# exit

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Command or Action Purpose

Step 10 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 11 max-redirect number Sets the number of times that a call can be redirected within
a Cisco CME system.
Example: • number—Range is from 5 to 20. Default is 5.
Router(config-telephony)# max-redirect 8
Step 12 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example defines peer ephone hunt group 1 with a pilot number 5601, a final number 6000,
and eight numbers in the list. After a call is redirected four times (makes four hops), it is redirected to
the final number.
ephone-hunt 1 peer
pilot 5601
list 5001, 5002, 5017, 5028, 5066, 5067, 5077, 5085
final 6000
hops 4
preference 1
timeout 30
no-reg

Configuring Longest-Idle Ephone Hunt Groups

In a longest-idle hunt group, the first ephone-dn to ring is the number that has been idle the longest
period of time.

SUMMARY STEPS

1. ephone-hunt hunt-tag longest idle

2. pilot pilot-number
3. list dn-number, dn-number[, dn-number...]
4. final final-number
5. preference preference-order
6. hops number
7. timeout seconds
8. no-reg
9. exit
10. telephony-service
11. max-redirect number

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12. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-hunt hunt-tag longest-idle Enters ephone-hunt configuration mode to define a
longest-idle ephone hunt group.
Example: • hunt-tag—Unique sequence number that identifies this
Router(config)# ephone-hunt 5 longest-idle hunt group during all configuration tasks. Range is
from 1 to 20.
Step 2 pilot pilot-number Defines the pilot number, which is the number that callers
dial to reach the hunt group.
Example: • pilot-number—E.164 number with a maximum length
Router(config-ephone-hunt)# pilot 5601 of 27 characters. The dial-plan pattern can be applied to
the pilot number.
Step 3 list dn-number, dn-number[, dn-number...] Defines the list of numbers to which the ephone hunt group
redirects the incoming calls that are made to the pilot
number.
Example:
Router(config-ephone-hunt)# list 5001, 5002, • Enter at least two and no more than ten numbers.
5017, 5028
• dn-number—An ephone-dn primary or secondary
number.
Step 4 final final-number Defines the last number in the ephone hunt group, after
which the call is no longer redirected.
Example: • This number can be an ephone-dn primary or secondary
Router(config-ephone-hunt)# final 6000 number, a voice-mail pilot number, a pilot number of
another hunt group, or a FXS caller ID number.
Note Once a final number is defined as a pilot number of
a second hunt group, the pilot number of the first
hunt group cannot be configured as a final number
in any other hunt group.
Step 5 preference preference-order Sets a preference order for the ephone-dn associated with
the hunt-group pilot number.
Example: • preference-order—Refer to CLI help for the range of
Router(config-ephone-hunt)# preference 1 numeric options, where 0 is the highest preference.
Step 6 hops number Sets the number of hops before a call proceeds to the final
number.
Example: • Range is from 2 to 20.
Router(config-ephone-hunt)# hops 7
Step 7 timeout seconds Sets the number of seconds after which a call that is not
answered at one number is redirected to the next number in
the hunt-group list.
Example:
Router(config-ephone-hunt)# timeout 30 • seconds—Number of seconds. Range is from 3 to
60000. Default is 180.

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Automatic Hunt Group Logout

Command or Action Purpose

Step 8 no-reg Prevents the hunt-group pilot number from registering with
an H.323 gatekeeper.
Example:
Router(config-ephone-hunt)# no-reg
Step 9 exit Exits ephone-hunt configuration mode.

Example:
Router(config-ephone-hunt)# exit
Step 10 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 11 max-redirect number Sets the number of times that a call can be redirected within
a Cisco CME system.
Example: • number—Range is from 5 to 20. Default is 5.
Router(config-telephony)# max-redirect 8
Step 12 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example defines longest-idle ephone hunt group 1 with a pilot number 7501, a final
number 8000, and 11 numbers in the list. After a call is redirected six times (makes six hops), it is
redirected to the final number, 8000.
ephone-hunt 1 longest-idle
pilot 7501
list 7001, 7002, 7023, 7028, 7045, 7062, 7067, 7072, 7079, 7085, 7099
final 8000
preference 1
hops 6
timeout 20
no-reg

Automatic Hunt Group Logout

For Cisco CME 3.2.1 and later versions, an ephone-dn of an ephone hunt group (see the “Ephone Hunt
Groups” section on page 273) can be logged out automatically when a call to the ephone-dn is
unanswered. A call is considered unanswered if it rings longer than the period of time configured in the
timeout command in ephone-hunt configuration mode. After an ephone-dn has been logged out, the
phone that it is assigned to will display the do not disturb (DND) indicator.
The automatic logout cabability is enabled with the auto logout command. You can enter the auto
logout command to any number of ephone hunt groups, but the ephone-dns must belong to one ephone.
Ephones can not share multiple ephone-dns.

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Automatic Hunt Group Logout

Restrictions
• Hunt-group logout is available on the following Cisco IP phones only:
– Cisco IP Phone 7905G
– Cisco IP Phone 7912G
– Cisco IP Phone 7940/7940G
– Cisco IP Phone 7960/7960G
• Auto logout does not support ephones with shared lines.

SUMMARY STEPS

1. ephone-hunt hunt-tag
2. auto logout
3. exit

Command or Action Purpose

Step 1 ephone-hunt hunt-tag Enters ephone-hunt configuration mode.
• hunt-tag—Unique sequence number that identifies a
Example: hunt group during all configuration tasks. Range is
Router(config)# ephone 10 from 1 to 20.
Step 2 auto logout Allows phone users to log out of the ephone hunt groups for
which their phones are configured.
Example:
Router(config-ephone-hunt)# auto logout
Step 3 exit Exits ephone-hunt configuration mode.

Example:
Router(config-ephone)# exit

Example

The following simplified ephone hunt group configuration allows auto logout for ephone 1 and ephone
2. If the phone users for ephone 1 and ephone 2 were away from their phones and each received a call
that rang for more than 60 seconds, their phones would automatically log out and display the DND
indicator. Upon returning to their desks, they could log onto their phones by pressing DND. In addition,
they would know that someone had called them and might have left voice mail.
ephone-dn 1
number 1002

ephone-dn 2
number 2001

ephone-hunt 1 peer
pilot 1111
list 1001, 1002
timeout 60
auto logout

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Call-Pickup Groups

ephone 1
button 1o1

ephone 2
button 1o2

Note that if the ephone configurations were as follows, auto logout would not work because shared lines
are used:
ephone 1
button 1o1,2

ephone 2
button 1o1,2

Call-Pickup Groups
Cisco CME allows administrators to associate pickup groups with individual ephone-dn entries, making
it easier for phone users to answer, or pick up, a call that is ringing on a different ephone-dn. If both
ephone-dns are in the same pickup group, the user presses fewer keys to pick up the call.
Call pickup has the following variations:
• Call pickup, explicit ringing extension—The phone user presses the PickUp soft key and then dials
the ephone-dn of the ringing telephone. Note that this method can also be used to pick up a call that
is on hold on another ephone-dn.
• Call pickup, explicit group ringing extension—The phone user presses the GPickUp soft key and
then dials the group number of the ringing telephone. If there is only one pickup group defined in
the entire Cisco CME system, the user needs only to press the GPickUp soft key.
• Call pickup, local group ringing extension—If the ringing telephone and the user’s phone are in the
same pickup group, the phone user presses the GPickUp soft key and presses the asterisk (*) key to
pick up the call on the ringing telephone.
Administrators can assign each ephone-dn independently to a maximum of one pickup group. A
ephone-dn that does not belong to any pickup group can still pick up a ringing call by dialing the
ephone-dn on which the call is ringing or the pickup group number of that ephone-dn.
Pickup group numbers may be of varying length, but must have unique leading digits. For example, you
cannot define pickup group 17 and pickup group 177 for the same Cisco CME system because a pickup
in group 17 will always be triggered before the user can enter the final 7 for 177.
There is no limit to the number of ephone-dns that can be assigned to a single pickup group, and there
is no limit to the number of pickup groups that can be defined in a Cisco CME system.
Figure 37 shows four call-pickup scenarios.

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Call-Pickup Groups

Figure 37 Call Pickup

Call Pickup, No Group or Unknown Group ephone-dn 55

number 5555
1 Extension 5555 rings. 2 User at phone 4 presses PickUp pickup-group 33
soft key and dials 5555.
ephone-dn 56
Phone 1 Phone 2 number 5556
Extension 5555 Extension 5556 pickup-group 33
IP IP
Pickup group 33 Pickup group 33
Phone 3 Phone 4 ephone-dn 57
Extension 5557 Extension 5558 number 5557
IP IP
Pickup group 44 No pickup group pickup-group 44

ephone-dn 58
Call Pickup in the Same Group number 5558
.
1 Extension 5555 rings. 2 User at phone 2 presses GPickUp .
soft key and * (asterisk). .
ephone 1
Phone 1 Phone 2 mac-address 1111.1111.1111
Extension 5555 Extension 5556 button 1:55
IP IP
Pickup group 33 Pickup group 33
Phone 3 Phone 4 ephone 2
Extension 5557 Extension 5558 mac-address 2222.2222.2222
IP IP
Pickup group 44 No pickup group button 1:56

ephone 3
mac-address 3333.3333.3333
button 1:57
Call Pickup from a Different Group
ephone 4
mac-address 4444.4444.4444
1 Extension 5555 rings. 2 User at phone 3 presses
button 1:58
GPickUp soft key and dials 33.
.
.
Phone 1 Phone 2
.
Extension 5555 Extension 5556
IP IP
Pickup group 33 Pickup group 33
Phone 3 Phone 4
IP Extension 5557 IP Extension 5558
Pickup group 44 No pickup group

Call Pickup, a Single Group for All Cisco CME Phones

1 Extension 5555 rings. 2 User at phone 2 presses

GPickUp soft key.

Phone 1 Phone 2
Extension 5555 Extension 5556
IP Pickup group 33 IP Pickup group 33
88954

This scenario assumes that every phone in the Cisco CME system is in pickup group
33, which differs slightly from the sample configuration shown to the right.

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SUMMARY STEPS

1. ephone-dn dn-tag
2. pickup-group number
3. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique identifier for the ephone-dn
Example: (extension) that you want to add to a pickup group.
Router(config)# ephone-dn 55
Step 2 pickup-group number Assigns this ephone-dn to a pickup group.
• number—Digit string of up to 32 characters. Group
Example: numbers in a Cisco CME system may be of varying
Router(config-ephone-dn)# pickup-group 2345 length, but they must have unique leading digits. For
example, if there is a group numbered 17, there cannot
also be a group numbered 177.
Step 3 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

The following example assigns the line that has an ephone-dn tag of 55 to pickup group 2345:
ephone-dn 55
number 2555
pickup-group 2345

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 Configuring Directories

This chapter describes features related to directory service in a Cisco CallManager Express
(Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Local Directory Order, page 289
• Additional Directory Entries, page 290
• Local Directory Disable, page 291

Local Directory Order

This task configures the naming order in the local directory—that is, whether the names appear with first
names first or last names first.

SUMMARY STEPS

1. telephony-service
2. directory {first-name-first | last-name-first}
3. exit

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Additional Directory Entries

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 directory {first-name-first | Defines the local directory naming order.
last-name-first}
Note The actual directory of names and phone numbers is
built using the name command and the number
Example: command in ephone-dn configuration mode.
Router(config-telephony)# directory
last-name-first When the command is set with the first-name-first keyword,
you see the directory information displayed on the phone, for
example, Jane E. Smith; and when the command is set with
the last-name-first keyword, you see the directory
information displayed on the phone as, for example,
Smith, Jane E.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example defines the naming order for the local directory on IP phones served by the
Cisco CME router:
telephony-service
directory last-name-first

Additional Directory Entries

The Cisco CME system automatically creates a local phone directory according to the telephone
numbers that are assigned during ephone-dn configuration. Additional entries to the local Cisco CME
directory can be made using the directory entry command. The additional numbers can be nonlocal
numbers, such as telephone numbers of other Cisco CME systems used by the same corporation.
Use the show telephony-service directory-entry command to display the entries made using the
directory entry command.

SUMMARY STEPS

1. telephony-service
2. directory entry {entry-tag number name name | clear}
3. exit

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Local Directory Disable

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 directory entry {entry-tag number name name | Creates a telephone directory entry that is displayed on an
clear} IP phone. Entries appear in the order in which they are
entered.
Example: • entry-tag—Unique sequence number that identifies this
Router(config-telephony)# directory entry 1 directory entry during all configuration tasks. Range is
5550111 name Sales
from 1 to 100.
• number—Telephone number or extension for the entry,
up to 32 characters.
• name name—Name of up to 24 characters that will
appear in the directory.
• clear—Removes all directory entries.
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example creates a directory of three telephone listings:

telephony-service
directory entry 1 14045550111 name Sales
directory entry 2 13125550122 name Marketing
directory entry 3 12135550144 name Support

Local Directory Disable

The local directory that is displayed on an IP phone (item 4 in the Local Services menu) is served as an
eXtensible Markup Language (XML) page that is accessed through HTTP without password protection.
The no service local-directory command disables the directory HTTP service to suppress the
availability of this directory.

SUMMARY STEPS

1. telephony-service
2. no service local-directory
3. exit

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DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 no service local-directory Disables local directory service on IP phones.

Example:
Router(config-telephony)# no service
local-directory
Step 3 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example disables the local directory on IP phones served by the Cisco CME router:
telephony-service
no service local-directory

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 Configuring Productivity Tools

This chapter describes several features that can help phone users be more productive in a
Cisco CallManager Express (Cisco CME) system.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Flash Soft Key for Hookflash Functionality, page 293
• Intercom, page 294
• Paging, page 298
• Account Code Entry by User, page 306
• Applications Integration with Cisco CME, page 306
• XML API, page 313
• Related Features, page 315

Flash Soft Key for Hookflash Functionality

Certain public switched telephone network (PSTN) services, such as three-way calling and call waiting,
require hookflash intervention from a phone user. A soft key, labeled Flash, provides this functionality
for Cisco IP Phones 7940 and 7940G and Cisco IP Phones 7960 and 7960G using foreign exchange
office (FXO) lines attached to the Cisco CME system. The Flash soft key is enabled using the fxo
hook-flash command.
Once a Flash soft key has been enabled on an IP phone, it is available to provide hookflash functionality
during all calls except for local IP-phone-to-IP-phone calls. Note that hookflash-controlled services can
be activated only if they are supported by the PSTN connection that is involved in the call and that the
availability of the Flash soft key does not guarantee that hookflash-based services are actually accessible
to the phone user.

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Intercom

SUMMARY STEPS

1. telephony-service
2. fxo hook-flash
3. restart all
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 fxo hook-flash Enables the Flash soft key on the Cisco IP Phones 7940 and
7940G or the Cisco IP Phones 7960 and 7960G that receive
calls from Dynamic Host Configuration Protocol (PSTN)
Example:
Router(config-telephony)# fxo hook-flash
phones through an FXO voice port.
The Flash soft key display is automatically disabled for
local IP-phone-to-IP-phone calls.
Step 3 restart all Performs a fast reboot of all phones associated with this
Cisco CME router. Does not contact the DHCP or TFTP
server for updated information.
Example:
Router(config-telephony)# restart all
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example enables the Flash soft key for the Cisco IP Phones 7940 and 7940G or the Cisco
IP Phones 7960 and 7960G that receive calls from PSTN phones through an FXO voice port.
telephony-service
fxo hook-flash

Intercom
Cisco CME supports intercom functionality for one-way and press-to-answer voice connections using a
dedicated pair of intercom ephone-dns on two phones that speed-dial each other.
When an intercom speed-dial button is pressed, a call is speed-dialed to the ephone-dn that is the other
half of the dedicated pair. The called ephone-dn automatically answers the call in speakerphone mode
with mute activated, which provides a one-way voice path from the initiator to the recipient. A beep is

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sounded when the call is auto-answered to alert the recipient to the incoming call. To respond to the
intercom call and open a two-way voice path, the recipient deactivates the mute function by taking one
of the following actions:
• On a multibutton phone, pressing the Mute button.
• On a Cisco IP Phone 7910, lifting the handset.
In Cisco 3.2.2 and later releases, the no-mute keyword can be used with the intercom command to
deactivate the speaker-mute function on inercom calls. For example, if phone user 1 makes an intercom
call to phone user 2, both users hear each other upon connection when no-mute has been configured. The
benefit is that people who receive intercom calls can be heard without having to disable the mute
function. The disadvantage is that people who receive intercom calls will have their conversations and
nearby background sounds heard the moment an intercom call to them is connected, regardless of
whether they are ready to take a call or not.
Intercom lines cannot be used in shared-line configurations. If an ephone-dn is configured for intercom
operation, it must be associated with one IP phone only. The intercom attribute causes an IP phone line
(ephone-dn) to operate as an autodial line for outbound calls and as an autoanswer-with-mute line for
inbound calls. Figure 38 shows an intercom between a receptionist and a manager.
To prevent an unauthorized phone from dialing an intercom line (and creating a situation in which a
phone automatically answers a non-intercom call), you can assign the intercom ephone-dn a dialing
string with an alphabetic character. No one can dial the alphabetic character from a normal phone, but
the phone at the other end of the intercom can be configured to dial the number that contains the
alphabetic character through the Cisco CME router. For example, the intercom ephone-dns in Figure 38
have been assigned numbers with alphabetic characters so that no one but the receptionist can call the
manager on his or her intercom line, and no one but the manager can call the receptionist on his or her
intercom line.

Note An intercom requires configuration of two ephone-dns, one each on a separate phone.

Figure 38 Intercom

ephone-dn 2
1 The receptionist at phone 6 2 Phone 7 beeps once and automatically number 2345
makes an intercom call to answers in speakerphone mode with
phone 7 by pressing button 2. mute activated. The manager hears the ephone-dn 3
receptionist’s voice and deactivates the number 4578
mute function to open a two-way voice
path for a reply. ephone-dn 18
number A5001
name "Intercom"
intercom A5002
IP IP
V ephone-dn 19
number A5002
Phone 6 - Receptionist Phone 7 - Manager name "Intercom"
Button 1 is extension 2345, a Button 1 is extension 4578, a intercom A5001
normal line. normal line.
Button 2 is extension A5001, a Button 2 is extension A5002, a ephone 6
dedicated intercom connection dedicated intercom connection to button 1:2 2:18
to intercom extension intercom extension
88952

A5002 on phone 7. A5001 on phone 6. ephone 7

button 1:3 2:19

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SUMMARY STEPS

1. ephone-dn dn-tag
2. number number
3. name name
4. intercom directory-number [barge-in | no-auto-answer] [label label] [no-mute]
5. exit
6. Repeat Steps 1 through 5 for the second ephone-dn.
7. ephone phone-tag
8. button button-number:dn-tag [[button-number:dn-tag] ...]
9. restart
10. exit
11. Repeat Steps 7 through 10 for the second phone.

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies
Example: this intercom ephone-dn during configuration
Router(config)# ephone-dn 11 tasks. Range is from 1 to 288.
Step 2 number number Assigns a valid intercom number.
• number—String that contains up to 16 numeric
Example: characters. If this ephone-dn is an intercom
Router(config-ephone-dn)# number A2345 ephone-dn, alphabetic characters can be included
for security. One or more alphabetic characters in
an intercom number ensures that the number can
only be dialed from the one other intercom
number that is programmed to dial this number.
The number cannot be dialed from a normal
phone if it contains an alphabetic character.
Step 3 name name Sets a name to be associated with the ephone-dn. This
name is used for caller-ID displays and also shows up
in the local directory associated with the ephone-dn.
Example:
Router(config-ephone-dn)# name intercom

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Command or Action Purpose

Step 4 intercom directory-number [barge-in | Defines the ephone-dn that is speed-dialed for the
no-auto-answer] [label label] [no-mute] intercom feature when this line is used.
• directory-number—Number to speed-dial for
Example: this intercom function.
Router(config-ephone-dn)# intercom A2346 label
Security • barge-in—(Optional) Specifies that an intercom
call on this ephone-dn will force an existing call
on the associated ephone into a call-hold state
and allow the intercom call to be immediately
answered.
• no-auto-answer—(Optional) Creates a
connection for the IP phone line resembling
private line automatic ringdown (PLAR).
• label label—(Optional) Defines a text label of up
to 24 characters for the intercom ephone-dn. This
label is used for caller-ID displays and directory
lists.
• no-mute—(Optional) Disables the speakerphone
mute function so that incoming intercom calls
can immediately hear the party being called.
Step 5 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 6 Repeat Steps 1 through 5 for the second ephone-dn. The intercom feature requires configuration of
separate ephone-dns at each end of the two-way voice
path.
Step 7 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that
Example: identifies the ephone that is to receive the
Router(config)# ephone 24 intercom ephone-dn.
Step 8 button button-number:dn-tag [[button-number:dn-tag] Assigns a button number to the intercom ephone-dn
...] that you just defined. Use the colon separator (:)
between the button number and the intercom
Example: ephone-dn tag to indicate a normal ring for the
Router(config-ephone)# button 1:1 2:4 3:14 intercom line.
For other keywords and arguments for this command,
refer to the Cisco CallManager Express 3.3
Command Reference.
Step 9 restart Performs a fast reboot of this ephone. Does not
contact the Dynamic Host Configuration Protocol
(DHCP) or TFTP server for updated information.
Example:
Router(config-ephone)# restart

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Command or Action Purpose

Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit
Step 11 Repeat Steps 7 through 10 for the second phone. The intercom feature requires configuration of an
ephone at each end of the two-way voice path.

Example

The following example shows an intercom between two Cisco IP phones. In this example, ephone-dn 2
and ephone-dn 4 are normal extensions, while ephone-dn 18 and ephone-dn 19 are set as an intercom
pair. Ephone-dn 18 is associated with line button 2 on Cisco IP phone 4. Ephone-dn 19 is associated with
line button 2 on Cisco IP phone 5. The two ephone-dns provide a two-way intercom between the two
Cisco IP phones.
ephone-dn 2
number 5333

ephone-dn 4
number 5222

ephone-dn 18
number 5001
name “intercom”
intercom 5002 barge-in

ephone-dn 19
name “intercom”
number 5002
intercom 5001 barge-in

ephone 4
button 1:2 2:18

ephone 5
button 1:4 2:19

Paging
Audio paging provides a one-way voice path to the phones that have been designated to receive paging.
It does not have a press-to-answer option like the intercom feature. A paging group is created using a
dummy ephone-dn, known as the paging ephone-dn, that can be associated with any number of local IP
phones. The paging ephone-dn can be dialed from anywhere, including on-net. Figure 39 shows a paging
group with two phones.
When a caller dials the paging number (ephone-dn), each idle IP phone that has been configured with
the paging number automatically answers using its speakerphone mode. Displays on the phones that
answer the page show the caller ID that has been set using the name command under the paging
ephone-dn. When the caller finishes speaking the message and hangs up, the phones are returned to their
idle states.
Several paging groups can be specified in a Cisco CME system and two or more paging groups can be
joined into a combined group.

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The paging mechanism supports audio distribution using IP multicast, replicated unicast, and a mixture
of both (so that multicast is used where possible, and unicast is used for specific phones that cannot be
reached using multicast).
To configure paging, complete one or both of the following tasks:
• Configuring Paging for a Single Group, page 300
• Configuring Paging for a Combined Group, page 303

Figure 39 Paging

1 To page all the phones in the shipping IP Any phone dials 4444.
department, a person at any phone dials
the number associated with the paging
ephone-dn for the shipping department.
The paging ephone-dn has a number that
does not appear on any phone (in this
example, extension 4444). Ephone-dn 4
Extension 4444
This is a paging ephone-dn; no physical phone
instrument is associated with this number.
2 A one-way voice connection is automatically 4444
made with all idle ephones that are
configured with paging ephone-dn 4. In this
example, that is phone 1 and phone 2. Both
phones answer the call in speakerphone V
mode. The voice of the calling party is heard
through the speaker, and the phone displays
the caller ID (name) of paging ephone-dn 4 Phone 1
("Paging Shipping"). Button 1 is extension 2121, a
IP
normal line.
This phone has a paging-dn to
receive pages.
ephone-dn 4
number 4444 Phone 2
name Paging Shipping Button 1 is extension 2222, a normal line.
IP This phone has a paging-dn to receive
paging ip 239.0.1.20 port 2000
pages.
ephone-dn 21
number 2121

ephone-dn 22 Note that paging-dns are not

number 2222 assigned to phone buttons.

ephone 1
mac-address 3662.0234.6ae2
button 1:21
paging-dn 4

ephone 2
88953

mac-address 9387.6738.2873
button 1:22
paging-dn 4

Restrictions
IP phones do not support multicast at 224.x.x.x addresses.

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Configuring Paging for a Single Group

The paging feature defines an ephone-dn that broadcasts unicast audio paging to a single group of idle
Cisco IP phones that have been associated with the paging ephone-dn tag.

SUMMARY STEPS

1. ephone-dn paging-dn-tag
2. number number
3. name name
4. paging [ip multicast-address port udp-port-number]
5. exit
6. ephone phone-tag
7. paging-dn paging-dn-tag {multicast | unicast}
8. exit
9. Repeat Steps 6 through 8 to add additional IP phones to the paging group.

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn paging-dn-tag Enters ephone-dn configuration mode.
• paging-dn-tag—A unique sequence number that identifies this
Example: paging ephone-dn during all configuration tasks. This is the
Router(config)# ephone-dn 42 ephone-dn that is dialed to initiate a page. This ephone-dn is not
associated with a physical phone. Range is from 1 to 288.
Step 2 number number Defines an extension number associated with the paging ephone-dn.
This is the number that people call to initiate a page.
Example:
Router(config-ephone-dn)# number 3556
Step 3 name name Assigns to the paging number a name to appear in caller-ID displays
and directories.
Example:
Router(config-ephone-dn)# name paging4

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Command or Action Purpose

Step 4 paging [ip multicast-address port Specifies that this ephone-dn is to be used to broadcast audio paging
udp-port-number] messages to the idle IP phones that are associated with the paging
dn-tag. If the optional keywords and arguments are not used, IP
Example: phones are paged individually using IP unicast transmission (to a
Router(config-ephone-dn)# paging ip maximum of ten IP phones). The optional keywords and arguments
239.1.1.10 port 2000 are as follows:
• ip multicast-address port udp-port-number—Specifies
multicast broadcast using the specified IP address and UDP
port. When multiple paging numbers are configured, each
paging number must use a unique IP multicast address. Port
2000 is recommended because it is already used for normal
nonmulitcast RTP media streams between phones and the
Cisco CME router.
Note IP phones do not support multicast at 224.x.x.x addresses.
Step 5 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 6 ephone phone-tag Enters ephone configuration mode to add IP phones to the paging
group.
Example: • phone-tag—Unique sequence number of a phone that should
Router(config)# ephone 2 receive audio pages from the paging ephone-dn.
Step 7 paging-dn paging-dn-tag {multicast | Associates this ephone with the ephone-dn tag used for the audio
unicast} paging ephone-dn. Note that the paging ephone-dn tag is not
associated with a line button on this ephone.
Example: The paging mechanism supports audio distribution using IP
Router(config-ephone)# paging-dn 42 multicast, replicated unicast, and a mixture of both (so that
multicast
multicast is used where possible and unicast is allowed to specific
phones that cannot be reached through multicast).
• paging-dn-tag—Unique sequence number that was assigned to
the paging ephone-dn.
• multicast—Multicast paging for groups. By default, audio
paging is transmitted to the Cisco IP phone using multicast.
• unicast—Unicast paging for a single Cisco IP phone. This
keyword indicates that the Cisco IP phone is not capable of
receiving audio paging through multicast and requests that the
phone receives the audio paging through a unicast transmission
directed to the individual phone.
Note The number of phones supported through unicast is limited
to a maximum of ten phones.

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Command or Action Purpose

Step 8 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit
Step 9 Repeat Steps 6 through 8 to add more phones —
to the paging group.

Examples

The following example sets up an ephone-dn for multicast paging. This example creates a paging number
for 5001 on ephone-dn 22 and adds ephone 4 as a member of the paging set. Multicast is set for the
paging-dn.
ephone-dn 22
name Paging Shipping
number 5001
paging ip 239.1.1.10 port 2000

ephone 4
mac-address 0030.94c3.8724
button 1:1 2:2
paging-dn 22 multicast

In this example, paging calls to 2000 are multicast to Cisco IP phones 1 and 2, and paging calls to 2001
go to Cisco IP phones 3 and 4. Note that the paging ephone-dns (20 and 21) are not assigned to any phone
buttons.
ephone-dn 20
number 2000
paging ip 239.0.1.20 port 2000

ephone-dn 21
number 2001
paging ip 239.0.1.21 port 2000

ephone 1
mac-address 3662.024.6ae2
button 1:1
paging-dn 20

ephone 2
mac-address 9387.678.2873
button 1:2
paging-dn 20

ephone 3
mac-address 0478.2a78.8640
button 1:3
paging-dn 21

ephone 4
mac-address 4398.b694.456
button 1:4
paging-dn 21

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Configuring Paging for a Combined Group

You can unit the groups that you created using the instructions in the “Configuring Paging for a Single
Group” section on page 300 into a combined group.
By configuring the ability to page combined groups in addition to single groups, you provide phone users
with the flexibility to page a small local paging group (for example, paging to four phones in a
company’s technical support department) or to page a combined set of several paging groups (for
example, by paging a group that consists of technical support and sales phones).

Prerequisites
You must specify two or more individual paging groups as explained in the “Configuring Paging for a
Single Group” section on page 300 before combining those groups in this task.

SUMMARY STEPS

1. ephone-dn group-paging-dn-tag
2. number number
3. name name
4. paging [ip multicast-address port udp-port-number]
5. paging group paging-dn-tag,paging-dn-tag[[,paging-dn-tag]...]
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn group-paging-dn-tag Enters ephone-dn configuration mode to create a paging
ephone-dn for the multiple group.
Example: • group-paging-dn-tag—Unique sequence number to identify
Router(config)# ephone-dn 20 this combined paging group during configuration tasks.
Step 2 number number Assigns an extension number to be dialed in order to send a page
to the combined paging group that is being defined.
Example:
Router(config-ephone-dn)# number 2000
Step 3 name name Assigns to the combined paging number a name to appear in
caller-ID displays and directories, up to 24 characters.
Example:
Router(config-ephone-dn)# name Paging
AllGroups

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Command or Action Purpose

Step 4 paging [ip multicast-address port Specifies that this ephone-dn is to be used to broadcast audio
udp-port-number] paging messages to the idle IP phones that are associated with the
paging dn-tag. If the optional keywords and arguments are not
Example: used, IP phones are paged individually using IP unicast
Router(config-ephone-dn)# paging ip transmission (to a maximum of ten IP phones). The optional
239.1.1.10 port 2000 keywords and arguments are as follows:
• ip multicast-address port udp-port-number—Specifies
multicast broadcast using the specified IP address and UDP
port. This is the recommended operation. When multiple
paging numbers are configured, each number must use a
unique IP multicast address.
Step 5 paging group paging-dn-tag,paging-dn-tag Sets the audio paging directory number for a combined group. The
[[,paging-dn-tag]...] paging group command combines the individual paging group
ephone-dns into a combined group so that a page can be sent to
Example: more than one paging group at one time.
Router(config-ephone-dn)# paging group Note Configure the paging command for all ephone-dns in a
20,21
group prior to configuring the paging group command for
that group.

• paging-dn-tag—Unique sequence number associated with the

paging number for an individual paging group, as described
in the “Configuring Paging for a Single Group” section on
page 300. List the paging-dn-tags of all the individual groups
that you want to include in this combined group, separated by
commas. You can include up to ten paging ephone-dn tags in
this command.
Step 6 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit

Example

This example sets the following paging behavior:

• When extension 2000 is dialed, a page is sent to ephones 1 and 2 (single paging group).
• When extension 2001 is dialed, a page is sent to ephones 3 and 4 (single paging group).
• When extension 2002 is dialed, a page is sent to ephones 1, 2, 3, 4, and 5 (combined paging group).
Ephones 1 and 2 are included in paging ephone-dn 22 through the membership of ephone-dn 20 in the
combined paging group. Ephones 3 and 4 are included in paging ephone-dn 22 through membership of
ephone-dn 21 in the combined paging group. Ephone 5 is directly subscribed to paging-dn 22.
ephone-dn 20
number 2000
paging ip 239.0.1.20 port 2000

ephone-dn 21
number 2001
paging ip 239.0.1.21 port 2000

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ephone-dn 22
number 2002
paging ip 239.0.2.22 port 2000
paging group 20,21

ephone-dn 6
number 1103
name user3

ephone-dn 7
number 1104
name user4

ephone-dn 8
number 1105
name user5

ephone-dn 9
number 1199

ephone-dn 10
number 1198

ephone 1
mac-address 1234.8903.2941
button 1:6
paging-dn 20

ephone 2
mac-address CFBA.321B.96FA
button 1:7
paging-dn 20

ephone 3
mac-address CFBB.3232.9611
button 1:8
paging-dn 21

ephone 4
mac-address 3928.3012.EE89
button 1:9
paging-dn 21

ephone 5
mac-address BB93.9345.0031
button 1:10
paging-dn 22

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Account Code Entry by User

Account Code Entry by User

The Cisco IP Phones 7940 and 7940G and the Cisco IP Phones 7960 and 7960G allow users to enter
account codes during call setup or while connected to an active call using the Acct soft key. Account
codes are inserted into call detail records (CDRs) on the Cisco CME router for later interpretation by
billing software.
An account code is visible in the output of the show call active command and the show call history
command for telephony call legs and is supported by the CISCO-VOICE-DIAL-CONTROL-MIB. The
account code also appears as a named string of “account-code” in RADIUS vendor-specific attribute
(VSA) Vendor Type field 1 for voice authentication, authorization, and accounting (AAA) accounting.
No configuration is required for this feature. To enter an account code during call setup or while in a
connected state, press the Acct soft key and enter the account code from the phone keypad.

Applications Integration with Cisco CME

Cisco CallManager Express (Cisco CME) provides an interface that enables simple one-to-one remote
control of a Cisco IP phone by an associated PC that is running Cisco IOS Telephony Services Provider
(TSP), a Telephony Application Programming Interface (TAPI) that supports the placing of outbound
calls from an application phone address book. This interface is intended to support only basic TAPI
services to enable caller-ID-based screen pop-ups for incoming calls and to support simple outgoing call
placement using one-click address-book-style speed-dialing from the PC application.
Cisco IOS TSP is a TAPI service provider software package that works as an interface between the TAPI
that is running on Microsoft Windows and the Cisco CME router. Cisco IOS TSP provides the following
functionality:
• Communication with the TAPI using the Telephony Service Provider Interface (TSPI).
• Implementation of a required set of APIs and interaction with TAPI.
• Enabling of other TAPI-based applications to provide call control to the Cisco IP phones connected
to the Cisco CME router.

Note Cisco CME implements only a small subset of TAPI functionality. It supports only TAPI clients that
operate on one phone line at a time. The Cisco IOS TSP support does not have full TAPI support for
multiple users or for complex features like automatic call distributor (ACD) or IP contact center (IPCC).
Also, this TAPI version does not have direct media- and voice-handling capabilities. Media and voice
are sent to the phone.

Cisco IOS TSP software increases personal productivity because users can handle call management from
a PC without picking up a phone handset or dialing numbers on the phone keypad. The following
functionality is available:
• Answering of incoming calls
• Forwarding of incoming calls to voice mail
• Dialing of address book entries (placing outbound calls from an address book)
• Providing of screen pop-ups with caller-ID displays
• Placing of calls on hold

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Applications Integration with Cisco CME

This section explains the following tasks that are related to applications integration with Cisco IOS TSP:
• Installing Cisco IOS TSP, page 307
• Modifying a TSP Configuration, page 309
• Removing Cisco IOS TSP, page 311
• Verifying Basic TAPI Operation, page 311
• Troubleshooting Cisco IOS TSP, page 312

Installing Cisco IOS TSP

This task prepares Cisco IOS TSP for use.

Prerequisites
• On the router, collect the following information for Cisco IOS TSP configuration:
– MAC address of the Cisco IP phone that is to be associated with and controlled by a TAPI client.
Use the show ephone command to find this information.
– Username and password for the Cisco IP phone. Use the show ephone command to find this
information.
– IP address or host name (if DNS is enabled) and port number of the Cisco CME router. Use the
show telephony-service command to find the IP address and port number or use the
show running-config command to determine the host name.
• Download the CiscoIOSTSP1.3.zip file from http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp
to your PC. Uncompress the file and complete the installation on a PC that is running Windows 2000
where you want to install the Cisco IOS TSP.

Note These instructions were written for version 1.3 of the Cisco IOS TSP application. If you download a
different version, note that the version number will change accordingly for all files.

SUMMARY STEPS

1. Ensure that there is network connectivity between your PC and the Cisco CME router.
2. Run CiscoIOSTspLite1.3.exe.
3. Complete the required fields in the Cisco IOS Telephony Services Provider dialog box.
4. At the prompt, restart your computer.
5. Before using Cisco IOS TSP for the first time, either restart the computer or select Cisco IOS TSP
from Phone and Modem Options.

DETAILED STEPS

Step 1 Ensure that there is network connectivity between your PC and the Cisco CME router. To verify the
network connectivity, enter the ping ip-address command on your PC, specifying the IP address of the
Cisco CME router.

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Step 2 Run CiscoIOSTspLite1.3.exe. The following Cisco IOS TSP dynamic linking library (DLL) files are
installed:
• CiscoIOSTSP.tsp
• CiscoIOSTUISP.dll
• LogTrace.dll
Step 3 Complete the required fields in the Cisco IOS Telephony Services Provider configuration dialog box
(Figure 40 on page 309), which appears after the DLL files are installed.
a. Enter the username and password that will be used by the Cisco IP phone user.
b. Enter the IP address and port number of the Cisco CME router.
c. Set the Synchronous Timeout response from the Cisco CME router to the desired value in seconds.
(The default is 3 seconds.)
d. If you are using a headset, check the Using HeadSet check box.
e. Check the Trace check box if you want to enable the trace function for troubleshooting. Note that
this function slows down the TAPI application considerably. For further information, see the
“Troubleshooting Cisco IOS TSP” section on page 312.

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Figure 40 Cisco IOS Telephony Service Provider Dialog Box

Step 4 At the prompt, restart your computer.

Modifying a TSP Configuration

The following task allows you to modify a TSP configuration that you have previously made on your PC.

SUMMARY STEPS

1. Choose Start > Settings > Control Panel > Phone and Modem Options.
2. Choose the Advanced tab to display telephony providers.
3. Choose Cisco IOS Telephony Services Provider and click Configure.
4. Make the changes that you desire in the Cisco IOS Telephony Services Provider dialog box.
5. Choose OK to save the changes and reboot your PC if prompted to do so.

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DETAILED STEPS

Step 1 Choose Start > Settings > Control Panel > Phone and Modem Options. The name of this option may
vary, depending on your operating system.
Step 2 Choose the Advanced tab in the Phone and Modem Options dialog box, which displays the Cisco IOS
Telephony Service Provider in the Providers list (Figure 41).

Figure 41 Cisco IOS Telephony Service Provider Selection Window

Step 3 Choose Cisco IOS Telephony Service Provider and click Configure, as shown in Figure 41.
The Cisco IOS Telephony Services Provider dialog box appears, as shown in Figure 40 on page 309.
Step 4 Make the changes that you desire in the Cisco IOS Telephony Service Provider dialog box.
Step 5 Choose OK to save your changes and reboot your PC if prompted to do so. If you change a username,
password, IP address, or port of the Cisco IOS Telephony Services Provider, you must restart any
running TAPI applications for the changes to take effect. If any services that depend on the telephony
service are running, such as Remote Access Connection Manager, you must also restart the system for
the changes to take effect. You might also get a prompt to reboot your system.

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Removing Cisco IOS TSP

This section explains how to remove Cisco IOS TSP from your computer.

Prerequisites
Ensure that the following conditions are met:
• The driver is not in use.
• The files are not in read-only mode.
• You have full control of the files in the C:\winnt\system32 directory.

SUMMARY STEPS

1. Choose Start > Settings > Control Panel > Add/Remove Programs.
2. Choose Cisco IOS Telephony Services Provider from the displayed list of programs and click the
Change/Remove button.
3. Follow the prompts to remove all TSP files from your PC.

DETAILED STEPS

1. Choose Start > Settings > Control Panel > Add/Remove Programs.
2. Choose Cisco IOS Telephony Services Provider from the displayed list of programs and click the
Change/Remove button.
3. Follow the prompts to remove all TSP files from your PC.

Verifying Basic TAPI Operation

This section explains how to verify that TAPI is working.

SUMMARY STEPS

1. Place an incoming call.

2. Place an outgoing call.

DETAILED STEPS

Step 1 Place an incoming call from another Cisco IP phone to the phone that you are verifying.
Step 2 Place an outgoing call from the Cisco IP phone that you are verifying.

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Troubleshooting Cisco IOS TSP

If you have problems with Cisco IOS TSP, perform the following tasks:
• Troubleshooting Cisco IOS TSP on the Cisco CME Router, page 312
• Troubleshooting Cisco IOS TSP on a PC, page 312

Troubleshooting Cisco IOS TSP on the Cisco CME Router

• Verify the IP address and the port number for the Cisco CME router, and make sure that they match
those as defined in the TSP Settings dialog box.
• Verify connectivity between the phone and the Cisco CME router by using the ping ip-address
command.
• If the Cisco IOS TSP fails to connect to the Cisco CME router, a message box with the title
“Cisco IOS Telephony Service TSP status” displays the cause for failure.
• If the Line or Address is not listed in the Preferences > Dialer dialog box, check the following
settings:
– Verify that the username and password have been configured correctly (they are case-sensitive)
and that Synchronous timeout is configured to a reasonable value (3 seconds).
– Verify that the IP address and port number in the dialog box match those on the Cisco CME
router.
– Verify connectivity between the Cisco IP phone and the Cisco CME router.
• Check the Trace check box, which you can see at the bottom of Figure 40 on page 309. The trace
feature runs the trace utility logs and saves the files in a specified trace file.

Note To save the trace utility logs, name a file that you can recognize to save the trace utility logs.
Each time you run the trace log, a new trace file is created, so you may need to regularly
remove the old trace files that are not required.

Troubleshooting Cisco IOS TSP on a PC

If the Cisco IOS Telephony Service Provider is not listed as one of the available providers in the
Control Panel > Phone and Modem Options > Add dialog box, verify whether the DLL files have been
installed in the PC’s system directory. During installation, the following DLL files should be added to
the PC’s system directory:
• CiscoIOSTSP.tsp
• CiscoIOSTUISP.dll
• LogTrace.dll

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XML API

XML API
An eXtensible Markup Language (XML) application program interface (API) is provided to supply data
from Cisco CME to management software. The Cisco IOS commands in this section allow you to specify
certain parameters associated with the XML API. For more information, refer to the XML Developer
Guide for Cisco CME/SRST.

SUMMARY STEPS

1. telephony-service
2. log password password-string
3. log table {max-size entries | retain-timer minutes}
4. xmlschema schema-url
5. xmltest
6. xmlthread number
7. exit
8. ephone phone-tag
9. keyphone
10. exit
11. exit
12. show fb-its-log [summary]

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 log password password-string Sets a local password for an XML API query. The local
password is used to authenticate XML API requests on the
network management server. If a password is not set, an
Example:
Router(config-telephony)# log password w2eiql
XML API query fails local authentication.

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Command or Action Purpose

Step 3 log table {max-size entries | retain-timer Sets the maximum number of events, or entries, that can be
minutes} stored in the event table and the length of time that the
events are retained before deletion from the table. The event
Example: table is an internal buffer that stores captured and
Router(config-telephony)# log table max-size time-stamped events, such as phones registering and
240 unregistering and extension status. One event equals one
entry in the table.
• max-size entries—Number of entries in the log table.
Range is from 0 to 1000. Default is 150.
• retain-timer minutes—Number of minutes to retain
entries in the log table before they are deleted. Range is
from 2 to 500. Default is 15.
Step 4 xmlschema schema-url Specifies the URL for an XML API schema to be used with
this Cisco CME system.
Example: • schema-url—Local or remote URL as defined in
Router(config-telephony)# xmlschema RFC 2396.
http://server2.example.com/
schema/schema1.xsd
Step 5 xmltest Specifies that the HTTP payload in XML API queries be
interpreted as having form format. The default is that XML
API queries are interpreted as having plain text (no form)
Example:
Router(config-telephony)# xmltest
format.
Step 6 xmlthread number Sets the maximum number of Cisco CME XML API
queries.
Example: • Maximum number of XML API queries. Range is from
Router(config-telephony)# xmlthread 4 1 to 5. Default is 2.
Step 7 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 8 ephone phone-tag Enters ephone configuration mode.
• phone-tag—Unique sequence number that identifies
Example: the ephone that is to receive the intercom ephone-dn.
Router(config)# ephone 24
Step 9 keyphone Marks a Cisco IP phone as a “key” phone to be tracked
while using the XML API. The XML API can be instructed
to report the status of only the key phones in the system for
Example:
Router(config-ephone)# keyphone
network management purposes, for example.
Step 10 exit Exits ephone configuration mode.

Example:
Router(config-ephone)# exit

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Command or Action Purpose

Step 11 exit Exits global configuration mode.

Example:
Router(config)# exit
Step 12 show fb-its-log [summary] Displays information about the Cisco CME XML API
configuration, statistics on XML API queries, and the XML
API event logs.
Example:
Router# show fb-its-log • summary—(Optional) Displays only the XML API
configuration and the statistics for queries and logs, and
not the logs themselves.

Related Features
Other features that can be used on Cisco IP phones to improve productivity include the following:
• Label support in the “Configuring an Attendant for Primary Call Coverage” chapter.
• Silent ring on Cisco IP Phones 7960 and 7960G and Cisco IP Phone Expansion Module 7914 in the
“Configuring an Attendant for Primary Call Coverage” chapter.

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 Monitoring and Managing a Cisco CME System

The commands in this chapter help you to monitor and manage a Cisco CallManager Express
(Cisco CME) system. For sample outputs and complete reference information, refer to the
Cisco CallManager Express 3.3 Command Reference.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

To view the entire running configuration, use this command:

• show running-config—Displays the running configuration for the router.
To view parameters associated with the entire Cisco CME system, use these commands:
• show telephony-service admin—Displays username and password for the system administrator
and whether the system administrator can make changes to ephone-dns and the system clock through
the GUI.
• show telephony-service all—Displays configurations for all ephones, ephone-dns, voice ports, and
dial peers.
• show telephony-service dial-peer—Displays information for the dial peers that are automatically
created by ephone-dns.
• show telephony-service directory-entry—Displays the entries that have been made using the
directory entry command.
• show telephony-service ephone—Displays configurations of all ephones.
• show telephony-service ephone-dn—Displays configurations of all ephone-dns.
• show telephony-service ephone-template—Displays configurations of all ephone-templates.
• show telephony-service tftp-bindings—Displays the current configuration files that are accessible
to IP phones.
• show telephony-service voice-port—Displays configurations of all virtual voice ports.
To view parameters associated with one or all extensions (ephone-dns), use these commands:
• show ephone-dn [dn-tag] [statistics]—Displays line status and detailed information for a specified
extension (ephone-dn) or all extensions. If the statistics keyword is used, provides statistics on voice
quality and far-end disconnect causes.
• show ephone-dn callback—Displays information about pending callbacks.

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• show ephone-dn loopback—Displays information about loopback-dn pairs that have been created.
• show ephone-dn summary—Displays brief information about extensions (ephone-dns).
To view parameters associated with one or all ephones, use these commands:
• show ephone [mac-address | phone-type]—Displays phone information, including associated
ephone-dns and their status, for a specified ephone, for a specified type of ephone, or for all ephones.
• show ephone cfa—Displays line status and brief information on the registered phones that have
call-forward-all set on one or more of their extensions (ephone-dns),
• show ephone dn [dn-tag]—Displays type of ephone-dn and identifies the phone on which an
ephone-dn tag appears, for a specified dn-tag or for all dn-tags.
• show ephone dnd—Displays brief information on the registered phones that have do not disturb set
on one or more of their extensions (ephone-dns).
• show ephone login—Displays whether an ephone has a personal identification number (PIN) and
whether its owner has logged in.
• show ephone offhook—Displays brief information, line status, and packet counts for phones that
are currently off hook.
• show ephone overlay—Displays brief information for phones with overlay sets assigned to them
and displays the contents of the overlay sets.
• show ephone phone-load—Displays device name, current phone firmware, previous phone
firmware, and reason for last reset for all ephones.
• show ephone registered—Displays information for phones that are currently registered with
Cisco CME.
• show ephone remote—Displays brief information for nonlocal phones (phones with no Address
Resolution Protocol [ARP] entry).
• show ephone ringing—Displays brief information on phones that are ringing and indicates which
buttons, ephone-dn tags, and numbers are ringing.
• show ephone summary—Displays brief information for all ephones.
• show ephone tapiclients—Displays status of ephone Telephony Application Programming
Interface (TAPI) clients.
• show ephone telephone-number number—Displays information for the phone associated with the
specified number. This is a good way to discover the phone on which a particular number appears.
• show ephone unregistered—Displays information for phones that are not currently registered with
Cisco CME.

Related commands that may be useful include the following:

• show call history voice—Displays the call history table for voice calls.
To specify call detail record (CDR) storage time, use the dial-control-mib command, which has two
sets of keyword and argument options. The max-size number option specifies the maximum size of
the CDR event table. The valid range is from 0-500, and the default is 50. Setting the value to 0
disables the CDR feature.
The retain-timer number option specifies the length of time in minutes that entries will remain in
the call history table. The valid range is from 0-2147483647 minutes, and the default is 15. Setting
the value to 0 prevents any call history from being retained.

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• show dial-peer voice summary—Displays brief information and status of all VoIP and POTS dial
peers on a router.
• show fb-its-log—Displays Cisco CME XML API configuration information, XML API query
statistics, and XML API event logs.
• show voice port summary—Displays brief information for all voice ports on a router.

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Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm. For more
information about voice and fax troubleshooting, refer to Cisco IOS Voice Troubleshooting and
Monitoring.

To troubleshoot the router in a Cisco CallManager Express (Cisco CME) system, perform the following
task.

Step 1 Enter the show ephone command to display all registered phones. If no phones are registered, perform
the following steps:
a. Configure the Cisco CME router.
b. Check DHCP configuration, including the default router and the TFTP server address (option 150).
c. Enter the dir command to check that the required files are in the router’s flash memory.
d. Check that the tftp-server command is set for the required files, and use the show
telephony-service tftp-bindings command to check which configuration files are accessible by IP
phones using TFTP. Use the debug tftp events command to monitor TFTP file access by the IP
phones during their registration attempts.
e. Enter the debug ephone register mac-address command to display Cisco IP phone registration
activity.
f. Enter the debug ip dhcp command to confirm DHCP operation.
Step 2 Enter the show ephone command to display all registered phones. If phones are registered and are
displayed, perform the following steps:
a. Check that the phone button binding to the directory number is correct.
b. Check that the Cisco IP phones show as registered.
c. Verify the IP parameter settings on the Cisco IP phone using the Settings display on the phone.
d. Check that the keepalive count is being updated when you enter the show ephone command.
e. Reset the phone and observe the reregistration by entering the debug ephone register mac-address
command to display the Cisco IP phones.
f. Enter the show ephone-dn summary command to check the state of the Cisco IP phone lines.

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g. Check the IP address of the phone, and attempt to ping the address.
Step 3 Use the show dialplan number command to display the number resolutions of a particular phone
number, which allows you to detect whether calls are going to unexpected destinations. This command
is useful for troubleshooting cases in which you dial a number but the expected phone does not ring.
Step 4 Enter the debug ephone keepalive command to set keepalive debugging for the Cisco IP phones.
Step 5 Enter the debug ephone state command to set state debugging for the Cisco IP phones.

To troubleshoot other areas of the Cisco CME router, use the commands listed in Table 8. For further
debugging, you can use the debug commands documented in the Cisco IOS Debug Command Reference,
Release 12.3 T.

Table 8 Troubleshooting Commands

Command Description
debug ephone detail Sets detail debugging for a Cisco IP phone
debug ephone error Sets error debugging for a Cisco IP phone.
debug ephone MWI Sets message-waiting indication (MWI) debugging for a Cisco IP phone.
debug ephone Sets call statistics debugging for a Cisco IP phone.
statistics
debug ephone pak Provides voice-packet-level debugging and display the contents of one voice
packet in every 1024 voice packets,
debug ephone raw Provides raw low-level protocol debugging display for all SCCP messages
debug ip http token Provides debugging for local directory search
debug ip http Troubleshoots HTTP authentication problems
authentication
debug ip http url Shows the URLs accessed from the router
debug ip http Displays HTTP server transaction processing
transaction

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 Appendix A: Configuring Loopback Call Routing

Loopback call routing in a Cisco CallManager Express (Cisco CME) system is provided through a
mechanism called loopback-dn, which provides a software-based limited emulation of back-to-back
physical voice ports connected together to provide a loopback call-routing path for voice calls.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• Information About Loopback Call Routing, page 323
• Configuring Loopback Call Routing, page 324

Information About Loopback Call Routing

The primary use of loopback call routing and loopback-dn is to restrict the passage of call-transfer and
call-forwarding supplementary service requests through the loopback. Instead of passing these requests
through, the loopback-dn mechanism attempts to service the requests locally. This allows loopback-dn
configurations to be used in call paths where one of the external devices does not support call transfer
or call forwarding (Cisco-proprietary or H.450-based). Control messages that request call transfer or call
forwarding are intercepted at the loopback virtual port and serviced on the local voice gateway. If
needed, this mechanism creates VoIP-to-VoIP call-routing paths.
Loopback call routing may be used for routing H.323 calls to Cisco Unity Express. For information
about using loopback call routing to integrate Cisco CME and Cisco Unity Express, refer to Integrating
Cisco CallManager Express with Cisco Unity Express at
http://www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/cmeinter/cue_cme1.htm. For
information about Cisco Unity Express configuration, go to the Cisco Unity Express documentation
index at http://www.cisco.com/univercd/cc/td/doc/product/voice/unityexp/index.htm.

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Configuring Loopback Call Routing

Note A preferred alternative to loopback call routing was introduced in Cisco CME 3.1. This alternative
blocks H.450-based supplementary service requests using the following CLI commands: no
supplementary-service h450.2, no supplementary-service h450.3, and supplementary-service
h450.12. For more information, see the “Configuring Call Transfer and Call Forwarding” chapter at
http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cme33/cme33sa/cme33bsc.htm.

Note Use of loopback-dn configurations within a VoIP network should be restricted to resolving critical
network interoperability service problems that cannot otherwise be solved. Loopback-dn configurations
are intended to be used in VoIP network interworking situations in which the only alternative would be
to make use of back-to-back-connected physical voice ports. Loopback-dn configurations emulate the
effect of a back-to-back physical voice-port arrangement without the expense of the physical voice-port
hardware. Because digital signal processors (DSPs) are not involved in loopback-dn arrangements, the
configuration does not support interworking or transcoding between calls that use different voice codecs.
In many cases, use of back-to-back physical voice ports that do involve DSPs to resolve VoIP network
interworking issues is preferred, because it introduces fewer restrictions in terms of supported codecs
and call flows. Also, loopback-dns do not support T.38 fax relay.

Configuring Loopback Call Routing

Loopback call routing requires two extensions (ephone-dns) to be separately configured, each as half of
a loopback-dn pair, as described in this task. In addition to defining the loopback-dn pair, you must
specify preference, huntstop, class of restriction (COR), and translation rules. Ephone-dns that are
defined using the loopback-dn command are never associated with a physical phone; they are used only
for loopback call routing.

SUMMARY STEPS

1. ephone-dn dn-tag
2. number number [secondary number] [no-reg [both | primary]]
3. caller-id {local | passthrough}
4. no huntstop
5. preference preference-order [secondary secondary-order]
6. cor {incoming | outgoing} cor-list-name
7. translate {called | calling} translation-rule-tag
8. loopback-dn dn-tag [forward number-of-digits | strip number-of-digits] [prefix
prefix-digit-string] [suffix suffix-digit-string] [retry seconds] [auto-con] [codec {g711alaw |
g711ulaw}]
9. exit
10. Repeat Steps 1 through 8 using a second ephone-dn to complete the loopback-dn pair.

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DETAILED STEPS

Command or Action Purpose

Step 1 ephone-dn dn-tag Enters ephone-dn configuration mode, creates an ephone-dn,
and optionally assigns it dual-line status.
Example: • dn-tag—Unique sequence number that identifies this
Router(config)# ephone-dn 15 ephone-dn during configuration tasks. Range is platform-
and version-dependent. Refer to command-line interface
(CLI) help.
Note Ephone-dns used for loopback cannot be dual-line
ephone-dns.
Step 2 number number [secondary number] [no-reg Associates a number with this extension (ephone-dn).
[both | primary]]
• number—String of up to 16 digits that represents a
telephone or extension number to be associated with this
Example: ephone-dn.
Router(config-ephone-dn)# number 2001
• secondary—(Optional) Allows you to associate a second
telephone number with an ephone-dn.
• no-reg—(Optional) Specifies that this number should not
register with the H.323 gatekeeper. The no-reg keyword by
itself indicates that only the secondary number should not
register. The no-reg both keywords indicate that both
numbers should not register, and the no-reg primary
keywords indicate that only the primary number should not
register.
Step 3 caller-id {local | passthrough} Specifies caller-ID treatment for outbound calls originated
from the ephone-dn. The default if this command is not used is
as follows. For transferred calls, caller ID is provided by the
Example:
Router(config-ephone-dn)# caller-id local
number and name fields from the outbound side of the
loopback-dn. For forwarded calls, caller ID is provided by the
original caller ID of the incoming call. Settings for the caller-id
block command and translation rules on the outbound side are
executed.
• local—Passes the local caller ID on redirected calls. This
is the preferred usage.
• passthrough—Passes the original caller ID on redirected
calls.
Step 4 no huntstop Disables huntstop and allows call hunting behavior for an
extension (ephone-dn).
Example:
Router(config-ephone-dn)# no huntstop

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Command or Action Purpose

Step 5 preference preference-order [secondary Sets dial-peer preference for an extension (ephone-dn).
secondary-order]
• preference-order—Preference order for the primary
number associated with an extension (ephone-dn). Range is
Example: from 0 to 10, where 0 is the highest preference and 10 is the
Router(config-ephone-dn)# preference 1 lowest preference. Default is 0.
• secondary secondary-order—(Optional) Preference order
for the secondary number associated with the ephone-dn.
Range is from 0 to 10, where 0 is the highest preference and
10 is the lowest preference. Default is 9.
Step 6 cor {incoming | outgoing} cor-list-name Applies a class of restriction (COR) to the dial peers associated
with an extension. COR is used to specify which incoming dial
peer can use which outgoing dial peer to make a call. Each dial
Example:
Router(config-ephone-dn)# cor incoming
peer can be provisioned with an incoming and an outgoing COR
corlist1 list.
For information about COR, see the “Configuring Dial Peer
Matching Features” section in “Dial Peer Configuration on
Voice Gateway Routers” at the Cisco IOS Voice Configuration
Library.
Step 7 translate {called | calling} Selects an existing translation rule and applies it to a calling
translation-rule-tag number or a number that has been called. This command
enables the manipulation of numbers as part of a dial plan to
Example: manage overlapping or nonconsecutive numbering schemes.
Router(config-ephone-dn)# translate called • called—Translates the called number.
1
• calling—Translates the calling number.
• translation-rule-tag—Unique sequence number of the
previously defined translation rule. Range is from
1 to 2147483647.
Note This command requires that you have previously
defined appropriate translation rules using the voice
translation-rule and rule commands.

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Command or Action Purpose

Step 8 loopback-dn dn-tag [forward Enables H.323 VoIP call transfer and call forwarding by using
number-of-digits | strip number-of-digits] hairpin call routing for VoIP endpoints that do not support the
[prefix prefix-digit-string] [suffix
suffix-digit-string] [retry seconds]
Cisco-proprietary call-transfer and call-forwarding
[auto-con] [codec {g711alaw | g711ulaw}] mechanism.
• dn-tag—Unique sequence number that identifies the
Example: ephone-dn that is being paired for loopback with the
Router(config-ephone-dn)# loopback-dn 24 ephone-dn that is currently being configured. The paired
forward 15 prefix 415353.... ephone-dn must be one that is already defined in the
system.
• forward number-of-digits—(Optional) Number of digits in
the original called number to forward to the other
ephone-dn in the loopback-dn pair. Range is from 1 to 32.
Default is to forward all digits.
• strip number-of-digits—(Optional) Number of leading
digits to be stripped from the original called number before
forwarding to the other ephone-dn in the loopback-dn pair.
Range is from 1 to 32. Default is to not strip any digits.
• prefix prefix-digit-string—(Optional) Defines a string of
digits to add in front of the forwarded called number.
Maximum number of digits in the string is 32. Default is
that no prefix is defined.
• suffix suffix-digit-string—(Optional) Defines a string of
digits to add to the end of the forwarded called number.
Maximum number of digits in the string is 32. Default is
that no suffix is defined. If you add a suffix that starts with
the pound character (#), the string must be enclosed in
quotation marks.
• retry seconds—(Optional) Number of seconds to wait
before retrying the loopback target when it is busy or
unavailable. Range is from 0 to 32767. Default is that retry
is disabled and appropriate call-progress tones are passed
to the call originator.
• auto-con—(Optional) Immediately connects the call and
provides in-band alerting while waiting for the far-end
destination to answer. Default is that automatic connection
is disabled.
• codec—(Optional) Explicitly forces the G.711 A-law or
G.711 mu-law voice coding type to be used for calls that
pass through the loopback-dn. This overrides the G.711
coding type that is negotiated for the call and provides
conversion from mu-law to A-law if needed. Default is that
Real-Time Transport Protocol (RTP) voice packets are
passed through the loopback-dn without considering the
G.711 coding type negotiated for the calls.
• g711alaw—G.711 A-law, 64000 bits per second, for T1.
• g711ulaw—G.711 mu-law, 64000 bits per second, for E1.

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 Appendix A: Configuring Loopback Call Routing
Configuring Loopback Call Routing

Command or Action Purpose

Step 9 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 10 Repeat steps 1 through 8 using a second —
ephone-dn to complete the loopback-dn pair.

Example

The following example uses ephone-dns 15 and 16 as a loopback-dn pair. Calls are routed through this
loopback ephone-dn pair in the following way:
• An incoming call to 4085552xxx enters the loopback pair through ephone-dn 16 and exits the
loopback via ephone-dn 15 as an outgoing call to 2xxx (based on the forward 4 digits setting).
• An incoming call to 6xxx enters the loopback pair through ephone-dn 15 and exits the loopback via
ephone-dn 16 as an outgoing call to 4157676xxx (based on the prefix 415767 setting).
ephone-dn 15
number 6...
loopback-dn 16 forward 4 prefix 415767
caller-id local
no huntstop
!
!
ephone-dn 16
number 4085552...
loopback-dn 15 forward 4
caller-id local
no huntstop
!

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 Appendix B: Providing Cisco CME Support for SIP

Cisco CallManager Express (Cisco CME) supports incoming and outgoing Session Initiation Protocol
(SIP) calls to and from IP phones and router voice gateway voice ports, but does not support direct
attachment of SIP phones to Cisco CME. Special configurations to support SIP calls are described in this
appendix.
For more information about SIP, refer to the Cisco IOS SIP Configuration Guide.

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/voice_c/vcl.htm.

Contents
• DTMF Relay for SIP Applications and Voice Mail, page 329
• SIP Register Support, page 333
• Call Transfer over SIP Networks, page 335
• Call Forwarding over SIP Networks, page 337

DTMF Relay for SIP Applications and Voice Mail

DTMF relay for SIP Applications can be used in two voice-mail situations:
• DTMF Relay Using SIP RFC 2833, page 329
• DTMF Relay Using SIP Notify (Nonstandard), page 331

DTMF Relay Using SIP RFC 2833

To use remote voice-mail or interactive voice response (IVR) applications on SIP networks from
Cisco CME phones, you must enable conversion of the out-of-band dual tone multifrequency (DTMF)
digits used by the Cisco CME phones to the RFC 2833 in-band DTMF relay mechanism used by SIP
phones.

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 Appendix B: Providing Cisco CME Support for SIP
DTMF Relay for SIP Applications and Voice Mail

Cisco Skinny Client Control Protocol (SCCP) phones, such as those used with Cisco CME systems, only
provide out-of-band DTMF digit indications. To enable SCCP phones to send digit information to
remote SIP-based IVR and voice mail applications, Cisco CME 3.2 and later versions provide conversion
from the out-of-band SCCP digit indication to the SIP standard for DTMF relay, which is RFC 2833.
You select this method in the SIP VoIP dial peer using the dtmf-relay rtp-nte command.
The SIP DTMF relay method is needed in the following situations:
• When SIP is used to connect a Cisco CME system to a remote SIP-based IVR or voice-mail
application.
• When SIP is used to connect a Cisco CME system to a remote SIP-PSTN voice gateway that goes
through the PSTN to a voice-mail or IVR application.
Note that the need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones
natively support in-band DTMF relay as specified in RFC 2833.
To enable SIP DTMF relay using RFC2833, the commands in this section must be used on both
originating and terminating gateways.

SUMMARY STEPS

1. dial-peer voice tag voip

2. dtmf-relay rtp-nte
3. exit
4. sip-ua
5. notify telephone-event max-duration time
6. exit

DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 2 voip
Step 2 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event
(NTE) payload type.
Example:
Router(config-dial-peer)# dtmf-relay sip-notify
Step 3 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 4 sip-ua Enables SIP user-agent configuration mode.

Example:
Router(config)# sip-ua

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 Appendix B: Providing Cisco CME Support for SIP
DTMF Relay for SIP Applications and Voice Mail

Command or Action Purpose

Step 5 notify telephone-event max-duration time Configures the maximum time interval allowed
between two consecutive NOTIFY messages for a
single DTMF event.
Example:
Router(config-sip-ua)# notify telephone-event • max-duration time—Time interval between
max-duration 2000 consecutive NOTIFY messages for a single
DTMF event, in milliseconds. Range is from 500
to 3000. Default is 2000.
Step 6 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit

Troubleshooting Tips

The dial-peer section of the show running-config command output displays DTMF relay status when it
is configured, as shown in this excerpt:
dial-peer voice 123 voip
destination-pattern [12]...
monitor probe icmp-ping
session protocol sipv2
session target ipv4:10.8.17.42
dtmf-relay rtp-nte

DTMF Relay Using SIP Notify (Nonstandard)

To use voice mail on a SIP network that connects to a Cisco Unity Express (CUE) system, use a
non-standard SIP Notify format. To configure the Notify format, use the sip-notify keyword with the
dtmf-relay command. Using the keyword sip-notify may be required for backward compatibility with
Cisco CME Versions 3.0 and 3.1.

SUMMARY STEPS

1. dial-peer voice tag voip

2. dtmf-relay sip-notify
3. exit
4. sip-ua
5. notify telephone-event max-duration time
6. exit

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 Appendix B: Providing Cisco CME Support for SIP
DTMF Relay for SIP Applications and Voice Mail

DETAILED STEPS

Command or Action Purpose

Step 1 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 2 voip
Step 2 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages.

Example:
Router(config-dial-peer)# dtmf-relay sip-notify
Step 3 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 4 sip-ua Enables SIP user-agent configuration mode.

Example:
Router(config)# sip-ua
Step 5 notify telephone-event max-duration time Configures the maximum time interval allowed
between two consecutive NOTIFY messages for a
single DTMF event.
Example:
Router(config-sip-ua)# notify telephone-event • max-duration time—Time interval between
max-duration 2000 consecutive NOTIFY messages for a single
DTMF event, in milliseconds. Range is from 500
to 3000. Default is 2000.
Step 6 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit

Troubleshooting Tips

The show sip-ua status command output displays the time interval between consecutive NOTIFY
messages for a telephone event. In the following example, the time interval is 2000 ms.
Router# show sip-ua status

SIP User Agent Status

SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
SIP User Agent bind status(signaling):DISABLED
SIP User Agent bind status(media):DISABLED
SIP early-media for 180 responses with SDP:ENABLED
SIP max-forwards :6
SIP DNS SRV version:2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP:NONE
Check media source packets:DISABLED
Maximum duration for a telephone-event in NOTIFYs:2000 ms
SIP support for ISDN SUSPEND/RESUME:ENABLED

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 Appendix B: Providing Cisco CME Support for SIP
SIP Register Support

Redirection (3xx) message handling:ENABLED

SDP application configuration:

Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported:audio image
Network types supported:IN
Address types supported:IP4
Transport types supported:RTP/AVP udptl

SIP Register Support

This section describes how to enable a SIP gateway to register E.164 numbers with a SIP proxy or SIP
registrar, similar to the way that H.323 gateways can register E.164 numbers with a gatekeeper. SIP
gateways allow registration of E.164 numbers to a SIP proxy or registrar on behalf of analog telephone
voice ports (FXS) and IP phone virtual voice ports (EFXS) for local SCCP phones.
When registering E.164 numbers in dial peers with an external registrar, you can also register them with
a secondary SIP proxy or registrar to provide redundancy. The secondary registration can be used if the
primary registrar fails.
For more detailed information, refer to SIP Gateway Enhancements, Cisco IOS Release 12.2(15)ZJ.

Note There are no commands that allow registration between the H.323 and SIP protocols.

By default, SIP gateways do not generate SIP Register messages, so the following steps are needed to
set up the gateway to register the gateway’s E.164 telephone numbers with an external SIP registrar.

SUMMARY STEPS

1. sip-ua
2. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]
3. retry register number
4. timers register time
5. exit

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 Appendix B: Providing Cisco CME Support for SIP
SIP Register Support

DETAILED STEPS

Command or Action Purpose

Step 1 sip-ua Enables SIP user-agent configuration mode.

Example:
Router(config)# sip-ua
Step 2 registrar {dns:host-name | ipv4:ip-address} Registers E.164 numbers on behalf of analog telephone
expires seconds [tcp] [secondary] voice ports (FXS) and IP phone virtual voice ports (EFXS)
with an external SIP proxy or SIP registrar server.
Example: • dns:host-name—Domain name server that resolves
Router(config-sip-ua)# registrar the name of the dial peer to receive calls.
ipv4:10.8.17.40 expires 3600 secondary
• ipv4:ip-address—IP address of the dial peer to receive
calls.
• expires seconds—Default registration time, in
seconds.
• tcp—(Optional) Sets the transport layer protocol to
TCP. UDP is the default.
• secondary—(Optional) Specifies registration with a
secondary SIP proxy or registrar for redundancy
purposes.
Step 3 retry register number Sets the total number of SIP Register messages that the
gateway should send.
Example: • number—Number of Register message retries. Range
Router(config-sip-ua)# retry register 10 is from 1 to 10. Default is 10.
Step 4 timers register time Sets how long the SIP user agent (UA) waits before sending
Register requests.
Example: • time—Waiting time, in milliseconds. Range is from
Router(config-sip-ua)# timers register 500 100 to 1000. Default is 500.
Step 5 exit Exits SIP user-agent configuration mode.

Example:
Router(config-sip-ua)# exit

Troubleshooting Tips

• Use the show sip-ua timers command to show the waiting time before Register requests are sent;
that is, the value that has been set with the timers register command.
• Use the show sip-ua register status command to show the status of local E.164 registrations.
• Use the show sip-ua statistics command to show the Register messages that have been sent.

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 Appendix B: Providing Cisco CME Support for SIP
Call Transfer over SIP Networks

Call Transfer over SIP Networks

Cisco CME supports all SIP Refer method call transfer scenarios. Before configuring the SIP Refer
method, follow the steps in this section to enable call transfer using H.450.2 standards. Note that the
transfer-system command must be configured with the full-blind or full-consult keyword for SIP Refer
to be invoked.

SUMMARY STEPS

1. telephony-service
2. transfer-system {full-blind | full-consult}
3. transfer-pattern transfer-pattern
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 transfer-system {full-blind | full-consult} Defines the call transfer method for all lines served by the
router.
Example: Note For SIP networks, use only the full-blind keyword
Router(config-telephony)# transfer-system or the full-consult keyword. For more information,
full-consult see the Cisco IOS SIP Configuration Guide.

• full-blind—Calls are transferred without consultation

using H.450.2 standard methods.
• full-consult—Calls are transferred with consultation
using a second phone line if available. The calls fall
back to full-blind if the second line is unavailable.

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 Appendix B: Providing Cisco CME Support for SIP
Call Transfer over SIP Networks

Command or Action Purpose

Step 3 transfer-pattern transfer-pattern Allows transfer of telephone calls by Cisco IP phones to
specified phone number patterns. If no transfer pattern is
set, the default is that transfers are permitted only to other
Example:
Router(config-telephony)# transfer-pattern
local IP phones.
52540.. • transfer-pattern—String of digits for permitted call
transfers. Wildcards are allowed.
Note When defining transfers to nonlocal numbers, it is
important to note that transfer-pattern digit
matching is performed before translation-rule
operations. Therefore, you should specify in this
command the digits that are actually entered by
phone users before they are translated. For more
information, refer to the “Translation Rules” section
in the “Setting Up Phones in a Cisco CME System”
chapter.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example specifies transfer with consultation using the H.450.2 standard for all IP phones
serviced by the router:
!
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
session protocol sipv2
session-target ipv4:1.1.1.1
!
telephony-service
transfer-pattern 4...
transfer-system full-consult

What to Do Next

After using the call transfer commands for Cisco CME, you need to configure SIP call transfer, which
is described in the “Configuring SIP Call Transfer” chapter of the Cisco IOS SIP Configuration Guide.

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 Appendix B: Providing Cisco CME Support for SIP
Call Forwarding over SIP Networks

Call Forwarding over SIP Networks

Call forwarding over SIP networks uses the 302 Moved Temporarily SIP response, which works in a
manner similar to the way in which the H.450.3 standard is used for H.323 networks. To enable call
forwarding, use the call-forward pattern command and specify a pattern that matches the calling-party
numbers of the calls that you want to be able to forward. Use the call-forward pattern command with
the .T pattern to allow all calls for all possible SIP calling parties to be forwarded using the SIP 302
response.

SUMMARY STEPS

1. telephony-service
2. call-forward pattern pattern
3. calling-number local
4. exit

DETAILED STEPS

Command or Action Purpose

Step 1 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 2 call-forward pattern pattern Specifies the H.450.3 standard or SIP 302 redirection
method for call forwarding. Calling-party numbers that do
not match the patterns defined with this command are
Example:
Router(config-telephony)# call-forward pattern
forwarded using Cisco-proprietary call forwarding for
4... backward compatibility (as described in the “Configuring
Call Forwarding” chapter in the Cisco IOS Telephony
Services V2.1 guide).
• pattern—Digits to match for call forwarding using the
H.450.3 standard or SIP 302 redirection method. A
pattern of .T matches all calling-party numbers.
Note When defining forwards to nonlocal numbers, it is
important to note that pattern-digit matching is
performed before translation-rule operations.
Therefore, you should specify in this command the
digits actually entered by phone users before they
are translated. For more information, see the
“Translation Rules” section in the “Setting Up
Phones in a Cisco CME System” chapter.

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 Appendix B: Providing Cisco CME Support for SIP
Call Forwarding over SIP Networks

Command or Action Purpose

Step 3 calling-number local (Optional) Replaces a calling-party number and name with
the forwarding-party (local) number and name.
Example: Note This command applies to hairpin-forwarded calls
Router(config-telephony)# calling-number local only and requires installation of the
app-h450-transfer.2.0.0.8.tcl script or a later
version. The local-hairpin attribute-value (AV) pair
must be set to 1.
Step 4 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit

Example

The following example enables call forwarding using the H.450.3 standard or SIP 302 response:
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
session protocol sipv2
session-target ipv4:1.1.1.1
!
telephony-service
call-forward pattern 4...

What to Do Next

After using the call forwarding commands for Cisco CME, you need to configure SIP call forwarding,
which is described in the “Configuring SIP Call Transfer” chapter of the Cisco IOS SIP Configuration
Guide.

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 INDEX

A C

abbreviated dialing 183 call back busy subscriber 178

account code entry 306 call blocking 251
after-hour exempt command 253 call blocking override 254
after-hours block pattern command 252 call coverage, secondary 261
after-hours date command 253 caller-id block command 214
after-hours day command 252 caller ID blocking 213
alerting (call stage) 218 caller-id command 325
allow-connections command 111 call forwarding
API (application programming interface), XML 313 CFwdAll soft key restriction 190
applications integration 306 Cisco-proprietary method 88
archives, downloading Cisco CME 12 configuring 103
audio file for MOH 16 H.450.3 standard 88, 89
authentication for HTTP server 147 SIP networks 337
auto assign command 56, 58 Tcl script 88
auto-cut-through command 173 transcoding between G.726 and G.711 126
auto-line command 188 using do not disturb (DND) soft keys 192
automatic line selection 187 verifying and troubleshooting 118
call-forward max-length command 191
call-forward pattern command 337
B
call notification, on-hold 244
beep, call waiting 198 call park 161
blocking caller ID 213 call pickup groups 286
blocking calls 251 call routing, loopback 323
busy monitor lamp 241 call transfer
busy subscriber callback 178 Cisco-proprietary method 88
button command 63 configuring 103
call-waiting beep allowed 243 consult transfer for direct station select 189
feature ring 63 H.450.2 standard 88, 89
monitor mode 242 SIP networks 335
normal ring 63 Tcl script 88
overlay ephone-dn 265, 269 transcoding between G.726 and G.711 126
silent ring 243 verifying and troubleshooting 118

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 Index

call-waiting beep 198 cor command 258

cautions create cnf-files command 48
usage in text xx create cnf-files command (Cisco IP Phone 7970G and
7971G-GE) 233
CFwdAll soft key restriction 190
customer administrator GUI access setup 152
Cisco BTS Softswitch (Cisco BTS) 89, 90, 91, 93, 95
using the CLI 157
Cisco CallManager
configuring to work with Cisco CME using the GUI 156
116
interworking with Cisco CME 112
network scenario 101, 102
D
no support for H.450 standards 90
Cisco CallManager Express (Cisco CME) date-format command 73

archives, tar 13 debug commands 321

automatic phone setup 50 debug dspfarm command 139

basic concepts 22 debug ephone mtp command 139

description 1 debug sccp command 139

downloading files 12 default-router command 40, 41

feature history 2 description command 217

installation sequence summary 20 DHCP (Dynamic Host Configuration Protocol)

service 38
license requirements 11
dial peers
phone firmware, downgrading 43
ephone-dns and 24
prerequisites 11
for call transfer and forwarding 118
CiscoIOSTSP.zip file 16, 307
dial-plan pattern 65
Cisco IP Phone 7970G and 7971G-GE settings 232
dialplan-pattern command 67
Cisco PSTN Gateway (Cisco PGW) 89, 90, 93, 95
dial tone, secondary 165
Cisco Unity Express
DID (Direct Inward Dialing) 31
transcoding between G.726 and G.711 126
direct 189
clid strip command 214
directory, local
clid strip name command 215
configuring name order 289
client identifier command 41
disabling 291
clock summer-time command 43
directory command 290
clock timezone command 43
directory entry command 291
command syntax conventions xix
direct station select 241
conference calls 194
direct station select for call transfer 189
transcoding between G.726 and G.711 for ad hoc
calls 126 display, phone

Conference Initiator Drop-Off Control 196 header bar 216

Configuring 87 idle system display message (idle URL) 223

connected (call stage) 218 idle system text message 222

COR (class of restriction) 257 URL provisioning for function buttons 225

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 Index

DND (do not disturb) 192, 256 ephone hunt groups 273
dn-webedit command 149 longest-idle 282
documentation conventions xix peer 280
documents, related 33 sequential 227, 277
downloading Cisco CME files 12 ephone-hunt peer command 280
DSPs (digital signal processors) 125, 127 ephone-hunt sequential command 278
DTMF integration with legacy voice-mail devices 236 ephones
DTMF relay 64 configuring 50
H.323 networks 64 definition 24
dtmf-relay command
H.323 networks 65
F
SIP networks 330, 332
dual-line ephone-dn 25 fastdial command 182
dual-number ephone-dns 27 fast transfer 241
feature ring (button command) 63
files, downloading Cisco CME 12
E
final command 278, 281, 283
E.164 number registration firmware
ephone-hunt group pilot number 279, 281, 284 phone files 14
H.323 gatekeeper (dialplan-pattern) 67 phone loads signed and unsigned 15
H.323 gatekeeper (number) 61 Flash soft key 293
SIP 333 foreign exchange office (FXO) trunk lines 227
ephone command 61 forwarding. See call forwarding
ephone-dn command 60 function button customized displays 225
ephone-dns fxo hook-flash command 294
assigning to phone (button command) 63
configuring 50
G
definition 24
dual-line 25 G.711, transcoding between G.726 and 125
dual-number 27 G.729, transcoding between G.711 and 125
hunt groups 273 gatekeeper, H.323
overlaid 262 not registering ephone hunt-group pilot number 279,
overlay 28 281, 284

preference 270 number format restrictions 31

shared 28 group call pickup 286

single-line 25 GUI (graphical user interface) 145

two ephone-dns with one number 26 customer administrator setup 152, 156

types 24 downloading files for 15

files to download 15

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 Index

how to access 150 hold-alert command 245

phone user setup 158 hookflash functionality 293
prerequisites 146 hops command 281, 283
restrictions 146 host command 40
setting up 147 HTTP path, setting 147
system administrator setup 148 HTTP server, enabling 147
hunt groups, ephone 273
huntstop 271
H
huntstop (ephone-dn) command 272
H.323-to-H.323 connections, enabling 111
H.450.12 supplementary services 88
I
description 92
enabling 108 idle (call stage) 219
H.450.2 supplementary services 89 idle message phone display 223
description 89 idle system text message 222
enabling 103 input gain command 173
handling non-H.450.2 calls 92 installation sequence for Cisco CME 20
network requirements 91 intercom 294
H.450.3 supplementary services 89 intercom command 297
description 89 interdigit timeout 167
enabling 103 interface command 42
handling non-H.450.3 calls 92 international language and tone support 72
network requirements 91 interoperability with systems other than Cisco CME 87
H.450 tandem gateways 88 intersite calling plan 30
description 95 ip dhcp pool command 39, 40
enabling H.323-to-H.323 connections 111 ip helper-address command 42
licensing 95 ip http authentication command 148
restrictions 97 ip http path flash command 148, 181
h225 h245-address on-connect (h323 voice-service) ip http server command 147, 181
command 114
IP phone header bar display 216
h225 h245-address on-connect (voice-class)
ip source-address command 48
command 115
hairpin call routing 88
description 93 K
enabling H.323-to-H.323 connections 111
keepalive command 47, 49
network requirements 94
keyphone command 314
restrictions 95
keyswitch model 32
Tcl script in Cisco CME 3.0 88
header bar phone display 216

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 Index

monitor mode for line button 63

L
multicast moh command 171
label command 240 music on hold (MOH)
label support 240 transcoding between G.726 and G.711 126
language support 72
license prerequisites 11
line selection, automatic 187
N
list command 278, 280, 283 name command 61
live-feed music on hold 172 network command 39
load command 48 network-locale command 73
local directory network prerequisites 11
configuring name order 289 night service 246
disabling 291 night-service bell (ephone) command 249
local speed dial 180 night-service bell (ephone-dn) command 249
login (telephony-service) command 255 night-service code command 248
log password command 313 night-service date command 248
log table command 314 night-service day command 248
loopback call routing 323 no-reg (ephone-hunt) command 279, 281, 284
loopback-dn command 327 normal ring (button command) 63
notes, usage in text xx
notify telephone-event command 331, 332
M
NTP (Network Time Protocol) 42
mac-address command 61 ntp server command 43
mapping numbers 31 number command 61
max-conferences command 195 number plan 30
max-dn command 48
max-ephones command 48
max-redirect command 279, 282, 284
O
MIBs 35 on-hold call notification 244
moh (ephone-dn) command 175 on-hook dialing 178
MOH (music on hold) 169 operation command 173
audio file to download 16 option 150 ip command 39, 41
from a live feed 172 overlay ephone-dns 28, 262
from an audio file 169
multicast 171, 175
moh (telephony-service) command 170
monitoring a Cisco CME system 317
monitor lamp 241
monitor-line button 179

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 Index

PIN (personal identification number) 254

P
pin command 255
packet voice/data modules (PVDM) 127 preference (ephone-dn) command 271
paging 298 preference (ephone-hunt) command 283
for a single group 300 preference order of ephone-dns 270
for multiple groups 303 private lines to Public Switched Telephone Network
paging command 301, 304 (PSTN) 227

paging-dn command 301

paging group command 304
R
parking calls 161
park-slot command 163 rebooting phones 75

pattern direct command 237 register support, SIP 333

pattern ext-to-ext busy command 237 registrar command 334

pattern ext-to-ext no-answer command 237 related documents 33

pattern trunk-to-ext busy command 238 relay, DTMF 64

pattern trunk-to-ext no-answer command 238 resetting phones

PBX model 31 description 75

peer ephone hunt groups 280 reset (ephone) command 76

personal speed dial 182 reset (telephony-service) command 76

phone display restarting phones

header bar 216 description 75

idle system display message 223 restart (ephone) command 78

idle system text message 222 restart (telephony-service) command 79

URL provisioning for function buttons 225 restrictions

phone number plan 30 H.450 tandem gateways 97

phones hairpin call routing 95

See ephone-dns and ephones language support 72

phone settings, Cisco IP Phone 7970G and retry register command 334
7971G-GE 232 RFCs 35
phone setup 45 ring, silent 243
automatic with Cisco CME setup tool 50 ringing timeout 165
manual using router CLI 59 routing, loopback 323
partially automated 56
phone user GUI access setup 158
using CLI 160 S
using GUI 158 secondary dial tone 165
pickup-group command 288 secondary-dialtone command 165
pickup groups 286 seized (call stage) 219
pilot command 283 sequential ephone hunt groups 227, 277

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 Index

service dhcp command 42 software prerequisites 12

service local-directory command 292 speed dial 178, 189, 191, 194
service phone command 232 buttons on phones 183
service phone command (Cisco IP Phone 7970G and local 180
7971G-GE) 233
monitor-line button 179
setting up a Cisco CME system 37
personal 182
summary 19
speeddial.xml file 179, 180
shared ephone-dns 28
speed-dial command 185
shared lines 30
standards 34
show call prompt-mem-usage command 119
station select 241
show commands for monitoring a Cisco CME system 317
supplementary-service h225-notify cid-update (dial-peer)
show ephone command 81 command 115
show ephone-dn callback command 178 supplementary-service h225-notify cid-update
(voice-service) command 114
show ephone-dn command 82
supplementary-service h450.12 (dial-peer) command 110
show ephone-dn park command 164
supplementary-service h450.12 (voice-service)
show ephone-hunt command 274
command 109
show fb-its-log command 315
supplementary-service h450.2 (dial-peer) command 107
show running-config command 80
supplementary-service h450.2 (voice-service)
show sccp command 139 command 106
show sccp connections details command 139 supplementary-service h450.3 (dial-peer) command 107
show sdspfarm command 139 supplementary-service h450.3 (voice-service)
show sdspfarm sessions command 129 command 107

show sdspfarm units command 129 system administrator GUI access setup 148

show sip-ua register status command 334 system display message 223

show sip-ua statistics command 334 system message command 223

show sip-ua status command 332 system text message 222

show sip-ua timers command 334

show telephony-service tftp-bindings command 81
T
show voice dsp command 129
show voip rtp connections command 119 TACACS authentication for HTTP server 147

signal immediate command 173 tandem gateways. See H.450 tandem gateways
silent ring 243 TAPI (Telephony Application Programming
Interface) 306
single in-line memory module (SIMM) sockets 127
telephony-service ccm-compatible (h323 voice-service)
single-line ephone-dn 25 command 114
SIP (Session Initiation Protocol) telephony-service ccm-compatible (voice-class)
call forwarding 337 command 115
call transfer 335 telephony-service command 52
register support 333 templates, soft key 218
soft key selection and order 218 TFTP server, changing address 74

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 Index

tftp-server flash command 48 user GUI access setup 158

three-party G.711 conference calls 194 user-locale command 73
time-format command 73 username command 160
timeout
interdigit 167
V
ringing 165
timeout (ephone-hunt) command 279, 281, 283 virtual voice port 24
timeouts busy command 166 vm-integration command 237
timeouts interdigit command 167 voice-mail
timeouts ringing (telephony-service) command 168 Cisco Unity 235
timers register command 334 legacy devices 236
time-webedit command 150 voice port, virtual 24
time-zone command 232 VoIP-to-VoIP connections 87
time-zone command (Cisco IP Phone 7970G and configuring 111
7971G-GE) 233
H.450 tandem gateways 95
toll bar 251
hairpin call routing 93
override 254
tone support, locale-specific 72
transcoding between G.726 and G.711 125 W
transfer, See call transfer
web admin customer command 158
transfer-pattern command 105, 336
web admin system command 149
transfer-system command 105, 335
web customize load command 154
translate command 68
translation-profile command 70
translation profiles 69 X
translation rules 68
XML API 313
troubleshooting 321
XML files
TSP (Telephony Service Provider) 306
downloading xml.template 16
downloading files 16
idle message use of XML file 224
interface 306
speeddial.xml for system speed dial 179, 180
prerequisites 307
XML configuration file for Cisco CME GUI 153
TSPI (Telephony Service Provider Interface) 306
xmlschema command 314
type command 62
xmltest command 314
xmlthread command 314
U

url command 226

url idle command 224
URL provisioning for customized function buttons 225

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